
From dworley@nortel.com  Mon Jun  1 11:56:57 2009
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From: "Dale Worley" <dworley@nortel.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [dispatch] Draft: CBUS (Condition-based URI Subscription)	requirements draft - draft-holmberg-dispatch-cbus-00.txt
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On Thu, 2009-05-28 at 08:53 +0200, Christer Holmberg wrote:
> Last, I don't agree to that the the functionality is a pure
> request/response protocol. It is a sub/not functionaltiy: you may not
> get the "response" (notification") immediately, and in some cases a
> single subscription may trigger multiple notifications.

That last phrase is key, and isn't something that's mentioned very
clearly in the draft.  That is, that once the criteria are established,
the "answer" may change with time and that the CBUS server is expected
to dynamically revise its answer.

Dale



From mramalho@cisco.com  Mon Jun  1 14:18:33 2009
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Thread-Topic: IETF "Royalty-Free" codecs
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From: "Michael Ramalho (mramalho)" <mramalho@cisco.com>
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Subject: [dispatch] IETF "Royalty-Free" codecs
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All,=20

=20

I have read, with some interest and comment, Jason Fischl's email
proposing a working group for codec development that would produce a
"royalty-free" codec designed "specifically for the Internet".

=20

While one cannot always know if there is IPR "lurking" in something that
someone has built (i.e., an "unknowing" infringement on a 3rd party
right), the goal of designing a codec with "old or otherwise
unencumbered" technology is an idea worth pursuing.

=20

I do note, as Roni Even did, that most of the of the "codec expertise"
resides in other SDO groups (such as ITU-T SG16, and ETSI). Most of
those subject matter experts are represented by companies that also
participate in the IEFT.

=20

I also remember that AVT has been down this road before with iLBC (the
result being an EXPERIMENTAL RFC  - not a standards track RFC).

=20

However, I have explicitly noted that other SDO groups such as ITU-T
SG16 have NO APPARENT INTEREST in developing a "royalty-free" designs.
In my observation the ITU-T SG16 tends to focus on bringing the "best
technology" from participating members to create a codec that meets a
given goal (specified by a Terms of Reference, ToR, document). Since the
requirements are written from the perspective of what "new technology
can do", they have the tendency to rule out "old/known/existing"
technology.

=20

Thus, it doesn't appear to me that the ITU-T or similar groups would
even strive to have a goal of "royalty-free" as a codec design criteria.
And that is OK, given the way they operate. They produce high-quality
results for the work they perform.

=20

Additionally, some SDOs explicitly request that  "royalty", or
"royalty-free" statements NOT be made at their meetings except in a
narrow-context of what their company position is given the SDOs IPR
policy. In other words, detailed discussion of the IPR dimensions of the
work are not to be discussed when developing the codec.

=20

So  given:

=20

Given 1: There are a bunch of folks that desire to specify a codec that
does not KNOWINGLY infringe on IPR/patents (perhaps by using "old or
previously disclosed" technology),

=20

Given 2: That no SDO organization that I know of that works on codec
issues explicitly allows the discussion of specific IPR issues or
"royalties" in their meetings, and

=20

Given 3: That some IETF participants desire a codec that does not
KNOWINGLY infringe on IPR and can be made "royalty-free" (except for
potentially lurking IPR).

=20

The questions (for me) become:

=20

Question 1: Where is such work to be performed?

=20

Corollary Question 1: Does anyone know of a SDO that would allow for the
criteria of "no known IPR" in the design?

=20

Question 2: Assuming that the IETF would allow such a "royalty-free"
criteria and that this work COULD be performed here, are there enough
"qualified people" in the IETF to perform such work?

=20

I don't know the answers to the above, but I would be interested in this
work.

=20

Regardds,

=20

Michael A. Ramalho, Ph.D.

=20

<Jason's initial email on dispatch@ietf.org follows>

=20

All,

=20

I would like to request agenda time inside the DISPATCH meeting to
propose

the formation of a new working group to define a Proposed Standard
wideband

audio codec.

=20

The text of the proposal is below. Comments, questions, and suggestions

welcomed.

=20

Regards,=20

Jason

=20

=20

Internet Wideband Audio Codec (IWAC)

Mailing Lists: TBD

Chairs: TBD

Area Directorate: Real Time Applications (RAI)

=20

Purpose:

=20

This new working group would be chartered with the purpose of collecting

expertise within the IETF in order to review the design of audio codecs

specifically for use with the Internet. Unlike other SDOs, these codecs

would be optimized for use on the Internet, and as much as possible
choose

technology that does not require paying patent royalties.

=20

The Internet Low Bit Rate Codec (iLBC)  work was done in AVT but it was
felt

that subsequent work should not be done in the AVT working group. This
new

working group will have as its primary purpose the standardization of a

multi-purpose audio codec that can be used in various situations on the

Internet. Some of the proposed Internet-specific requirements include:

* scalable and adaptive bit rate;

* various sampling rate profiles from narrow-band to super-wideband;

* scalable complexity;

* low latency; and=20

* resilience to packet loss.

=20

There are a number of wide-band capable codecs defined by other SDOs.
For

instance, G.722 is seeing adoption in Enterprise applications since it
is

relatively simple and low-cost to deploy. However, it has a high, fixed

bitrate and is not appropriate for mobile applications where spectrum

efficiency is important or in consumer applications where available

bandwidth is fluctuating or limited. G.722.2 (AMR-wideband) has been
adopted

by the 3GPP as a wideband standard for mobile applications. G.722.2 is

relatively high cost due to patent royalties and is seeing minimal

deployments outside of mobile handsets making it challenging to create

wideband experiences on Internet-capable mobile devices when extending

beyond the mobile network. In other cases, proprietary codecs are being

adopted which further create islands with no interoperability unless

widespread transcoding is performed. Transcoding leads to higher costs
and

lower quality.=20

=20

The goal of this working group is to define a single codec with multiple

profiles which can be made available on a wide variety of
Internet-capable

devices including low-power, mobile devices as well as devices capable
of

utilizing high quality, high bitrate audio.

=20

Proposed Deliverables:

=20

1) Requirements for wideband, Internet audio codec(s).

2) Algorithm description for wideband, Internet audio codec(s) as
Proposed

Standard.=20

3) Specification of payload format(s) for defined codecs as Proposed

Standard

=20


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<div class=3DSection1>

<p class=3DMsoNormal>All, <o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>I have read, with some interest and comment, Jason =
Fischl&#8217;s
email proposing a working group for codec development that would produce =
a &#8220;royalty-free&#8221;
codec designed &#8220;specifically for the =
Internet&#8221;.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>While one cannot always know if there is IPR =
&#8220;lurking&#8221;
in something that someone has built (i.e., an &#8220;unknowing&#8221;
infringement on a 3<sup>rd</sup> party right), the goal of designing a =
codec
with &#8220;old or otherwise unencumbered&#8221; technology is an idea =
worth
pursuing.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>I do note, as Roni Even did, that most of the of =
the &#8220;codec
expertise&#8221; resides in other SDO groups (such as ITU-T SG16, and =
ETSI).
Most of those subject matter experts are represented by companies that =
also
participate in the IEFT.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>I also remember that AVT has been down this road =
before with
iLBC (the result being an EXPERIMENTAL RFC &nbsp;- not a standards track =
RFC).<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>However, I have explicitly noted that other SDO =
groups such
as ITU-T SG16 have NO APPARENT INTEREST in developing a =
&#8220;royalty-free&#8221;
designs. In my observation the ITU-T SG16 tends to focus on bringing the =
&#8220;best
technology&#8221; from participating members to create a codec that =
meets a
given goal (specified by a Terms of Reference, ToR, document). Since the =
requirements
are written from the perspective of what &#8220;new technology can =
do&#8221;,
they have the tendency to rule out &#8220;old/known/existing&#8221; =
technology.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Thus, it doesn&#8217;t appear to me that the ITU-T =
or
similar groups would even strive to have a goal of =
&#8220;royalty-free&#8221;
as a codec design criteria. &nbsp;And that is OK, given the way they =
operate.
They produce high-quality results for the work they =
perform.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Additionally, some SDOs explicitly request that =
&nbsp;&#8220;royalty&#8221;,
or &#8220;royalty-free&#8221; statements NOT be made at their meetings =
except
in a narrow-context of what their company position is given the SDOs IPR =
policy.
In other words, detailed discussion of the IPR dimensions of the work =
are not
to be discussed when developing the codec.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>So &nbsp;given:<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Given 1: There are a bunch of folks that desire to =
specify a
codec that does not KNOWINGLY infringe on IPR/patents (perhaps by using =
&#8220;old
or previously disclosed&#8221; technology),<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Given 2: That no SDO organization that I know of =
that works
on codec issues explicitly allows the discussion of specific IPR issues =
or &#8220;royalties&#8221;
in their meetings, and<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Given 3: That some IETF participants desire a codec =
that
does not KNOWINGLY infringe on IPR and can be made =
&#8220;royalty-free&#8221;
(except for potentially lurking IPR).<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>The questions (for me) become:<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Question 1: Where is such work to be =
performed?<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Corollary Question 1: Does anyone know of a SDO =
that would
allow for the criteria of &#8220;no known IPR&#8221; in the =
design?<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Question 2: Assuming that the IETF would allow such =
a &#8220;royalty-free&#8221;
criteria and that this work COULD be performed here, are there enough =
&#8220;qualified
people&#8221; in the IETF to perform such work?<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>I don&#8217;t know the answers to the above, but I =
would be
interested in this work.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Regardds,<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Michael A. Ramalho, Ph.D.<o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>&lt;Jason&#8217;s
initial email on <a =
href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a>
follows&gt;<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'>All,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>I
would like to request agenda time inside the DISPATCH meeting to =
propose<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>the
formation of a new working group to define a Proposed Standard =
wideband<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>audio
codec.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>The
text of the proposal is below. Comments, questions, and =
suggestions<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'>welcomed.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>Regards,
<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'>Jason<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>Internet
Wideband Audio Codec (IWAC)<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>Mailing
Lists: TBD<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>Chairs:
TBD<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>Area
Directorate: Real Time Applications (RAI)<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'>Purpose:<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>This
new working group would be chartered with the purpose of =
collecting<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>expertise
within the IETF in order to review the design of audio =
codecs<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>specifically
for use with the Internet. Unlike other SDOs, these =
codecs<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>would
be optimized for use on the Internet, and as much as possible =
choose<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>technology
that does not require paying patent royalties.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>The
Internet Low Bit Rate Codec (iLBC)&nbsp; work was done in AVT but it was =
felt<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>that
subsequent work should not be done in the AVT working group. This =
new<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>working
group will have as its primary purpose the standardization of =
a<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>multi-purpose
audio codec that can be used in various situations on =
the<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>Internet.
Some of the proposed Internet-specific requirements =
include:<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>*
scalable and adaptive bit rate;<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>*
various sampling rate profiles from narrow-band to =
super-wideband;<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>*
scalable complexity;<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>*
low latency; and <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>*
resilience to packet loss.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>There
are a number of wide-band capable codecs defined by other SDOs. =
For<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>instance,
G.722 is seeing adoption in Enterprise applications since it =
is<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>relatively
simple and low-cost to deploy. However, it has a high, =
fixed<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>bitrate
and is not appropriate for mobile applications where =
spectrum<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>efficiency
is important or in consumer applications where =
available<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>bandwidth
is fluctuating or limited. G.722.2 (AMR-wideband) has been =
adopted<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>by
the 3GPP as a wideband standard for mobile applications. G.722.2 =
is<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>relatively
high cost due to patent royalties and is seeing =
minimal<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>deployments
outside of mobile handsets making it challenging to =
create<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>wideband
experiences on Internet-capable mobile devices when =
extending<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>beyond
the mobile network. In other cases, proprietary codecs are =
being<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>adopted
which further create islands with no interoperability =
unless<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>widespread
transcoding is performed. Transcoding leads to higher costs =
and<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>lower
quality. <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>The
goal of this working group is to define a single codec with =
multiple<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>profiles
which can be made available on a wide variety of =
Internet-capable<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>devices
including low-power, mobile devices as well as devices capable =
of<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>utilizing
high quality, high bitrate audio.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>Proposed
Deliverables:<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>1)
Requirements for wideband, Internet audio =
codec(s).<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>2)
Algorithm description for wideband, Internet audio codec(s) as =
Proposed<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>Standard.
<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>3)
Specification of payload format(s) for defined codecs as =
Proposed<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'>Standard<o:p></o:p></span></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

</body>

</html>

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A New Internet-Draft is available from the on-line Internet-Drafts directories.
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	Title           : Managing Client Initiated Connections in the Session Initiation Protocol (SIP)
	Author(s)       : C. Jennings
	Filename        : draft-ietf-sip-outbound-19.txt
	Pages           : 49
	Date            : 2009-06-01

The Session Initiation Protocol (SIP) allows proxy servers to
initiate TCP connections or to send asynchronous UDP datagrams to
User Agents in order to deliver requests.  However, in a large number
of real deployments, many practical considerations, such as the
existence of firewalls and Network Address Translators (NATs) or the
use of TLS with server-provided certificates, prevent servers from
connecting to User Agents in this way.  This specification defines
behaviors for User Agents, registrars and proxy servers that allow
requests to be delivered on existing connections established by the
User Agent.  It also defines keep alive behaviors needed to keep NAT
bindings open and specifies the usage of multiple connections from
the User Agent to its Registrar.

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Subject: Re: [dispatch] Proposal to form Internet Wideband Audio Codec WG
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I again have to point out, as I did with iLBC, that just because  
something is Open Source and submitted Without IPR and, as proposed  
here, done in a non-IPR regime, does NOT necessarily mean the result  
is without encumbrances.

Since I like to keep people away from trouble, I won't quote existing  
IPR. Let's look at some expired IPR instead.

If your proposed codec used LPC, it would violate a Philips patent.  
For that matter, it would step on a handful of TI patents.

I would guess (but I have NOT done a search), if the proposed codec  
used FFT, DCT, physiological modeling, filtering, etc., it would step  
on a handful of patents.

I would offer the math work for developing a codec is something  
outside the IETF area of expertise.  I suppose it is right and proper  
to ask the question, as to counter Joel's take-it-to-the-logical- 
conclusion the IETF should do everything (not), one could ask if the  
IETF should not ever do anything new (I would offer: not).

I would also offer the protocol work for such a codec is something the  
IETF really, really should do.  If that work is somehow different than  
the AVT work, for example, if such a codec required a transport that  
RTP/SRTP cannot provide, then a new work group would be in order.

My vote is the IETF does not have the expertise to do the math part.   
My vote is the IETF has the best expertise for the protocol part.

On May 29, 2009, at 11:45 AM, Henry Sinnreich wrote:

> Spencer,
>
> Yes, your points show exactly why we need a new WG in the IETF  
> chartered specifically for a standard for an Internet voice (and why  
> not video as well) codec.
>
> The _global_ standard nature, royalty free, open source aspects will  
> most likely warm the heart of most Internet folks.
>
> The determining factor will be if there are contributors willing to  
> do the work of writing the I-Ds for the requirements, algorithm,  
> code and last but not least show  measurements about performance. I  
> see signs that we may have very valuable contributions.
>
> We can gage the interest for such a WG by the attendance in the BOF  
> at the 75 IETF in July.
> Please just give it a chance.
> We may soon be in the fortunate position to finally have an Internet  
> voice codec standard!
>
> Henry
>
>
> On 5/29/09 9:56 AM, "Spencer Dawkins" <spencer@wonderhamster.org>  
> wrote:
>
> Hi, Henry,
>
> I may have misunderstood Roni's point, but I thought that he was  
> saying that audio codec types don't participate in the IETF today,  
> because the IETF does not develop audio codecs (and AVT is  
> prohibited by charter from producing one, because of a stated belief  
> that we don't have the expertise in the IETF to do this work).
>
> Thanks,
>
> Spencer
>
> ----- Original Message -----
>
> From:  Henry  Sinnreich <mailto:hsinnrei@adobe.com>
>
> To: Roni Even <mailto:ron.even.tlv@gmail.com>  ; Jason  Fischl <mailto:jason.fischl@skype.net 
> >  ; dispatch@ietf.org
>
> Sent: Thursday, May 28, 2009 10:28  AM
>
> Subject: Re: [dispatch] Proposal to form  Internet Wideband Audio  
> Codec WG
>
>
> Roni,
>
> Sorry, we have here a fundamental  disagreement.
> The IETF is chartered for Internet standards and may or may  not  
> chose solutions that apply to ITU-T networks.
> The Internet has  different criteria than ITU-T networks may have.
> A worldwide Internet  standard for a wideband codec will be very  
> beneficial  IMO.
>
> Henry
>
>
> On 5/28/09 9:45 AM, "Roni Even" <ron.even.tlv@gmail.com>  wrote:
>
>
> Hi,
> Like you mention other SDOs like ITU-T are  doing just that. They  
> have the
> expertise to specify, and evaluate the  result. These SDOs can receive
> requirements and select a proper codec  based on the requirements.
>
>
> As for the other reasons:
>
> 1.  Defining a codec in the IETF or even in MPEG / ITU-T does not  
> make it  a
> mandatory part of a system solution, this is done by other standard   
> bodies
> like 3GPP, ETSI.
>
> 2. The IETF, similar to other standard  bodies is not rubber  
> stamping a
> specific solution, so you will most  probably see in the final  
> result some
> technology that carry  IPR.
>
> 3. If this group will be established, you will probably see here   
> the audio
> experts now working in ITU-T arguing the same issues since they  are  
> the
> expertise you need and they work for the same companies that are   
> already
> members of IETF.
>
> I think that if you have a specific codec  in mind you  can make it  
> publicly
> available maybe with quality  results and standardized in AVT a  
> payload
> specification.
>
> BTW: The  ITU is keeping a list of codecs (Not only ITU-T ones) in a  
> table
> that  describes their features.
>
> Regards
> Roni Even
>
> -----Original  Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org]   
> On Behalf
> Of Jason Fischl
> Sent: Wednesday, May 27, 2009 2:18 AM
> To:  dispatch@ietf.org
> Subject: [dispatch]  Proposal to form Internet Wideband Audio Codec WG
>
> All,
>
> I would  like to request agenda time inside the DISPATCH meeting to  
> propose
> the  formation of a new working group to define a Proposed Standard   
> wideband
> audio codec.
>
> The text of the proposal is below. Comments,  questions, and   
> suggestions
> welcomed.
>
> Regards,
> Jason
>
>
> Internet  Wideband Audio Codec (IWAC)
> Mailing Lists: TBD
> Chairs: TBD
> Area  Directorate: Real Time Applications (RAI)
>
> Purpose:
>
> This new  working group would be chartered with the purpose of  
> collecting
> expertise  within the IETF in order to review the design of audio   
> codecs
> specifically for use with the Internet. Unlike other SDOs, these   
> codecs
> would be optimized for use on the Internet, and as much as  possible  
> choose
> technology that does not require paying patent  royalties.
>
> The Internet Low Bit Rate Codec (iLBC)  work was done  in AVT but it  
> was felt
> that subsequent work should not be done in the AVT  working group.  
> This new
> working group will have as its primary purpose  the standardization  
> of a
> multi-purpose audio codec that can be used in  various situations on  
> the
> Internet. Some of the proposed  Internet-specific requirements  
> include:
> * scalable and adaptive bit  rate;
> * various sampling rate profiles from narrow-band to  super-wideband;
> * scalable complexity;
> * low latency; and
> *  resilience to packet loss.
>
> There are a number of wide-band capable  codecs defined by other  
> SDOs. For
> instance, G.722 is seeing adoption in  Enterprise applications since  
> it is
> relatively simple and low-cost to  deploy. However, it has a high,  
> fixed
> bitrate and is not appropriate for  mobile applications where spectrum
> efficiency is important or in consumer  applications where available
> bandwidth is fluctuating or limited. G.722.2  (AMR-wideband) has  
> been adopted
> by the 3GPP as a wideband standard for  mobile applications. G.722.2  
> is
> relatively high cost due to patent  royalties and is seeing minimal
> deployments outside of mobile handsets  making it challenging to  
> create
> wideband experiences on Internet-capable  mobile devices when  
> extending
> beyond the mobile network. In other cases,  proprietary codecs are  
> being
> adopted which further create islands with no  interoperability unless
> widespread transcoding is performed. Transcoding  leads to higher  
> costs and
> lower quality.
>
> The goal of this working  group is to define a single codec with  
> multiple
> profiles which can be  made available on a wide variety of Internet- 
> capable
> devices including  low-power, mobile devices as well as devices  
> capable of
> utilizing high  quality, high bitrate audio.
>
> Proposed Deliverables:
>
> 1)  Requirements for wideband, Internet audio codec(s).
> 2) Algorithm  description for wideband, Internet audio codec(s) as   
> Proposed
> Standard.
> 3) Specification of payload format(s) for defined  codecs as  Proposed
> Standard
>
> _______________________________________________
> dispatch  mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>
> _______________________________________________
> dispatch  mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>
>
>
>
>
>
> _______________________________________________
> dispatch mailing  list
> dispatch@ietf.org
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>
> _______________________________________________
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Cc: Henry Sinnreich <hsinnrei@adobe.com>, dispatch@ietf.org
Subject: Re: [dispatch] Proposal to form Internet Wideband Audio Codec WG
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On Jun 1, 2009, at 4:02 PM, Eric Burger wrote:

> I again have to point out, as I did with iLBC, that just because  
> something is Open Source and submitted Without IPR and, as proposed  
> here, done in a non-IPR regime, does NOT necessarily mean the result  
> is without encumbrances.
>
> Since I like to keep people away from trouble, I won't quote  
> existing IPR. Let's look at some expired IPR instead.
>
> If your proposed codec used LPC, it would violate a Philips patent.  
> For that matter, it would step on a handful of TI patents.

That's an interesting assertion given LPC was published in the late  
60's and research goes back earlier than that. Would you like to  
comment on "valid patents" :-)


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Subject: Re: [dispatch] Proposal to form Internet Wideband Audio Codec WG
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Michael Ramalho \(mramalho\)'" <mramalho@cisco.com>, <dispatch@ietf.org>
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Date: Tue, 2 Jun 2009 08:25:51 +0300
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Subject: Re: [dispatch] IETF "Royalty-Free" codecs
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Michael,

I think your observation is valid. As a matter of fact even at the ITU some
companies wanted to have "royalty free" codecs and as far as I know some of
the audio codecs like G.722.1C are royalty free.  (I am not sure about your
ITU WP3 work)

I think that you cannot write as part of the term of reference (or charter)
that the codec must be "royalty free" since this is a business decision for
the technology contributor and not a technical one and will probably be
against some of the competition laws in some countries.

 

Roni Even

 

 

 

From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
Of Michael Ramalho (mramalho)
Sent: Tuesday, June 02, 2009 12:19 AM
To: dispatch@ietf.org
Subject: [dispatch] IETF "Royalty-Free" codecs
Importance: Low

 

All, 

 

I have read, with some interest and comment, Jason Fischl's email proposing
a working group for codec development that would produce a "royalty-free"
codec designed "specifically for the Internet".

 

While one cannot always know if there is IPR "lurking" in something that
someone has built (i.e., an "unknowing" infringement on a 3rd party right),
the goal of designing a codec with "old or otherwise unencumbered"
technology is an idea worth pursuing.

 

I do note, as Roni Even did, that most of the of the "codec expertise"
resides in other SDO groups (such as ITU-T SG16, and ETSI). Most of those
subject matter experts are represented by companies that also participate in
the IEFT.

 

I also remember that AVT has been down this road before with iLBC (the
result being an EXPERIMENTAL RFC  - not a standards track RFC).

 

However, I have explicitly noted that other SDO groups such as ITU-T SG16
have NO APPARENT INTEREST in developing a "royalty-free" designs. In my
observation the ITU-T SG16 tends to focus on bringing the "best technology"
from participating members to create a codec that meets a given goal
(specified by a Terms of Reference, ToR, document). Since the requirements
are written from the perspective of what "new technology can do", they have
the tendency to rule out "old/known/existing" technology.

 

Thus, it doesn't appear to me that the ITU-T or similar groups would even
strive to have a goal of "royalty-free" as a codec design criteria.  And
that is OK, given the way they operate. They produce high-quality results
for the work they perform.

 

Additionally, some SDOs explicitly request that  "royalty", or
"royalty-free" statements NOT be made at their meetings except in a
narrow-context of what their company position is given the SDOs IPR policy.
In other words, detailed discussion of the IPR dimensions of the work are
not to be discussed when developing the codec.

 

So  given:

 

Given 1: There are a bunch of folks that desire to specify a codec that does
not KNOWINGLY infringe on IPR/patents (perhaps by using "old or previously
disclosed" technology),

 

Given 2: That no SDO organization that I know of that works on codec issues
explicitly allows the discussion of specific IPR issues or "royalties" in
their meetings, and

 

Given 3: That some IETF participants desire a codec that does not KNOWINGLY
infringe on IPR and can be made "royalty-free" (except for potentially
lurking IPR).

 

The questions (for me) become:

 

Question 1: Where is such work to be performed?

 

Corollary Question 1: Does anyone know of a SDO that would allow for the
criteria of "no known IPR" in the design?

 

Question 2: Assuming that the IETF would allow such a "royalty-free"
criteria and that this work COULD be performed here, are there enough
"qualified people" in the IETF to perform such work?

 

I don't know the answers to the above, but I would be interested in this
work.

 

Regardds,

 

Michael A. Ramalho, Ph.D.

 

<Jason's initial email on dispatch@ietf.org follows>

 

All,

 

I would like to request agenda time inside the DISPATCH meeting to propose

the formation of a new working group to define a Proposed Standard wideband

audio codec.

 

The text of the proposal is below. Comments, questions, and suggestions

welcomed.

 

Regards, 

Jason

 

 

Internet Wideband Audio Codec (IWAC)

Mailing Lists: TBD

Chairs: TBD

Area Directorate: Real Time Applications (RAI)

 

Purpose:

 

This new working group would be chartered with the purpose of collecting

expertise within the IETF in order to review the design of audio codecs

specifically for use with the Internet. Unlike other SDOs, these codecs

would be optimized for use on the Internet, and as much as possible choose

technology that does not require paying patent royalties.

 

The Internet Low Bit Rate Codec (iLBC)  work was done in AVT but it was felt

that subsequent work should not be done in the AVT working group. This new

working group will have as its primary purpose the standardization of a

multi-purpose audio codec that can be used in various situations on the

Internet. Some of the proposed Internet-specific requirements include:

* scalable and adaptive bit rate;

* various sampling rate profiles from narrow-band to super-wideband;

* scalable complexity;

* low latency; and 

* resilience to packet loss.

 

There are a number of wide-band capable codecs defined by other SDOs. For

instance, G.722 is seeing adoption in Enterprise applications since it is

relatively simple and low-cost to deploy. However, it has a high, fixed

bitrate and is not appropriate for mobile applications where spectrum

efficiency is important or in consumer applications where available

bandwidth is fluctuating or limited. G.722.2 (AMR-wideband) has been adopted

by the 3GPP as a wideband standard for mobile applications. G.722.2 is

relatively high cost due to patent royalties and is seeing minimal

deployments outside of mobile handsets making it challenging to create

wideband experiences on Internet-capable mobile devices when extending

beyond the mobile network. In other cases, proprietary codecs are being

adopted which further create islands with no interoperability unless

widespread transcoding is performed. Transcoding leads to higher costs and

lower quality. 

 

The goal of this working group is to define a single codec with multiple

profiles which can be made available on a wide variety of Internet-capable

devices including low-power, mobile devices as well as devices capable of

utilizing high quality, high bitrate audio.

 

Proposed Deliverables:

 

1) Requirements for wideband, Internet audio codec(s).

2) Algorithm description for wideband, Internet audio codec(s) as Proposed

Standard. 

3) Specification of payload format(s) for defined codecs as Proposed

Standard

 


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<div class=3DSection1>

<p class=3DMsoNormal><span =
style=3D'color:#1F497D'>Michael,<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>I think your =
observation is
valid. As a matter of fact even at the ITU some companies wanted to have =
&quot;royalty
free&quot; codecs and as far as I know some of the audio codecs like =
G.722.1C
are royalty free. &nbsp;(I am not sure about your ITU WP3 =
work)<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>I think that you =
cannot write as
part of the term of reference (or charter) that the codec must be =
&quot;royalty
free&quot; since this is a business decision for the technology =
contributor and
not a technical one and will probably be against some of the competition =
laws
in some countries.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>Roni =
Even<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] <b>On =
Behalf Of </b>Michael
Ramalho (mramalho)<br>
<b>Sent:</b> Tuesday, June 02, 2009 12:19 AM<br>
<b>To:</b> dispatch@ietf.org<br>
<b>Subject:</b> [dispatch] IETF &quot;Royalty-Free&quot; codecs<br>
<b>Importance:</b> Low<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>All, <o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>I have read, with some interest and comment, Jason
Fischl&#8217;s email proposing a working group for codec development =
that would
produce a &#8220;royalty-free&#8221; codec designed &#8220;specifically =
for the
Internet&#8221;.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>While one cannot always know if there is IPR =
&#8220;lurking&#8221;
in something that someone has built (i.e., an &#8220;unknowing&#8221;
infringement on a 3<sup>rd</sup> party right), the goal of designing a =
codec
with &#8220;old or otherwise unencumbered&#8221; technology is an idea =
worth
pursuing.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>I do note, as Roni Even did, that most of the of =
the
&#8220;codec expertise&#8221; resides in other SDO groups (such as ITU-T =
SG16,
and ETSI). Most of those subject matter experts are represented by =
companies
that also participate in the IEFT.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>I also remember that AVT has been down this road =
before with
iLBC (the result being an EXPERIMENTAL RFC &nbsp;- not a standards track =
RFC).<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>However, I have explicitly noted that other SDO =
groups such
as ITU-T SG16 have NO APPARENT INTEREST in developing a
&#8220;royalty-free&#8221; designs. In my observation the ITU-T SG16 =
tends to
focus on bringing the &#8220;best technology&#8221; from participating =
members
to create a codec that meets a given goal (specified by a Terms of =
Reference,
ToR, document). Since the requirements are written from the perspective =
of what
&#8220;new technology can do&#8221;, they have the tendency to rule out
&#8220;old/known/existing&#8221; technology.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Thus, it doesn&#8217;t appear to me that the ITU-T =
or
similar groups would even strive to have a goal of =
&#8220;royalty-free&#8221;
as a codec design criteria. &nbsp;And that is OK, given the way they =
operate.
They produce high-quality results for the work they =
perform.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Additionally, some SDOs explicitly request that
&nbsp;&#8220;royalty&#8221;, or &#8220;royalty-free&#8221; statements =
NOT be
made at their meetings except in a narrow-context of what their company
position is given the SDOs IPR policy. In other words, detailed =
discussion of
the IPR dimensions of the work are not to be discussed when developing =
the
codec.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>So &nbsp;given:<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Given 1: There are a bunch of folks that desire to =
specify a
codec that does not KNOWINGLY infringe on IPR/patents (perhaps by using
&#8220;old or previously disclosed&#8221; technology),<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Given 2: That no SDO organization that I know of =
that works
on codec issues explicitly allows the discussion of specific IPR issues =
or
&#8220;royalties&#8221; in their meetings, and<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Given 3: That some IETF participants desire a codec =
that
does not KNOWINGLY infringe on IPR and can be made =
&#8220;royalty-free&#8221;
(except for potentially lurking IPR).<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>The questions (for me) become:<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Question 1: Where is such work to be =
performed?<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Corollary Question 1: Does anyone know of a SDO =
that would
allow for the criteria of &#8220;no known IPR&#8221; in the =
design?<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Question 2: Assuming that the IETF would allow such =
a
&#8220;royalty-free&#8221; criteria and that this work COULD be =
performed here,
are there enough &#8220;qualified people&#8221; in the IETF to perform =
such
work?<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>I don&#8217;t know the answers to the above, but I =
would be
interested in this work.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Regardds,<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Michael A. Ramalho, Ph.D.<o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>&lt;Jason&#8217;s
initial email on <a =
href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a>
follows&gt;<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'>All,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>I
would like to request agenda time inside the DISPATCH meeting to =
propose<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>the
formation of a new working group to define a Proposed Standard =
wideband<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>audio
codec.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>The
text of the proposal is below. Comments, questions, and =
suggestions<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'>welcomed.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>Regards,
<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'>Jason<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>Internet
Wideband Audio Codec (IWAC)<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>Mailing
Lists: TBD<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>Chairs:
TBD<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>Area
Directorate: Real Time Applications (RAI)<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'>Purpose:<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>This
new working group would be chartered with the purpose of =
collecting<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>expertise
within the IETF in order to review the design of audio =
codecs<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>specifically
for use with the Internet. Unlike other SDOs, these =
codecs<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>would
be optimized for use on the Internet, and as much as possible =
choose<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>technology
that does not require paying patent royalties.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>The
Internet Low Bit Rate Codec (iLBC)&nbsp; work was done in AVT but it was =
felt<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>that
subsequent work should not be done in the AVT working group. This =
new<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>working
group will have as its primary purpose the standardization of =
a<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>multi-purpose
audio codec that can be used in various situations on =
the<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>Internet.
Some of the proposed Internet-specific requirements =
include:<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>*
scalable and adaptive bit rate;<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>*
various sampling rate profiles from narrow-band to =
super-wideband;<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>*
scalable complexity;<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>*
low latency; and <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>*
resilience to packet loss.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>There
are a number of wide-band capable codecs defined by other SDOs. =
For<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>instance,
G.722 is seeing adoption in Enterprise applications since it =
is<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>relatively
simple and low-cost to deploy. However, it has a high, =
fixed<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>bitrate
and is not appropriate for mobile applications where =
spectrum<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>efficiency
is important or in consumer applications where =
available<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>bandwidth
is fluctuating or limited. G.722.2 (AMR-wideband) has been =
adopted<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>by
the 3GPP as a wideband standard for mobile applications. G.722.2 =
is<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>relatively
high cost due to patent royalties and is seeing =
minimal<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>deployments
outside of mobile handsets making it challenging to =
create<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>wideband
experiences on Internet-capable mobile devices when =
extending<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>beyond
the mobile network. In other cases, proprietary codecs are =
being<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>adopted
which further create islands with no interoperability =
unless<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>widespread
transcoding is performed. Transcoding leads to higher costs =
and<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>lower
quality. <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>The
goal of this working group is to define a single codec with =
multiple<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>profiles
which can be made available on a wide variety of =
Internet-capable<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>devices
including low-power, mobile devices as well as devices capable =
of<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>utilizing
high quality, high bitrate audio.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>Proposed
Deliverables:<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>1)
Requirements for wideband, Internet audio =
codec(s).<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>2)
Algorithm description for wideband, Internet audio codec(s) as =
Proposed<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>Standard.
<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>3)
Specification of payload format(s) for defined codecs as =
Proposed<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'>Standard<o:p></o:p></span></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

</body>

</html>

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Thread-Topic: [dispatch] Proposal to form Internet Wideband Audio Codec WG
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References: <4A24B17A.5000408@ericsson.com> <C64A98E2.3E3D%hsinnrei@adobe.com>
From: "Slava Borilin" <Borilin@spiritdsp.com>
To: "Roni Even" <ron.even.tlv@gmail.com>, "Jason Fischl" <jason.fischl@skype.net>, "Mary Barnes" <mary.barnes@nortel.com>, "Peterson, Jon" <jon.peterson@neustar.biz>, "Cullen Jennings" <fluffy@cisco.com>, "Gonzalo Camarillo" <Gonzalo.Camarillo@ericsson.com>
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Cc: Henry Sinnreich <hsinnrei@adobe.com>, dispatch@ietf.org
Subject: Re: [dispatch] Proposal to form Internet Wideband Audio Codec WG
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Dear Gonzalo,
=20
by "to convince them may be to have IETF people with knowledge in this
area volunteer to
work on this and review the results." Gonzalo means find IETF voluntiers
to review the idea of the WG?"

you mean to find the IETF people who will vounteers to review the idea
of such Wideband WG?
=20
regards,
Slava Borilin
=20
------ Forwarded Message
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
Date: Mon, 1 Jun 2009 21:58:34 -0700
To: Adobe Systems <hsinnrei@adobe.com>
Cc: Roni Even <ron.even.tlv@gmail.com>, Jason Fischl
<jason.fischl@skype.net>, Mary Barnes <mary.barnes@nortel.com>,
"Peterson, Jon" <jon.peterson@neustar.biz>, Cullen Jennings
<fluffy@cisco.com>
Subject: Re: [dispatch] Proposal to form Internet Wideband Audio Codec
WG

Hi Henry,

given that some people on the list think that the IETF does not have
that type of expertise, we need to have those list discussions so that
the WG reaches consensus (one way or the other). It seems that when iLBC
was developed, it was difficult to find IETF people to review it. That
is probably why some people are hesitant about this. A way to convince
them may be to have IETF people with knowledge in this area volunteer to
work on this and review the results.

Thanks,

Gonzalo

Henry Sinnreich wrote:
> (list deleted)
>
> Gonzalo,
>
>>whether or not the IETF in general and RAI in particular have the
> expertise to
>>do this or at least review it when it is done. Comments?
>
> The folks from Skype driving this proposal have without any doubt the
> best qualifications for building voice codecs, just as have many other
> IETF AVT contributors, such as for the iLBC, SPEEX, IP-MR and other
codecs.
>
> The pushback for the proposal for a WG for an Internet Audio Codec, is
> quite amazing, especially the argument on available expertise.
> (You certainly understand my arguments why the better expertise for an
> _Internet_ codec is just in the IETF, and certainly not in the ITU-T,
> ETSI, OMA, etc.).
>
> I don't plan therefore to participate any more in this discussion.
>
> Henry
>
>
> On 5/30/09 1:17 AM, "Gonzalo Camarillo"
<Gonzalo.Camarillo@ericsson.com>
> wrote:
>
>     Hi,
>
>     I think Roni touches upon an important point here. The first
question we
>     need to answer (not only for this proposal; for any proposal) is
whether
>     or not the IETF in general and RAI in particular have the
expertise to
>     do this or at least review it when it is done. Comments?
>
>     Cheers,
>
>     Gonzalo
>
>     Roni Even wrote:
>     >  Hi,
>     >  Like you mention other SDOs like ITU-T are doing just that.
They
>     have the
>     >  expertise to specify, and evaluate the result. These SDOs can
receive
>     >  requirements and select a proper codec based on the
requirements.
>     >
>     >
>     >  As for the other reasons:
>     >
>     >  1. Defining a codec in the IETF or even in MPEG / ITU-T does
not
>     make it a
>     >  mandatory part of a system solution, this is done by other
>     standard bodies
>     >  like 3GPP, ETSI.
>     >
>     >  2. The IETF, similar to other standard bodies is not rubber
stamping a
>     >  specific solution, so you will most probably see in the final
>     result some
>     >  technology that carry IPR.
>     >
>     >  3. If this group will be established, you will probably see
here
>     the audio
>     >  experts now working in ITU-T arguing the same issues since they
>     are the
>     >  expertise you need and they work for the same companies that
are
>     already
>     >  members of IETF.
>     >
>     >  I think that if you have a specific codec in mind you  can make
it
>     publicly
>     >  available maybe with quality results and standardized in AVT a
payload
>     >  specification.
>     >
>     >  BTW: The ITU is keeping a list of codecs (Not only ITU-T ones)
in
>     a table
>     >  that describes their features.
>     >
>     >  Regards
>     >  Roni Even
>     >
>     >  -----Original Message-----
>     >  From: dispatch-bounces@ietf.org
[mailto:dispatch-bounces@ietf.org]
>     On Behalf
>     >  Of Jason Fischl
>     >  Sent: Wednesday, May 27, 2009 2:18 AM
>     >  To: dispatch@ietf.org
>     >  Subject: [dispatch] Proposal to form Internet Wideband Audio
Codec WG
>     >
>     >  All,
>     >
>     >  I would like to request agenda time inside the DISPATCH meeting
to
>     propose
>     >  the formation of a new working group to define a Proposed
Standard
>     wideband
>     >  audio codec.
>     >
>     >  The text of the proposal is below. Comments, questions, and
>     suggestions
>     >  welcomed.
>     >
>     >  Regards,
>     >  Jason
>     >
>     >
>     >  Internet Wideband Audio Codec (IWAC)
>     >  Mailing Lists: TBD
>     >  Chairs: TBD
>     >  Area Directorate: Real Time Applications (RAI)
>     >
>     >  Purpose:
>     >
>     >  This new working group would be chartered with the purpose of
>     collecting
>     >  expertise within the IETF in order to review the design of
audio
>     codecs
>     >  specifically for use with the Internet. Unlike other SDOs,
these
>     codecs
>     >  would be optimized for use on the Internet, and as much as
>     possible choose
>     >  technology that does not require paying patent royalties.
>     >
>     >  The Internet Low Bit Rate Codec (iLBC)  work was done in AVT
but
>     it was felt
>     >  that subsequent work should not be done in the AVT working
group.
>     This new
>     >  working group will have as its primary purpose the
standardization
>     of a
>     >  multi-purpose audio codec that can be used in various
situations
>     on the
>     >  Internet. Some of the proposed Internet-specific requirements
include:
>     >  * scalable and adaptive bit rate;
>     >  * various sampling rate profiles from narrow-band to
super-wideband;
>     >  * scalable complexity;
>     >  * low latency; and
>     >  * resilience to packet loss.
>     >
>     >  There are a number of wide-band capable codecs defined by other
>     SDOs. For
>     >  instance, G.722 is seeing adoption in Enterprise applications
>     since it is
>     >  relatively simple and low-cost to deploy. However, it has a
high,
>     fixed
>     >  bitrate and is not appropriate for mobile applications where
spectrum
>     >  efficiency is important or in consumer applications where
available
>     >  bandwidth is fluctuating or limited. G.722.2 (AMR-wideband) has
>     been adopted
>     >  by the 3GPP as a wideband standard for mobile applications.
G.722.2 is
>     >  relatively high cost due to patent royalties and is seeing
minimal
>     >  deployments outside of mobile handsets making it challenging to
create
>     >  wideband experiences on Internet-capable mobile devices when
extending
>     >  beyond the mobile network. In other cases, proprietary codecs
are
>     being
>     >  adopted which further create islands with no interoperability
unless
>     >  widespread transcoding is performed. Transcoding leads to
higher
>     costs and
>     >  lower quality.
>     >
>     >  The goal of this working group is to define a single codec with
>     multiple
>     >  profiles which can be made available on a wide variety of
>     Internet-capable
>     >  devices including low-power, mobile devices as well as devices
>     capable of
>     >  utilizing high quality, high bitrate audio.
>     >
>     >  Proposed Deliverables:
>     >
>     >  1) Requirements for wideband, Internet audio codec(s).
>     >  2) Algorithm description for wideband, Internet audio codec(s)
as
>     Proposed
>     >  Standard.
>     >  3) Specification of payload format(s) for defined codecs as
Proposed
>     >  Standard
>     >
>     >  _______________________________________________
>     >  dispatch mailing list
>     >  dispatch@ietf.org
>     >  https://www.ietf.org/mailman/listinfo/dispatch
>     >
>     >  _______________________________________________
>     >  dispatch mailing list
>     >  dispatch@ietf.org
>     >  https://www.ietf.org/mailman/listinfo/dispatch
>     >
>
>     _______________________________________________
>     dispatch mailing list
>     dispatch@ietf.org
>     https://www.ietf.org/mailman/listinfo/dispatch
>



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<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
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class=3D441190014-02062009>Dear Gonzalo,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
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<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D441190014-02062009>by "<FONT face=3DCalibri color=3D#000000 =
size=3D5>to convince=20
them may be to have IETF people with knowledge in this area volunteer =
to<BR>work=20
on this and review the results." Gonzalo means find IETF voluntiers to =
review=20
the idea of the WG?"</FONT></SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D441190014-02062009><FONT face=3DCalibri color=3D#000000 =
size=3D5></FONT><FONT=20
face=3DCalibri color=3D#000000 size=3D5></FONT><FONT face=3DCalibri =
color=3D#000000=20
size=3D5></FONT><BR>you mean to find the IETF people who will vounteers =
to review=20
the idea of such Wideband WG?</SPAN></FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV align=3Dleft><FONT face=3DArial size=3D2>regards,</FONT></DIV>
<DIV align=3Dleft><FONT face=3DArial size=3D2>Slava Borilin</FONT></DIV>
<DIV align=3Dleft><SPAN lang=3DEN-US=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"></SPAN>&nbsp;</DIV>
<DIV><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SPAN=20
style=3D"FONT-SIZE: 18pt">------ Forwarded Message<BR><B>From: =
</B>Gonzalo=20
Camarillo &lt;<A=20
href=3D"Gonzalo.Camarillo@ericsson.com">Gonzalo.Camarillo@ericsson.com</A=
>&gt;<BR><B>Date:=20
</B>Mon, 1 Jun 2009 21:58:34 -0700<BR><B>To: </B>Adobe Systems &lt;<A=20
href=3D"hsinnrei@adobe.com">hsinnrei@adobe.com</A>&gt;<BR><B>Cc: =
</B>Roni Even=20
&lt;<A href=3D"ron.even.tlv@gmail.com">ron.even.tlv@gmail.com</A>&gt;, =
Jason=20
Fischl &lt;<A =
href=3D"jason.fischl@skype.net">jason.fischl@skype.net</A>&gt;, Mary=20
Barnes &lt;<A =
href=3D"mary.barnes@nortel.com">mary.barnes@nortel.com</A>&gt;,=20
"Peterson, Jon" &lt;<A=20
href=3D"jon.peterson@neustar.biz">jon.peterson@neustar.biz</A>&gt;, =
Cullen=20
Jennings &lt;<A =
href=3D"fluffy@cisco.com">fluffy@cisco.com</A>&gt;<BR><B>Subject:=20
</B>Re: [dispatch] Proposal to form Internet Wideband Audio Codec =
WG<BR><BR>Hi=20
Henry,<BR><BR>given that some people on the list think that the IETF =
does not=20
have<BR>that type of expertise, we need to have those list discussions =
so=20
that<BR>the WG reaches consensus (one way or the other). It seems that =
when=20
iLBC<BR>was developed, it was difficult to find IETF people to review =
it.=20
That<BR>is probably why some people are hesitant about this. A way to=20
convince<BR>them may be to have IETF people with knowledge in this area=20
volunteer to<BR>work on this and review the=20
results.<BR><BR>Thanks,<BR><BR>Gonzalo<BR><BR>Henry Sinnreich =
wrote:<BR>&gt;=20
(list deleted)<BR>&gt;<BR>&gt; Gonzalo,<BR>&gt;<BR>&gt;&gt;whether or =
not the=20
IETF in general and RAI in particular have the<BR>&gt; expertise=20
to<BR>&gt;&gt;do this or at least review it when it is done.=20
Comments?<BR>&gt;<BR>&gt; The folks from Skype driving this proposal =
have=20
without any doubt the<BR>&gt; best qualifications for building voice =
codecs,=20
just as have many other<BR>&gt; IETF AVT contributors, such as for the =
iLBC,=20
SPEEX, IP-MR and other codecs.<BR>&gt;<BR>&gt; The pushback for the =
proposal for=20
a WG for an Internet Audio Codec, is<BR>&gt; quite amazing, especially =
the=20
argument on available expertise.<BR>&gt; (You certainly understand my =
arguments=20
why the better expertise for an<BR>&gt; _Internet_ codec is just in the =
IETF,=20
and certainly not in the ITU-T,<BR>&gt; ETSI, OMA, =
etc.).<BR>&gt;<BR>&gt; I=20
don&#8217;t plan therefore to participate any more in this =
discussion.<BR>&gt;<BR>&gt;=20
Henry<BR>&gt;<BR>&gt;<BR>&gt; On 5/30/09 1:17 AM, "Gonzalo Camarillo" =
&lt;<A=20
href=3D"Gonzalo.Camarillo@ericsson.com">Gonzalo.Camarillo@ericsson.com</A=
>&gt;<BR>&gt;=20
wrote:<BR>&gt;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;Hi,<BR>&gt;<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;I think Roni touches upon an important point =
here. The=20
first question we<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;need to answer (not =
only for=20
this proposal; for any proposal) is whether<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;or=20
not the IETF in general and RAI in particular have the expertise =
to<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;do this or at least review it when it is done.=20
Comments?<BR>&gt;<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;Cheers,<BR>&gt;<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;Gonzalo<BR>&gt;<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;Roni=20
Even wrote:<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Hi,<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Like you mention other SDOs like =
ITU-T are=20
doing just that. They<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;have the<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;expertise to specify, and evaluate =
the=20
result. These SDOs can receive<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
&nbsp;requirements and select a proper codec based on the =
requirements.<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;As for the other reasons:<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;1.=20
Defining a codec in the IETF or even in MPEG / ITU-T does not<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;make it a<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
&nbsp;mandatory part of a system solution, this is done by other<BR>&gt; =

&nbsp;&nbsp;&nbsp;&nbsp;standard bodies<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
&nbsp;like 3GPP, ETSI.<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;2. The IETF, similar to other =
standard bodies=20
is not rubber stamping a<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;specific=20
solution, so you will most probably see in the final<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;result some<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; =

&nbsp;technology that carry IPR.<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;3. If this group will be established, =
you=20
will probably see here<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;the audio<BR>&gt; =

&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;experts now working in ITU-T arguing =
the same=20
issues since they<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;are the<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;expertise you need and they work for =
the same=20
companies that are<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;already<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;members of IETF.<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;I think=20
that if you have a specific codec in mind you &nbsp;can make it<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;publicly<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
&nbsp;available maybe with quality results and standardized in AVT a=20
payload<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;specification.<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;BTW: The=20
ITU is keeping a list of codecs (Not only ITU-T ones) in<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;a table<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;that=20
describes their features.<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Regards<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
&nbsp;Roni Even<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;-----Original Message-----<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;From: <A=20
href=3D"dispatch-bounces@ietf.org">dispatch-bounces@ietf.org</A> [<A=20
href=3D"mailto:dispatch-bounces@ietf.org">mailto:dispatch-bounces@ietf.or=
g</A>]<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;On Behalf<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;Of=20
Jason Fischl<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Sent: Wednesday, =
May 27,=20
2009 2:18 AM<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;To: <A=20
href=3D"dispatch@ietf.org">dispatch@ietf.org</A><BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Subject: [dispatch] Proposal to form =
Internet=20
Wideband Audio Codec WG<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;All,<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;I would=20
like to request agenda time inside the DISPATCH meeting to<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;propose<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;the=20
formation of a new working group to define a Proposed Standard<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;wideband<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
&nbsp;audio codec.<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;The text of the proposal is below. =
Comments,=20
questions, and<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;suggestions<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;welcomed.<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
&nbsp;Regards,<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Jason<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Internet Wideband Audio Codec =
(IWAC)<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Mailing Lists: TBD<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Chairs: TBD<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Area Directorate: Real Time =
Applications=20
(RAI)<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
&nbsp;Purpose:<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;This new working group would be =
chartered=20
with the purpose of<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;collecting<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;expertise within the IETF in order to =
review=20
the design of audio<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;codecs<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;specifically for use with the =
Internet.=20
Unlike other SDOs, these<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;codecs<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;would be optimized for use on the =
Internet,=20
and as much as<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;possible choose<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;technology that does not require =
paying=20
patent royalties.<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;The Internet Low Bit Rate Codec =
(iLBC)=20
&nbsp;work was done in AVT but<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;it was=20
felt<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;that subsequent work =
should not=20
be done in the AVT working group.<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;This=20
new<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;working group will have =
as its=20
primary purpose the standardization<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;of =
a<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;multi-purpose audio codec that can be =
used in=20
various situations<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;on the<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Internet. Some of the proposed=20
Internet-specific requirements include:<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
&nbsp;* scalable and adaptive bit rate;<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
&nbsp;* various sampling rate profiles from narrow-band to=20
super-wideband;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;* scalable=20
complexity;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;* low latency;=20
and<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;* resilience to packet=20
loss.<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
&nbsp;There are a number of wide-band capable codecs defined by =
other<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;SDOs. For<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
&nbsp;instance, G.722 is seeing adoption in Enterprise =
applications<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;since it is<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; =

&nbsp;relatively simple and low-cost to deploy. However, it has a =
high,<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;fixed<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;bitrate=20
and is not appropriate for mobile applications where spectrum<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;efficiency is important or in =
consumer=20
applications where available<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
&nbsp;bandwidth is fluctuating or limited. G.722.2 (AMR-wideband) =
has<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;been adopted<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
&nbsp;by the 3GPP as a wideband standard for mobile applications. =
G.722.2=20
is<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;relatively high cost due =
to patent=20
royalties and is seeing minimal<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
&nbsp;deployments outside of mobile handsets making it challenging to=20
create<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;wideband experiences =
on=20
Internet-capable mobile devices when extending<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;beyond the mobile network. In other =
cases,=20
proprietary codecs are<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;being<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;adopted which further create islands =
with no=20
interoperability unless<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;widespread=20
transcoding is performed. Transcoding leads to higher<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;costs and<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
&nbsp;lower quality.<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;The goal of this working group is to =
define a=20
single codec with<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;multiple<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;profiles which can be made available =
on a=20
wide variety of<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;Internet-capable<BR>&gt; =

&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;devices including low-power, mobile =
devices=20
as well as devices<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;capable of<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;utilizing high quality, high bitrate=20
audio.<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
&nbsp;Proposed Deliverables:<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;1) Requirements for wideband, =
Internet audio=20
codec(s).<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;2) Algorithm =
description=20
for wideband, Internet audio codec(s) as<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;Proposed<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
&nbsp;Standard.<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;3) =
Specification of=20
payload format(s) for defined codecs as Proposed<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Standard<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
&nbsp;_______________________________________________<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;dispatch mailing list<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;<A=20
href=3D"dispatch@ietf.org">dispatch@ietf.org</A><BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;<A=20
href=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf.=
org/mailman/listinfo/dispatch</A><BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
&nbsp;_______________________________________________<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;dispatch mailing list<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;<A=20
href=3D"dispatch@ietf.org">dispatch@ietf.org</A><BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;<A=20
href=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf.=
org/mailman/listinfo/dispatch</A><BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;<BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;_______________________________________________<B=
R>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;dispatch mailing list<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;<A=20
href=3D"dispatch@ietf.org">dispatch@ietf.org</A><BR>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;<A=20
href=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf.=
org/mailman/listinfo/dispatch</A><BR>&gt;<BR><BR><BR><BR>------=20
End of Forwarded Message<BR></SPAN></FONT></DIV></BODY></HTML>

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From jean-marc.valin@octasic.com  Tue Jun  2 08:13:39 2009
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Date: Tue, 02 Jun 2009 11:14:01 -0400
From: Jean-Marc Valin <jean-marc.valin@octasic.com>
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Subject: Re: [dispatch] [AVT] Proposal to form Internet Wideband Audio Codec WG
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Hi,

Roni Even wrote:
>
> I am not sure what prevented you from doing it today at the ITU or 
> MPEG, why do you see the IETF handling it differently.
>
> I would also like to remind you and Jean-Marc that once you are 
> bringing work to a standard body it may require collaboration with 
> other people who will have other proposals that will also address the 
> same requirements and you may need to invest money in comparative 
> testing by independent listening labs.
>
> I also think that you will need to supply the codec in source code and 
> provide copy right to the IETF.
>

I am well aware that bringing work to the IETF would require 
collaboration with others. I am not seeking control over the work I am 
currently doing and would really welcome such collaboration. The idea is 
only to have the best wideband codec possible without IPR issues. Given 
the ITU and MPEG track record, I think it would be very unlikely for any 
of those organisations to work on an IPR-free codec.

I also agree with Henry that "the Internet has different criteria than 
ITU-T networks may have". Internet adoption follows different patterns 
than adoption in the ITU primary target markets. For example, the 
Internet has more consumer-reconfigurable software, while the ITU has to 
deal with a lot of fixed hardware deployments. At the ITU, it makes 
sense to invest large sums of money into testing and characterisation of 
codecs because the codecs deployed there usually stay around for a long 
time and the infrastructure investments are usually very large. On the 
other hand, I would say the investments in codecs for the Internet are 
comparatively smaller and, while testing is still important, it is not 
as critical as it is for the ITU.

It's also a choice one has to make. It is unlikely that companies would 
invest money in expensive testing if they are not going to obtain 
royalties in return. However, I think we may be able to define some more 
lightweight (collaborative?) testing that is sufficient and doesn't 
involve as much investments as what the ITU does. For the Internet, I 
believe an IPR-free codec that everyone agrees performs well is better 
than a patent-encumbered codec that has had more extensive testing. This 
is again another difference with the ITU: patent-encumbered codecs tend 
to hurt a lot more on the Internet because many applications (e.g. 
giving away the client) are very hard (or impossible) when one has to 
pay per-channel royalties.

As for the source code issue you pointed out, all the Xiph codecs are 
already published under a very permissive open source license (BSD), so 
this would not really change.

    Jean-Marc

> Roni
>
>  
>
> *From:* avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] *On Behalf 
> Of *Slava Borilin
> *Sent:* Monday, June 01, 2009 11:50 PM
> *To:* jean-marc.valin@octasic.com
> *Cc:* avt@ietf.org; Jason Fischl
> *Subject:* Re: [AVT] [dispatch] Proposal to form Internet Wideband 
> Audio Codec WG
>
>  
>
> Agree with Jean-Marc.
> SPIRIT is interested to contribute as well - having a dozen of proprietary codecs developed, including 
> one specifically desgined for internet (SPIRIT IP-MR, which is now under WGLC on draft-ietf-avt-rtp-ipmr-04) - 
> multi-rate, scalable, adaptive, wideband codec.
>  
> We can also continue this work with IETF.
>  
> Moreover, most if not all efforts coming from ITU on codecs unfortunately are NOT really focused on 
> internet-specific codecs (that's why several companies have had to invent) - as ITU preference is mainly 
> specific (i.e. cellular) networks at first.
>  
> however, as we see the greant rise of pure "internet-basd communications" (skype, webex/citrix, and many others) - 
> we all (and users) are suffering from inefficiency in all currently "standard" codecs and ambiguity in the choice of 
> internet-targeted ones.
>  
> Again, we probably can put together enough number of contributors to the WG to have the expertise.
>  
> regards,
> Slava Borilin
>  
> --
> John Lazzaro wrote:
>
>     A traditional response to this type of request is to note that the IETF
>
>     really doesn't have much in the way of expertise in audio codec design.
>
>     I don't see many of the regulars who present at the AES codec paper sessions
>
>     posting here on AVT (ditto ICASSP paper sessions for voice codecs). It's
>
>     a full-time job to keep up-to-date and contribute to that
>
>     signal-processing lore.
>
>      
>
> Well, there's no reason that the IETF cannot build such an expertise 
> in audio codecs. This is actually something in which I'd be interested 
> in getting involved and I'm sure others at Xiph.Org would be 
> interested as well. We have several people with audio codec expertise 
> from working on Vorbis, Speex and (more recently) CELT. It turns out 
> that the CELT codec currently under development at Xiph actually meets 
> most of the requirements from the original proposal in being a very 
> low delay codec with adaptive bit-rate and sampling rate (up to 48 
> kHz), scalable complexity, and good robustness to packet loss. We'd be 
> willing to continue the development with the IETF. Even if not with 
> CELT, it's still like to be involved in such a new WG.
>
> Cheers,
>  
>    Jean-Marc


From hsinnrei@adobe.com  Tue Jun  2 09:12:50 2009
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From: Henry Sinnreich <hsinnrei@adobe.com>
To: Roni Even <Even.roni@huawei.com>, Jean-Marc Valin <jean-marc.valin@octasic.com>
Date: Tue, 2 Jun 2009 09:12:26 -0700
Thread-Topic: [AVT] [dispatch] Proposal to form Internet Wideband Audio Codec WG
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Subject: Re: [dispatch] [AVT] Proposal to form Internet Wideband Audio Codec WG
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>This leaves the other reasons I heard for doing it at the IETF which is th=
e
>price of participating

This is an important point.
Given the travel constraints for some of the most valuable potential contri=
butors and reviewers, could we envisage online work until the economy impro=
ves, so that an eventual BOF to start with should not be starved of attende=
es? An online BOF?

There are several free online meeting tools available...

How can this be done within the IETF policy?

Henry


On 6/2/09 10:57 AM, "Roni Even" <Even.roni@huawei.com> wrote:

Hi,
I do not want to sound like someone who is for IPR (I am not), but why stop
at codec, let's require it for all IETF work. There are IPR on IETF work
which is must simpler, in my view, than wide band audio codecs.

I think that we can start with "royalty free" even though I am not sure tha=
t
it will accepted as part of the charter of any other work group so why pick
on codecs which require more work.

This leaves the other reasons I heard for doing it at the IETF which is the
price of participating (cheaper than being an ITU-T member) and maybe desig=
n
less expensive characterization tests.

Roni


-----Original Message-----
From: Jean-Marc Valin [mailto:jean-marc.valin@octasic.com]
Sent: Tuesday, June 02, 2009 6:14 PM
To: Roni Even
Cc: 'Slava Borilin'; avt@ietf.org; 'Jason Fischl'; dispatch@ietf.org;
hsinnrei@adobe.com
Subject: Re: [AVT] [dispatch] Proposal to form Internet Wideband Audio Code=
c
WG

Hi,

Roni Even wrote:
>
> I am not sure what prevented you from doing it today at the ITU or
> MPEG, why do you see the IETF handling it differently.
>
> I would also like to remind you and Jean-Marc that once you are
> bringing work to a standard body it may require collaboration with
> other people who will have other proposals that will also address the
> same requirements and you may need to invest money in comparative
> testing by independent listening labs.
>
> I also think that you will need to supply the codec in source code and
> provide copy right to the IETF.
>

I am well aware that bringing work to the IETF would require
collaboration with others. I am not seeking control over the work I am
currently doing and would really welcome such collaboration. The idea is
only to have the best wideband codec possible without IPR issues. Given
the ITU and MPEG track record, I think it would be very unlikely for any
of those organisations to work on an IPR-free codec.

I also agree with Henry that "the Internet has different criteria than
ITU-T networks may have". Internet adoption follows different patterns
than adoption in the ITU primary target markets. For example, the
Internet has more consumer-reconfigurable software, while the ITU has to
deal with a lot of fixed hardware deployments. At the ITU, it makes
sense to invest large sums of money into testing and characterisation of
codecs because the codecs deployed there usually stay around for a long
time and the infrastructure investments are usually very large. On the
other hand, I would say the investments in codecs for the Internet are
comparatively smaller and, while testing is still important, it is not
as critical as it is for the ITU.

It's also a choice one has to make. It is unlikely that companies would
invest money in expensive testing if they are not going to obtain
royalties in return. However, I think we may be able to define some more
lightweight (collaborative?) testing that is sufficient and doesn't
involve as much investments as what the ITU does. For the Internet, I
believe an IPR-free codec that everyone agrees performs well is better
than a patent-encumbered codec that has had more extensive testing. This
is again another difference with the ITU: patent-encumbered codecs tend
to hurt a lot more on the Internet because many applications (e.g.
giving away the client) are very hard (or impossible) when one has to
pay per-channel royalties.

As for the source code issue you pointed out, all the Xiph codecs are
already published under a very permissive open source license (BSD), so
this would not really change.

    Jean-Marc

> Roni
>
>
>
> *From:* avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] *On Behalf
> Of *Slava Borilin
> *Sent:* Monday, June 01, 2009 11:50 PM
> *To:* jean-marc.valin@octasic.com
> *Cc:* avt@ietf.org; Jason Fischl
> *Subject:* Re: [AVT] [dispatch] Proposal to form Internet Wideband
> Audio Codec WG
>
>
>
> Agree with Jean-Marc.
> SPIRIT is interested to contribute as well - having a dozen of proprietar=
y
codecs developed, including
> one specifically desgined for internet (SPIRIT IP-MR, which is now under
WGLC on draft-ietf-avt-rtp-ipmr-04) -
> multi-rate, scalable, adaptive, wideband codec.
>
> We can also continue this work with IETF.
>
> Moreover, most if not all efforts coming from ITU on codecs unfortunately
are NOT really focused on
> internet-specific codecs (that's why several companies have had to invent=
)
- as ITU preference is mainly
> specific (i.e. cellular) networks at first.
>
> however, as we see the greant rise of pure "internet-basd communications"
(skype, webex/citrix, and many others) -
> we all (and users) are suffering from inefficiency in all currently
"standard" codecs and ambiguity in the choice of
> internet-targeted ones.
>
> Again, we probably can put together enough number of contributors to the
WG to have the expertise.
>
> regards,
> Slava Borilin
>
> --
> John Lazzaro wrote:
>
>     A traditional response to this type of request is to note that the
IETF
>
>     really doesn't have much in the way of expertise in audio codec
design.
>
>     I don't see many of the regulars who present at the AES codec paper
sessions
>
>     posting here on AVT (ditto ICASSP paper sessions for voice codecs).
It's
>
>     a full-time job to keep up-to-date and contribute to that
>
>     signal-processing lore.
>
>
>
> Well, there's no reason that the IETF cannot build such an expertise
> in audio codecs. This is actually something in which I'd be interested
> in getting involved and I'm sure others at Xiph.Org would be
> interested as well. We have several people with audio codec expertise
> from working on Vorbis, Speex and (more recently) CELT. It turns out
> that the CELT codec currently under development at Xiph actually meets
> most of the requirements from the original proposal in being a very
> low delay codec with adaptive bit-rate and sampling rate (up to 48
> kHz), scalable complexity, and good robustness to packet loss. We'd be
> willing to continue the development with the IETF. Even if not with
> CELT, it's still like to be involved in such a new WG.
>
> Cheers,
>
>    Jean-Marc



--_000_C64AB99A3E55hsinnreiadobecom_
Content-Type: text/html; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<HTML>
<HEAD>
<TITLE>Re: [AVT] [dispatch] Proposal to form Internet Wideband Audio Codec =
WG</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
18pt'>&gt;This leaves the other reasons I heard for doing it at the IETF wh=
ich is the <BR>
&gt;price of participating <BR>
<BR>
This is an important point.<BR>
Given the travel constraints for some of the most valuable potential contri=
butors and reviewers, could we envisage online work until the economy impro=
ves, so that an eventual BOF to start with should not be starved of attende=
es? An online BOF?<BR>
<BR>
There are several free online meeting tools available...<BR>
<BR>
How can this be done within the IETF policy?<BR>
<BR>
Henry<BR>
<BR>
<BR>
On 6/2/09 10:57 AM, &quot;Roni Even&quot; &lt;<a href=3D"Even.roni@huawei.c=
om">Even.roni@huawei.com</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:18pt'>Hi,<BR>
I do not want to sound like someone who is for IPR (I am not), but why stop=
<BR>
at codec, let's require it for all IETF work. There are IPR on IETF work<BR=
>
which is must simpler, in my view, than wide band audio codecs.<BR>
<BR>
I think that we can start with &quot;royalty free&quot; even though I am no=
t sure that<BR>
it will accepted as part of the charter of any other work group so why pick=
<BR>
on codecs which require more work.<BR>
<BR>
This leaves the other reasons I heard for doing it at the IETF which is the=
<BR>
price of participating (cheaper than being an ITU-T member) and maybe desig=
n<BR>
less expensive characterization tests.<BR>
<BR>
Roni<BR>
<BR>
<BR>
-----Original Message-----<BR>
From: Jean-Marc Valin [<a href=3D"mailto:jean-marc.valin@octasic.com">mailt=
o:jean-marc.valin@octasic.com</a>]<BR>
Sent: Tuesday, June 02, 2009 6:14 PM<BR>
To: Roni Even<BR>
Cc: 'Slava Borilin'; <a href=3D"avt@ietf.org">avt@ietf.org</a>; 'Jason Fisc=
hl'; <a href=3D"dispatch@ietf.org">dispatch@ietf.org</a>;<BR>
<a href=3D"hsinnrei@adobe.com">hsinnrei@adobe.com</a><BR>
Subject: Re: [AVT] [dispatch] Proposal to form Internet Wideband Audio Code=
c<BR>
WG<BR>
<BR>
Hi,<BR>
<BR>
Roni Even wrote:<BR>
&gt;<BR>
&gt; I am not sure what prevented you from doing it today at the ITU or<BR>
&gt; MPEG, why do you see the IETF handling it differently.<BR>
&gt;<BR>
&gt; I would also like to remind you and Jean-Marc that once you are<BR>
&gt; bringing work to a standard body it may require collaboration with<BR>
&gt; other people who will have other proposals that will also address the<=
BR>
&gt; same requirements and you may need to invest money in comparative<BR>
&gt; testing by independent listening labs.<BR>
&gt;<BR>
&gt; I also think that you will need to supply the codec in source code and=
<BR>
&gt; provide copy right to the IETF.<BR>
&gt;<BR>
<BR>
I am well aware that bringing work to the IETF would require<BR>
collaboration with others. I am not seeking control over the work I am<BR>
currently doing and would really welcome such collaboration. The idea is<BR=
>
only to have the best wideband codec possible without IPR issues. Given<BR>
the ITU and MPEG track record, I think it would be very unlikely for any<BR=
>
of those organisations to work on an IPR-free codec.<BR>
<BR>
I also agree with Henry that &quot;the Internet has different criteria than=
<BR>
ITU-T networks may have&quot;. Internet adoption follows different patterns=
<BR>
than adoption in the ITU primary target markets. For example, the<BR>
Internet has more consumer-reconfigurable software, while the ITU has to<BR=
>
deal with a lot of fixed hardware deployments. At the ITU, it makes<BR>
sense to invest large sums of money into testing and characterisation of<BR=
>
codecs because the codecs deployed there usually stay around for a long<BR>
time and the infrastructure investments are usually very large. On the<BR>
other hand, I would say the investments in codecs for the Internet are<BR>
comparatively smaller and, while testing is still important, it is not<BR>
as critical as it is for the ITU.<BR>
<BR>
It's also a choice one has to make. It is unlikely that companies would<BR>
invest money in expensive testing if they are not going to obtain<BR>
royalties in return. However, I think we may be able to define some more<BR=
>
lightweight (collaborative?) testing that is sufficient and doesn't<BR>
involve as much investments as what the ITU does. For the Internet, I<BR>
believe an IPR-free codec that everyone agrees performs well is better<BR>
than a patent-encumbered codec that has had more extensive testing. This<BR=
>
is again another difference with the ITU: patent-encumbered codecs tend<BR>
to hurt a lot more on the Internet because many applications (e.g.<BR>
giving away the client) are very hard (or impossible) when one has to<BR>
pay per-channel royalties.<BR>
<BR>
As for the source code issue you pointed out, all the Xiph codecs are<BR>
already published under a very permissive open source license (BSD), so<BR>
this would not really change.<BR>
<BR>
&nbsp;&nbsp;&nbsp;&nbsp;Jean-Marc<BR>
<BR>
&gt; Roni<BR>
&gt;<BR>
&gt; <BR>
&gt;<BR>
&gt; *From:* <a href=3D"avt-bounces@ietf.org">avt-bounces@ietf.org</a> [<a =
href=3D"mailto:avt-bounces@ietf.org">mailto:avt-bounces@ietf.org</a>] *On B=
ehalf<BR>
&gt; Of *Slava Borilin<BR>
&gt; *Sent:* Monday, June 01, 2009 11:50 PM<BR>
&gt; *To:* <a href=3D"jean-marc.valin@octasic.com">jean-marc.valin@octasic.=
com</a><BR>
&gt; *Cc:* <a href=3D"avt@ietf.org">avt@ietf.org</a>; Jason Fischl<BR>
&gt; *Subject:* Re: [AVT] [dispatch] Proposal to form Internet Wideband<BR>
&gt; Audio Codec WG<BR>
&gt;<BR>
&gt; <BR>
&gt;<BR>
&gt; Agree with Jean-Marc.<BR>
&gt; SPIRIT is interested to contribute as well - having a dozen of proprie=
tary<BR>
codecs developed, including<BR>
&gt; one specifically desgined for internet (SPIRIT IP-MR, which is now und=
er<BR>
WGLC on draft-ietf-avt-rtp-ipmr-04) -<BR>
&gt; multi-rate, scalable, adaptive, wideband codec.<BR>
&gt; <BR>
&gt; We can also continue this work with IETF.<BR>
&gt; <BR>
&gt; Moreover, most if not all efforts coming from ITU on codecs unfortunat=
ely<BR>
are NOT really focused on<BR>
&gt; internet-specific codecs (that's why several companies have had to inv=
ent)<BR>
- as ITU preference is mainly<BR>
&gt; specific (i.e. cellular) networks at first.<BR>
&gt; <BR>
&gt; however, as we see the greant rise of pure &quot;internet-basd communi=
cations&quot;<BR>
(skype, webex/citrix, and many others) -<BR>
&gt; we all (and users) are suffering from inefficiency in all currently<BR=
>
&quot;standard&quot; codecs and ambiguity in the choice of<BR>
&gt; internet-targeted ones.<BR>
&gt; <BR>
&gt; Again, we probably can put together enough number of contributors to t=
he<BR>
WG to have the expertise.<BR>
&gt; <BR>
&gt; regards,<BR>
&gt; Slava Borilin<BR>
&gt; <BR>
&gt; --<BR>
&gt; John Lazzaro wrote:<BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;A traditional response to this type of request=
 is to note that the<BR>
IETF<BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;really doesn't have much in the way of experti=
se in audio codec<BR>
design.<BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;I don't see many of the regulars who present a=
t the AES codec paper<BR>
sessions<BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;posting here on AVT (ditto ICASSP paper sessio=
ns for voice codecs).<BR>
It's<BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;a full-time job to keep up-to-date and contrib=
ute to that<BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;signal-processing lore.<BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;<BR>
&gt;<BR>
&gt; Well, there's no reason that the IETF cannot build such an expertise<B=
R>
&gt; in audio codecs. This is actually something in which I'd be interested=
<BR>
&gt; in getting involved and I'm sure others at Xiph.Org would be<BR>
&gt; interested as well. We have several people with audio codec expertise<=
BR>
&gt; from working on Vorbis, Speex and (more recently) CELT. It turns out<B=
R>
&gt; that the CELT codec currently under development at Xiph actually meets=
<BR>
&gt; most of the requirements from the original proposal in being a very<BR=
>
&gt; low delay codec with adaptive bit-rate and sampling rate (up to 48<BR>
&gt; kHz), scalable complexity, and good robustness to packet loss. We'd be=
<BR>
&gt; willing to continue the development with the IETF. Even if not with<BR=
>
&gt; CELT, it's still like to be involved in such a new WG.<BR>
&gt;<BR>
&gt; Cheers,<BR>
&gt; <BR>
&gt; &nbsp;&nbsp;&nbsp;Jean-Marc<BR>
<BR>
<BR>
</SPAN></FONT></BLOCKQUOTE>
</BODY>
</HTML>


--_000_C64AB99A3E55hsinnreiadobecom_--

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Date: Tue, 02 Jun 2009 18:57:16 +0300
From: Roni Even <Even.roni@huawei.com>
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Subject: Re: [dispatch] [AVT] Proposal to form Internet Wideband Audio Codec WG
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Hi,
I do not want to sound like someone who is for IPR (I am not), but why stop
at codec, let's require it for all IETF work. There are IPR on IETF work
which is must simpler, in my view, than wide band audio codecs.

I think that we can start with "royalty free" even though I am not sure that
it will accepted as part of the charter of any other work group so why pick
on codecs which require more work.

This leaves the other reasons I heard for doing it at the IETF which is the
price of participating (cheaper than being an ITU-T member) and maybe design
less expensive characterization tests.

Roni


-----Original Message-----
From: Jean-Marc Valin [mailto:jean-marc.valin@octasic.com] 
Sent: Tuesday, June 02, 2009 6:14 PM
To: Roni Even
Cc: 'Slava Borilin'; avt@ietf.org; 'Jason Fischl'; dispatch@ietf.org;
hsinnrei@adobe.com
Subject: Re: [AVT] [dispatch] Proposal to form Internet Wideband Audio Codec
WG

Hi,

Roni Even wrote:
>
> I am not sure what prevented you from doing it today at the ITU or 
> MPEG, why do you see the IETF handling it differently.
>
> I would also like to remind you and Jean-Marc that once you are 
> bringing work to a standard body it may require collaboration with 
> other people who will have other proposals that will also address the 
> same requirements and you may need to invest money in comparative 
> testing by independent listening labs.
>
> I also think that you will need to supply the codec in source code and 
> provide copy right to the IETF.
>

I am well aware that bringing work to the IETF would require 
collaboration with others. I am not seeking control over the work I am 
currently doing and would really welcome such collaboration. The idea is 
only to have the best wideband codec possible without IPR issues. Given 
the ITU and MPEG track record, I think it would be very unlikely for any 
of those organisations to work on an IPR-free codec.

I also agree with Henry that "the Internet has different criteria than 
ITU-T networks may have". Internet adoption follows different patterns 
than adoption in the ITU primary target markets. For example, the 
Internet has more consumer-reconfigurable software, while the ITU has to 
deal with a lot of fixed hardware deployments. At the ITU, it makes 
sense to invest large sums of money into testing and characterisation of 
codecs because the codecs deployed there usually stay around for a long 
time and the infrastructure investments are usually very large. On the 
other hand, I would say the investments in codecs for the Internet are 
comparatively smaller and, while testing is still important, it is not 
as critical as it is for the ITU.

It's also a choice one has to make. It is unlikely that companies would 
invest money in expensive testing if they are not going to obtain 
royalties in return. However, I think we may be able to define some more 
lightweight (collaborative?) testing that is sufficient and doesn't 
involve as much investments as what the ITU does. For the Internet, I 
believe an IPR-free codec that everyone agrees performs well is better 
than a patent-encumbered codec that has had more extensive testing. This 
is again another difference with the ITU: patent-encumbered codecs tend 
to hurt a lot more on the Internet because many applications (e.g. 
giving away the client) are very hard (or impossible) when one has to 
pay per-channel royalties.

As for the source code issue you pointed out, all the Xiph codecs are 
already published under a very permissive open source license (BSD), so 
this would not really change.

    Jean-Marc

> Roni
>
>  
>
> *From:* avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] *On Behalf 
> Of *Slava Borilin
> *Sent:* Monday, June 01, 2009 11:50 PM
> *To:* jean-marc.valin@octasic.com
> *Cc:* avt@ietf.org; Jason Fischl
> *Subject:* Re: [AVT] [dispatch] Proposal to form Internet Wideband 
> Audio Codec WG
>
>  
>
> Agree with Jean-Marc.
> SPIRIT is interested to contribute as well - having a dozen of proprietary
codecs developed, including 
> one specifically desgined for internet (SPIRIT IP-MR, which is now under
WGLC on draft-ietf-avt-rtp-ipmr-04) - 
> multi-rate, scalable, adaptive, wideband codec.
>  
> We can also continue this work with IETF.
>  
> Moreover, most if not all efforts coming from ITU on codecs unfortunately
are NOT really focused on 
> internet-specific codecs (that's why several companies have had to invent)
- as ITU preference is mainly 
> specific (i.e. cellular) networks at first.
>  
> however, as we see the greant rise of pure "internet-basd communications"
(skype, webex/citrix, and many others) - 
> we all (and users) are suffering from inefficiency in all currently
"standard" codecs and ambiguity in the choice of 
> internet-targeted ones.
>  
> Again, we probably can put together enough number of contributors to the
WG to have the expertise.
>  
> regards,
> Slava Borilin
>  
> --
> John Lazzaro wrote:
>
>     A traditional response to this type of request is to note that the
IETF
>
>     really doesn't have much in the way of expertise in audio codec
design.
>
>     I don't see many of the regulars who present at the AES codec paper
sessions
>
>     posting here on AVT (ditto ICASSP paper sessions for voice codecs).
It's
>
>     a full-time job to keep up-to-date and contribute to that
>
>     signal-processing lore.
>
>      
>
> Well, there's no reason that the IETF cannot build such an expertise 
> in audio codecs. This is actually something in which I'd be interested 
> in getting involved and I'm sure others at Xiph.Org would be 
> interested as well. We have several people with audio codec expertise 
> from working on Vorbis, Speex and (more recently) CELT. It turns out 
> that the CELT codec currently under development at Xiph actually meets 
> most of the requirements from the original proposal in being a very 
> low delay codec with adaptive bit-rate and sampling rate (up to 48 
> kHz), scalable complexity, and good robustness to packet loss. We'd be 
> willing to continue the development with the IETF. Even if not with 
> CELT, it's still like to be involved in such a new WG.
>
> Cheers,
>  
>    Jean-Marc


From hgs@cs.columbia.edu  Tue Jun  2 09:16:31 2009
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Subject: Re: [dispatch] [AVT] Proposal to form Internet Wideband Audio Codec WG
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I view this as a trade-off. If we pursue this, there are risks:

- nothing may come of it since there are no experts willing to help
- somebody will claim IPR on the resulting work
- the quality of the codec may not be competitive

However, if we don't do this, we are stuck with the status quo, which  
is not all that satisfactory. Thus, unless there are significant costs  
for "innocent bystanders", I see this as a risk worth taking. In the  
worst case, we are no worse off than we are today. In all other cases,  
we'll have an additional choice for a wideband codec, even if it's not  
"the best", just "good enough". After all, most people use G.711  
today, which has a really hard time making that claim.

Most real work in the IETF is done by very small teams, typically less  
than 10, so as long as we have a handful of people that are willing to  
contribute, this can work. It might even work better, since you may  
get fewer people who have half-baked opinions - we may skip the binary  
vs. XML debates...

We can set some ground rules ("must be tested with packet loss of 5%")  
and then see what happens. Compared to most network protocols, the  
likely negative impacts (such as security or congestion control  
issues) of even a badly-designed codec are pretty limited.

Henning

On Jun 2, 2009, at 11:57 AM, Roni Even wrote:

> Hi,
> I do not want to sound like someone who is for IPR (I am not), but  
> why stop
> at codec, let's require it for all IETF work. There are IPR on IETF  
> work
> which is must simpler, in my view, than wide band audio codecs.
>
> I think that we can start with "royalty free" even though I am not  
> sure that
> it will accepted as part of the charter of any other work group so  
> why pick
> on codecs which require more work.
>
> This leaves the other reasons I heard for doing it at the IETF which  
> is the
> price of participating (cheaper than being an ITU-T member) and  
> maybe design
> less expensive characterization tests.
>
> Roni
>
>
> -----Original Message-----
> From: Jean-Marc Valin [mailto:jean-marc.valin@octasic.com]
> Sent: Tuesday, June 02, 2009 6:14 PM
> To: Roni Even
> Cc: 'Slava Borilin'; avt@ietf.org; 'Jason Fischl'; dispatch@ietf.org;
> hsinnrei@adobe.com
> Subject: Re: [AVT] [dispatch] Proposal to form Internet Wideband  
> Audio Codec
> WG
>
> Hi,
>
> Roni Even wrote:
>>
>> I am not sure what prevented you from doing it today at the ITU or
>> MPEG, why do you see the IETF handling it differently.
>>
>> I would also like to remind you and Jean-Marc that once you are
>> bringing work to a standard body it may require collaboration with
>> other people who will have other proposals that will also address the
>> same requirements and you may need to invest money in comparative
>> testing by independent listening labs.
>>
>> I also think that you will need to supply the codec in source code  
>> and
>> provide copy right to the IETF.
>>
>
> I am well aware that bringing work to the IETF would require
> collaboration with others. I am not seeking control over the work I am
> currently doing and would really welcome such collaboration. The  
> idea is
> only to have the best wideband codec possible without IPR issues.  
> Given
> the ITU and MPEG track record, I think it would be very unlikely for  
> any
> of those organisations to work on an IPR-free codec.
>
> I also agree with Henry that "the Internet has different criteria than
> ITU-T networks may have". Internet adoption follows different patterns
> than adoption in the ITU primary target markets. For example, the
> Internet has more consumer-reconfigurable software, while the ITU  
> has to
> deal with a lot of fixed hardware deployments. At the ITU, it makes
> sense to invest large sums of money into testing and  
> characterisation of
> codecs because the codecs deployed there usually stay around for a  
> long
> time and the infrastructure investments are usually very large. On the
> other hand, I would say the investments in codecs for the Internet are
> comparatively smaller and, while testing is still important, it is not
> as critical as it is for the ITU.
>
> It's also a choice one has to make. It is unlikely that companies  
> would
> invest money in expensive testing if they are not going to obtain
> royalties in return. However, I think we may be able to define some  
> more
> lightweight (collaborative?) testing that is sufficient and doesn't
> involve as much investments as what the ITU does. For the Internet, I
> believe an IPR-free codec that everyone agrees performs well is better
> than a patent-encumbered codec that has had more extensive testing.  
> This
> is again another difference with the ITU: patent-encumbered codecs  
> tend
> to hurt a lot more on the Internet because many applications (e.g.
> giving away the client) are very hard (or impossible) when one has to
> pay per-channel royalties.
>
> As for the source code issue you pointed out, all the Xiph codecs are
> already published under a very permissive open source license (BSD),  
> so
> this would not really change.
>
>    Jean-Marc
>
>> Roni
>>
>>
>>
>> *From:* avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] *On Behalf
>> Of *Slava Borilin
>> *Sent:* Monday, June 01, 2009 11:50 PM
>> *To:* jean-marc.valin@octasic.com
>> *Cc:* avt@ietf.org; Jason Fischl
>> *Subject:* Re: [AVT] [dispatch] Proposal to form Internet Wideband
>> Audio Codec WG
>>
>>
>>
>> Agree with Jean-Marc.
>> SPIRIT is interested to contribute as well - having a dozen of  
>> proprietary
> codecs developed, including
>> one specifically desgined for internet (SPIRIT IP-MR, which is now  
>> under
> WGLC on draft-ietf-avt-rtp-ipmr-04) -
>> multi-rate, scalable, adaptive, wideband codec.
>>
>> We can also continue this work with IETF.
>>
>> Moreover, most if not all efforts coming from ITU on codecs  
>> unfortunately
> are NOT really focused on
>> internet-specific codecs (that's why several companies have had to  
>> invent)
> - as ITU preference is mainly
>> specific (i.e. cellular) networks at first.
>>
>> however, as we see the greant rise of pure "internet-basd  
>> communications"
> (skype, webex/citrix, and many others) -
>> we all (and users) are suffering from inefficiency in all currently
> "standard" codecs and ambiguity in the choice of
>> internet-targeted ones.
>>
>> Again, we probably can put together enough number of contributors  
>> to the
> WG to have the expertise.
>>
>> regards,
>> Slava Borilin
>>
>> --
>> John Lazzaro wrote:
>>
>>    A traditional response to this type of request is to note that the
> IETF
>>
>>    really doesn't have much in the way of expertise in audio codec
> design.
>>
>>    I don't see many of the regulars who present at the AES codec  
>> paper
> sessions
>>
>>    posting here on AVT (ditto ICASSP paper sessions for voice  
>> codecs).
> It's
>>
>>    a full-time job to keep up-to-date and contribute to that
>>
>>    signal-processing lore.
>>
>>
>>
>> Well, there's no reason that the IETF cannot build such an expertise
>> in audio codecs. This is actually something in which I'd be  
>> interested
>> in getting involved and I'm sure others at Xiph.Org would be
>> interested as well. We have several people with audio codec expertise
>> from working on Vorbis, Speex and (more recently) CELT. It turns out
>> that the CELT codec currently under development at Xiph actually  
>> meets
>> most of the requirements from the original proposal in being a very
>> low delay codec with adaptive bit-rate and sampling rate (up to 48
>> kHz), scalable complexity, and good robustness to packet loss. We'd  
>> be
>> willing to continue the development with the IETF. Even if not with
>> CELT, it's still like to be involved in such a new WG.
>>
>> Cheers,
>>
>>   Jean-Marc
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
>


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Subject: Re: [dispatch] [AVT] Proposal to form Internet Wideband Audio Codec WG
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At 11:12 AM 6/2/2009, Henry Sinnreich wrote:
>Content-Language: en
>Content-Type: multipart/alternative;
>         boundary="_000_C64AB99A3E55hsinnreiadobecom_"
>
> >This leaves the other reasons I heard for doing it at the IETF which is the
> >price of participating
>
>This is an important point.
>Given the travel constraints for some of the most valuable potential 
>contributors and reviewers, could we envisage online work until the 
>economy improves, so that an eventual BOF to start with should not 
>be starved of attendees?

Henry

this economic constraint will exist no matter which SDO this work 
eventually gets done in.

>An online BOF?
>
>There are several free online meeting tools available...
>
>How can this be done within the IETF policy?
>
>Henry
>
>
>On 6/2/09 10:57 AM, "Roni Even" 
><<Even.roni@huawei.htm>Even.roni@huawei.com> wrote:
>
>Hi,
>I do not want to sound like someone who is for IPR (I am not), but why stop
>at codec, let's require it for all IETF work. There are IPR on IETF work
>which is must simpler, in my view, than wide band audio codecs.
>
>I think that we can start with "royalty free" even though I am not sure that
>it will accepted as part of the charter of any other work group so why pick
>on codecs which require more work.
>
>This leaves the other reasons I heard for doing it at the IETF which is the
>price of participating (cheaper than being an ITU-T member) and maybe design
>less expensive characterization tests.
>
>Roni
>
>
>-----Original Message-----
>From: Jean-Marc Valin 
>[<mailto:jean-marc.valin@octasic.com>mailto:jean-marc.valin@octasic.com]
>Sent: Tuesday, June 02, 2009 6:14 PM
>To: Roni Even
>Cc: 'Slava Borilin'; <avt@ietf.htm>avt@ietf.org; 'Jason Fischl'; 
><dispatch@ietf.htm>dispatch@ietf.org;
><hsinnrei@adobe.htm>hsinnrei@adobe.com
>Subject: Re: [AVT] [dispatch] Proposal to form Internet Wideband Audio Codec
>WG
>
>Hi,
>
>Roni Even wrote:
> >
> > I am not sure what prevented you from doing it today at the ITU or
> > MPEG, why do you see the IETF handling it differently.
> >
> > I would also like to remind you and Jean-Marc that once you are
> > bringing work to a standard body it may require collaboration with
> > other people who will have other proposals that will also address the
> > same requirements and you may need to invest money in comparative
> > testing by independent listening labs.
> >
> > I also think that you will need to supply the codec in source code and
> > provide copy right to the IETF.
> >
>
>I am well aware that bringing work to the IETF would require
>collaboration with others. I am not seeking control over the work I am
>currently doing and would really welcome such collaboration. The idea is
>only to have the best wideband codec possible without IPR issues. Given
>the ITU and MPEG track record, I think it would be very unlikely for any
>of those organisations to work on an IPR-free codec.
>
>I also agree with Henry that "the Internet has different criteria than
>ITU-T networks may have". Internet adoption follows different patterns
>than adoption in the ITU primary target markets. For example, the
>Internet has more consumer-reconfigurable software, while the ITU has to
>deal with a lot of fixed hardware deployments. At the ITU, it makes
>sense to invest large sums of money into testing and characterisation of
>codecs because the codecs deployed there usually stay around for a long
>time and the infrastructure investments are usually very large. On the
>other hand, I would say the investments in codecs for the Internet are
>comparatively smaller and, while testing is still important, it is not
>as critical as it is for the ITU.
>
>It's also a choice one has to make. It is unlikely that companies would
>invest money in expensive testing if they are not going to obtain
>royalties in return. However, I think we may be able to define some more
>lightweight (collaborative?) testing that is sufficient and doesn't
>involve as much investments as what the ITU does. For the Internet, I
>believe an IPR-free codec that everyone agrees performs well is better
>than a patent-encumbered codec that has had more extensive testing. This
>is again another difference with the ITU: patent-encumbered codecs tend
>to hurt a lot more on the Internet because many applications (e.g.
>giving away the client) are very hard (or impossible) when one has to
>pay per-channel royalties.
>
>As for the source code issue you pointed out, all the Xiph codecs are
>already published under a very permissive open source license (BSD), so
>this would not really change.
>
>     Jean-Marc
>
> > Roni
> >
> >
> >
> > *From:* <avt-bounces@ietf.htm>avt-bounces@ietf.org 
> [mailto:avt-bounces@ietf.org] *On Behalf
> > Of *Slava Borilin
> > *Sent:* Monday, June 01, 2009 11:50 PM
> > *To:* <jean-marc.valin@octasic.htm>jean-marc.valin@octasic.com
> > *Cc:* <avt@ietf.htm>avt@ietf.org; Jason Fischl
> > *Subject:* Re: [AVT] [dispatch] Proposal to form Internet Wideband
> > Audio Codec WG
> >
> >
> >
> > Agree with Jean-Marc.
> > SPIRIT is interested to contribute as well - having a dozen of proprietary
>codecs developed, including
> > one specifically desgined for internet (SPIRIT IP-MR, which is now under
>WGLC on draft-ietf-avt-rtp-ipmr-04) -
> > multi-rate, scalable, adaptive, wideband codec.
> >
> > We can also continue this work with IETF.
> >
> > Moreover, most if not all efforts coming from ITU on codecs unfortunately
>are NOT really focused on
> > internet-specific codecs (that's why several companies have had to invent)
>- as ITU preference is mainly
> > specific (i.e. cellular) networks at first.
> >
> > however, as we see the greant rise of pure "internet-basd communications"
>(skype, webex/citrix, and many others) -
> > we all (and users) are suffering from inefficiency in all currently
>"standard" codecs and ambiguity in the choice of
> > internet-targeted ones.
> >
> > Again, we probably can put together enough number of contributors to the
>WG to have the expertise.
> >
> > regards,
> > Slava Borilin
> >
> > --
> > John Lazzaro wrote:
> >
> >     A traditional response to this type of request is to note that the
>IETF
> >
> >     really doesn't have much in the way of expertise in audio codec
>design.
> >
> >     I don't see many of the regulars who present at the AES codec paper
>sessions
> >
> >     posting here on AVT (ditto ICASSP paper sessions for voice codecs).
>It's
> >
> >     a full-time job to keep up-to-date and contribute to that
> >
> >     signal-processing lore.
> >
> >
> >
> > Well, there's no reason that the IETF cannot build such an expertise
> > in audio codecs. This is actually something in which I'd be interested
> > in getting involved and I'm sure others at Xiph.Org would be
> > interested as well. We have several people with audio codec expertise
> > from working on Vorbis, Speex and (more recently) CELT. It turns out
> > that the CELT codec currently under development at Xiph actually meets
> > most of the requirements from the original proposal in being a very
> > low delay codec with adaptive bit-rate and sampling rate (up to 48
> > kHz), scalable complexity, and good robustness to packet loss. We'd be
> > willing to continue the development with the IETF. Even if not with
> > CELT, it's still like to be involved in such a new WG.
> >
> > Cheers,
> >
> >    Jean-Marc
>
>
>_______________________________________________
>dispatch mailing list
>dispatch@ietf.org
>https://www.ietf.org/mailman/listinfo/dispatch


From spencer@wonderhamster.org  Tue Jun  2 09:46:32 2009
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From: "Spencer Dawkins" <spencer@wonderhamster.org>
To: "Henry Sinnreich" <hsinnrei@adobe.com>, "Roni Even" <Even.roni@huawei.com>, "Jean-Marc Valin" <jean-marc.valin@octasic.com>
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Date: Tue, 2 Jun 2009 11:44:51 -0500
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Cc: dispatch@ietf.org, Slava Borilin <Borilin@spiritdsp.com>, avt@ietf.org
Subject: Re: [dispatch] [AVT] Proposal to form Internet Wideband Audio Codec WG
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Re: [AVT] [dispatch] Proposal to form Internet Wideband Audio Codec WGI =
am not an area director, nor do i play one on IPTV, but my suggestion =
for anyone interested would be to=20

- request a non-WG mailing list (from either of the RAI area directors),

- announce it on this mailing list (and others that seem to be =
appropriate, and=20

- start discussions, which the area directors typically look at when =
judging whether there's enough interest and enough clue to justify =
approving a BOF and/or chartering a working group.

I didn't see a mailing list included in Jason's proposal - if I missed =
one, my apologies.

If people at the IETF can do this work, and want to do this work, IETF =
process policies should not get in the way of that happening.

And I haven't heard of an AD turning down a request for a non-WG mailing =
list yet, keeping in mind that one of said lists is ietf-sailors, for =
people interested in sailing to/from/at IETF meetings - the bar is =
(appropriately) low.

Thanks,

Spencer
  ----- Original Message -----=20
  From: Henry Sinnreich=20
  To: Roni Even ; Jean-Marc Valin=20
  Cc: dispatch@ietf.org ; Slava Borilin ; avt@ietf.org=20
  Sent: Tuesday, June 02, 2009 11:12 AM
  Subject: Re: [dispatch] [AVT] Proposal to form Internet Wideband Audio =
Codec WG


  >This leaves the other reasons I heard for doing it at the IETF which =
is the=20
  >price of participating=20

  This is an important point.
  Given the travel constraints for some of the most valuable potential =
contributors and reviewers, could we envisage online work until the =
economy improves, so that an eventual BOF to start with should not be =
starved of attendees? An online BOF?

  There are several free online meeting tools available...

  How can this be done within the IETF policy?

  Henry


  On 6/2/09 10:57 AM, "Roni Even" <Even.roni@huawei.com> wrote:


    Hi,
    I do not want to sound like someone who is for IPR (I am not), but =
why stop
    at codec, let's require it for all IETF work. There are IPR on IETF =
work
    which is must simpler, in my view, than wide band audio codecs.

    I think that we can start with "royalty free" even though I am not =
sure that
    it will accepted as part of the charter of any other work group so =
why pick
    on codecs which require more work.

    This leaves the other reasons I heard for doing it at the IETF which =
is the
    price of participating (cheaper than being an ITU-T member) and =
maybe design
    less expensive characterization tests.

    Roni


    -----Original Message-----
    From: Jean-Marc Valin [mailto:jean-marc.valin@octasic.com]
    Sent: Tuesday, June 02, 2009 6:14 PM
    To: Roni Even
    Cc: 'Slava Borilin'; avt@ietf.org; 'Jason Fischl'; =
dispatch@ietf.org;
    hsinnrei@adobe.com
    Subject: Re: [AVT] [dispatch] Proposal to form Internet Wideband =
Audio Codec
    WG

    Hi,

    Roni Even wrote:
    >
    > I am not sure what prevented you from doing it today at the ITU or
    > MPEG, why do you see the IETF handling it differently.
    >
    > I would also like to remind you and Jean-Marc that once you are
    > bringing work to a standard body it may require collaboration with
    > other people who will have other proposals that will also address =
the
    > same requirements and you may need to invest money in comparative
    > testing by independent listening labs.
    >
    > I also think that you will need to supply the codec in source code =
and
    > provide copy right to the IETF.
    >

    I am well aware that bringing work to the IETF would require
    collaboration with others. I am not seeking control over the work I =
am
    currently doing and would really welcome such collaboration. The =
idea is
    only to have the best wideband codec possible without IPR issues. =
Given
    the ITU and MPEG track record, I think it would be very unlikely for =
any
    of those organisations to work on an IPR-free codec.

    I also agree with Henry that "the Internet has different criteria =
than
    ITU-T networks may have". Internet adoption follows different =
patterns
    than adoption in the ITU primary target markets. For example, the
    Internet has more consumer-reconfigurable software, while the ITU =
has to
    deal with a lot of fixed hardware deployments. At the ITU, it makes
    sense to invest large sums of money into testing and =
characterisation of
    codecs because the codecs deployed there usually stay around for a =
long
    time and the infrastructure investments are usually very large. On =
the
    other hand, I would say the investments in codecs for the Internet =
are
    comparatively smaller and, while testing is still important, it is =
not
    as critical as it is for the ITU.

    It's also a choice one has to make. It is unlikely that companies =
would
    invest money in expensive testing if they are not going to obtain
    royalties in return. However, I think we may be able to define some =
more
    lightweight (collaborative?) testing that is sufficient and doesn't
    involve as much investments as what the ITU does. For the Internet, =
I
    believe an IPR-free codec that everyone agrees performs well is =
better
    than a patent-encumbered codec that has had more extensive testing. =
This
    is again another difference with the ITU: patent-encumbered codecs =
tend
    to hurt a lot more on the Internet because many applications (e.g.
    giving away the client) are very hard (or impossible) when one has =
to
    pay per-channel royalties.

    As for the source code issue you pointed out, all the Xiph codecs =
are
    already published under a very permissive open source license (BSD), =
so
    this would not really change.

        Jean-Marc

    > Roni
    >
    >=20
    >
    > *From:* avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] *On =
Behalf
    > Of *Slava Borilin
    > *Sent:* Monday, June 01, 2009 11:50 PM
    > *To:* jean-marc.valin@octasic.com
    > *Cc:* avt@ietf.org; Jason Fischl
    > *Subject:* Re: [AVT] [dispatch] Proposal to form Internet Wideband
    > Audio Codec WG
    >
    >=20
    >
    > Agree with Jean-Marc.
    > SPIRIT is interested to contribute as well - having a dozen of =
proprietary
    codecs developed, including
    > one specifically desgined for internet (SPIRIT IP-MR, which is now =
under
    WGLC on draft-ietf-avt-rtp-ipmr-04) -
    > multi-rate, scalable, adaptive, wideband codec.
    >=20
    > We can also continue this work with IETF.
    >=20
    > Moreover, most if not all efforts coming from ITU on codecs =
unfortunately
    are NOT really focused on
    > internet-specific codecs (that's why several companies have had to =
invent)
    - as ITU preference is mainly
    > specific (i.e. cellular) networks at first.
    >=20
    > however, as we see the greant rise of pure "internet-basd =
communications"
    (skype, webex/citrix, and many others) -
    > we all (and users) are suffering from inefficiency in all =
currently
    "standard" codecs and ambiguity in the choice of
    > internet-targeted ones.
    >=20
    > Again, we probably can put together enough number of contributors =
to the
    WG to have the expertise.
    >=20
    > regards,
    > Slava Borilin
    >=20
    > --
    > John Lazzaro wrote:
    >
    >     A traditional response to this type of request is to note that =
the
    IETF
    >
    >     really doesn't have much in the way of expertise in audio =
codec
    design.
    >
    >     I don't see many of the regulars who present at the AES codec =
paper
    sessions
    >
    >     posting here on AVT (ditto ICASSP paper sessions for voice =
codecs).
    It's
    >
    >     a full-time job to keep up-to-date and contribute to that
    >
    >     signal-processing lore.
    >
    >    =20
    >
    > Well, there's no reason that the IETF cannot build such an =
expertise
    > in audio codecs. This is actually something in which I'd be =
interested
    > in getting involved and I'm sure others at Xiph.Org would be
    > interested as well. We have several people with audio codec =
expertise
    > from working on Vorbis, Speex and (more recently) CELT. It turns =
out
    > that the CELT codec currently under development at Xiph actually =
meets
    > most of the requirements from the original proposal in being a =
very
    > low delay codec with adaptive bit-rate and sampling rate (up to 48
    > kHz), scalable complexity, and good robustness to packet loss. =
We'd be
    > willing to continue the development with the IETF. Even if not =
with
    > CELT, it's still like to be involved in such a new WG.
    >
    > Cheers,
    >=20
    >    Jean-Marc





-------------------------------------------------------------------------=
-----


  _______________________________________________
  dispatch mailing list
  dispatch@ietf.org
  https://www.ietf.org/mailman/listinfo/dispatch

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>Re: [AVT] [dispatch] Proposal to form Internet =
Wideband Audio Codec WG</TITLE>
<META content=3D"text/html; charset=3Diso-8859-1" =
http-equiv=3DContent-Type>
<META name=3DGENERATOR content=3D"MSHTML 8.00.6001.18702">
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT size=3D2>I am not an area director, nor do i play one on =
IPTV, but my=20
suggestion for anyone interested would be to </FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>- request a non-WG mailing list (from either of the =
RAI area=20
directors),</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>- announce it on this mailing list (and others that =
seem to be=20
appropriate, and </FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>- start discussions, which the area directors =
typically look=20
at when judging whether there's enough interest and enough clue to =
justify=20
approving a BOF and/or chartering a working group.</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>I didn't see a mailing list included in Jason's =
proposal - if=20
I missed one, my apologies.</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>If people at the IETF can do this work, and want to =
do this=20
work, IETF process policies should not get in the way of that=20
happening.</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>And I haven't heard of an AD turning down a request =
for a=20
non-WG mailing list yet, keeping in mind that one of said lists is =
ietf-sailors,=20
for people interested in sailing to/from/at IETF meetings - the bar is=20
(appropriately) low.</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>Thanks,</FONT></DIV>
<DIV><FONT size=3D2><BR>Spencer</FONT></DIV>
<BLOCKQUOTE=20
style=3D"BORDER-LEFT: #000000 2px solid; PADDING-LEFT: 5px; =
PADDING-RIGHT: 0px; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px">
  <DIV style=3D"FONT: 10pt arial">----- Original Message ----- </DIV>
  <DIV=20
  style=3D"FONT: 10pt arial; BACKGROUND: #e4e4e4; font-color: =
black"><B>From:</B>=20
  <A title=3Dhsinnrei@adobe.com href=3D"mailto:hsinnrei@adobe.com">Henry =

  Sinnreich</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A =
title=3DEven.roni@huawei.com=20
  href=3D"mailto:Even.roni@huawei.com">Roni Even</A> ; <A=20
  title=3Djean-marc.valin@octasic.com=20
  href=3D"mailto:jean-marc.valin@octasic.com">Jean-Marc Valin</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Cc:</B> <A =
title=3Ddispatch@ietf.org=20
  href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</A> ; <A=20
  title=3DBorilin@spiritdsp.com =
href=3D"mailto:Borilin@spiritdsp.com">Slava=20
  Borilin</A> ; <A title=3Davt@ietf.org=20
  href=3D"mailto:avt@ietf.org">avt@ietf.org</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Sent:</B> Tuesday, June 02, 2009 =
11:12=20
  AM</DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Subject:</B> Re: [dispatch] [AVT] =
Proposal to=20
  form Internet Wideband Audio Codec WG</DIV>
  <DIV><BR></DIV><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SPAN =

  style=3D"FONT-SIZE: 18pt">&gt;This leaves the other reasons I heard =
for doing it=20
  at the IETF which is the <BR>&gt;price of participating <BR><BR>This =
is an=20
  important point.<BR>Given the travel constraints for some of the most =
valuable=20
  potential contributors and reviewers, could we envisage online work =
until the=20
  economy improves, so that an eventual BOF to start with should not be =
starved=20
  of attendees? An online BOF?<BR><BR>There are several free online =
meeting=20
  tools available...<BR><BR>How can this be done within the IETF=20
  policy?<BR><BR>Henry<BR><BR><BR>On 6/2/09 10:57 AM, "Roni Even" &lt;<A =

  href=3D"mailto:Even.roni@huawei.com">Even.roni@huawei.com</A>&gt;=20
  wrote:<BR><BR></SPAN></FONT>
  <BLOCKQUOTE><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SPAN=20
    style=3D"FONT-SIZE: 18pt">Hi,<BR>I do not want to sound like someone =
who is=20
    for IPR (I am not), but why stop<BR>at codec, let's require it for =
all IETF=20
    work. There are IPR on IETF work<BR>which is must simpler, in my =
view, than=20
    wide band audio codecs.<BR><BR>I think that we can start with =
"royalty free"=20
    even though I am not sure that<BR>it will accepted as part of the =
charter of=20
    any other work group so why pick<BR>on codecs which require more=20
    work.<BR><BR>This leaves the other reasons I heard for doing it at =
the IETF=20
    which is the<BR>price of participating (cheaper than being an ITU-T =
member)=20
    and maybe design<BR>less expensive characterization=20
    tests.<BR><BR>Roni<BR><BR><BR>-----Original Message-----<BR>From: =
Jean-Marc=20
    Valin [<A=20
    =
href=3D"mailto:jean-marc.valin@octasic.com">mailto:jean-marc.valin@octasi=
c.com</A>]<BR>Sent:=20
    Tuesday, June 02, 2009 6:14 PM<BR>To: Roni Even<BR>Cc: 'Slava =
Borilin'; <A=20
    href=3D"avt@ietf.org">avt@ietf.org</A>; 'Jason Fischl'; <A=20
    href=3D"dispatch@ietf.org">dispatch@ietf.org</A>;<BR><A=20
    href=3D"hsinnrei@adobe.com">hsinnrei@adobe.com</A><BR>Subject: Re: =
[AVT]=20
    [dispatch] Proposal to form Internet Wideband Audio=20
    Codec<BR>WG<BR><BR>Hi,<BR><BR>Roni Even wrote:<BR>&gt;<BR>&gt; I am =
not sure=20
    what prevented you from doing it today at the ITU or<BR>&gt; MPEG, =
why do=20
    you see the IETF handling it differently.<BR>&gt;<BR>&gt; I would =
also like=20
    to remind you and Jean-Marc that once you are<BR>&gt; bringing work =
to a=20
    standard body it may require collaboration with<BR>&gt; other people =
who=20
    will have other proposals that will also address the<BR>&gt; same=20
    requirements and you may need to invest money in comparative<BR>&gt; =
testing=20
    by independent listening labs.<BR>&gt;<BR>&gt; I also think that you =
will=20
    need to supply the codec in source code and<BR>&gt; provide copy =
right to=20
    the IETF.<BR>&gt;<BR><BR>I am well aware that bringing work to the =
IETF=20
    would require<BR>collaboration with others. I am not seeking control =
over=20
    the work I am<BR>currently doing and would really welcome such=20
    collaboration. The idea is<BR>only to have the best wideband codec =
possible=20
    without IPR issues. Given<BR>the ITU and MPEG track record, I think =
it would=20
    be very unlikely for any<BR>of those organisations to work on an =
IPR-free=20
    codec.<BR><BR>I also agree with Henry that "the Internet has =
different=20
    criteria than<BR>ITU-T networks may have". Internet adoption follows =

    different patterns<BR>than adoption in the ITU primary target =
markets. For=20
    example, the<BR>Internet has more consumer-reconfigurable software, =
while=20
    the ITU has to<BR>deal with a lot of fixed hardware deployments. At =
the ITU,=20
    it makes<BR>sense to invest large sums of money into testing and=20
    characterisation of<BR>codecs because the codecs deployed there =
usually stay=20
    around for a long<BR>time and the infrastructure investments are =
usually=20
    very large. On the<BR>other hand, I would say the investments in =
codecs for=20
    the Internet are<BR>comparatively smaller and, while testing is =
still=20
    important, it is not<BR>as critical as it is for the =
ITU.<BR><BR>It's also a=20
    choice one has to make. It is unlikely that companies =
would<BR>invest money=20
    in expensive testing if they are not going to obtain<BR>royalties in =
return.=20
    However, I think we may be able to define some more<BR>lightweight=20
    (collaborative?) testing that is sufficient and doesn't<BR>involve =
as much=20
    investments as what the ITU does. For the Internet, I<BR>believe an =
IPR-free=20
    codec that everyone agrees performs well is better<BR>than a=20
    patent-encumbered codec that has had more extensive testing. =
This<BR>is=20
    again another difference with the ITU: patent-encumbered codecs =
tend<BR>to=20
    hurt a lot more on the Internet because many applications =
(e.g.<BR>giving=20
    away the client) are very hard (or impossible) when one has =
to<BR>pay=20
    per-channel royalties.<BR><BR>As for the source code issue you =
pointed out,=20
    all the Xiph codecs are<BR>already published under a very permissive =
open=20
    source license (BSD), so<BR>this would not really=20
    change.<BR><BR>&nbsp;&nbsp;&nbsp;&nbsp;Jean-Marc<BR><BR>&gt;=20
    Roni<BR>&gt;<BR>&gt; <BR>&gt;<BR>&gt; *From:* <A=20
    href=3D"avt-bounces@ietf.org">avt-bounces@ietf.org</A> [<A=20
    =
href=3D"mailto:avt-bounces@ietf.org">mailto:avt-bounces@ietf.org</A>] =
*On=20
    Behalf<BR>&gt; Of *Slava Borilin<BR>&gt; *Sent:* Monday, June 01, =
2009 11:50=20
    PM<BR>&gt; *To:* <A=20
    =
href=3D"jean-marc.valin@octasic.com">jean-marc.valin@octasic.com</A><BR>&=
gt;=20
    *Cc:* <A href=3D"avt@ietf.org">avt@ietf.org</A>; Jason =
Fischl<BR>&gt;=20
    *Subject:* Re: [AVT] [dispatch] Proposal to form Internet =
Wideband<BR>&gt;=20
    Audio Codec WG<BR>&gt;<BR>&gt; <BR>&gt;<BR>&gt; Agree with=20
    Jean-Marc.<BR>&gt; SPIRIT is interested to contribute as well - =
having a=20
    dozen of proprietary<BR>codecs developed, including<BR>&gt; one =
specifically=20
    desgined for internet (SPIRIT IP-MR, which is now under<BR>WGLC on=20
    draft-ietf-avt-rtp-ipmr-04) -<BR>&gt; multi-rate, scalable, =
adaptive,=20
    wideband codec.<BR>&gt; <BR>&gt; We can also continue this work with =

    IETF.<BR>&gt; <BR>&gt; Moreover, most if not all efforts coming from =
ITU on=20
    codecs unfortunately<BR>are NOT really focused on<BR>&gt; =
internet-specific=20
    codecs (that's why several companies have had to invent)<BR>- as ITU =

    preference is mainly<BR>&gt; specific (i.e. cellular) networks at=20
    first.<BR>&gt; <BR>&gt; however, as we see the greant rise of pure=20
    "internet-basd communications"<BR>(skype, webex/citrix, and many =
others)=20
    -<BR>&gt; we all (and users) are suffering from inefficiency in all=20
    currently<BR>"standard" codecs and ambiguity in the choice =
of<BR>&gt;=20
    internet-targeted ones.<BR>&gt; <BR>&gt; Again, we probably can put =
together=20
    enough number of contributors to the<BR>WG to have the =
expertise.<BR>&gt;=20
    <BR>&gt; regards,<BR>&gt; Slava Borilin<BR>&gt; <BR>&gt; --<BR>&gt; =
John=20
    Lazzaro wrote:<BR>&gt;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;A traditional =

    response to this type of request is to note that =
the<BR>IETF<BR>&gt;<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;really doesn't have much in the way of =
expertise in=20
    audio codec<BR>design.<BR>&gt;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;I =
don't see=20
    many of the regulars who present at the AES codec=20
    paper<BR>sessions<BR>&gt;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;posting =
here on=20
    AVT (ditto ICASSP paper sessions for voice =
codecs).<BR>It's<BR>&gt;<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;a full-time job to keep up-to-date and =
contribute to=20
    that<BR>&gt;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;signal-processing=20
    lore.<BR>&gt;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;<BR>&gt;<BR>&gt; Well, =
there's=20
    no reason that the IETF cannot build such an expertise<BR>&gt; in =
audio=20
    codecs. This is actually something in which I'd be =
interested<BR>&gt; in=20
    getting involved and I'm sure others at Xiph.Org would be<BR>&gt; =
interested=20
    as well. We have several people with audio codec expertise<BR>&gt; =
from=20
    working on Vorbis, Speex and (more recently) CELT. It turns =
out<BR>&gt; that=20
    the CELT codec currently under development at Xiph actually =
meets<BR>&gt;=20
    most of the requirements from the original proposal in being a =
very<BR>&gt;=20
    low delay codec with adaptive bit-rate and sampling rate (up to =
48<BR>&gt;=20
    kHz), scalable complexity, and good robustness to packet loss. We'd=20
    be<BR>&gt; willing to continue the development with the IETF. Even =
if not=20
    with<BR>&gt; CELT, it's still like to be involved in such a new=20
    WG.<BR>&gt;<BR>&gt; Cheers,<BR>&gt; <BR>&gt;=20
    &nbsp;&nbsp;&nbsp;Jean-Marc<BR><BR><BR></SPAN></FONT></BLOCKQUOTE>
  <P>
  <HR>

  <P></P>_______________________________________________<BR>dispatch =
mailing=20
  =
list<BR>dispatch@ietf.org<BR>https://www.ietf.org/mailman/listinfo/dispat=
ch<BR></BLOCKQUOTE></BODY></HTML>

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From: "Slava Borilin" <Borilin@spiritdsp.com>
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I do not beleive the one that will come wil be low quality.
at least people from the potential contributors (at least Skype, Speex,
SPIRIT) are already pretty-good in the commercially exploiting their own
codecs on the market.
i think this is probably false alert.
=20
regards,
Slava Borilin
=20
=20

________________________________

From: stephen botzko [mailto:stephen.botzko@gmail.com]=20
Sent: Tuesday, June 02, 2009 9:32 PM
To: Henning Schulzrinne
Cc: Roni Even; dispatch@ietf.org; Jason Fischl; avt@ietf.org;
hsinnrei@adobe.com; Slava Borilin
Subject: Re: [AVT] [dispatch] Proposal to form Internet Wideband Audio
Codec WG


>>> - the quality of the codec may not be competitive

I think its very important that the codec quality be competitive.
People expect excellence from IETF standards  Standardizing
non-competitive codecs because they are cheap does not seem to be a good
choce.

Steve B.


On Tue, Jun 2, 2009 at 12:15 PM, Henning Schulzrinne
<hgs@cs.columbia.edu> wrote:


	I view this as a trade-off. If we pursue this, there are risks:
=09
	- nothing may come of it since there are no experts willing to
help
	- somebody will claim IPR on the resulting work
	- the quality of the codec may not be competitive
=09
	However, if we don't do this, we are stuck with the status quo,
which is not all that satisfactory. Thus, unless there are significant
costs for "innocent bystanders", I see this as a risk worth taking. In
the worst case, we are no worse off than we are today. In all other
cases, we'll have an additional choice for a wideband codec, even if
it's not "the best", just "good enough". After all, most people use
G.711 today, which has a really hard time making that claim.
=09
	Most real work in the IETF is done by very small teams,
typically less than 10, so as long as we have a handful of people that
are willing to contribute, this can work. It might even work better,
since you may get fewer people who have half-baked opinions - we may
skip the binary vs. XML debates...
=09
	We can set some ground rules ("must be tested with packet loss
of 5%") and then see what happens. Compared to most network protocols,
the likely negative impacts (such as security or congestion control
issues) of even a badly-designed codec are pretty limited.
=09
	Henning=20


	On Jun 2, 2009, at 11:57 AM, Roni Even wrote:
=09
=09

		Hi,
		I do not want to sound like someone who is for IPR (I am
not), but why stop
		at codec, let's require it for all IETF work. There are
IPR on IETF work
		which is must simpler, in my view, than wide band audio
codecs.
	=09
		I think that we can start with "royalty free" even
though I am not sure that
		it will accepted as part of the charter of any other
work group so why pick
		on codecs which require more work.
	=09
		This leaves the other reasons I heard for doing it at
the IETF which is the
		price of participating (cheaper than being an ITU-T
member) and maybe design
		less expensive characterization tests.
	=09
		Roni
	=09
	=09
		-----Original Message-----
		From: Jean-Marc Valin
[mailto:jean-marc.valin@octasic.com]
		Sent: Tuesday, June 02, 2009 6:14 PM
		To: Roni Even
		Cc: 'Slava Borilin'; avt@ietf.org; 'Jason Fischl';
dispatch@ietf.org;
		hsinnrei@adobe.com
		Subject: Re: [AVT] [dispatch] Proposal to form Internet
Wideband Audio Codec
		WG
	=09
		Hi,
	=09
		Roni Even wrote:
	=09


			I am not sure what prevented you from doing it
today at the ITU or
			MPEG, why do you see the IETF handling it
differently.
		=09
			I would also like to remind you and Jean-Marc
that once you are
			bringing work to a standard body it may require
collaboration with
			other people who will have other proposals that
will also address the
			same requirements and you may need to invest
money in comparative
			testing by independent listening labs.
		=09
			I also think that you will need to supply the
codec in source code and
			provide copy right to the IETF.
		=09
		=09


		I am well aware that bringing work to the IETF would
require
		collaboration with others. I am not seeking control over
the work I am
		currently doing and would really welcome such
collaboration. The idea is
		only to have the best wideband codec possible without
IPR issues. Given
		the ITU and MPEG track record, I think it would be very
unlikely for any
		of those organisations to work on an IPR-free codec.
	=09
		I also agree with Henry that "the Internet has different
criteria than
		ITU-T networks may have". Internet adoption follows
different patterns
		than adoption in the ITU primary target markets. For
example, the
		Internet has more consumer-reconfigurable software,
while the ITU has to
		deal with a lot of fixed hardware deployments. At the
ITU, it makes
		sense to invest large sums of money into testing and
characterisation of
		codecs because the codecs deployed there usually stay
around for a long
		time and the infrastructure investments are usually very
large. On the
		other hand, I would say the investments in codecs for
the Internet are
		comparatively smaller and, while testing is still
important, it is not
		as critical as it is for the ITU.
	=09
		It's also a choice one has to make. It is unlikely that
companies would
		invest money in expensive testing if they are not going
to obtain
		royalties in return. However, I think we may be able to
define some more
		lightweight (collaborative?) testing that is sufficient
and doesn't
		involve as much investments as what the ITU does. For
the Internet, I
		believe an IPR-free codec that everyone agrees performs
well is better
		than a patent-encumbered codec that has had more
extensive testing. This
		is again another difference with the ITU:
patent-encumbered codecs tend
		to hurt a lot more on the Internet because many
applications (e.g.
		giving away the client) are very hard (or impossible)
when one has to
		pay per-channel royalties.
	=09
		As for the source code issue you pointed out, all the
Xiph codecs are
		already published under a very permissive open source
license (BSD), so
		this would not really change.
	=09
		  Jean-Marc
	=09
	=09

			Roni
		=09
		=09
		=09
			*From:* avt-bounces@ietf.org
[mailto:avt-bounces@ietf.org] *On Behalf
			Of *Slava Borilin
			*Sent:* Monday, June 01, 2009 11:50 PM
			*To:* jean-marc.valin@octasic.com
			*Cc:* avt@ietf.org; Jason Fischl
			*Subject:* Re: [AVT] [dispatch] Proposal to form
Internet Wideband
			Audio Codec WG
		=09
		=09
		=09
			Agree with Jean-Marc.
			SPIRIT is interested to contribute as well -
having a dozen of proprietary
		=09

		codecs developed, including
	=09

			one specifically desgined for internet (SPIRIT
IP-MR, which is now under
		=09

		WGLC on draft-ietf-avt-rtp-ipmr-04) -
	=09

			multi-rate, scalable, adaptive, wideband codec.
		=09
			We can also continue this work with IETF.
		=09
			Moreover, most if not all efforts coming from
ITU on codecs unfortunately
		=09

		are NOT really focused on
	=09

			internet-specific codecs (that's why several
companies have had to invent)
		=09

		- as ITU preference is mainly
	=09

			specific (i.e. cellular) networks at first.
		=09
			however, as we see the greant rise of pure
"internet-basd communications"
		=09

		(skype, webex/citrix, and many others) -
	=09

			we all (and users) are suffering from
inefficiency in all currently
		=09

		"standard" codecs and ambiguity in the choice of
	=09

			internet-targeted ones.
		=09
			Again, we probably can put together enough
number of contributors to the
		=09

		WG to have the expertise.
	=09


			regards,
			Slava Borilin
		=09
			--
			John Lazzaro wrote:
		=09
			  A traditional response to this type of request
is to note that the
		=09

		IETF
	=09


			  really doesn't have much in the way of
expertise in audio codec
		=09

		design.
	=09


			  I don't see many of the regulars who present
at the AES codec paper
		=09

		sessions
	=09


			  posting here on AVT (ditto ICASSP paper
sessions for voice codecs).
		=09

		It's
	=09


			  a full-time job to keep up-to-date and
contribute to that
		=09
			  signal-processing lore.
		=09
		=09
		=09
			Well, there's no reason that the IETF cannot
build such an expertise
			in audio codecs. This is actually something in
which I'd be interested
			in getting involved and I'm sure others at
Xiph.Org would be
			interested as well. We have several people with
audio codec expertise
			from working on Vorbis, Speex and (more
recently) CELT. It turns out
			that the CELT codec currently under development
at Xiph actually meets
			most of the requirements from the original
proposal in being a very
			low delay codec with adaptive bit-rate and
sampling rate (up to 48
			kHz), scalable complexity, and good robustness
to packet loss. We'd be
			willing to continue the development with the
IETF. Even if not with
			CELT, it's still like to be involved in such a
new WG.
		=09
			Cheers,
		=09
			 Jean-Marc
		=09


		_______________________________________________
		Audio/Video Transport Working Group
		avt@ietf.org
		https://www.ietf.org/mailman/listinfo/avt
	=09
	=09


	_______________________________________________
	Audio/Video Transport Working Group
	avt@ietf.org
	https://www.ietf.org/mailman/listinfo/avt
=09



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<DIV dir=3Dltr align=3Dleft><SPAN class=3D203403217-02062009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>I do not beleive the one that will come wil be =
low=20
quality.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D203403217-02062009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>at least people from the potential contributors =
(at least=20
Skype, Speex, SPIRIT) are already pretty-good in the commercially =
exploiting=20
their own codecs on the market.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D203403217-02062009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>i think this is probably false =
alert.</FONT></SPAN></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV align=3Dleft><FONT face=3DArial size=3D2>regards,</FONT></DIV>
<DIV align=3Dleft><FONT face=3DArial size=3D2>Slava Borilin</FONT></DIV>
<DIV align=3Dleft><SPAN lang=3DEN-US=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"></SPAN>&nbsp;</DIV>
<DIV>&nbsp;</DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> stephen botzko=20
[mailto:stephen.botzko@gmail.com] <BR><B>Sent:</B> Tuesday, June 02, =
2009 9:32=20
PM<BR><B>To:</B> Henning Schulzrinne<BR><B>Cc:</B> Roni Even; =
dispatch@ietf.org;=20
Jason Fischl; avt@ietf.org; hsinnrei@adobe.com; Slava =
Borilin<BR><B>Subject:</B>=20
Re: [AVT] [dispatch] Proposal to form Internet Wideband Audio Codec=20
WG<BR></FONT><BR></DIV>
<DIV></DIV>&gt;&gt;&gt; - the quality of the codec may not be=20
competitive<BR><BR>I think its very important that the codec quality be=20
competitive.&nbsp; People expect excellence from IETF standards&nbsp;=20
Standardizing non-competitive codecs because they are cheap does not =
seem to be=20
a good choce.<BR><BR>Steve B.<BR><BR>
<DIV class=3Dgmail_quote>On Tue, Jun 2, 2009 at 12:15 PM, Henning =
Schulzrinne=20
<SPAN dir=3Dltr>&lt;<A=20
href=3D"mailto:hgs@cs.columbia.edu">hgs@cs.columbia.edu</A>&gt;</SPAN> =
wrote:<BR>
<BLOCKQUOTE class=3Dgmail_quote=20
style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: =
rgb(204,204,204) 1px solid">I=20
  view this as a trade-off. If we pursue this, there are risks:<BR><BR>- =
nothing=20
  may come of it since there are no experts willing to help<BR>- =
somebody will=20
  claim IPR on the resulting work<BR>- the quality of the codec may not =
be=20
  competitive<BR><BR>However, if we don't do this, we are stuck with the =
status=20
  quo, which is not all that satisfactory. Thus, unless there are =
significant=20
  costs for "innocent bystanders", I see this as a risk worth taking. In =
the=20
  worst case, we are no worse off than we are today. In all other cases, =
we'll=20
  have an additional choice for a wideband codec, even if it's not "the =
best",=20
  just "good enough". After all, most people use G.711 today, which has =
a really=20
  hard time making that claim.<BR><BR>Most real work in the IETF is done =
by very=20
  small teams, typically less than 10, so as long as we have a handful =
of people=20
  that are willing to contribute, this can work. It might even work =
better,=20
  since you may get fewer people who have half-baked opinions - we may =
skip the=20
  binary vs. XML debates...<BR><BR>We can set some ground rules ("must =
be tested=20
  with packet loss of 5%") and then see what happens. Compared to most =
network=20
  protocols, the likely negative impacts (such as security or congestion =
control=20
  issues) of even a badly-designed codec are pretty limited.<BR><FONT=20
  color=3D#888888><BR>Henning</FONT>
  <DIV>
  <DIV></DIV>
  <DIV class=3Dh5><BR><BR>On Jun 2, 2009, at 11:57 AM, Roni Even =
wrote:<BR><BR>
  <BLOCKQUOTE class=3Dgmail_quote=20
  style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: =
rgb(204,204,204) 1px solid">Hi,<BR>I=20
    do not want to sound like someone who is for IPR (I am not), but why =

    stop<BR>at codec, let's require it for all IETF work. There are IPR =
on IETF=20
    work<BR>which is must simpler, in my view, than wide band audio=20
    codecs.<BR><BR>I think that we can start with "royalty free" even =
though I=20
    am not sure that<BR>it will accepted as part of the charter of any =
other=20
    work group so why pick<BR>on codecs which require more =
work.<BR><BR>This=20
    leaves the other reasons I heard for doing it at the IETF which is=20
    the<BR>price of participating (cheaper than being an ITU-T member) =
and maybe=20
    design<BR>less expensive characterization=20
    tests.<BR><BR>Roni<BR><BR><BR>-----Original Message-----<BR>From: =
Jean-Marc=20
    Valin [mailto:<A href=3D"mailto:jean-marc.valin@octasic.com"=20
    target=3D_blank>jean-marc.valin@octasic.com</A>]<BR>Sent: Tuesday, =
June 02,=20
    2009 6:14 PM<BR>To: Roni Even<BR>Cc: 'Slava Borilin'; <A=20
    href=3D"mailto:avt@ietf.org" target=3D_blank>avt@ietf.org</A>; =
'Jason Fischl';=20
    <A href=3D"mailto:dispatch@ietf.org"=20
    target=3D_blank>dispatch@ietf.org</A>;<BR><A =
href=3D"mailto:hsinnrei@adobe.com"=20
    target=3D_blank>hsinnrei@adobe.com</A><BR>Subject: Re: [AVT] =
[dispatch]=20
    Proposal to form Internet Wideband Audio =
Codec<BR>WG<BR><BR>Hi,<BR><BR>Roni=20
    Even wrote:<BR>
    <BLOCKQUOTE class=3Dgmail_quote=20
    style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: =
rgb(204,204,204) 1px solid"><BR>I=20
      am not sure what prevented you from doing it today at the ITU =
or<BR>MPEG,=20
      why do you see the IETF handling it differently.<BR><BR>I would =
also like=20
      to remind you and Jean-Marc that once you are<BR>bringing work to =
a=20
      standard body it may require collaboration with<BR>other people =
who will=20
      have other proposals that will also address the<BR>same =
requirements and=20
      you may need to invest money in comparative<BR>testing by =
independent=20
      listening labs.<BR><BR>I also think that you will need to supply =
the codec=20
      in source code and<BR>provide copy right to the=20
    IETF.<BR><BR></BLOCKQUOTE><BR>I am well aware that bringing work to =
the IETF=20
    would require<BR>collaboration with others. I am not seeking control =
over=20
    the work I am<BR>currently doing and would really welcome such=20
    collaboration. The idea is<BR>only to have the best wideband codec =
possible=20
    without IPR issues. Given<BR>the ITU and MPEG track record, I think =
it would=20
    be very unlikely for any<BR>of those organisations to work on an =
IPR-free=20
    codec.<BR><BR>I also agree with Henry that "the Internet has =
different=20
    criteria than<BR>ITU-T networks may have". Internet adoption follows =

    different patterns<BR>than adoption in the ITU primary target =
markets. For=20
    example, the<BR>Internet has more consumer-reconfigurable software, =
while=20
    the ITU has to<BR>deal with a lot of fixed hardware deployments. At =
the ITU,=20
    it makes<BR>sense to invest large sums of money into testing and=20
    characterisation of<BR>codecs because the codecs deployed there =
usually stay=20
    around for a long<BR>time and the infrastructure investments are =
usually=20
    very large. On the<BR>other hand, I would say the investments in =
codecs for=20
    the Internet are<BR>comparatively smaller and, while testing is =
still=20
    important, it is not<BR>as critical as it is for the =
ITU.<BR><BR>It's also a=20
    choice one has to make. It is unlikely that companies =
would<BR>invest money=20
    in expensive testing if they are not going to obtain<BR>royalties in =
return.=20
    However, I think we may be able to define some more<BR>lightweight=20
    (collaborative?) testing that is sufficient and doesn't<BR>involve =
as much=20
    investments as what the ITU does. For the Internet, I<BR>believe an =
IPR-free=20
    codec that everyone agrees performs well is better<BR>than a=20
    patent-encumbered codec that has had more extensive testing. =
This<BR>is=20
    again another difference with the ITU: patent-encumbered codecs =
tend<BR>to=20
    hurt a lot more on the Internet because many applications =
(e.g.<BR>giving=20
    away the client) are very hard (or impossible) when one has =
to<BR>pay=20
    per-channel royalties.<BR><BR>As for the source code issue you =
pointed out,=20
    all the Xiph codecs are<BR>already published under a very permissive =
open=20
    source license (BSD), so<BR>this would not really =
change.<BR><BR>&nbsp;=20
    Jean-Marc<BR><BR>
    <BLOCKQUOTE class=3Dgmail_quote=20
    style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: =
rgb(204,204,204) 1px solid">Roni<BR><BR><BR><BR>*From:*=20
      <A href=3D"mailto:avt-bounces@ietf.org"=20
      target=3D_blank>avt-bounces@ietf.org</A> [mailto:<A=20
      href=3D"mailto:avt-bounces@ietf.org" =
target=3D_blank>avt-bounces@ietf.org</A>]=20
      *On Behalf<BR>Of *Slava Borilin<BR>*Sent:* Monday, June 01, 2009 =
11:50=20
      PM<BR>*To:* <A href=3D"mailto:jean-marc.valin@octasic.com"=20
      target=3D_blank>jean-marc.valin@octasic.com</A><BR>*Cc:* <A=20
      href=3D"mailto:avt@ietf.org" target=3D_blank>avt@ietf.org</A>; =
Jason=20
      Fischl<BR>*Subject:* Re: [AVT] [dispatch] Proposal to form =
Internet=20
      Wideband<BR>Audio Codec WG<BR><BR><BR><BR>Agree with =
Jean-Marc.<BR>SPIRIT=20
      is interested to contribute as well - having a dozen of=20
    proprietary<BR></BLOCKQUOTE>codecs developed, including<BR>
    <BLOCKQUOTE class=3Dgmail_quote=20
    style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: =
rgb(204,204,204) 1px solid">one=20
      specifically desgined for internet (SPIRIT IP-MR, which is now=20
    under<BR></BLOCKQUOTE>WGLC on draft-ietf-avt-rtp-ipmr-04) -<BR>
    <BLOCKQUOTE class=3Dgmail_quote=20
    style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: =
rgb(204,204,204) 1px solid">multi-rate,=20
      scalable, adaptive, wideband codec.<BR><BR>We can also continue =
this work=20
      with IETF.<BR><BR>Moreover, most if not all efforts coming from =
ITU on=20
      codecs unfortunately<BR></BLOCKQUOTE>are NOT really focused on<BR>
    <BLOCKQUOTE class=3Dgmail_quote=20
    style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: =
rgb(204,204,204) 1px solid">internet-specific=20
      codecs (that's why several companies have had to =
invent)<BR></BLOCKQUOTE>-=20
    as ITU preference is mainly<BR>
    <BLOCKQUOTE class=3Dgmail_quote=20
    style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: =
rgb(204,204,204) 1px solid">specific=20
      (i.e. cellular) networks at first.<BR><BR>however, as we see the =
greant=20
      rise of pure "internet-basd =
communications"<BR></BLOCKQUOTE>(skype,=20
    webex/citrix, and many others) -<BR>
    <BLOCKQUOTE class=3Dgmail_quote=20
    style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: =
rgb(204,204,204) 1px solid">we=20
      all (and users) are suffering from inefficiency in all=20
    currently<BR></BLOCKQUOTE>"standard" codecs and ambiguity in the =
choice=20
    of<BR>
    <BLOCKQUOTE class=3Dgmail_quote=20
    style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: =
rgb(204,204,204) 1px solid">internet-targeted=20
      ones.<BR><BR>Again, we probably can put together enough number of=20
      contributors to the<BR></BLOCKQUOTE>WG to have the expertise.<BR>
    <BLOCKQUOTE class=3Dgmail_quote=20
    style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: =
rgb(204,204,204) 1px solid"><BR>regards,<BR>Slava=20
      Borilin<BR><BR>--<BR>John Lazzaro wrote:<BR><BR>&nbsp; A =
traditional=20
      response to this type of request is to note that=20
the<BR></BLOCKQUOTE>IETF<BR>
    <BLOCKQUOTE class=3Dgmail_quote=20
    style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: =
rgb(204,204,204) 1px solid"><BR>&nbsp;=20
      really doesn't have much in the way of expertise in audio=20
    codec<BR></BLOCKQUOTE>design.<BR>
    <BLOCKQUOTE class=3Dgmail_quote=20
    style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: =
rgb(204,204,204) 1px solid"><BR>&nbsp;=20
      I don't see many of the regulars who present at the AES codec=20
    paper<BR></BLOCKQUOTE>sessions<BR>
    <BLOCKQUOTE class=3Dgmail_quote=20
    style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: =
rgb(204,204,204) 1px solid"><BR>&nbsp;=20
      posting here on AVT (ditto ICASSP paper sessions for voice=20
    codecs).<BR></BLOCKQUOTE>It's<BR>
    <BLOCKQUOTE class=3Dgmail_quote=20
    style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: =
rgb(204,204,204) 1px solid"><BR>&nbsp;=20
      a full-time job to keep up-to-date and contribute to =
that<BR><BR>&nbsp;=20
      signal-processing lore.<BR><BR><BR><BR>Well, there's no reason =
that the=20
      IETF cannot build such an expertise<BR>in audio codecs. This is =
actually=20
      something in which I'd be interested<BR>in getting involved and =
I'm sure=20
      others at Xiph.Org would be<BR>interested as well. We have several =
people=20
      with audio codec expertise<BR>from working on Vorbis, Speex and =
(more=20
      recently) CELT. It turns out<BR>that the CELT codec currently =
under=20
      development at Xiph actually meets<BR>most of the requirements =
from the=20
      original proposal in being a very<BR>low delay codec with adaptive =

      bit-rate and sampling rate (up to 48<BR>kHz), scalable complexity, =
and=20
      good robustness to packet loss. We'd be<BR>willing to continue the =

      development with the IETF. Even if not with<BR>CELT, it's still =
like to be=20
      involved in such a new=20
    =
WG.<BR><BR>Cheers,<BR><BR>&nbsp;Jean-Marc<BR></BLOCKQUOTE><BR>___________=
____________________________________<BR>Audio/Video=20
    Transport Working Group<BR><A href=3D"mailto:avt@ietf.org"=20
    target=3D_blank>avt@ietf.org</A><BR><A=20
    href=3D"https://www.ietf.org/mailman/listinfo/avt"=20
    =
target=3D_blank>https://www.ietf.org/mailman/listinfo/avt</A><BR><BR></BL=
OCKQUOTE><BR>_______________________________________________<BR>Audio/Vid=
eo=20
  Transport Working Group<BR><A href=3D"mailto:avt@ietf.org"=20
  target=3D_blank>avt@ietf.org</A><BR><A=20
  href=3D"https://www.ietf.org/mailman/listinfo/avt"=20
  =
target=3D_blank>https://www.ietf.org/mailman/listinfo/avt</A><BR></DIV></=
DIV></BLOCKQUOTE></DIV><BR></BODY></HTML>

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>>> - the quality of the codec may not be competitive

I think its very important that the codec quality be competitive.  People
expect excellence from IETF standards  Standardizing non-competitive codecs
because they are cheap does not seem to be a good choce.

Steve B.

On Tue, Jun 2, 2009 at 12:15 PM, Henning Schulzrinne <hgs@cs.columbia.edu>wrote:

> I view this as a trade-off. If we pursue this, there are risks:
>
> - nothing may come of it since there are no experts willing to help
> - somebody will claim IPR on the resulting work
> - the quality of the codec may not be competitive
>
> However, if we don't do this, we are stuck with the status quo, which is
> not all that satisfactory. Thus, unless there are significant costs for
> "innocent bystanders", I see this as a risk worth taking. In the worst case,
> we are no worse off than we are today. In all other cases, we'll have an
> additional choice for a wideband codec, even if it's not "the best", just
> "good enough". After all, most people use G.711 today, which has a really
> hard time making that claim.
>
> Most real work in the IETF is done by very small teams, typically less than
> 10, so as long as we have a handful of people that are willing to
> contribute, this can work. It might even work better, since you may get
> fewer people who have half-baked opinions - we may skip the binary vs. XML
> debates...
>
> We can set some ground rules ("must be tested with packet loss of 5%") and
> then see what happens. Compared to most network protocols, the likely
> negative impacts (such as security or congestion control issues) of even a
> badly-designed codec are pretty limited.
>
> Henning
>
>
> On Jun 2, 2009, at 11:57 AM, Roni Even wrote:
>
>  Hi,
>> I do not want to sound like someone who is for IPR (I am not), but why
>> stop
>> at codec, let's require it for all IETF work. There are IPR on IETF work
>> which is must simpler, in my view, than wide band audio codecs.
>>
>> I think that we can start with "royalty free" even though I am not sure
>> that
>> it will accepted as part of the charter of any other work group so why
>> pick
>> on codecs which require more work.
>>
>> This leaves the other reasons I heard for doing it at the IETF which is
>> the
>> price of participating (cheaper than being an ITU-T member) and maybe
>> design
>> less expensive characterization tests.
>>
>> Roni
>>
>>
>> -----Original Message-----
>> From: Jean-Marc Valin [mailto:jean-marc.valin@octasic.com]
>> Sent: Tuesday, June 02, 2009 6:14 PM
>> To: Roni Even
>> Cc: 'Slava Borilin'; avt@ietf.org; 'Jason Fischl'; dispatch@ietf.org;
>> hsinnrei@adobe.com
>> Subject: Re: [AVT] [dispatch] Proposal to form Internet Wideband Audio
>> Codec
>> WG
>>
>> Hi,
>>
>> Roni Even wrote:
>>
>>>
>>> I am not sure what prevented you from doing it today at the ITU or
>>> MPEG, why do you see the IETF handling it differently.
>>>
>>> I would also like to remind you and Jean-Marc that once you are
>>> bringing work to a standard body it may require collaboration with
>>> other people who will have other proposals that will also address the
>>> same requirements and you may need to invest money in comparative
>>> testing by independent listening labs.
>>>
>>> I also think that you will need to supply the codec in source code and
>>> provide copy right to the IETF.
>>>
>>>
>> I am well aware that bringing work to the IETF would require
>> collaboration with others. I am not seeking control over the work I am
>> currently doing and would really welcome such collaboration. The idea is
>> only to have the best wideband codec possible without IPR issues. Given
>> the ITU and MPEG track record, I think it would be very unlikely for any
>> of those organisations to work on an IPR-free codec.
>>
>> I also agree with Henry that "the Internet has different criteria than
>> ITU-T networks may have". Internet adoption follows different patterns
>> than adoption in the ITU primary target markets. For example, the
>> Internet has more consumer-reconfigurable software, while the ITU has to
>> deal with a lot of fixed hardware deployments. At the ITU, it makes
>> sense to invest large sums of money into testing and characterisation of
>> codecs because the codecs deployed there usually stay around for a long
>> time and the infrastructure investments are usually very large. On the
>> other hand, I would say the investments in codecs for the Internet are
>> comparatively smaller and, while testing is still important, it is not
>> as critical as it is for the ITU.
>>
>> It's also a choice one has to make. It is unlikely that companies would
>> invest money in expensive testing if they are not going to obtain
>> royalties in return. However, I think we may be able to define some more
>> lightweight (collaborative?) testing that is sufficient and doesn't
>> involve as much investments as what the ITU does. For the Internet, I
>> believe an IPR-free codec that everyone agrees performs well is better
>> than a patent-encumbered codec that has had more extensive testing. This
>> is again another difference with the ITU: patent-encumbered codecs tend
>> to hurt a lot more on the Internet because many applications (e.g.
>> giving away the client) are very hard (or impossible) when one has to
>> pay per-channel royalties.
>>
>> As for the source code issue you pointed out, all the Xiph codecs are
>> already published under a very permissive open source license (BSD), so
>> this would not really change.
>>
>>   Jean-Marc
>>
>>  Roni
>>>
>>>
>>>
>>> *From:* avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] *On Behalf
>>> Of *Slava Borilin
>>> *Sent:* Monday, June 01, 2009 11:50 PM
>>> *To:* jean-marc.valin@octasic.com
>>> *Cc:* avt@ietf.org; Jason Fischl
>>> *Subject:* Re: [AVT] [dispatch] Proposal to form Internet Wideband
>>> Audio Codec WG
>>>
>>>
>>>
>>> Agree with Jean-Marc.
>>> SPIRIT is interested to contribute as well - having a dozen of
>>> proprietary
>>>
>> codecs developed, including
>>
>>> one specifically desgined for internet (SPIRIT IP-MR, which is now under
>>>
>> WGLC on draft-ietf-avt-rtp-ipmr-04) -
>>
>>> multi-rate, scalable, adaptive, wideband codec.
>>>
>>> We can also continue this work with IETF.
>>>
>>> Moreover, most if not all efforts coming from ITU on codecs unfortunately
>>>
>> are NOT really focused on
>>
>>> internet-specific codecs (that's why several companies have had to
>>> invent)
>>>
>> - as ITU preference is mainly
>>
>>> specific (i.e. cellular) networks at first.
>>>
>>> however, as we see the greant rise of pure "internet-basd communications"
>>>
>> (skype, webex/citrix, and many others) -
>>
>>> we all (and users) are suffering from inefficiency in all currently
>>>
>> "standard" codecs and ambiguity in the choice of
>>
>>> internet-targeted ones.
>>>
>>> Again, we probably can put together enough number of contributors to the
>>>
>> WG to have the expertise.
>>
>>>
>>> regards,
>>> Slava Borilin
>>>
>>> --
>>> John Lazzaro wrote:
>>>
>>>   A traditional response to this type of request is to note that the
>>>
>> IETF
>>
>>>
>>>   really doesn't have much in the way of expertise in audio codec
>>>
>> design.
>>
>>>
>>>   I don't see many of the regulars who present at the AES codec paper
>>>
>> sessions
>>
>>>
>>>   posting here on AVT (ditto ICASSP paper sessions for voice codecs).
>>>
>> It's
>>
>>>
>>>   a full-time job to keep up-to-date and contribute to that
>>>
>>>   signal-processing lore.
>>>
>>>
>>>
>>> Well, there's no reason that the IETF cannot build such an expertise
>>> in audio codecs. This is actually something in which I'd be interested
>>> in getting involved and I'm sure others at Xiph.Org would be
>>> interested as well. We have several people with audio codec expertise
>>> from working on Vorbis, Speex and (more recently) CELT. It turns out
>>> that the CELT codec currently under development at Xiph actually meets
>>> most of the requirements from the original proposal in being a very
>>> low delay codec with adaptive bit-rate and sampling rate (up to 48
>>> kHz), scalable complexity, and good robustness to packet loss. We'd be
>>> willing to continue the development with the IETF. Even if not with
>>> CELT, it's still like to be involved in such a new WG.
>>>
>>> Cheers,
>>>
>>>  Jean-Marc
>>>
>>
>> _______________________________________________
>> Audio/Video Transport Working Group
>> avt@ietf.org
>> https://www.ietf.org/mailman/listinfo/avt
>>
>>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
>

--00163641759b5f3983046b60ea82
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

&gt;&gt;&gt; - the quality of the codec may not be competitive<br><br>I thi=
nk its very important that the codec quality be competitive.=A0 People expe=
ct excellence from IETF standards=A0 Standardizing non-competitive codecs b=
ecause they are cheap does not seem to be a good choce.<br>
<br>Steve B.<br><br><div class=3D"gmail_quote">On Tue, Jun 2, 2009 at 12:15=
 PM, Henning Schulzrinne <span dir=3D"ltr">&lt;<a href=3D"mailto:hgs@cs.col=
umbia.edu">hgs@cs.columbia.edu</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, 204, 204); margin=
: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
I view this as a trade-off. If we pursue this, there are risks:<br>
<br>
- nothing may come of it since there are no experts willing to help<br>
- somebody will claim IPR on the resulting work<br>
- the quality of the codec may not be competitive<br>
<br>
However, if we don&#39;t do this, we are stuck with the status quo, which i=
s not all that satisfactory. Thus, unless there are significant costs for &=
quot;innocent bystanders&quot;, I see this as a risk worth taking. In the w=
orst case, we are no worse off than we are today. In all other cases, we&#3=
9;ll have an additional choice for a wideband codec, even if it&#39;s not &=
quot;the best&quot;, just &quot;good enough&quot;. After all, most people u=
se G.711 today, which has a really hard time making that claim.<br>

<br>
Most real work in the IETF is done by very small teams, typically less than=
 10, so as long as we have a handful of people that are willing to contribu=
te, this can work. It might even work better, since you may get fewer peopl=
e who have half-baked opinions - we may skip the binary vs. XML debates...<=
br>

<br>
We can set some ground rules (&quot;must be tested with packet loss of 5%&q=
uot;) and then see what happens. Compared to most network protocols, the li=
kely negative impacts (such as security or congestion control issues) of ev=
en a badly-designed codec are pretty limited.<br>
<font color=3D"#888888">
<br>
Henning</font><div><div></div><div class=3D"h5"><br>
<br>
On Jun 2, 2009, at 11:57 AM, Roni Even wrote:<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hi,<br>
I do not want to sound like someone who is for IPR (I am not), but why stop=
<br>
at codec, let&#39;s require it for all IETF work. There are IPR on IETF wor=
k<br>
which is must simpler, in my view, than wide band audio codecs.<br>
<br>
I think that we can start with &quot;royalty free&quot; even though I am no=
t sure that<br>
it will accepted as part of the charter of any other work group so why pick=
<br>
on codecs which require more work.<br>
<br>
This leaves the other reasons I heard for doing it at the IETF which is the=
<br>
price of participating (cheaper than being an ITU-T member) and maybe desig=
n<br>
less expensive characterization tests.<br>
<br>
Roni<br>
<br>
<br>
-----Original Message-----<br>
From: Jean-Marc Valin [mailto:<a href=3D"mailto:jean-marc.valin@octasic.com=
" target=3D"_blank">jean-marc.valin@octasic.com</a>]<br>
Sent: Tuesday, June 02, 2009 6:14 PM<br>
To: Roni Even<br>
Cc: &#39;Slava Borilin&#39;; <a href=3D"mailto:avt@ietf.org" target=3D"_bla=
nk">avt@ietf.org</a>; &#39;Jason Fischl&#39;; <a href=3D"mailto:dispatch@ie=
tf.org" target=3D"_blank">dispatch@ietf.org</a>;<br>
<a href=3D"mailto:hsinnrei@adobe.com" target=3D"_blank">hsinnrei@adobe.com<=
/a><br>
Subject: Re: [AVT] [dispatch] Proposal to form Internet Wideband Audio Code=
c<br>
WG<br>
<br>
Hi,<br>
<br>
Roni Even wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>
I am not sure what prevented you from doing it today at the ITU or<br>
MPEG, why do you see the IETF handling it differently.<br>
<br>
I would also like to remind you and Jean-Marc that once you are<br>
bringing work to a standard body it may require collaboration with<br>
other people who will have other proposals that will also address the<br>
same requirements and you may need to invest money in comparative<br>
testing by independent listening labs.<br>
<br>
I also think that you will need to supply the codec in source code and<br>
provide copy right to the IETF.<br>
<br>
</blockquote>
<br>
I am well aware that bringing work to the IETF would require<br>
collaboration with others. I am not seeking control over the work I am<br>
currently doing and would really welcome such collaboration. The idea is<br=
>
only to have the best wideband codec possible without IPR issues. Given<br>
the ITU and MPEG track record, I think it would be very unlikely for any<br=
>
of those organisations to work on an IPR-free codec.<br>
<br>
I also agree with Henry that &quot;the Internet has different criteria than=
<br>
ITU-T networks may have&quot;. Internet adoption follows different patterns=
<br>
than adoption in the ITU primary target markets. For example, the<br>
Internet has more consumer-reconfigurable software, while the ITU has to<br=
>
deal with a lot of fixed hardware deployments. At the ITU, it makes<br>
sense to invest large sums of money into testing and characterisation of<br=
>
codecs because the codecs deployed there usually stay around for a long<br>
time and the infrastructure investments are usually very large. On the<br>
other hand, I would say the investments in codecs for the Internet are<br>
comparatively smaller and, while testing is still important, it is not<br>
as critical as it is for the ITU.<br>
<br>
It&#39;s also a choice one has to make. It is unlikely that companies would=
<br>
invest money in expensive testing if they are not going to obtain<br>
royalties in return. However, I think we may be able to define some more<br=
>
lightweight (collaborative?) testing that is sufficient and doesn&#39;t<br>
involve as much investments as what the ITU does. For the Internet, I<br>
believe an IPR-free codec that everyone agrees performs well is better<br>
than a patent-encumbered codec that has had more extensive testing. This<br=
>
is again another difference with the ITU: patent-encumbered codecs tend<br>
to hurt a lot more on the Internet because many applications (e.g.<br>
giving away the client) are very hard (or impossible) when one has to<br>
pay per-channel royalties.<br>
<br>
As for the source code issue you pointed out, all the Xiph codecs are<br>
already published under a very permissive open source license (BSD), so<br>
this would not really change.<br>
<br>
 =A0 Jean-Marc<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Roni<br>
<br>
<br>
<br>
*From:* <a href=3D"mailto:avt-bounces@ietf.org" target=3D"_blank">avt-bounc=
es@ietf.org</a> [mailto:<a href=3D"mailto:avt-bounces@ietf.org" target=3D"_=
blank">avt-bounces@ietf.org</a>] *On Behalf<br>
Of *Slava Borilin<br>
*Sent:* Monday, June 01, 2009 11:50 PM<br>
*To:* <a href=3D"mailto:jean-marc.valin@octasic.com" target=3D"_blank">jean=
-marc.valin@octasic.com</a><br>
*Cc:* <a href=3D"mailto:avt@ietf.org" target=3D"_blank">avt@ietf.org</a>; J=
ason Fischl<br>
*Subject:* Re: [AVT] [dispatch] Proposal to form Internet Wideband<br>
Audio Codec WG<br>
<br>
<br>
<br>
Agree with Jean-Marc.<br>
SPIRIT is interested to contribute as well - having a dozen of proprietary<=
br>
</blockquote>
codecs developed, including<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
one specifically desgined for internet (SPIRIT IP-MR, which is now under<br=
>
</blockquote>
WGLC on draft-ietf-avt-rtp-ipmr-04) -<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
multi-rate, scalable, adaptive, wideband codec.<br>
<br>
We can also continue this work with IETF.<br>
<br>
Moreover, most if not all efforts coming from ITU on codecs unfortunately<b=
r>
</blockquote>
are NOT really focused on<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
internet-specific codecs (that&#39;s why several companies have had to inve=
nt)<br>
</blockquote>
- as ITU preference is mainly<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
specific (i.e. cellular) networks at first.<br>
<br>
however, as we see the greant rise of pure &quot;internet-basd communicatio=
ns&quot;<br>
</blockquote>
(skype, webex/citrix, and many others) -<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
we all (and users) are suffering from inefficiency in all currently<br>
</blockquote>
&quot;standard&quot; codecs and ambiguity in the choice of<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
internet-targeted ones.<br>
<br>
Again, we probably can put together enough number of contributors to the<br=
>
</blockquote>
WG to have the expertise.<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>
regards,<br>
Slava Borilin<br>
<br>
--<br>
John Lazzaro wrote:<br>
<br>
 =A0 A traditional response to this type of request is to note that the<br>
</blockquote>
IETF<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>
 =A0 really doesn&#39;t have much in the way of expertise in audio codec<br=
>
</blockquote>
design.<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>
 =A0 I don&#39;t see many of the regulars who present at the AES codec pape=
r<br>
</blockquote>
sessions<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>
 =A0 posting here on AVT (ditto ICASSP paper sessions for voice codecs).<br=
>
</blockquote>
It&#39;s<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>
 =A0 a full-time job to keep up-to-date and contribute to that<br>
<br>
 =A0 signal-processing lore.<br>
<br>
<br>
<br>
Well, there&#39;s no reason that the IETF cannot build such an expertise<br=
>
in audio codecs. This is actually something in which I&#39;d be interested<=
br>
in getting involved and I&#39;m sure others at Xiph.Org would be<br>
interested as well. We have several people with audio codec expertise<br>
from working on Vorbis, Speex and (more recently) CELT. It turns out<br>
that the CELT codec currently under development at Xiph actually meets<br>
most of the requirements from the original proposal in being a very<br>
low delay codec with adaptive bit-rate and sampling rate (up to 48<br>
kHz), scalable complexity, and good robustness to packet loss. We&#39;d be<=
br>
willing to continue the development with the IETF. Even if not with<br>
CELT, it&#39;s still like to be involved in such a new WG.<br>
<br>
Cheers,<br>
<br>
 =A0Jean-Marc<br>
</blockquote>
<br>
_______________________________________________<br>
Audio/Video Transport Working Group<br>
<a href=3D"mailto:avt@ietf.org" target=3D"_blank">avt@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/avt" target=3D"_blank">htt=
ps://www.ietf.org/mailman/listinfo/avt</a><br>
<br>
</blockquote>
<br>
_______________________________________________<br>
Audio/Video Transport Working Group<br>
<a href=3D"mailto:avt@ietf.org" target=3D"_blank">avt@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/avt" target=3D"_blank">htt=
ps://www.ietf.org/mailman/listinfo/avt</a><br>
</div></div></blockquote></div><br>

--00163641759b5f3983046b60ea82--

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Subject: Re: [dispatch] [AVT] Proposal to form Internet Wideband Audio Codec WG
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>>> i think this is probably false alert.
I would hope so.  I think its important that the characterization testing
(however it is done) prevent the standardization of a codec that is
sub-standard.

So I'd prefer Henning's last risk item to be that "a standardizable codec
might not result".

Steve B.

On Tue, Jun 2, 2009 at 1:33 PM, Slava Borilin <Borilin@spiritdsp.com> wrote:

>  I do not beleive the one that will come wil be low quality.
> at least people from the potential contributors (at least Skype, Speex,
> SPIRIT) are already pretty-good in the commercially exploiting their own
> codecs on the market.
> i think this is probably false alert.
>
> regards,
> Slava Borilin
>
>
>
>  ------------------------------
> *From:* stephen botzko [mailto:stephen.botzko@gmail.com]
> *Sent:* Tuesday, June 02, 2009 9:32 PM
> *To:* Henning Schulzrinne
> *Cc:* Roni Even; dispatch@ietf.org; Jason Fischl; avt@ietf.org;
> hsinnrei@adobe.com; Slava Borilin
>
> *Subject:* Re: [AVT] [dispatch] Proposal to form Internet Wideband Audio
> Codec WG
>
> >>> - the quality of the codec may not be competitive
>
> I think its very important that the codec quality be competitive.  People
> expect excellence from IETF standards  Standardizing non-competitive codecs
> because they are cheap does not seem to be a good choce.
>
> Steve B.
>
> On Tue, Jun 2, 2009 at 12:15 PM, Henning Schulzrinne <hgs@cs.columbia.edu>wrote:
>
>> I view this as a trade-off. If we pursue this, there are risks:
>>
>> - nothing may come of it since there are no experts willing to help
>> - somebody will claim IPR on the resulting work
>> - the quality of the codec may not be competitive
>>
>> However, if we don't do this, we are stuck with the status quo, which is
>> not all that satisfactory. Thus, unless there are significant costs for
>> "innocent bystanders", I see this as a risk worth taking. In the worst case,
>> we are no worse off than we are today. In all other cases, we'll have an
>> additional choice for a wideband codec, even if it's not "the best", just
>> "good enough". After all, most people use G.711 today, which has a really
>> hard time making that claim.
>>
>> Most real work in the IETF is done by very small teams, typically less
>> than 10, so as long as we have a handful of people that are willing to
>> contribute, this can work. It might even work better, since you may get
>> fewer people who have half-baked opinions - we may skip the binary vs. XML
>> debates...
>>
>> We can set some ground rules ("must be tested with packet loss of 5%") and
>> then see what happens. Compared to most network protocols, the likely
>> negative impacts (such as security or congestion control issues) of even a
>> badly-designed codec are pretty limited.
>>
>> Henning
>>
>>
>> On Jun 2, 2009, at 11:57 AM, Roni Even wrote:
>>
>> Hi,
>>> I do not want to sound like someone who is for IPR (I am not), but why
>>> stop
>>> at codec, let's require it for all IETF work. There are IPR on IETF work
>>> which is must simpler, in my view, than wide band audio codecs.
>>>
>>> I think that we can start with "royalty free" even though I am not sure
>>> that
>>> it will accepted as part of the charter of any other work group so why
>>> pick
>>> on codecs which require more work.
>>>
>>> This leaves the other reasons I heard for doing it at the IETF which is
>>> the
>>> price of participating (cheaper than being an ITU-T member) and maybe
>>> design
>>> less expensive characterization tests.
>>>
>>> Roni
>>>
>>>
>>> -----Original Message-----
>>> From: Jean-Marc Valin [mailto:jean-marc.valin@octasic.com]
>>> Sent: Tuesday, June 02, 2009 6:14 PM
>>> To: Roni Even
>>> Cc: 'Slava Borilin'; avt@ietf.org; 'Jason Fischl'; dispatch@ietf.org;
>>> hsinnrei@adobe.com
>>> Subject: Re: [AVT] [dispatch] Proposal to form Internet Wideband Audio
>>> Codec
>>> WG
>>>
>>> Hi,
>>>
>>> Roni Even wrote:
>>>
>>>>
>>>> I am not sure what prevented you from doing it today at the ITU or
>>>> MPEG, why do you see the IETF handling it differently.
>>>>
>>>> I would also like to remind you and Jean-Marc that once you are
>>>> bringing work to a standard body it may require collaboration with
>>>> other people who will have other proposals that will also address the
>>>> same requirements and you may need to invest money in comparative
>>>> testing by independent listening labs.
>>>>
>>>> I also think that you will need to supply the codec in source code and
>>>> provide copy right to the IETF.
>>>>
>>>>
>>> I am well aware that bringing work to the IETF would require
>>> collaboration with others. I am not seeking control over the work I am
>>> currently doing and would really welcome such collaboration. The idea is
>>> only to have the best wideband codec possible without IPR issues. Given
>>> the ITU and MPEG track record, I think it would be very unlikely for any
>>> of those organisations to work on an IPR-free codec.
>>>
>>> I also agree with Henry that "the Internet has different criteria than
>>> ITU-T networks may have". Internet adoption follows different patterns
>>> than adoption in the ITU primary target markets. For example, the
>>> Internet has more consumer-reconfigurable software, while the ITU has to
>>> deal with a lot of fixed hardware deployments. At the ITU, it makes
>>> sense to invest large sums of money into testing and characterisation of
>>> codecs because the codecs deployed there usually stay around for a long
>>> time and the infrastructure investments are usually very large. On the
>>> other hand, I would say the investments in codecs for the Internet are
>>> comparatively smaller and, while testing is still important, it is not
>>> as critical as it is for the ITU.
>>>
>>> It's also a choice one has to make. It is unlikely that companies would
>>> invest money in expensive testing if they are not going to obtain
>>> royalties in return. However, I think we may be able to define some more
>>> lightweight (collaborative?) testing that is sufficient and doesn't
>>> involve as much investments as what the ITU does. For the Internet, I
>>> believe an IPR-free codec that everyone agrees performs well is better
>>> than a patent-encumbered codec that has had more extensive testing. This
>>> is again another difference with the ITU: patent-encumbered codecs tend
>>> to hurt a lot more on the Internet because many applications (e.g.
>>> giving away the client) are very hard (or impossible) when one has to
>>> pay per-channel royalties.
>>>
>>> As for the source code issue you pointed out, all the Xiph codecs are
>>> already published under a very permissive open source license (BSD), so
>>> this would not really change.
>>>
>>>   Jean-Marc
>>>
>>> Roni
>>>>
>>>>
>>>>
>>>> *From:* avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] *On Behalf
>>>> Of *Slava Borilin
>>>> *Sent:* Monday, June 01, 2009 11:50 PM
>>>> *To:* jean-marc.valin@octasic.com
>>>> *Cc:* avt@ietf.org; Jason Fischl
>>>> *Subject:* Re: [AVT] [dispatch] Proposal to form Internet Wideband
>>>> Audio Codec WG
>>>>
>>>>
>>>>
>>>> Agree with Jean-Marc.
>>>> SPIRIT is interested to contribute as well - having a dozen of
>>>> proprietary
>>>>
>>> codecs developed, including
>>>
>>>> one specifically desgined for internet (SPIRIT IP-MR, which is now under
>>>>
>>> WGLC on draft-ietf-avt-rtp-ipmr-04) -
>>>
>>>> multi-rate, scalable, adaptive, wideband codec.
>>>>
>>>> We can also continue this work with IETF.
>>>>
>>>> Moreover, most if not all efforts coming from ITU on codecs
>>>> unfortunately
>>>>
>>> are NOT really focused on
>>>
>>>> internet-specific codecs (that's why several companies have had to
>>>> invent)
>>>>
>>> - as ITU preference is mainly
>>>
>>>> specific (i.e. cellular) networks at first.
>>>>
>>>> however, as we see the greant rise of pure "internet-basd
>>>> communications"
>>>>
>>> (skype, webex/citrix, and many others) -
>>>
>>>> we all (and users) are suffering from inefficiency in all currently
>>>>
>>> "standard" codecs and ambiguity in the choice of
>>>
>>>> internet-targeted ones.
>>>>
>>>> Again, we probably can put together enough number of contributors to the
>>>>
>>> WG to have the expertise.
>>>
>>>>
>>>> regards,
>>>> Slava Borilin
>>>>
>>>> --
>>>> John Lazzaro wrote:
>>>>
>>>>   A traditional response to this type of request is to note that the
>>>>
>>> IETF
>>>
>>>>
>>>>   really doesn't have much in the way of expertise in audio codec
>>>>
>>> design.
>>>
>>>>
>>>>   I don't see many of the regulars who present at the AES codec paper
>>>>
>>> sessions
>>>
>>>>
>>>>   posting here on AVT (ditto ICASSP paper sessions for voice codecs).
>>>>
>>> It's
>>>
>>>>
>>>>   a full-time job to keep up-to-date and contribute to that
>>>>
>>>>   signal-processing lore.
>>>>
>>>>
>>>>
>>>> Well, there's no reason that the IETF cannot build such an expertise
>>>> in audio codecs. This is actually something in which I'd be interested
>>>> in getting involved and I'm sure others at Xiph.Org would be
>>>> interested as well. We have several people with audio codec expertise
>>>> from working on Vorbis, Speex and (more recently) CELT. It turns out
>>>> that the CELT codec currently under development at Xiph actually meets
>>>> most of the requirements from the original proposal in being a very
>>>> low delay codec with adaptive bit-rate and sampling rate (up to 48
>>>> kHz), scalable complexity, and good robustness to packet loss. We'd be
>>>> willing to continue the development with the IETF. Even if not with
>>>> CELT, it's still like to be involved in such a new WG.
>>>>
>>>> Cheers,
>>>>
>>>>  Jean-Marc
>>>>
>>>
>>> _______________________________________________
>>> Audio/Video Transport Working Group
>>> avt@ietf.org
>>> https://www.ietf.org/mailman/listinfo/avt
>>>
>>>
>> _______________________________________________
>> Audio/Video Transport Working Group
>> avt@ietf.org
>> https://www.ietf.org/mailman/listinfo/avt
>>
>
>

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&gt;&gt;&gt; <span><font size=3D"2" color=3D"#0000ff" face=3D"Arial">i thin=
k this is probably false alert.<br>I
would hope so.=A0 I think its important that the characterization testing
(however it is done) prevent the standardization of a codec that is
sub-standard.=A0 <br>
<br>So I&#39;d prefer Henning&#39;s last risk item to be that &quot;a stand=
ardizable codec might not result&quot;.<br><br>Steve B.<br></font></span><b=
r><div class=3D"gmail_quote">On Tue, Jun 2, 2009 at 1:33 PM, Slava Borilin =
<span dir=3D"ltr">&lt;<a href=3D"mailto:Borilin@spiritdsp.com">Borilin@spir=
itdsp.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">



<div>
<div dir=3D"ltr" align=3D"left"><span><font size=3D"2" color=3D"#0000ff" fa=
ce=3D"Arial">I do not beleive the one that will come wil be low=20
quality.</font></span></div>
<div dir=3D"ltr" align=3D"left"><span><font size=3D"2" color=3D"#0000ff" fa=
ce=3D"Arial">at least people from the potential contributors (at least=20
Skype, Speex, SPIRIT) are already pretty-good in the commercially exploitin=
g=20
their own codecs on the market.</font></span></div>
<div dir=3D"ltr" align=3D"left"><span><font size=3D"2" color=3D"#0000ff" fa=
ce=3D"Arial">i think this is probably false alert.</font></span></div>
<div><font size=3D"2" color=3D"#0000ff" face=3D"Arial"></font>=A0</div>
<div align=3D"left"><font size=3D"2" face=3D"Arial">regards,</font></div>
<div align=3D"left"><font size=3D"2" face=3D"Arial">Slava Borilin</font></d=
iv>
<div align=3D"left"><span style=3D"font-size: 10pt; font-family: Arial;" la=
ng=3D"EN-US"></span>=A0</div>
<div>=A0</div><br>
<div dir=3D"ltr" align=3D"left" lang=3D"en-us">
<hr>
<font size=3D"2" face=3D"Tahoma"><b>From:</b> stephen botzko=20
[mailto:<a href=3D"mailto:stephen.botzko@gmail.com" target=3D"_blank">steph=
en.botzko@gmail.com</a>] <br><b>Sent:</b> Tuesday, June 02, 2009 9:32=20
PM<br><b>To:</b> Henning Schulzrinne<br><b>Cc:</b> Roni Even; <a href=3D"ma=
ilto:dispatch@ietf.org" target=3D"_blank">dispatch@ietf.org</a>;=20
Jason Fischl; <a href=3D"mailto:avt@ietf.org" target=3D"_blank">avt@ietf.or=
g</a>; <a href=3D"mailto:hsinnrei@adobe.com" target=3D"_blank">hsinnrei@ado=
be.com</a>; Slava Borilin<div><div></div><div class=3D"h5"><br><b>Subject:<=
/b>=20
Re: [AVT] [dispatch] Proposal to form Internet Wideband Audio Codec=20
WG<br></div></div></font><br></div><div><div></div><div class=3D"h5">
<div></div>&gt;&gt;&gt; - the quality of the codec may not be=20
competitive<br><br>I think its very important that the codec quality be=20
competitive.=A0 People expect excellence from IETF standards=A0=20
Standardizing non-competitive codecs because they are cheap does not seem t=
o be=20
a good choce.<br><br>Steve B.<br><br>
<div class=3D"gmail_quote">On Tue, Jun 2, 2009 at 12:15 PM, Henning Schulzr=
inne=20
<span dir=3D"ltr">&lt;<a href=3D"mailto:hgs@cs.columbia.edu" target=3D"_bla=
nk">hgs@cs.columbia.edu</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">I=20
  view this as a trade-off. If we pursue this, there are risks:<br><br>- no=
thing=20
  may come of it since there are no experts willing to help<br>- somebody w=
ill=20
  claim IPR on the resulting work<br>- the quality of the codec may not be=
=20
  competitive<br><br>However, if we don&#39;t do this, we are stuck with th=
e status=20
  quo, which is not all that satisfactory. Thus, unless there are significa=
nt=20
  costs for &quot;innocent bystanders&quot;, I see this as a risk worth tak=
ing. In the=20
  worst case, we are no worse off than we are today. In all other cases, we=
&#39;ll=20
  have an additional choice for a wideband codec, even if it&#39;s not &quo=
t;the best&quot;,=20
  just &quot;good enough&quot;. After all, most people use G.711 today, whi=
ch has a really=20
  hard time making that claim.<br><br>Most real work in the IETF is done by=
 very=20
  small teams, typically less than 10, so as long as we have a handful of p=
eople=20
  that are willing to contribute, this can work. It might even work better,=
=20
  since you may get fewer people who have half-baked opinions - we may skip=
 the=20
  binary vs. XML debates...<br><br>We can set some ground rules (&quot;must=
 be tested=20
  with packet loss of 5%&quot;) and then see what happens. Compared to most=
 network=20
  protocols, the likely negative impacts (such as security or congestion co=
ntrol=20
  issues) of even a badly-designed codec are pretty limited.<br><font color=
=3D"#888888"><br>Henning</font>
  <div>
  <div></div>
  <div><br><br>On Jun 2, 2009, at 11:57 AM, Roni Even wrote:<br><br>
  <blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204=
, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hi,<br>I=20
    do not want to sound like someone who is for IPR (I am not), but why=20
    stop<br>at codec, let&#39;s require it for all IETF work. There are IPR=
 on IETF=20
    work<br>which is must simpler, in my view, than wide band audio=20
    codecs.<br><br>I think that we can start with &quot;royalty free&quot; =
even though I=20
    am not sure that<br>it will accepted as part of the charter of any othe=
r=20
    work group so why pick<br>on codecs which require more work.<br><br>Thi=
s=20
    leaves the other reasons I heard for doing it at the IETF which is=20
    the<br>price of participating (cheaper than being an ITU-T member) and =
maybe=20
    design<br>less expensive characterization=20
    tests.<br><br>Roni<br><br><br>-----Original Message-----<br>From: Jean-=
Marc=20
    Valin [mailto:<a href=3D"mailto:jean-marc.valin@octasic.com" target=3D"=
_blank">jean-marc.valin@octasic.com</a>]<br>Sent: Tuesday, June 02,=20
    2009 6:14 PM<br>To: Roni Even<br>Cc: &#39;Slava Borilin&#39;; <a href=
=3D"mailto:avt@ietf.org" target=3D"_blank">avt@ietf.org</a>; &#39;Jason Fis=
chl&#39;;=20
    <a href=3D"mailto:dispatch@ietf.org" target=3D"_blank">dispatch@ietf.or=
g</a>;<br><a href=3D"mailto:hsinnrei@adobe.com" target=3D"_blank">hsinnrei@=
adobe.com</a><br>Subject: Re: [AVT] [dispatch]=20
    Proposal to form Internet Wideband Audio Codec<br>WG<br><br>Hi,<br><br>=
Roni=20
    Even wrote:<br>
    <blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(2=
04, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>I=20
      am not sure what prevented you from doing it today at the ITU or<br>M=
PEG,=20
      why do you see the IETF handling it differently.<br><br>I would also =
like=20
      to remind you and Jean-Marc that once you are<br>bringing work to a=
=20
      standard body it may require collaboration with<br>other people who w=
ill=20
      have other proposals that will also address the<br>same requirements =
and=20
      you may need to invest money in comparative<br>testing by independent=
=20
      listening labs.<br><br>I also think that you will need to supply the =
codec=20
      in source code and<br>provide copy right to the=20
    IETF.<br><br></blockquote><br>I am well aware that bringing work to the=
 IETF=20
    would require<br>collaboration with others. I am not seeking control ov=
er=20
    the work I am<br>currently doing and would really welcome such=20
    collaboration. The idea is<br>only to have the best wideband codec poss=
ible=20
    without IPR issues. Given<br>the ITU and MPEG track record, I think it =
would=20
    be very unlikely for any<br>of those organisations to work on an IPR-fr=
ee=20
    codec.<br><br>I also agree with Henry that &quot;the Internet has diffe=
rent=20
    criteria than<br>ITU-T networks may have&quot;. Internet adoption follo=
ws=20
    different patterns<br>than adoption in the ITU primary target markets. =
For=20
    example, the<br>Internet has more consumer-reconfigurable software, whi=
le=20
    the ITU has to<br>deal with a lot of fixed hardware deployments. At the=
 ITU,=20
    it makes<br>sense to invest large sums of money into testing and=20
    characterisation of<br>codecs because the codecs deployed there usually=
 stay=20
    around for a long<br>time and the infrastructure investments are usuall=
y=20
    very large. On the<br>other hand, I would say the investments in codecs=
 for=20
    the Internet are<br>comparatively smaller and, while testing is still=
=20
    important, it is not<br>as critical as it is for the ITU.<br><br>It&#39=
;s also a=20
    choice one has to make. It is unlikely that companies would<br>invest m=
oney=20
    in expensive testing if they are not going to obtain<br>royalties in re=
turn.=20
    However, I think we may be able to define some more<br>lightweight=20
    (collaborative?) testing that is sufficient and doesn&#39;t<br>involve =
as much=20
    investments as what the ITU does. For the Internet, I<br>believe an IPR=
-free=20
    codec that everyone agrees performs well is better<br>than a=20
    patent-encumbered codec that has had more extensive testing. This<br>is=
=20
    again another difference with the ITU: patent-encumbered codecs tend<br=
>to=20
    hurt a lot more on the Internet because many applications (e.g.<br>givi=
ng=20
    away the client) are very hard (or impossible) when one has to<br>pay=
=20
    per-channel royalties.<br><br>As for the source code issue you pointed =
out,=20
    all the Xiph codecs are<br>already published under a very permissive op=
en=20
    source license (BSD), so<br>this would not really change.<br><br>=A0=20
    Jean-Marc<br><br>
    <blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(2=
04, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Roni<br><br><=
br><br>*From:*=20
      <a href=3D"mailto:avt-bounces@ietf.org" target=3D"_blank">avt-bounces=
@ietf.org</a> [mailto:<a href=3D"mailto:avt-bounces@ietf.org" target=3D"_bl=
ank">avt-bounces@ietf.org</a>]=20
      *On Behalf<br>Of *Slava Borilin<br>*Sent:* Monday, June 01, 2009 11:5=
0=20
      PM<br>*To:* <a href=3D"mailto:jean-marc.valin@octasic.com" target=3D"=
_blank">jean-marc.valin@octasic.com</a><br>*Cc:* <a href=3D"mailto:avt@ietf=
.org" target=3D"_blank">avt@ietf.org</a>; Jason=20
      Fischl<br>*Subject:* Re: [AVT] [dispatch] Proposal to form Internet=
=20
      Wideband<br>Audio Codec WG<br><br><br><br>Agree with Jean-Marc.<br>SP=
IRIT=20
      is interested to contribute as well - having a dozen of=20
    proprietary<br></blockquote>codecs developed, including<br>
    <blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(2=
04, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">one=20
      specifically desgined for internet (SPIRIT IP-MR, which is now=20
    under<br></blockquote>WGLC on draft-ietf-avt-rtp-ipmr-04) -<br>
    <blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(2=
04, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">multi-rate,=
=20
      scalable, adaptive, wideband codec.<br><br>We can also continue this =
work=20
      with IETF.<br><br>Moreover, most if not all efforts coming from ITU o=
n=20
      codecs unfortunately<br></blockquote>are NOT really focused on<br>
    <blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(2=
04, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">internet-spec=
ific=20
      codecs (that&#39;s why several companies have had to invent)<br></blo=
ckquote>-=20
    as ITU preference is mainly<br>
    <blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(2=
04, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">specific=20
      (i.e. cellular) networks at first.<br><br>however, as we see the grea=
nt=20
      rise of pure &quot;internet-basd communications&quot;<br></blockquote=
>(skype,=20
    webex/citrix, and many others) -<br>
    <blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(2=
04, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">we=20
      all (and users) are suffering from inefficiency in all=20
    currently<br></blockquote>&quot;standard&quot; codecs and ambiguity in =
the choice=20
    of<br>
    <blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(2=
04, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">internet-targ=
eted=20
      ones.<br><br>Again, we probably can put together enough number of=20
      contributors to the<br></blockquote>WG to have the expertise.<br>
    <blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(2=
04, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>regards,<=
br>Slava=20
      Borilin<br><br>--<br>John Lazzaro wrote:<br><br>=A0 A traditional=20
      response to this type of request is to note that=20
the<br></blockquote>IETF<br>
    <blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(2=
04, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>=A0=20
      really doesn&#39;t have much in the way of expertise in audio=20
    codec<br></blockquote>design.<br>
    <blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(2=
04, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>=A0=20
      I don&#39;t see many of the regulars who present at the AES codec=20
    paper<br></blockquote>sessions<br>
    <blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(2=
04, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>=A0=20
      posting here on AVT (ditto ICASSP paper sessions for voice=20
    codecs).<br></blockquote>It&#39;s<br>
    <blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(2=
04, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>=A0=20
      a full-time job to keep up-to-date and contribute to that<br><br>=A0=
=20
      signal-processing lore.<br><br><br><br>Well, there&#39;s no reason th=
at the=20
      IETF cannot build such an expertise<br>in audio codecs. This is actua=
lly=20
      something in which I&#39;d be interested<br>in getting involved and I=
&#39;m sure=20
      others at Xiph.Org would be<br>interested as well. We have several pe=
ople=20
      with audio codec expertise<br>from working on Vorbis, Speex and (more=
=20
      recently) CELT. It turns out<br>that the CELT codec currently under=
=20
      development at Xiph actually meets<br>most of the requirements from t=
he=20
      original proposal in being a very<br>low delay codec with adaptive=20
      bit-rate and sampling rate (up to 48<br>kHz), scalable complexity, an=
d=20
      good robustness to packet loss. We&#39;d be<br>willing to continue th=
e=20
      development with the IETF. Even if not with<br>CELT, it&#39;s still l=
ike to be=20
      involved in such a new=20
    WG.<br><br>Cheers,<br><br>=A0Jean-Marc<br></blockquote><br>____________=
___________________________________<br>Audio/Video=20
    Transport Working Group<br><a href=3D"mailto:avt@ietf.org" target=3D"_b=
lank">avt@ietf.org</a><br><a href=3D"https://www.ietf.org/mailman/listinfo/=
avt" target=3D"_blank">https://www.ietf.org/mailman/listinfo/avt</a><br><br=
></blockquote>
<br>_______________________________________________<br>Audio/Video=20
  Transport Working Group<br><a href=3D"mailto:avt@ietf.org" target=3D"_bla=
nk">avt@ietf.org</a><br><a href=3D"https://www.ietf.org/mailman/listinfo/av=
t" target=3D"_blank">https://www.ietf.org/mailman/listinfo/avt</a><br></div=
></div>
</blockquote></div><br></div></div></div>
</blockquote></div><br>

--001636b430c8fa044c046b617ead--

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Subject: Re: [dispatch] [AVT] Proposal to form Internet Wideband Audio Codec WG
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On Jun 2, 2009, at 1:31 PM, stephen botzko wrote:

> >>> - the quality of the codec may not be competitive
>
> I think its very important that the codec quality be competitive.   
> People expect excellence from IETF standards  Standardizing non- 
> competitive codecs because they are cheap does not seem to be a good  
> choce.

This is an engineering trade-off. Ogg Vorbis is probably not the  
"best" codec out there, but fills a need.

From stephen.botzko@gmail.com  Tue Jun  2 11:32:27 2009
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Subject: Re: [dispatch] [AVT] Proposal to form Internet Wideband Audio Codec WG
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--0016363ba72076dbd1046b61c3d5
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Of course there are multiple dimensions in the engineering - including
complexity, delay, bitrate, audio bandwidth, and fidelity.

I'm not saying there is a "best" codec, let alone that the IETF should
attempt to build it.

But I think any codec standardized by IETF should be competitive, and
hopefully would have some advantage (not just price) over other available
codecs for its stated application(s).

Steve B.

On Tue, Jun 2, 2009 at 2:12 PM, Henning Schulzrinne <hgs@cs.columbia.edu>wrote:

>
> On Jun 2, 2009, at 1:31 PM, stephen botzko wrote:
>
>  >>> - the quality of the codec may not be competitive
>>
>> I think its very important that the codec quality be competitive.  People
>> expect excellence from IETF standards  Standardizing non-competitive codecs
>> because they are cheap does not seem to be a good choce.
>>
>
> This is an engineering trade-off. Ogg Vorbis is probably not the "best"
> codec out there, but fills a need.
>

--0016363ba72076dbd1046b61c3d5
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Of course there are multiple dimensions in the engineering - including comp=
lexity, delay, bitrate, audio bandwidth, and fidelity. <br><br>I&#39;m not =
saying there is a &quot;best&quot; codec, let alone that the IETF should at=
tempt to build it.=A0 <br>
<br>But I think any codec standardized by IETF should be competitive, and h=
opefully would have some advantage (not just price) over other available co=
decs for its stated application(s).<br><br>Steve B.<br><br><div class=3D"gm=
ail_quote">
On Tue, Jun 2, 2009 at 2:12 PM, Henning Schulzrinne <span dir=3D"ltr">&lt;<=
a href=3D"mailto:hgs@cs.columbia.edu">hgs@cs.columbia.edu</a>&gt;</span> wr=
ote:<br><blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid r=
gb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div class=3D"im"><br>
On Jun 2, 2009, at 1:31 PM, stephen botzko wrote:<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
&gt;&gt;&gt; - the quality of the codec may not be competitive<br>
<br>
I think its very important that the codec quality be competitive. =A0People=
 expect excellence from IETF standards =A0Standardizing non-competitive cod=
ecs because they are cheap does not seem to be a good choce.<br>
</blockquote>
<br></div>
This is an engineering trade-off. Ogg Vorbis is probably not the &quot;best=
&quot; codec out there, but fills a need.<br>
</blockquote></div><br>

--0016363ba72076dbd1046b61c3d5--

From hsinnrei@adobe.com  Tue Jun  2 12:11:00 2009
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From: Henry Sinnreich <hsinnrei@adobe.com>
To: Slava Borilin <Borilin@spiritdsp.com>, stephen botzko <stephen.botzko@gmail.com>, Henning Schulzrinne <hgs@cs.columbia.edu>
Date: Tue, 2 Jun 2009 12:10:38 -0700
Thread-Topic: [AVT] [dispatch] Proposal to form Internet Wideband Audio Codec WG
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Subject: Re: [dispatch] [AVT] Proposal to form Internet Wideband Audio Codec WG
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--_000_C64AE35E3E6Ehsinnreiadobecom_
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>at least people from the potential contributors (at least Skype, Speex, SP=
IRIT) are already pretty-good

As a user of all three codecs and some other as well:
The seem to me clearly among the best technology and in wide-spread use as =
well.

This should put to rest the questions of expertise, quality and availabilit=
y of contributors and reviewers.

Henry


On 6/2/09 12:33 PM, "Slava Borilin" <Borilin@spiritdsp.com> wrote:

I do not beleive the one that will come wil be low quality.
at least people from the potential contributors (at least Skype, Speex, SPI=
RIT) are already pretty-good in the commercially exploiting their own codec=
s on the market.
i think this is probably false alert.

regards,
Slava Borilin



________________________________
From: stephen botzko [mailto:stephen.botzko@gmail.com]
Sent: Tuesday, June 02, 2009 9:32 PM
To: Henning Schulzrinne
Cc: Roni Even; dispatch@ietf.org; Jason Fischl; avt@ietf.org; hsinnrei@adob=
e.com; Slava Borilin
Subject: Re: [AVT] [dispatch] Proposal to form Internet Wideband Audio Code=
c WG

>>> - the quality of the codec may not be competitive

I think its very important that the codec quality be competitive.  People e=
xpect excellence from IETF standards  Standardizing non-competitive codecs =
because they are cheap does not seem to be a good choce.

Steve B.

On Tue, Jun 2, 2009 at 12:15 PM, Henning Schulzrinne <hgs@cs.columbia.edu> =
wrote:
I  view this as a trade-off. If we pursue this, there are risks:

- nothing  may come of it since there are no experts willing to help
- somebody will  claim IPR on the resulting work
- the quality of the codec may not be  competitive

However, if we don't do this, we are stuck with the status  quo, which is n=
ot all that satisfactory. Thus, unless there are significant  costs for "in=
nocent bystanders", I see this as a risk worth taking. In the  worst case, =
we are no worse off than we are today. In all other cases, we'll  have an a=
dditional choice for a wideband codec, even if it's not "the best",  just "=
good enough". After all, most people use G.711 today, which has a really  h=
ard time making that claim.

Most real work in the IETF is done by very  small teams, typically less tha=
n 10, so as long as we have a handful of people  that are willing to contri=
bute, this can work. It might even work better,  since you may get fewer pe=
ople who have half-baked opinions - we may skip the  binary vs. XML debates=
...

We can set some ground rules ("must be tested  with packet loss of 5%") and=
 then see what happens. Compared to most network  protocols, the likely neg=
ative impacts (such as security or congestion control  issues) of even a ba=
dly-designed codec are pretty limited.

Henning




On Jun 2, 2009, at 11:57 AM, Roni Even wrote:


Hi,
I  do not want to sound like someone who is for IPR (I am not), but why  st=
op
at codec, let's require it for all IETF work. There are IPR on IETF  work
which is must simpler, in my view, than wide band audio  codecs.

I think that we can start with "royalty free" even though I  am not sure th=
at
it will accepted as part of the charter of any other  work group so why pic=
k
on codecs which require more work.

This  leaves the other reasons I heard for doing it at the IETF which is  t=
he
price of participating (cheaper than being an ITU-T member) and maybe  desi=
gn
less expensive characterization  tests.

Roni


-----Original Message-----
From: Jean-Marc  Valin [mailto:jean-marc.valin@octasic.com]
Sent: Tuesday, June 02,  2009 6:14 PM
To: Roni Even
Cc: 'Slava Borilin'; avt@ietf.org; 'Jason Fischl';  dispatch@ietf.org;
hsinnrei@adobe.com
Subject: Re: [AVT] [dispatch]  Proposal to form Internet Wideband Audio Cod=
ec
WG

Hi,

Roni  Even wrote:


I  am not sure what prevented you from doing it today at the ITU or
MPEG,  why do you see the IETF handling it differently.

I would also like  to remind you and Jean-Marc that once you are
bringing work to a  standard body it may require collaboration with
other people who will  have other proposals that will also address the
same requirements and  you may need to invest money in comparative
testing by independent  listening labs.

I also think that you will need to supply the codec  in source code and
provide copy right to the  IETF.


I am well aware that bringing work to the IETF  would require
collaboration with others. I am not seeking control over  the work I am
currently doing and would really welcome such  collaboration. The idea is
only to have the best wideband codec possible  without IPR issues. Given
the ITU and MPEG track record, I think it would  be very unlikely for any
of those organisations to work on an IPR-free  codec.

I also agree with Henry that "the Internet has different  criteria than
ITU-T networks may have". Internet adoption follows  different patterns
than adoption in the ITU primary target markets. For  example, the
Internet has more consumer-reconfigurable software, while  the ITU has to
deal with a lot of fixed hardware deployments. At the ITU,  it makes
sense to invest large sums of money into testing and  characterisation of
codecs because the codecs deployed there usually stay  around for a long
time and the infrastructure investments are usually  very large. On the
other hand, I would say the investments in codecs for  the Internet are
comparatively smaller and, while testing is still  important, it is not
as critical as it is for the ITU.

It's also a  choice one has to make. It is unlikely that companies would
invest money  in expensive testing if they are not going to obtain
royalties in return.  However, I think we may be able to define some more
lightweight  (collaborative?) testing that is sufficient and doesn't
involve as much  investments as what the ITU does. For the Internet, I
believe an IPR-free  codec that everyone agrees performs well is better
than a  patent-encumbered codec that has had more extensive testing. This
is  again another difference with the ITU: patent-encumbered codecs tend
to  hurt a lot more on the Internet because many applications (e.g.
giving  away the client) are very hard (or impossible) when one has to
pay  per-channel royalties.

As for the source code issue you pointed out,  all the Xiph codecs are
already published under a very permissive open  source license (BSD), so
this would not really change.

   Jean-Marc


Roni



*From:*  avt-bounces@ietf.org [mailto:avt-bounces@ietf.org]  *On Behalf
Of *Slava Borilin
*Sent:* Monday, June 01, 2009 11:50  PM
*To:* jean-marc.valin@octasic.com
*Cc:* avt@ietf.org; Jason  Fischl
*Subject:* Re: [AVT] [dispatch] Proposal to form Internet  Wideband
Audio Codec WG



Agree with Jean-Marc.
SPIRIT  is interested to contribute as well - having a dozen of  proprietar=
y
codecs developed, including

one  specifically desgined for internet (SPIRIT IP-MR, which is now  under
WGLC on draft-ietf-avt-rtp-ipmr-04) -

multi-rate,  scalable, adaptive, wideband codec.

We can also continue this work  with IETF.

Moreover, most if not all efforts coming from ITU on  codecs unfortunately
are NOT really focused on

internet-specific  codecs (that's why several companies have had to invent)
-  as ITU preference is mainly

specific  (i.e. cellular) networks at first.

however, as we see the greant  rise of pure "internet-basd communications"
(skype,  webex/citrix, and many others) -

we  all (and users) are suffering from inefficiency in all  currently
"standard" codecs and ambiguity in the choice  of

internet-targeted  ones.

Again, we probably can put together enough number of  contributors to the
WG to have the expertise.


regards,
Slava  Borilin

--
John Lazzaro wrote:

  A traditional  response to this type of request is to note that the
IETF


   really doesn't have much in the way of expertise in audio  codec
design.


   I don't see many of the regulars who present at the AES codec  paper
sessions


   posting here on AVT (ditto ICASSP paper sessions for voice  codecs).
It's


   a full-time job to keep up-to-date and contribute to that

   signal-processing lore.



Well, there's no reason that the  IETF cannot build such an expertise
in audio codecs. This is actually  something in which I'd be interested
in getting involved and I'm sure  others at Xiph.Org would be
interested as well. We have several people  with audio codec expertise
from working on Vorbis, Speex and (more  recently) CELT. It turns out
that the CELT codec currently under  development at Xiph actually meets
most of the requirements from the  original proposal in being a very
low delay codec with adaptive  bit-rate and sampling rate (up to 48
kHz), scalable complexity, and  good robustness to packet loss. We'd be
willing to continue the  development with the IETF. Even if not with
CELT, it's still like to be  involved in such a new  WG.

Cheers,

 Jean-Marc

_______________________________________________
Audio/Video  Transport Working Group
avt@ietf.org
https://www.ietf.org/mailman/listinfo/avt


_______________________________________________
Audio/Video  Transport Working Group
avt@ietf.org
https://www.ietf.org/mailman/listinfo/avt



--_000_C64AE35E3E6Ehsinnreiadobecom_
Content-Type: text/html; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<HTML>
<HEAD>
<TITLE>Re: [AVT] [dispatch] Proposal to form Internet Wideband Audio Codec =
WG</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
18pt'>&gt;</SPAN></FONT><SPAN STYLE=3D'font-size:18pt'><FONT COLOR=3D"#0000=
FE"><FONT FACE=3D"Arial">at least people from the potential contributors (a=
t least Skype, Speex, SPIRIT) are already pretty-good<BR>
<BR>
</FONT></FONT><FONT FACE=3D"Arial">As a user of all three codecs and some o=
ther as well:<BR>
The seem to me clearly among the best technology and in wide-spread use as =
well.<BR>
<BR>
This should put to rest the questions of expertise, quality and availabilit=
y of contributors and reviewers.<BR>
<BR>
Henry<BR>
</FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><BR>
<BR>
On 6/2/09 12:33 PM, &quot;Slava Borilin&quot; &lt;<a href=3D"Borilin@spirit=
dsp.com">Borilin@spiritdsp.com</a>&gt; wrote:<BR>
<BR>
</FONT></SPAN><BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT COLOR=3D"#00=
00FF"><FONT FACE=3D"Arial">I do not beleive the one that will come wil be l=
ow quality.<BR>
at least people from the potential contributors (at least Skype, Speex, SPI=
RIT) are already pretty-good in the commercially exploiting their own codec=
s on the market.<BR>
i think this is probably false alert.<BR>
</FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"> <BR>
</FONT><FONT FACE=3D"Arial">regards,<BR>
Slava Borilin<BR>
</FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"> <BR>
&nbsp;<BR>
<BR>
<HR ALIGN=3DCENTER SIZE=3D"3" WIDTH=3D"100%"></FONT><FONT FACE=3D"Tahoma, V=
erdana, Helvetica, Arial"><B>From:</B> stephen botzko [<a href=3D"mailto:st=
ephen.botzko@gmail.com">mailto:stephen.botzko@gmail.com</a>] <BR>
<B>Sent:</B> Tuesday, June 02, 2009 9:32 PM<BR>
<B>To:</B> Henning Schulzrinne<BR>
<B>Cc:</B> Roni Even; <a href=3D"dispatch@ietf.org">dispatch@ietf.org</a>; =
Jason Fischl; <a href=3D"avt@ietf.org">avt@ietf.org</a>; <a href=3D"hsinnre=
i@adobe.com">hsinnrei@adobe.com</a>; Slava Borilin<BR>
<B>Subject:</B> Re: [AVT] [dispatch] Proposal to form Internet Wideband Aud=
io Codec WG<BR>
</FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><BR>
&gt;&gt;&gt; - the quality of the codec may not be competitive<BR>
<BR>
I think its very important that the codec quality be competitive. &nbsp;Peo=
ple expect excellence from IETF standards &nbsp;Standardizing non-competiti=
ve codecs because they are cheap does not seem to be a good choce.<BR>
<BR>
Steve B.<BR>
<BR>
On Tue, Jun 2, 2009 at 12:15 PM, Henning Schulzrinne &lt;<a href=3D"hgs@cs.=
columbia.edu">hgs@cs.columbia.edu</a>&gt; wrote:<BR>
</FONT></SPAN><BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cali=
bri, Verdana, Helvetica, Arial">I &nbsp;view this as a trade-off. If we pur=
sue this, there are risks:<BR>
<BR>
- nothing &nbsp;may come of it since there are no experts willing to help<B=
R>
- somebody will &nbsp;claim IPR on the resulting work<BR>
- the quality of the codec may not be &nbsp;competitive<BR>
<BR>
However, if we don't do this, we are stuck with the status &nbsp;quo, which=
 is not all that satisfactory. Thus, unless there are significant &nbsp;cos=
ts for &quot;innocent bystanders&quot;, I see this as a risk worth taking. =
In the &nbsp;worst case, we are no worse off than we are today. In all othe=
r cases, we'll &nbsp;have an additional choice for a wideband codec, even i=
f it's not &quot;the best&quot;, &nbsp;just &quot;good enough&quot;. After =
all, most people use G.711 today, which has a really &nbsp;hard time making=
 that claim.<BR>
<BR>
Most real work in the IETF is done by very &nbsp;small teams, typically les=
s than 10, so as long as we have a handful of people &nbsp;that are willing=
 to contribute, this can work. It might even work better, &nbsp;since you m=
ay get fewer people who have half-baked opinions - we may skip the &nbsp;bi=
nary vs. XML debates...<BR>
<BR>
We can set some ground rules (&quot;must be tested &nbsp;with packet loss o=
f 5%&quot;) and then see what happens. Compared to most network &nbsp;proto=
cols, the likely negative impacts (such as security or congestion control &=
nbsp;issues) of even a badly-designed codec are pretty limited.<BR>
<FONT COLOR=3D"#888888"><BR>
Henning</FONT> <BR>
&nbsp;<BR>
&nbsp;<BR>
<BR>
<BR>
On Jun 2, 2009, at 11:57 AM, Roni Even wrote:<BR>
<BR>
&nbsp;<BR>
</FONT></SPAN><BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cali=
bri, Verdana, Helvetica, Arial">Hi,<BR>
I &nbsp;do not want to sound like someone who is for IPR (I am not), but wh=
y &nbsp;stop<BR>
at codec, let's require it for all IETF work. There are IPR on IETF &nbsp;w=
ork<BR>
which is must simpler, in my view, than wide band audio &nbsp;codecs.<BR>
<BR>
I think that we can start with &quot;royalty free&quot; even though I &nbsp=
;am not sure that<BR>
it will accepted as part of the charter of any other &nbsp;work group so wh=
y pick<BR>
on codecs which require more work.<BR>
<BR>
This &nbsp;leaves the other reasons I heard for doing it at the IETF which =
is &nbsp;the<BR>
price of participating (cheaper than being an ITU-T member) and maybe &nbsp=
;design<BR>
less expensive characterization &nbsp;tests.<BR>
<BR>
Roni<BR>
<BR>
<BR>
-----Original Message-----<BR>
From: Jean-Marc &nbsp;Valin [<a href=3D"mailto:jean-marc.valin@octasic.com"=
>mailto:jean-marc.valin@octasic.com</a>]<BR>
Sent: Tuesday, June 02, &nbsp;2009 6:14 PM<BR>
To: Roni Even<BR>
Cc: 'Slava Borilin'; <a href=3D"avt@ietf.org">avt@ietf.org</a>; 'Jason Fisc=
hl'; &nbsp;<a href=3D"dispatch@ietf.org">dispatch@ietf.org</a>;<BR>
<a href=3D"hsinnrei@adobe.com">hsinnrei@adobe.com</a><BR>
Subject: Re: [AVT] [dispatch] &nbsp;Proposal to form Internet Wideband Audi=
o Codec<BR>
WG<BR>
<BR>
Hi,<BR>
<BR>
Roni &nbsp;Even wrote:<BR>
&nbsp;<BR>
</FONT></SPAN><BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cali=
bri, Verdana, Helvetica, Arial"><BR>
I &nbsp;am not sure what prevented you from doing it today at the ITU or<BR=
>
MPEG, &nbsp;why do you see the IETF handling it differently.<BR>
<BR>
I would also like &nbsp;to remind you and Jean-Marc that once you are<BR>
bringing work to a &nbsp;standard body it may require collaboration with<BR=
>
other people who will &nbsp;have other proposals that will also address the=
<BR>
same requirements and &nbsp;you may need to invest money in comparative<BR>
testing by independent &nbsp;listening labs.<BR>
<BR>
I also think that you will need to supply the codec &nbsp;in source code an=
d<BR>
provide copy right to the &nbsp;IETF.<BR>
<BR>
</FONT></SPAN></BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cal=
ibri, Verdana, Helvetica, Arial"><BR>
I am well aware that bringing work to the IETF &nbsp;would require<BR>
collaboration with others. I am not seeking control over &nbsp;the work I a=
m<BR>
currently doing and would really welcome such &nbsp;collaboration. The idea=
 is<BR>
only to have the best wideband codec possible &nbsp;without IPR issues. Giv=
en<BR>
the ITU and MPEG track record, I think it would &nbsp;be very unlikely for =
any<BR>
of those organisations to work on an IPR-free &nbsp;codec.<BR>
<BR>
I also agree with Henry that &quot;the Internet has different &nbsp;criteri=
a than<BR>
ITU-T networks may have&quot;. Internet adoption follows &nbsp;different pa=
tterns<BR>
than adoption in the ITU primary target markets. For &nbsp;example, the<BR>
Internet has more consumer-reconfigurable software, while &nbsp;the ITU has=
 to<BR>
deal with a lot of fixed hardware deployments. At the ITU, &nbsp;it makes<B=
R>
sense to invest large sums of money into testing and &nbsp;characterisation=
 of<BR>
codecs because the codecs deployed there usually stay &nbsp;around for a lo=
ng<BR>
time and the infrastructure investments are usually &nbsp;very large. On th=
e<BR>
other hand, I would say the investments in codecs for &nbsp;the Internet ar=
e<BR>
comparatively smaller and, while testing is still &nbsp;important, it is no=
t<BR>
as critical as it is for the ITU.<BR>
<BR>
It's also a &nbsp;choice one has to make. It is unlikely that companies wou=
ld<BR>
invest money &nbsp;in expensive testing if they are not going to obtain<BR>
royalties in return. &nbsp;However, I think we may be able to define some m=
ore<BR>
lightweight &nbsp;(collaborative?) testing that is sufficient and doesn't<B=
R>
involve as much &nbsp;investments as what the ITU does. For the Internet, I=
<BR>
believe an IPR-free &nbsp;codec that everyone agrees performs well is bette=
r<BR>
than a &nbsp;patent-encumbered codec that has had more extensive testing. T=
his<BR>
is &nbsp;again another difference with the ITU: patent-encumbered codecs te=
nd<BR>
to &nbsp;hurt a lot more on the Internet because many applications (e.g.<BR=
>
giving &nbsp;away the client) are very hard (or impossible) when one has to=
<BR>
pay &nbsp;per-channel royalties.<BR>
<BR>
As for the source code issue you pointed out, &nbsp;all the Xiph codecs are=
<BR>
already published under a very permissive open &nbsp;source license (BSD), =
so<BR>
this would not really change.<BR>
<BR>
&nbsp;&nbsp;&nbsp;Jean-Marc<BR>
<BR>
&nbsp;<BR>
</FONT></SPAN><BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cali=
bri, Verdana, Helvetica, Arial">Roni<BR>
<BR>
<BR>
<BR>
*From:* &nbsp;<a href=3D"avt-bounces@ietf.org">avt-bounces@ietf.org</a> [<a=
 href=3D"mailto:avt-bounces@ietf.org">mailto:avt-bounces@ietf.org</a>] &nbs=
p;*On Behalf<BR>
Of *Slava Borilin<BR>
*Sent:* Monday, June 01, 2009 11:50 &nbsp;PM<BR>
*To:* <a href=3D"jean-marc.valin@octasic.com">jean-marc.valin@octasic.com</=
a><BR>
*Cc:* <a href=3D"avt@ietf.org">avt@ietf.org</a>; Jason &nbsp;Fischl<BR>
*Subject:* Re: [AVT] [dispatch] Proposal to form Internet &nbsp;Wideband<BR=
>
Audio Codec WG<BR>
<BR>
<BR>
<BR>
Agree with Jean-Marc.<BR>
SPIRIT &nbsp;is interested to contribute as well - having a dozen of &nbsp;=
proprietary<BR>
</FONT></SPAN></BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cal=
ibri, Verdana, Helvetica, Arial">codecs developed, including<BR>
&nbsp;<BR>
</FONT></SPAN><BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cali=
bri, Verdana, Helvetica, Arial">one &nbsp;specifically desgined for interne=
t (SPIRIT IP-MR, which is now &nbsp;under<BR>
</FONT></SPAN></BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cal=
ibri, Verdana, Helvetica, Arial">WGLC on draft-ietf-avt-rtp-ipmr-04) -<BR>
&nbsp;<BR>
</FONT></SPAN><BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cali=
bri, Verdana, Helvetica, Arial">multi-rate, &nbsp;scalable, adaptive, wideb=
and codec.<BR>
<BR>
We can also continue this work &nbsp;with IETF.<BR>
<BR>
Moreover, most if not all efforts coming from ITU on &nbsp;codecs unfortuna=
tely<BR>
</FONT></SPAN></BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cal=
ibri, Verdana, Helvetica, Arial">are NOT really focused on<BR>
&nbsp;<BR>
</FONT></SPAN><BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cali=
bri, Verdana, Helvetica, Arial">internet-specific &nbsp;codecs (that's why =
several companies have had to invent)<BR>
</FONT></SPAN></BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cal=
ibri, Verdana, Helvetica, Arial">- &nbsp;as ITU preference is mainly<BR>
&nbsp;<BR>
</FONT></SPAN><BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cali=
bri, Verdana, Helvetica, Arial">specific &nbsp;(i.e. cellular) networks at =
first.<BR>
<BR>
however, as we see the greant &nbsp;rise of pure &quot;internet-basd commun=
ications&quot;<BR>
</FONT></SPAN></BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cal=
ibri, Verdana, Helvetica, Arial">(skype, &nbsp;webex/citrix, and many other=
s) -<BR>
&nbsp;<BR>
</FONT></SPAN><BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cali=
bri, Verdana, Helvetica, Arial">we &nbsp;all (and users) are suffering from=
 inefficiency in all &nbsp;currently<BR>
</FONT></SPAN></BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cal=
ibri, Verdana, Helvetica, Arial">&quot;standard&quot; codecs and ambiguity =
in the choice &nbsp;of<BR>
&nbsp;<BR>
</FONT></SPAN><BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cali=
bri, Verdana, Helvetica, Arial">internet-targeted &nbsp;ones.<BR>
<BR>
Again, we probably can put together enough number of &nbsp;contributors to =
the<BR>
</FONT></SPAN></BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cal=
ibri, Verdana, Helvetica, Arial">WG to have the expertise.<BR>
&nbsp;<BR>
</FONT></SPAN><BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cali=
bri, Verdana, Helvetica, Arial"><BR>
regards,<BR>
Slava &nbsp;Borilin<BR>
<BR>
--<BR>
John Lazzaro wrote:<BR>
<BR>
&nbsp;&nbsp;A traditional &nbsp;response to this type of request is to note=
 that the<BR>
</FONT></SPAN></BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cal=
ibri, Verdana, Helvetica, Arial">IETF<BR>
&nbsp;<BR>
</FONT></SPAN><BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cali=
bri, Verdana, Helvetica, Arial"><BR>
&nbsp;&nbsp;&nbsp;really doesn't have much in the way of expertise in audio=
 &nbsp;codec<BR>
</FONT></SPAN></BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cal=
ibri, Verdana, Helvetica, Arial">design.<BR>
&nbsp;<BR>
</FONT></SPAN><BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cali=
bri, Verdana, Helvetica, Arial"><BR>
&nbsp;&nbsp;&nbsp;I don't see many of the regulars who present at the AES c=
odec &nbsp;paper<BR>
</FONT></SPAN></BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cal=
ibri, Verdana, Helvetica, Arial">sessions<BR>
&nbsp;<BR>
</FONT></SPAN><BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cali=
bri, Verdana, Helvetica, Arial"><BR>
&nbsp;&nbsp;&nbsp;posting here on AVT (ditto ICASSP paper sessions for voic=
e &nbsp;codecs).<BR>
</FONT></SPAN></BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cal=
ibri, Verdana, Helvetica, Arial">It's<BR>
&nbsp;<BR>
</FONT></SPAN><BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cali=
bri, Verdana, Helvetica, Arial"><BR>
&nbsp;&nbsp;&nbsp;a full-time job to keep up-to-date and contribute to that=
<BR>
<BR>
&nbsp;&nbsp;&nbsp;signal-processing lore.<BR>
<BR>
<BR>
<BR>
Well, there's no reason that the &nbsp;IETF cannot build such an expertise<=
BR>
in audio codecs. This is actually &nbsp;something in which I'd be intereste=
d<BR>
in getting involved and I'm sure &nbsp;others at Xiph.Org would be<BR>
interested as well. We have several people &nbsp;with audio codec expertise=
<BR>
from working on Vorbis, Speex and (more &nbsp;recently) CELT. It turns out<=
BR>
that the CELT codec currently under &nbsp;development at Xiph actually meet=
s<BR>
most of the requirements from the &nbsp;original proposal in being a very<B=
R>
low delay codec with adaptive &nbsp;bit-rate and sampling rate (up to 48<BR=
>
kHz), scalable complexity, and &nbsp;good robustness to packet loss. We'd b=
e<BR>
willing to continue the &nbsp;development with the IETF. Even if not with<B=
R>
CELT, it's still like to be &nbsp;involved in such a new &nbsp;WG.<BR>
<BR>
Cheers,<BR>
<BR>
&nbsp;Jean-Marc<BR>
</FONT></SPAN></BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cal=
ibri, Verdana, Helvetica, Arial"><BR>
_______________________________________________<BR>
Audio/Video &nbsp;Transport Working Group<BR>
<a href=3D"avt@ietf.org">avt@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/avt">https://www.ietf.org/=
mailman/listinfo/avt</a><BR>
<BR>
</FONT></SPAN></BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cal=
ibri, Verdana, Helvetica, Arial"><BR>
_______________________________________________<BR>
Audio/Video &nbsp;Transport Working Group<BR>
<a href=3D"avt@ietf.org">avt@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/avt">https://www.ietf.org/=
mailman/listinfo/avt</a><BR>
</FONT></SPAN></BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cal=
ibri, Verdana, Helvetica, Arial"><BR>
<BR>
</FONT></SPAN></BLOCKQUOTE>
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Subject: Re: [dispatch] [AVT] Proposal to form Internet Wideband Audio Codec	WG
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It would also be hard to 'object' to transport work being done in 
traditional 'codec shops' (such as the ITU and MPEG) if this goes 
ahead.

A quality codec standard is usually backed by
* testing
* conformance files
* reference software

and a fairly rigorous design process.  When SMPTE took on VC-1 from 
Microsoft, I think both they and Microsoft were surprised by the 
amount of work involved.

FWIW, 3GPP SA4 is also a place with interest in codecs for packet 
networks, and has standardized codecs in the past.  Opening a new 
work item there is not trivial, but can be done.
-- 
David Singer
Multimedia Standards, Apple Inc.

From singer@apple.com  Tue Jun  2 12:32:15 2009
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Date: Tue, 2 Jun 2009 12:31:53 -0700
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Subject: Re: [dispatch] [AVT] Proposal to form Internet Wideband Audio Codec	WG
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At 15:23  -0400 2/06/09, Randell Jesup wrote:
>  >FWIW, 3GPP SA4 is also a place with interest in codecs for packet networks,
>>and has standardized codecs in the past.  Opening a new work item there is
>>not trivial, but can be done.
>
>Again, I imagine even getting involved in 3GPP is tough with many hoops,
>let alone getting the rest of them to sanction a new codec not designed for
>radio use specifically.  You can't even directly apply to join; you have to
>go through an "Organizational Partner" like ATIS.  Not going to happen here...

The only hoop is an organizational partner.  For large companies, it 
can be expensive (e.g. ETSI sets your dues based on your telecoms 
related revenue, if I recall correctly).

On a spectrum of desirability of process etc., I'd guess that
* codecs from international bodies with a procedure and track record 
for doing them thoroughly, are preferred (MPEG, ITU, 3GPP)
* codecs from international bodies not normally in the codec field 
offer at least a stable specification and the opportunity for 
critique and refinement, leading to that specification (e.g. IETF)
* codecs developed in open-source  are at least open to anyone, but 
they have the issue that they lack a formal reference, a formal 
consensus process, a stable publication 'home', IPR rules for 
contributors, and so on
* commercial codecs have the backing of a company, and generally 
stability, interop, and so on are well managed (if they want to 
succeed), but are typically confidential and 'take it or leave it'.


And yes, it gets a little cheaper each time we move down a rung here, 
which is the trade-off.
-- 
David Singer
Multimedia Standards, Apple Inc.

From rjesup@wgate.com  Tue Jun  2 12:23:00 2009
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From: Randell Jesup <rjesup@wgate.com>
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Subject: Re: [dispatch] [AVT] Proposal to form Internet Wideband Audio Codec	WG
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David Singer <singer@apple.com> writes:
>It would also be hard to 'object' to transport work being done in
>traditional 'codec shops' (such as the ITU and MPEG) if this goes ahead.
>
>A quality codec standard is usually backed by
>* testing
>* conformance files
>* reference software
>
>and a fairly rigorous design process.  When SMPTE took on VC-1 from
>Microsoft, I think both they and Microsoft were surprised by the amount of
>work involved.

All relevant and indicative of the level of work required, but not
disqualifying I think.

>FWIW, 3GPP SA4 is also a place with interest in codecs for packet networks,
>and has standardized codecs in the past.  Opening a new work item there is
>not trivial, but can be done.

Again, I imagine even getting involved in 3GPP is tough with many hoops,
let alone getting the rest of them to sanction a new codec not designed for
radio use specifically.  You can't even directly apply to join; you have to
go through an "Organizational Partner" like ATIS.  Not going to happen here...

-- 
Randell Jesup, Worldgate (developers of the Ojo videophone), ex-Amiga OS team
rjesup@wgate.com
"The fetters imposed on liberty at home have ever been forged out of the weapons
provided for defence against real, pretended, or imaginary dangers from abroad."
		- James Madison, 4th US president (1751-1836)


From Borilin@spiritdsp.com  Tue Jun  2 13:28:24 2009
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From: "Slava Borilin" <Borilin@spiritdsp.com>
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Subject: Re: [dispatch] [AVT] Proposal to form Internet Wideband Audio Codec	WG
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David-

SA4 is definitely the option, but they are focused, again, on LTE only,
around EVRC family, so they has a differetnt roots and different agendas
from IETF now.

regards,
Slava Borilin
=20

-----Original Message-----
From: David Singer [mailto:singer@apple.com]=20
Sent: Tuesday, June 02, 2009 10:37 PM
To: Henning Schulzrinne; stephen botzko
Cc: dispatch@ietf.org; Jason Fischl; avt@ietf.org; Roni Even;
hsinnrei@adobe.com; Slava Borilin
Subject: Re: [AVT] [dispatch] Proposal to form Internet Wideband Audio
Codec WG

It would also be hard to 'object' to transport work being done in
traditional 'codec shops' (such as the ITU and MPEG) if this goes ahead.

A quality codec standard is usually backed by
* testing
* conformance files
* reference software

and a fairly rigorous design process.  When SMPTE took on VC-1 from
Microsoft, I think both they and Microsoft were surprised by the amount
of work involved.

FWIW, 3GPP SA4 is also a place with interest in codecs for packet
networks, and has standardized codecs in the past.  Opening a new work
item there is not trivial, but can be done.
--
David Singer
Multimedia Standards, Apple Inc.

From eburger@standardstrack.com  Tue Jun  2 18:50:37 2009
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Subject: Re: [dispatch] [AVT] Proposal to form Internet Wideband Audio Codec WG
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To directly answer Henry, having a mail list and the initial (and  
possibly detailed) discussion hosted at the IETF costs less than the  
discussion we are having here.  If a codec comes out of it, without  
having a BOF or WG, that's great!  If not, at least there was a forum  
for doing it.

I can also offer a SIP Forum list if people still really, really don't  
like the idea of even discussing the topic under even the most  
faintest of auspices of the IETF.

If you think I am schizophrenic by offering we can have a discussion  
in the IETF and at the same time I doubt we could ever have a formal  
work group to create the codec, you would be only slightly  
correct :-)  But seriously: the IETF (AVT in particular) can pickup  
the wire format for a codec, once the codec exists. Having a mail list  
WITHOUT A WORK GROUP give people who want to work on this a venue to  
do the work, without burdening the AD's, IESG, and the rest of the  
community. If a codec pops out, then we can run with it.

On Jun 2, 2009, at 12:44 PM, Spencer Dawkins wrote:

> I am not an area director, nor do i play one on IPTV, but my  
> suggestion for anyone interested would be to
>
> - request a non-WG mailing list (from either of the RAI area  
> directors),
>
> - announce it on this mailing list (and others that seem to be  
> appropriate, and
>
> - start discussions, which the area directors typically look at when  
> judging whether there's enough interest and enough clue to justify  
> approving a BOF and/or chartering a working group.
>
> I didn't see a mailing list included in Jason's proposal - if I  
> missed one, my apologies.
>
> If people at the IETF can do this work, and want to do this work,  
> IETF process policies should not get in the way of that happening.
>
> And I haven't heard of an AD turning down a request for a non-WG  
> mailing list yet, keeping in mind that one of said lists is ietf- 
> sailors, for people interested in sailing to/from/at IETF meetings -  
> the bar is (appropriately) low.
>
> Thanks,
>
> Spencer
> ----- Original Message -----
> From: Henry Sinnreich
> To: Roni Even ; Jean-Marc Valin
> Cc: dispatch@ietf.org ; Slava Borilin ; avt@ietf.org
> Sent: Tuesday, June 02, 2009 11:12 AM
> Subject: Re: [dispatch] [AVT] Proposal to form Internet Wideband  
> Audio Codec WG
>
> >This leaves the other reasons I heard for doing it at the IETF  
> which is the
> >price of participating
>
> This is an important point.
> Given the travel constraints for some of the most valuable potential  
> contributors and reviewers, could we envisage online work until the  
> economy improves, so that an eventual BOF to start with should not  
> be starved of attendees? An online BOF?
>
> There are several free online meeting tools available...
>
> How can this be done within the IETF policy?
>
> Henry
>
>
> On 6/2/09 10:57 AM, "Roni Even" <Even.roni@huawei.com> wrote:
>
> Hi,
> I do not want to sound like someone who is for IPR (I am not), but  
> why stop
> at codec, let's require it for all IETF work. There are IPR on IETF  
> work
> which is must simpler, in my view, than wide band audio codecs.
>
> I think that we can start with "royalty free" even though I am not  
> sure that
> it will accepted as part of the charter of any other work group so  
> why pick
> on codecs which require more work.
>
> This leaves the other reasons I heard for doing it at the IETF which  
> is the
> price of participating (cheaper than being an ITU-T member) and  
> maybe design
> less expensive characterization tests.
>
> Roni
>
>
> -----Original Message-----
> From: Jean-Marc Valin [mailto:jean-marc.valin@octasic.com]
> Sent: Tuesday, June 02, 2009 6:14 PM
> To: Roni Even
> Cc: 'Slava Borilin'; avt@ietf.org; 'Jason Fischl';dispatch@ietf.org;
> hsinnrei@adobe.com
> Subject: Re: [AVT] [dispatch] Proposal to form Internet Wideband  
> Audio Codec
> WG
>
> Hi,
>
> Roni Even wrote:
> >
> > I am not sure what prevented you from doing it today at the ITU or
> > MPEG, why do you see the IETF handling it differently.
> >
> > I would also like to remind you and Jean-Marc that once you are
> > bringing work to a standard body it may require collaboration with
> > other people who will have other proposals that will also address  
> the
> > same requirements and you may need to invest money in comparative
> > testing by independent listening labs.
> >
> > I also think that you will need to supply the codec in source code  
> and
> > provide copy right to the IETF.
> >
>
> I am well aware that bringing work to the IETF would require
> collaboration with others. I am not seeking control over the work I am
> currently doing and would really welcome such collaboration. The  
> idea is
> only to have the best wideband codec possible without IPR issues.  
> Given
> the ITU and MPEG track record, I think it would be very unlikely for  
> any
> of those organisations to work on an IPR-free codec.
>
> I also agree with Henry that "the Internet has different criteria than
> ITU-T networks may have". Internet adoption follows different patterns
> than adoption in the ITU primary target markets. For example, the
> Internet has more consumer-reconfigurable software, while the ITU  
> has to
> deal with a lot of fixed hardware deployments. At the ITU, it makes
> sense to invest large sums of money into testing and  
> characterisation of
> codecs because the codecs deployed there usually stay around for a  
> long
> time and the infrastructure investments are usually very large. On the
> other hand, I would say the investments in codecs for the Internet are
> comparatively smaller and, while testing is still important, it is not
> as critical as it is for the ITU.
>
> It's also a choice one has to make. It is unlikely that companies  
> would
> invest money in expensive testing if they are not going to obtain
> royalties in return. However, I think we may be able to define some  
> more
> lightweight (collaborative?) testing that is sufficient and doesn't
> involve as much investments as what the ITU does. For the Internet, I
> believe an IPR-free codec that everyone agrees performs well is better
> than a patent-encumbered codec that has had more extensive testing.  
> This
> is again another difference with the ITU: patent-encumbered codecs  
> tend
> to hurt a lot more on the Internet because many applications (e.g.
> giving away the client) are very hard (or impossible) when one has to
> pay per-channel royalties.
>
> As for the source code issue you pointed out, all the Xiph codecs are
> already published under a very permissive open source license (BSD),  
> so
> this would not really change.
>
>     Jean-Marc
>
> > Roni
> >
> >
> >
> > *From:* avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] *On  
> Behalf
> > Of *Slava Borilin
> > *Sent:* Monday, June 01, 2009 11:50 PM
> > *To:* jean-marc.valin@octasic.com
> > *Cc:* avt@ietf.org; Jason Fischl
> > *Subject:* Re: [AVT] [dispatch] Proposal to form Internet Wideband
> > Audio Codec WG
> >
> >
> >
> > Agree with Jean-Marc.
> > SPIRIT is interested to contribute as well - having a dozen of  
> proprietary
> codecs developed, including
> > one specifically desgined for internet (SPIRIT IP-MR, which is now  
> under
> WGLC on draft-ietf-avt-rtp-ipmr-04) -
> > multi-rate, scalable, adaptive, wideband codec.
> >
> > We can also continue this work with IETF.
> >
> > Moreover, most if not all efforts coming from ITU on codecs  
> unfortunately
> are NOT really focused on
> > internet-specific codecs (that's why several companies have had to  
> invent)
> - as ITU preference is mainly
> > specific (i.e. cellular) networks at first.
> >
> > however, as we see the greant rise of pure "internet-basd  
> communications"
> (skype, webex/citrix, and many others) -
> > we all (and users) are suffering from inefficiency in all currently
> "standard" codecs and ambiguity in the choice of
> > internet-targeted ones.
> >
> > Again, we probably can put together enough number of contributors  
> to the
> WG to have the expertise.
> >
> > regards,
> > Slava Borilin
> >
> > --
> > John Lazzaro wrote:
> >
> >     A traditional response to this type of request is to note that  
> the
> IETF
> >
> >     really doesn't have much in the way of expertise in audio codec
> design.
> >
> >     I don't see many of the regulars who present at the AES codec  
> paper
> sessions
> >
> >     posting here on AVT (ditto ICASSP paper sessions for voice  
> codecs).
> It's
> >
> >     a full-time job to keep up-to-date and contribute to that
> >
> >     signal-processing lore.
> >
> >
> >
> > Well, there's no reason that the IETF cannot build such an expertise
> > in audio codecs. This is actually something in which I'd be  
> interested
> > in getting involved and I'm sure others at Xiph.Org would be
> > interested as well. We have several people with audio codec  
> expertise
> > from working on Vorbis, Speex and (more recently) CELT. It turns out
> > that the CELT codec currently under development at Xiph actually  
> meets
> > most of the requirements from the original proposal in being a very
> > low delay codec with adaptive bit-rate and sampling rate (up to 48
> > kHz), scalable complexity, and good robustness to packet loss.  
> We'd be
> > willing to continue the development with the IETF. Even if not with
> > CELT, it's still like to be involved in such a new WG.
> >
> > Cheers,
> >
> >    Jean-Marc
>
>
>
>
> _______________________________________________
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> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
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Subject: Re: [dispatch] [AVT] Proposal to form Internet Wideband Audio Codec WG
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I would strike
> - somebody will claim IPR on the resulting work
from the list of risks. The *resulting* work would clearly be  
predicated by both prior art and Note Well.

On Jun 2, 2009, at 12:15 PM, Henning Schulzrinne wrote:

> I view this as a trade-off. If we pursue this, there are risks:
>
> - nothing may come of it since there are no experts willing to help
> - somebody will claim IPR on the resulting work
> - the quality of the codec may not be competitive
>
> However, if we don't do this, we are stuck with the status quo,  
> which is not all that satisfactory. Thus, unless there are  
> significant costs for "innocent bystanders", I see this as a risk  
> worth taking. In the worst case, we are no worse off than we are  
> today. In all other cases, we'll have an additional choice for a  
> wideband codec, even if it's not "the best", just "good enough".  
> After all, most people use G.711 today, which has a really hard time  
> making that claim.
>
> Most real work in the IETF is done by very small teams, typically  
> less than 10, so as long as we have a handful of people that are  
> willing to contribute, this can work. It might even work better,  
> since you may get fewer people who have half-baked opinions - we may  
> skip the binary vs. XML debates...
>
> We can set some ground rules ("must be tested with packet loss of  
> 5%") and then see what happens. Compared to most network protocols,  
> the likely negative impacts (such as security or congestion control  
> issues) of even a badly-designed codec are pretty limited.
>
> Henning

> [snip]

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Date: Wed, 03 Jun 2009 20:14:04 +0300
From: Roni Even <Even.roni@huawei.com>
In-reply-to: <021701c9e030$0a5deef0$1f19ccd0$%roni@huawei.com>
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Cc: avt@ietf.org
Subject: Re: [dispatch] [AVT] Proposal to form Internet Wideband Audio Codec WG
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Hi,

I would like to clarify that I am not against the discussion of codecs at
the IETF  but I think that before deciding to standardize one at the IETF we
should write requirements and analyze current codecs (standard as well as
non-standards) and see if they provide a good fit. 

 

One of my points is that having an IETF standard track codec does not make
it widely deployed as long as the IETF do not mandate codecs. I think that
you can look at the discussion in the SIP forum about specifying a codec.

 

I also think (and I will keep mentioning it) that one of the advantages of
voice on IP or over the Internet is that you can have a higher quality user
experience by using a wideband codec, it is a pity that G.711 is still
widely used taking bigger bandwidth than current wideband audio with lower
quality.

 

Roni Even

 

 

 

All,

 

I would like to request agenda time inside the DISPATCH meeting to propose
the formation of a new working group to define a Proposed Standard wideband
audio codec.

 

The text of the proposal is below. Comments, questions, and suggestions
welcomed.

 

Regards,

Jason

 

 

Internet Wideband Audio Codec (IWAC)

Mailing Lists: TBD

Chairs: TBD

Area Directorate: Real Time Applications (RAI)

 

Purpose:

 

This new working group would be chartered with the purpose of collecting

expertise within the IETF in order to review the design of audio codecs

specifically for use with the Internet. Unlike other SDOs, these codecs

would be optimized for use on the Internet, and as much as possible choose

technology that does not require paying patent royalties.

 

The Internet Low Bit Rate Codec (iLBC)  work was done in AVT but it was felt

that subsequent work should not be done in the AVT working group. This new

working group will have as its primary purpose the standardization of a

multi-purpose audio codec that can be used in various situations on the

Internet. Some of the proposed Internet-specific requirements include:

* scalable and adaptive bit rate;

* various sampling rate profiles from narrow-band to super-wideband;

* scalable complexity;

* low latency; and 

* resilience to packet loss.

 

There are a number of wide-band capable codecs defined by other SDOs. For

instance, G.722 is seeing adoption in Enterprise applications since it is

relatively simple and low-cost to deploy. However, it has a high, fixed

bitrate and is not appropriate for mobile applications where spectrum

efficiency is important or in consumer applications where available

bandwidth is fluctuating or limited. G.722.2 (AMR-wideband) has been adopted

by the 3GPP as a wideband standard for mobile applications. G.722.2 is

relatively high cost due to patent royalties and is seeing minimal

deployments outside of mobile handsets making it challenging to create

wideband experiences on Internet-capable mobile devices when extending

beyond the mobile network. In other cases, proprietary codecs are being

adopted which further create islands with no interoperability unless

widespread transcoding is performed. Transcoding leads to higher costs and

lower quality. 

 

The goal of this working group is to define a single codec with multiple

profiles which can be made available on a wide variety of Internet-capable

devices including low-power, mobile devices as well as devices capable of

utilizing high quality, high bitrate audio.

 

Proposed Deliverables:

 

1) Requirements for wideband, Internet audio codec(s).

2) Algorithm description for wideband, Internet audio codec(s) as Proposed

Standard. 

3) Specification of payload format(s) for defined codecs as Proposed

Standard

 

_______________________________________________

dispatch mailing list

dispatch@ietf.org

https://www.ietf.org/mailman/listinfo/dispatch

 


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<p class=3DMsoNormal><span =
style=3D'color:#1F497D'>Hi,<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>I would like to =
clarify that I am
not against the discussion of codecs at the IETF &nbsp;but I think that =
before
deciding to standardize one at the IETF we should write requirements and =
analyze
current codecs (standard as well as non-standards) and see if they =
provide a
good fit. <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>One of my points is =
that having
an IETF standard track codec does not make it widely deployed as long as =
the
IETF do not mandate codecs. I think that you can look at the discussion =
in the
SIP forum about specifying a codec.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>I also think (and I =
will keep
mentioning it) that one of the advantages of voice on IP or over the =
Internet
is that you can have a higher quality user experience by using a =
wideband codec,
it is a pity that G.711 is still widely used taking bigger bandwidth =
than
current wideband audio with lower quality.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>Roni =
Even<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoPlainText>All,<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>I would like to request agenda time inside the =
DISPATCH meeting
to propose the formation of a new working group to define a Proposed =
Standard
wideband audio codec.<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>The text of the proposal is below. Comments, =
questions,
and suggestions welcomed.<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>Regards,<o:p></o:p></p>

<p class=3DMsoPlainText>Jason<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>Internet Wideband Audio Codec =
(IWAC)<o:p></o:p></p>

<p class=3DMsoPlainText>Mailing Lists: TBD<o:p></o:p></p>

<p class=3DMsoPlainText>Chairs: TBD<o:p></o:p></p>

<p class=3DMsoPlainText>Area Directorate: Real Time Applications =
(RAI)<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>Purpose:<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>This new working group would be chartered with =
the
purpose of collecting<o:p></o:p></p>

<p class=3DMsoPlainText>expertise within the IETF in order to review the =
design
of audio codecs<o:p></o:p></p>

<p class=3DMsoPlainText>specifically for use with the Internet. Unlike =
other
SDOs, these codecs<o:p></o:p></p>

<p class=3DMsoPlainText>would be optimized for use on the Internet, and =
as much
as possible choose<o:p></o:p></p>

<p class=3DMsoPlainText>technology that does not require paying patent =
royalties.<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>The Internet Low Bit Rate Codec (iLBC)&nbsp; =
work was
done in AVT but it was felt<o:p></o:p></p>

<p class=3DMsoPlainText>that subsequent work should not be done in the =
AVT
working group. This new<o:p></o:p></p>

<p class=3DMsoPlainText>working group will have as its primary purpose =
the
standardization of a<o:p></o:p></p>

<p class=3DMsoPlainText>multi-purpose audio codec that can be used in =
various
situations on the<o:p></o:p></p>

<p class=3DMsoPlainText>Internet. Some of the proposed Internet-specific
requirements include:<o:p></o:p></p>

<p class=3DMsoPlainText>* scalable and adaptive bit rate;<o:p></o:p></p>

<p class=3DMsoPlainText>* various sampling rate profiles from =
narrow-band to
super-wideband;<o:p></o:p></p>

<p class=3DMsoPlainText>* scalable complexity;<o:p></o:p></p>

<p class=3DMsoPlainText>* low latency; and <o:p></o:p></p>

<p class=3DMsoPlainText>* resilience to packet loss.<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>There are a number of wide-band capable codecs =
defined by
other SDOs. For<o:p></o:p></p>

<p class=3DMsoPlainText>instance, G.722 is seeing adoption in Enterprise
applications since it is<o:p></o:p></p>

<p class=3DMsoPlainText>relatively simple and low-cost to deploy. =
However, it has
a high, fixed<o:p></o:p></p>

<p class=3DMsoPlainText>bitrate and is not appropriate for mobile =
applications
where spectrum<o:p></o:p></p>

<p class=3DMsoPlainText>efficiency is important or in consumer =
applications where
available<o:p></o:p></p>

<p class=3DMsoPlainText>bandwidth is fluctuating or limited. G.722.2
(AMR-wideband) has been adopted<o:p></o:p></p>

<p class=3DMsoPlainText>by the 3GPP as a wideband standard for mobile
applications. G.722.2 is<o:p></o:p></p>

<p class=3DMsoPlainText>relatively high cost due to patent royalties and =
is
seeing minimal<o:p></o:p></p>

<p class=3DMsoPlainText>deployments outside of mobile handsets making it
challenging to create<o:p></o:p></p>

<p class=3DMsoPlainText>wideband experiences on Internet-capable mobile =
devices
when extending<o:p></o:p></p>

<p class=3DMsoPlainText>beyond the mobile network. In other cases, =
proprietary
codecs are being<o:p></o:p></p>

<p class=3DMsoPlainText>adopted which further create islands with no
interoperability unless<o:p></o:p></p>

<p class=3DMsoPlainText>widespread transcoding is performed. Transcoding =
leads to
higher costs and<o:p></o:p></p>

<p class=3DMsoPlainText>lower quality. <o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>The goal of this working group is to define a =
single
codec with multiple<o:p></o:p></p>

<p class=3DMsoPlainText>profiles which can be made available on a wide =
variety of
Internet-capable<o:p></o:p></p>

<p class=3DMsoPlainText>devices including low-power, mobile devices as =
well as
devices capable of<o:p></o:p></p>

<p class=3DMsoPlainText>utilizing high quality, high bitrate =
audio.<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>Proposed Deliverables:<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>1) Requirements for wideband, Internet audio =
codec(s).<o:p></o:p></p>

<p class=3DMsoPlainText>2) Algorithm description for wideband, Internet =
audio
codec(s) as Proposed<o:p></o:p></p>

<p class=3DMsoPlainText>Standard. <o:p></o:p></p>

<p class=3DMsoPlainText>3) Specification of payload format(s) for =
defined codecs
as Proposed<o:p></o:p></p>

<p class=3DMsoPlainText>Standard<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p =
class=3DMsoPlainText>_______________________________________________<o:p>=
</o:p></p>

<p class=3DMsoPlainText>dispatch mailing list<o:p></o:p></p>

<p class=3DMsoPlainText><a =
href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a><o:p></o:p></p>

<p class=3DMsoPlainText><a =
href=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf.=
org/mailman/listinfo/dispatch</a><o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

</body>

</html>

--Boundary_(ID_lpXgMXCdnGW+Yj9iFz4pwA)--

From jmh@joelhalpern.com  Wed Jun  3 10:40:24 2009
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Subject: Re: [dispatch] [AVT] Proposal to form Internet Wideband Audio Codec WG
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To be blunt, without some better explanation of why the IETF ought to do 
this, and how the IETF is likely to be successful in doing this, I think 
it would be a disservice to the folks who want to work on this to 
encourage them to believe that the only obstacle is a clear problem 
definition.

To be clear, I have absolutely no problem with a community using an IETF 
   based email list to discuss their codec ideas, work out why they want 
to work on a new one, organize thier ideas for where that work should be 
done, etc.

So far, what strikes me is that most of the folks who would be called 
upon to evaluate a requirements document, or an actual protocol, have 
little or no expertise in codec design either from a mathematical or a 
software perspective (there are a few exceptions in both those regards, 
folks like Henning), and have little or no experience in the observed 
difficulties in actually standardizing codecs.  (Folks I ahve talked to 
who have been down that road have indicated that it involves many 
aspects that are foreign to IETF experience and practice.)

Yours,
Joel M. Halpern

Roni Even wrote:
> 
> 
> Hi,
> 
> I would like to clarify that I am not against the discussion of codecs 
> at the IETF  but I think that before deciding to standardize one at the 
> IETF we should write requirements and analyze current codecs (standard 
> as well as non-standards) and see if they provide a good fit.
> 
>  
> 
> One of my points is that having an IETF standard track codec does not 
> make it widely deployed as long as the IETF do not mandate codecs. I 
> think that you can look at the discussion in the SIP forum about 
> specifying a codec.
> 
>  
> 
> I also think (and I will keep mentioning it) that one of the advantages 
> of voice on IP or over the Internet is that you can have a higher 
> quality user experience by using a wideband codec, it is a pity that 
> G.711 is still widely used taking bigger bandwidth than current wideband 
> audio with lower quality.
> 
>  
> 
> Roni Even
> 
>  
> 
>  
> 
>  
> 
> All,
> 
>  
> 
> I would like to request agenda time inside the DISPATCH meeting to 
> propose the formation of a new working group to define a Proposed 
> Standard wideband audio codec.
> 
>  
> 
> The text of the proposal is below. Comments, questions, and suggestions 
> welcomed.
> 
>  
> 
> Regards,
> 
> Jason
> 
>  
> 
>  
> 
> Internet Wideband Audio Codec (IWAC)
> 
> Mailing Lists: TBD
> 
> Chairs: TBD
> 
> Area Directorate: Real Time Applications (RAI)
> 
>  
> 
> Purpose:
> 
>  
> 
> This new working group would be chartered with the purpose of collecting
> 
> expertise within the IETF in order to review the design of audio codecs
> 
> specifically for use with the Internet. Unlike other SDOs, these codecs
> 
> would be optimized for use on the Internet, and as much as possible choose
> 
> technology that does not require paying patent royalties.
> 
>  
> 
> The Internet Low Bit Rate Codec (iLBC)  work was done in AVT but it was 
> felt
> 
> that subsequent work should not be done in the AVT working group. This new
> 
> working group will have as its primary purpose the standardization of a
> 
> multi-purpose audio codec that can be used in various situations on the
> 
> Internet. Some of the proposed Internet-specific requirements include:
> 
> * scalable and adaptive bit rate;
> 
> * various sampling rate profiles from narrow-band to super-wideband;
> 
> * scalable complexity;
> 
> * low latency; and
> 
> * resilience to packet loss.
> 
>  
> 
> There are a number of wide-band capable codecs defined by other SDOs. For
> 
> instance, G.722 is seeing adoption in Enterprise applications since it is
> 
> relatively simple and low-cost to deploy. However, it has a high, fixed
> 
> bitrate and is not appropriate for mobile applications where spectrum
> 
> efficiency is important or in consumer applications where available
> 
> bandwidth is fluctuating or limited. G.722.2 (AMR-wideband) has been 
> adopted
> 
> by the 3GPP as a wideband standard for mobile applications. G.722.2 is
> 
> relatively high cost due to patent royalties and is seeing minimal
> 
> deployments outside of mobile handsets making it challenging to create
> 
> wideband experiences on Internet-capable mobile devices when extending
> 
> beyond the mobile network. In other cases, proprietary codecs are being
> 
> adopted which further create islands with no interoperability unless
> 
> widespread transcoding is performed. Transcoding leads to higher costs and
> 
> lower quality.
> 
>  
> 
> The goal of this working group is to define a single codec with multiple
> 
> profiles which can be made available on a wide variety of Internet-capable
> 
> devices including low-power, mobile devices as well as devices capable of
> 
> utilizing high quality, high bitrate audio.
> 
>  
> 
> Proposed Deliverables:
> 
>  
> 
> 1) Requirements for wideband, Internet audio codec(s).
> 
> 2) Algorithm description for wideband, Internet audio codec(s) as Proposed
> 
> Standard.
> 
> 3) Specification of payload format(s) for defined codecs as Proposed
> 
> Standard
> 
>  
> 
> _______________________________________________
> 
> dispatch mailing list
> 
> dispatch@ietf.org <mailto:dispatch@ietf.org>
> 
> https://www.ietf.org/mailman/listinfo/dispatch
> 
>  
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

From jean-marc.valin@octasic.com  Wed Jun  3 11:47:39 2009
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> So far, what strikes me is that most of the folks who would be called 
> upon to evaluate a requirements document, or an actual protocol, have 
> little or no expertise in codec design either from a mathematical or a 
> software perspective (there are a few exceptions in both those 
> regards, folks like Henning), and have little or no experience in the 
> observed difficulties in actually standardizing codecs. (Folks I ahve 
> talked to who have been down that road have indicated that it involves 
> many aspects that are foreign to IETF experience and practice.) 

Let me start by stating that codec design is no rocket science. There 
are many people out there who know how to write audio codecs and many 
more who have sufficient skills to contribute. At this point, there is 
already interest from several people with actual codec design experience 
at Skype, Xiph.Org and SPIRIT. The internet community has clear and 
considerable needs which are not being adequately met by the existing 
standards bodies. We have parties experienced with codec development who 
want to work together within the IETF. Why do you believe the risk of 
not succeeding justifies not even attempting the effort?

Cheers,

    Jean-Marc

From dyork@voxeo.com  Wed Jun  3 12:45:51 2009
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Subject: Re: [dispatch] [AVT] Proposal to form Internet Wideband Audio Codec WG
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I've been reading this thread with great interest as for the past  
probably 6 years or so I've been saying to anyone who will listen (and  
to those who won't ;-) that one factor that will help "IP  
communications/VoIP/UC/whatever-marketing-term-we-want-to-rename-our- 
products-to-this-week" to really find wide adoption is to provide a  
"rich communication experience" better than what people are used to now.

Wideband audio is, to me, a huge part of that.

However, I completely agree with Henning on this point:

> On Jun 2, 2009, at 12:15 PM, Henning Schulzrinne wrote:
>
>> However, if we don't do this, we are stuck with the status quo,  
>> which is not all that satisfactory.


The status quo is not satisfactory.  If you want to add wideband audio  
to your product, you've got an alphabet soup of mutually incompatible  
options with vastly different licensing terms/fees/options and source  
code availability.

I think we ultimately need a "G.711 for wideband", i.e. a wideband  
codec that anyone can implement in any device.  I'd argue that a large  
part of the reason we are "stuck" with so many people in VoIP using G. 
711 is precisely because ANYONE can implement it in basically ANY  
device.  You can buy a G.711 codec implementation or you can write  
your own or find some code on the Internet (google "G.711 source  
code"). You can implement G.711 in your big softswitch or IP-PBX... or  
you can implement it in some small embedded device.

And it interoperates.

We need that kind of codec for wideband.... which I would say is  
"royalty-free" and needs to have "open source" implementations  
available.   (And yes, I know Speex is out there - I don't have a view  
as to why it is not more widely implemented.  And yes, I know Skype is  
giving away SILK royalty-free, but it's not open source.)

Can the IETF help with this?  I don't know... but I think it's worth a  
shot.  If there is a group that is passionate about this (and there  
seems to be) then lets get the mailing list set up and have it go the  
regular List -> BOF -> Working Group trajectory.

In the best case perhaps we come out with the codec we want/need.  At  
the very worst I think we'd wind up with at least creating some good  
discussions and requirements for what we do need.

My 2 cents,
Dan

-- 
Dan York, Director of Conversations
Voxeo Corporation   http://www.voxeo.com  dyork@voxeo.com
Phone: +1-407-455-5859    Skype: danyork

Join the Voxeo conversation:
Blogs: http://blogs.voxeo.com  http://www.disruptivetelephony.com
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Let's be clear.  I am not trying to argue that the existing codecs are 
good enough.  I hear everyone saying that they are not.
However, the IETF is not the answer to all the worlds standards problems.

In this case, there are a number of existing bodies addressing these 
problems.  There also are apparently existing solutions with near the 
right properties.
1) The IETF is structurally not well suited to dealing with the kinds of 
patent problems that we are dealing with.
2) The IETF, based on history, is not very good at dealing with market 
acceptance issues (questions such as why isn't Speex or SILK being 
accepted.)
3) IETF WGs are very bad at working out what the real problems are. 
THey are good at working out protocol issues.  But this is not primarily 
a protocol issue, from what everyone is saying.

I have heard rumors of several other bodies that are working in this 
space.  Given that we have lately been working very hard to keep other 
folks from encroaching on our patch, we ought to be sensitive to 
encroaching on theirs.

And now, before I start repeating myself, I will try to stay quite for a 
few days and keep listening.
Yours,
Joel


Dan York wrote:
> I've been reading this thread with great interest as for the past 
> probably 6 years or so I've been saying to anyone who will listen (and 
> to those who won't ;-) that one factor that will help "IP 
> communications/VoIP/UC/whatever-marketing-term-we-want-to-rename-our-products-to-this-week" 
> to really find wide adoption is to provide a "rich communication 
> experience" better than what people are used to now.
> 
> Wideband audio is, to me, a huge part of that.
> 
> However, I completely agree with Henning on this point:
> 
>> On Jun 2, 2009, at 12:15 PM, Henning Schulzrinne wrote:
>>
>>> However, if we don't do this, we are stuck with the status quo, which 
>>> is not all that satisfactory.
> 
> 
> The status quo is not satisfactory.  If you want to add wideband audio 
> to your product, you've got an alphabet soup of mutually incompatible 
> options with vastly different licensing terms/fees/options and source 
> code availability.
> 
> I think we ultimately need a "G.711 for wideband", i.e. a wideband codec 
> that anyone can implement in any device.  I'd argue that a large part of 
> the reason we are "stuck" with so many people in VoIP using G.711 is 
> precisely because ANYONE can implement it in basically ANY device.  You 
> can buy a G.711 codec implementation or you can write your own or find 
> some code on the Internet (google "G.711 source code"). You can 
> implement G.711 in your big softswitch or IP-PBX... or you can implement 
> it in some small embedded device.
> 
> And it interoperates.
> 
> We need that kind of codec for wideband.... which I would say is 
> "royalty-free" and needs to have "open source" implementations 
> available.   (And yes, I know Speex is out there - I don't have a view 
> as to why it is not more widely implemented.  And yes, I know Skype is 
> giving away SILK royalty-free, but it's not open source.)
> 
> Can the IETF help with this?  I don't know... but I think it's worth a 
> shot.  If there is a group that is passionate about this (and there 
> seems to be) then lets get the mailing list set up and have it go the 
> regular List -> BOF -> Working Group trajectory.
> 
> In the best case perhaps we come out with the codec we want/need.  At 
> the very worst I think we'd wind up with at least creating some good 
> discussions and requirements for what we do need.
> 
> My 2 cents,
> Dan
> 

From hsinnrei@adobe.com  Wed Jun  3 13:24:08 2009
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From: Henry Sinnreich <hsinnrei@adobe.com>
To: "Joel M. Halpern" <jmh@joelhalpern.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Wed, 3 Jun 2009 13:23:26 -0700
Thread-Topic: [dispatch] [AVT] Proposal to form Internet Wideband Audio Codec WG
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Joel,

>And now, before I start repeating myself

I will also repeat here that it makes no sense to have this huge IETF work =
for voice (mainly) signaling (SIP, XMPP, NSIS) and no standard for an _Inte=
rnet_ voice codec.

Going by your guidance for avoiding a voice codec, the IETF may also just g=
ive up signaling for voice and close down all the voice related signaling W=
Gs. And since we are at it, why not all the A/V codec profiles in AVT? :-)
This is hard to rationalize.

Henry


On 6/3/09 2:57 PM, "Joel M. Halpern" <jmh@joelhalpern.com> wrote:

Let's be clear.  I am not trying to argue that the existing codecs are
good enough.  I hear everyone saying that they are not.
However, the IETF is not the answer to all the worlds standards problems.

In this case, there are a number of existing bodies addressing these
problems.  There also are apparently existing solutions with near the
right properties.
1) The IETF is structurally not well suited to dealing with the kinds of
patent problems that we are dealing with.
2) The IETF, based on history, is not very good at dealing with market
acceptance issues (questions such as why isn't Speex or SILK being
accepted.)
3) IETF WGs are very bad at working out what the real problems are.
THey are good at working out protocol issues.  But this is not primarily
a protocol issue, from what everyone is saying.

I have heard rumors of several other bodies that are working in this
space.  Given that we have lately been working very hard to keep other
folks from encroaching on our patch, we ought to be sensitive to
encroaching on theirs.

And now, before I start repeating myself, I will try to stay quite for a
few days and keep listening.
Yours,
Joel


Dan York wrote:
> I've been reading this thread with great interest as for the past
> probably 6 years or so I've been saying to anyone who will listen (and
> to those who won't ;-) that one factor that will help "IP
> communications/VoIP/UC/whatever-marketing-term-we-want-to-rename-our-prod=
ucts-to-this-week"
> to really find wide adoption is to provide a "rich communication
> experience" better than what people are used to now.
>
> Wideband audio is, to me, a huge part of that.
>
> However, I completely agree with Henning on this point:
>
>> On Jun 2, 2009, at 12:15 PM, Henning Schulzrinne wrote:
>>
>>> However, if we don't do this, we are stuck with the status quo, which
>>> is not all that satisfactory.
>
>
> The status quo is not satisfactory.  If you want to add wideband audio
> to your product, you've got an alphabet soup of mutually incompatible
> options with vastly different licensing terms/fees/options and source
> code availability.
>
> I think we ultimately need a "G.711 for wideband", i.e. a wideband codec
> that anyone can implement in any device.  I'd argue that a large part of
> the reason we are "stuck" with so many people in VoIP using G.711 is
> precisely because ANYONE can implement it in basically ANY device.  You
> can buy a G.711 codec implementation or you can write your own or find
> some code on the Internet (google "G.711 source code"). You can
> implement G.711 in your big softswitch or IP-PBX... or you can implement
> it in some small embedded device.
>
> And it interoperates.
>
> We need that kind of codec for wideband.... which I would say is
> "royalty-free" and needs to have "open source" implementations
> available.   (And yes, I know Speex is out there - I don't have a view
> as to why it is not more widely implemented.  And yes, I know Skype is
> giving away SILK royalty-free, but it's not open source.)
>
> Can the IETF help with this?  I don't know... but I think it's worth a
> shot.  If there is a group that is passionate about this (and there
> seems to be) then lets get the mailing list set up and have it go the
> regular List -> BOF -> Working Group trajectory.
>
> In the best case perhaps we come out with the codec we want/need.  At
> the very worst I think we'd wind up with at least creating some good
> discussions and requirements for what we do need.
>
> My 2 cents,
> Dan
>
_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch


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<HTML>
<HEAD>
<TITLE>Re: [dispatch] [AVT] Proposal to form Internet Wideband Audio Codec =
WG</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
18pt'>Joel,<BR>
<BR>
&gt;And now, before I start repeating myself<BR>
<BR>
I will also repeat here that it makes no sense to have this huge IETF work =
for voice (mainly) signaling (SIP, XMPP, NSIS) and no standard for an _Inte=
rnet_<I> </I> voice codec. <BR>
<BR>
Going by your guidance for avoiding a voice codec, the IETF may also just g=
ive up signaling for voice and close down all the voice related signaling W=
Gs. And since we are at it, why not all the A/V codec profiles in AVT? :-) =
<BR>
This is hard to rationalize.<BR>
<BR>
Henry<BR>
<BR>
<BR>
On 6/3/09 2:57 PM, &quot;Joel M. Halpern&quot; &lt;<a href=3D"jmh@joelhalpe=
rn.com">jmh@joelhalpern.com</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:18pt'>Let's be clear. &nbsp;I am not trying to ar=
gue that the existing codecs are<BR>
good enough. &nbsp;I hear everyone saying that they are not.<BR>
However, the IETF is not the answer to all the worlds standards problems.<B=
R>
<BR>
In this case, there are a number of existing bodies addressing these<BR>
problems. &nbsp;There also are apparently existing solutions with near the<=
BR>
right properties.<BR>
1) The IETF is structurally not well suited to dealing with the kinds of<BR=
>
patent problems that we are dealing with.<BR>
2) The IETF, based on history, is not very good at dealing with market<BR>
acceptance issues (questions such as why isn't Speex or SILK being<BR>
accepted.)<BR>
3) IETF WGs are very bad at working out what the real problems are.<BR>
THey are good at working out protocol issues. &nbsp;But this is not primari=
ly<BR>
a protocol issue, from what everyone is saying.<BR>
<BR>
I have heard rumors of several other bodies that are working in this<BR>
space. &nbsp;Given that we have lately been working very hard to keep other=
<BR>
folks from encroaching on our patch, we ought to be sensitive to<BR>
encroaching on theirs.<BR>
<BR>
And now, before I start repeating myself, I will try to stay quite for a<BR=
>
few days and keep listening.<BR>
Yours,<BR>
Joel<BR>
<BR>
<BR>
Dan York wrote:<BR>
&gt; I've been reading this thread with great interest as for the past<BR>
&gt; probably 6 years or so I've been saying to anyone who will listen (and=
<BR>
&gt; to those who won't ;-) that one factor that will help &quot;IP<BR>
&gt; communications/VoIP/UC/whatever-marketing-term-we-want-to-rename-our-p=
roducts-to-this-week&quot;<BR>
&gt; to really find wide adoption is to provide a &quot;rich communication<=
BR>
&gt; experience&quot; better than what people are used to now.<BR>
&gt;<BR>
&gt; Wideband audio is, to me, a huge part of that.<BR>
&gt;<BR>
&gt; However, I completely agree with Henning on this point:<BR>
&gt;<BR>
&gt;&gt; On Jun 2, 2009, at 12:15 PM, Henning Schulzrinne wrote:<BR>
&gt;&gt;<BR>
&gt;&gt;&gt; However, if we don't do this, we are stuck with the status quo=
, which<BR>
&gt;&gt;&gt; is not all that satisfactory.<BR>
&gt;<BR>
&gt;<BR>
&gt; The status quo is not satisfactory. &nbsp;If you want to add wideband =
audio<BR>
&gt; to your product, you've got an alphabet soup of mutually incompatible<=
BR>
&gt; options with vastly different licensing terms/fees/options and source<=
BR>
&gt; code availability.<BR>
&gt;<BR>
&gt; I think we ultimately need a &quot;G.711 for wideband&quot;, i.e. a wi=
deband codec<BR>
&gt; that anyone can implement in any device. &nbsp;I'd argue that a large =
part of<BR>
&gt; the reason we are &quot;stuck&quot; with so many people in VoIP using =
G.711 is<BR>
&gt; precisely because ANYONE can implement it in basically ANY device. &nb=
sp;You<BR>
&gt; can buy a G.711 codec implementation or you can write your own or find=
<BR>
&gt; some code on the Internet (google &quot;G.711 source code&quot;). You =
can<BR>
&gt; implement G.711 in your big softswitch or IP-PBX... or you can impleme=
nt<BR>
&gt; it in some small embedded device.<BR>
&gt;<BR>
&gt; And it interoperates.<BR>
&gt;<BR>
&gt; We need that kind of codec for wideband.... which I would say is<BR>
&gt; &quot;royalty-free&quot; and needs to have &quot;open source&quot; imp=
lementations<BR>
&gt; available. &nbsp;&nbsp;(And yes, I know Speex is out there - I don't h=
ave a view<BR>
&gt; as to why it is not more widely implemented. &nbsp;And yes, I know Sky=
pe is<BR>
&gt; giving away SILK royalty-free, but it's not open source.)<BR>
&gt;<BR>
&gt; Can the IETF help with this? &nbsp;I don't know... but I think it's wo=
rth a<BR>
&gt; shot. &nbsp;If there is a group that is passionate about this (and the=
re<BR>
&gt; seems to be) then lets get the mailing list set up and have it go the<=
BR>
&gt; regular List -&gt; BOF -&gt; Working Group trajectory.<BR>
&gt;<BR>
&gt; In the best case perhaps we come out with the codec we want/need. &nbs=
p;At<BR>
&gt; the very worst I think we'd wind up with at least creating some good<B=
R>
&gt; discussions and requirements for what we do need.<BR>
&gt;<BR>
&gt; My 2 cents,<BR>
&gt; Dan<BR>
&gt;<BR>
_______________________________________________<BR>
dispatch mailing list<BR>
<a href=3D"dispatch@ietf.org">dispatch@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf=
.org/mailman/listinfo/dispatch</a><BR>
<BR>
</SPAN></FONT></BLOCKQUOTE>
</BODY>
</HTML>


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From jason.fischl@skype.net  Wed Jun  3 15:09:12 2009
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Date: Wed, 03 Jun 2009 15:08:41 -0700
From: Jason Fischl <jason.fischl@skype.net>
To: Eric Burger <eburger@standardstrack.com>, Henry Sinnreich <hsinnrei@adobe.com>
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Subject: Re: [dispatch] [AVT] Proposal to form Internet Wideband Audio Codec WG
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I've requested a mailing list and it has been approved. It should be live in
a day or so. I'll post the details once it is live.

Jason



On 6/2/09 6:50 PM, "Eric Burger" <eburger@standardstrack.com> wrote:

> To directly answer Henry, having a mail list and the initial (and
> possibly detailed) discussion hosted at the IETF costs less than the
> discussion we are having here.  If a codec comes out of it, without
> having a BOF or WG, that's great!  If not, at least there was a forum
> for doing it.
> 
> I can also offer a SIP Forum list if people still really, really don't
> like the idea of even discussing the topic under even the most
> faintest of auspices of the IETF.
> 
> If you think I am schizophrenic by offering we can have a discussion
> in the IETF and at the same time I doubt we could ever have a formal
> work group to create the codec, you would be only slightly
> correct :-)  But seriously: the IETF (AVT in particular) can pickup
> the wire format for a codec, once the codec exists. Having a mail list
> WITHOUT A WORK GROUP give people who want to work on this a venue to
> do the work, without burdening the AD's, IESG, and the rest of the
> community. If a codec pops out, then we can run with it.
> 
> On Jun 2, 2009, at 12:44 PM, Spencer Dawkins wrote:
> 
>> I am not an area director, nor do i play one on IPTV, but my
>> suggestion for anyone interested would be to
>> 
>> - request a non-WG mailing list (from either of the RAI area
>> directors),
>> 
>> - announce it on this mailing list (and others that seem to be
>> appropriate, and
>> 
>> - start discussions, which the area directors typically look at when
>> judging whether there's enough interest and enough clue to justify
>> approving a BOF and/or chartering a working group.
>> 
>> I didn't see a mailing list included in Jason's proposal - if I
>> missed one, my apologies.
>> 
>> If people at the IETF can do this work, and want to do this work,
>> IETF process policies should not get in the way of that happening.
>> 
>> And I haven't heard of an AD turning down a request for a non-WG
>> mailing list yet, keeping in mind that one of said lists is ietf-
>> sailors, for people interested in sailing to/from/at IETF meetings -
>> the bar is (appropriately) low.
>> 
>> Thanks,
>> 
>> Spencer
>> ----- Original Message -----
>> From: Henry Sinnreich
>> To: Roni Even ; Jean-Marc Valin
>> Cc: dispatch@ietf.org ; Slava Borilin ; avt@ietf.org
>> Sent: Tuesday, June 02, 2009 11:12 AM
>> Subject: Re: [dispatch] [AVT] Proposal to form Internet Wideband
>> Audio Codec WG
>> 
>>> This leaves the other reasons I heard for doing it at the IETF
>> which is the
>>> price of participating
>> 
>> This is an important point.
>> Given the travel constraints for some of the most valuable potential
>> contributors and reviewers, could we envisage online work until the
>> economy improves, so that an eventual BOF to start with should not
>> be starved of attendees? An online BOF?
>> 
>> There are several free online meeting tools available...
>> 
>> How can this be done within the IETF policy?
>> 
>> Henry
>> 
>> 
>> On 6/2/09 10:57 AM, "Roni Even" <Even.roni@huawei.com> wrote:
>> 
>> Hi,
>> I do not want to sound like someone who is for IPR (I am not), but
>> why stop
>> at codec, let's require it for all IETF work. There are IPR on IETF
>> work
>> which is must simpler, in my view, than wide band audio codecs.
>> 
>> I think that we can start with "royalty free" even though I am not
>> sure that
>> it will accepted as part of the charter of any other work group so
>> why pick
>> on codecs which require more work.
>> 
>> This leaves the other reasons I heard for doing it at the IETF which
>> is the
>> price of participating (cheaper than being an ITU-T member) and
>> maybe design
>> less expensive characterization tests.
>> 
>> Roni
>> 
>> 
>> -----Original Message-----
>> From: Jean-Marc Valin [mailto:jean-marc.valin@octasic.com]
>> Sent: Tuesday, June 02, 2009 6:14 PM
>> To: Roni Even
>> Cc: 'Slava Borilin'; avt@ietf.org; 'Jason Fischl';dispatch@ietf.org;
>> hsinnrei@adobe.com
>> Subject: Re: [AVT] [dispatch] Proposal to form Internet Wideband
>> Audio Codec
>> WG
>> 
>> Hi,
>> 
>> Roni Even wrote:
>>> 
>>> I am not sure what prevented you from doing it today at the ITU or
>>> MPEG, why do you see the IETF handling it differently.
>>> 
>>> I would also like to remind you and Jean-Marc that once you are
>>> bringing work to a standard body it may require collaboration with
>>> other people who will have other proposals that will also address
>> the
>>> same requirements and you may need to invest money in comparative
>>> testing by independent listening labs.
>>> 
>>> I also think that you will need to supply the codec in source code
>> and
>>> provide copy right to the IETF.
>>> 
>> 
>> I am well aware that bringing work to the IETF would require
>> collaboration with others. I am not seeking control over the work I am
>> currently doing and would really welcome such collaboration. The
>> idea is
>> only to have the best wideband codec possible without IPR issues.
>> Given
>> the ITU and MPEG track record, I think it would be very unlikely for
>> any
>> of those organisations to work on an IPR-free codec.
>> 
>> I also agree with Henry that "the Internet has different criteria than
>> ITU-T networks may have". Internet adoption follows different patterns
>> than adoption in the ITU primary target markets. For example, the
>> Internet has more consumer-reconfigurable software, while the ITU
>> has to
>> deal with a lot of fixed hardware deployments. At the ITU, it makes
>> sense to invest large sums of money into testing and
>> characterisation of
>> codecs because the codecs deployed there usually stay around for a
>> long
>> time and the infrastructure investments are usually very large. On the
>> other hand, I would say the investments in codecs for the Internet are
>> comparatively smaller and, while testing is still important, it is not
>> as critical as it is for the ITU.
>> 
>> It's also a choice one has to make. It is unlikely that companies
>> would
>> invest money in expensive testing if they are not going to obtain
>> royalties in return. However, I think we may be able to define some
>> more
>> lightweight (collaborative?) testing that is sufficient and doesn't
>> involve as much investments as what the ITU does. For the Internet, I
>> believe an IPR-free codec that everyone agrees performs well is better
>> than a patent-encumbered codec that has had more extensive testing.
>> This
>> is again another difference with the ITU: patent-encumbered codecs
>> tend
>> to hurt a lot more on the Internet because many applications (e.g.
>> giving away the client) are very hard (or impossible) when one has to
>> pay per-channel royalties.
>> 
>> As for the source code issue you pointed out, all the Xiph codecs are
>> already published under a very permissive open source license (BSD),
>> so
>> this would not really change.
>> 
>>     Jean-Marc
>> 
>>> Roni
>>> 
>>> 
>>> 
>>> *From:* avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] *On
>> Behalf
>>> Of *Slava Borilin
>>> *Sent:* Monday, June 01, 2009 11:50 PM
>>> *To:* jean-marc.valin@octasic.com
>>> *Cc:* avt@ietf.org; Jason Fischl
>>> *Subject:* Re: [AVT] [dispatch] Proposal to form Internet Wideband
>>> Audio Codec WG
>>> 
>>> 
>>> 
>>> Agree with Jean-Marc.
>>> SPIRIT is interested to contribute as well - having a dozen of
>> proprietary
>> codecs developed, including
>>> one specifically desgined for internet (SPIRIT IP-MR, which is now
>> under
>> WGLC on draft-ietf-avt-rtp-ipmr-04) -
>>> multi-rate, scalable, adaptive, wideband codec.
>>> 
>>> We can also continue this work with IETF.
>>> 
>>> Moreover, most if not all efforts coming from ITU on codecs
>> unfortunately
>> are NOT really focused on
>>> internet-specific codecs (that's why several companies have had to
>> invent)
>> - as ITU preference is mainly
>>> specific (i.e. cellular) networks at first.
>>> 
>>> however, as we see the greant rise of pure "internet-basd
>> communications"
>> (skype, webex/citrix, and many others) -
>>> we all (and users) are suffering from inefficiency in all currently
>> "standard" codecs and ambiguity in the choice of
>>> internet-targeted ones.
>>> 
>>> Again, we probably can put together enough number of contributors
>> to the
>> WG to have the expertise.
>>> 
>>> regards,
>>> Slava Borilin
>>> 
>>> --
>>> John Lazzaro wrote:
>>> 
>>>     A traditional response to this type of request is to note that
>> the
>> IETF
>>> 
>>>     really doesn't have much in the way of expertise in audio codec
>> design.
>>> 
>>>     I don't see many of the regulars who present at the AES codec
>> paper
>> sessions
>>> 
>>>     posting here on AVT (ditto ICASSP paper sessions for voice
>> codecs).
>> It's
>>> 
>>>     a full-time job to keep up-to-date and contribute to that
>>> 
>>>     signal-processing lore.
>>> 
>>> 
>>> 
>>> Well, there's no reason that the IETF cannot build such an expertise
>>> in audio codecs. This is actually something in which I'd be
>> interested
>>> in getting involved and I'm sure others at Xiph.Org would be
>>> interested as well. We have several people with audio codec
>> expertise
>>> from working on Vorbis, Speex and (more recently) CELT. It turns out
>>> that the CELT codec currently under development at Xiph actually
>> meets
>>> most of the requirements from the original proposal in being a very
>>> low delay codec with adaptive bit-rate and sampling rate (up to 48
>>> kHz), scalable complexity, and good robustness to packet loss.
>> We'd be
>>> willing to continue the development with the IETF. Even if not with
>>> CELT, it's still like to be involved in such a new WG.
>>> 
>>> Cheers,
>>> 
>>>    Jean-Marc
>> 
>> 
>> 
>> 
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


From dean.willis@softarmor.com  Wed Jun  3 15:16:30 2009
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Subject: Re: [dispatch] [AVT] Proposal to form Internet Wideband Audio Codec WG
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On Jun 3, 2009, at 2:45 PM, Dan York wrote:

> I think we ultimately need a "G.711 for wideband", i.e. a wideband  
> codec that anyone can implement in any device.  I'd argue that a  
> large part of the reason we are "stuck" with so many people in VoIP  
> using G.711 is precisely because ANYONE can implement it in  
> basically ANY device.  You can buy a G.711 codec implementation or  
> you can write your own or find some code on the Internet (google "G. 
> 711 source code"). You can implement G.711 in your big softswitch or  
> IP-PBX... or you can implement it in some small embedded device.

We have such a codec: 16 bit sampling at 44.1 khz, aka "what your CD  
player does in stereo", or L16 audio as defined in RFC 1890.

While there might be some IPR on it, it clearly hasn't kept everybody  
and their dog from playing the audio format on devices ranging from a  
$3 CD player on up.


We also have RFC 3190, which does RTP format for 12-bit DAT audio and  
20 and 24 bit linear sampled audio.


So what is it that we need for a "wideband G.711" that isn't already  
in there somewhere?


It sounds like what you want is more a wideband iLBC, which is an  
intellectually much larger challenge.

--
Dean


From xiphmont@gmail.com  Wed Jun  3 19:12:50 2009
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Subject: Re: [dispatch] [AVT] Proposal to form Internet Wideband Audio Codec WG
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I've been following this discussion through the archives, and it seems
time to dip a toe in. Apologies for the vaguely formal language, I'm
speaking for my organization.

I'm happy to see more IETF interest in taking up the subject of
royalty-free codecs for the Internet.  Several of our Xiph.Org members
have been quietly advocating for this within IETF for some time, and
it's nice to see the call coming from other sources as well. It's no
coincidence that the Internet's greatest success stories have all been
unencumbered.  Any new technology that rolls out with a royalty
pricetag has a steep uphill climb to do anything more than fragment
and muddy both social and business interoperability.

The Xiph.Org Foundation is a registered non-profit created toward
designing and deploying unencumered media codecs.  I'd like to
stand up and say officially that we would very much like to work with
the IETF on roaylty-free codecs.

Xiph.Org does two things: we research fundamental codec technology to
further the state of the art, then develop and deploy this research
without royalty or licensing restrictions.  We've created and made
available the Ogg container format and the Vorbis, Speex, FLAC, Theora
and CELT codecs.  Ogg-related protocols are already covered by a few
RFCs.  We exert no IP claims on our specifications and our reference
implementations are all BSD licensed.  Our codecs are a means to an
end, not an end in and of themselves.

Unfortunately, unencumbered access to basic media technology (a level
playing field) has meant reinventing and reimplementing from the
ground up. On the plus side, reinventing and reimplementing has
allowed us to use new [and rediscovered] techniques to better fit the
needs of the Internet.

...so we'd like to toss our hat into this ring and volunteer the
experience our own organization has gained in the past 15 years of
building royalty-free, unencumbered media tech from scratch. We
look forward to working with others who recognize the importance of
open and free media technology.

Monty
Xiph.Org

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I would argue that G.711 is implemented, not because anyone can implement i=
t, but because everyone else has implemented it. It essentially forms the l=
owest common denominator for interoperability. If fancy codec x doesn't wor=
k, then assuming you have the bandwidth availabile, G.711 probably will. An=
d I suspect it is royalty free not because it was always so, but because an=
y IPR that existed has pretty much expired by now.

G.711 is also the one codec that is probably the most neutral to transcodin=
g. I can do to codec A to G.711 and back to codec A with less impact that w=
ith any other intermediate codec.

For wideband there probably is not any one codec that has achieved that pos=
ition.

I don't believe building a better, even royalty free codec, will guarantee =
a position in the market place. The world is littered with better solutions=
 that never made it. You need a market where everyone chooses to implement =
it so that you can use that codec without having to go to transcoding. Past=
ing IETF on the front cover does not achieve that.

I don't really have a problem with people trying to sit down and design a n=
ew codec and IETF then publishing it. Without some other selling point howe=
ver it just becomes yet another codec competing for market place. As such, =
put it somewhere where it does not interfere with other work. Sticking it o=
n a separate mailing list, and not letting it compete for valuable IETF fac=
e to face slots as a working group is fine.

regards

Keith
=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Dan York
> Sent: Wednesday, June 03, 2009 8:46 PM
> To: dispatch@ietf.org
> Cc: avt@ietf.org
> Subject: Re: [dispatch] [AVT] Proposal to form Internet=20
> Wideband Audio Codec WG
>=20
> I've been reading this thread with great interest as for the=20
> past probably 6 years or so I've been saying to anyone who=20
> will listen (and to those who won't ;-) that one factor that=20
> will help "IP
> communications/VoIP/UC/whatever-marketing-term-we-want-to-rename-our-
> products-to-this-week" to really find wide adoption is to=20
> provide a "rich communication experience" better than what=20
> people are used to now.
>=20
> Wideband audio is, to me, a huge part of that.
>=20
> However, I completely agree with Henning on this point:
>=20
> > On Jun 2, 2009, at 12:15 PM, Henning Schulzrinne wrote:
> >
> >> However, if we don't do this, we are stuck with the status=20
> quo, which=20
> >> is not all that satisfactory.
>=20
>=20
> The status quo is not satisfactory.  If you want to add=20
> wideband audio =20
> to your product, you've got an alphabet soup of mutually=20
> incompatible =20
> options with vastly different licensing terms/fees/options=20
> and source =20
> code availability.
>=20
> I think we ultimately need a "G.711 for wideband", i.e. a wideband =20
> codec that anyone can implement in any device.  I'd argue=20
> that a large =20
> part of the reason we are "stuck" with so many people in VoIP=20
> using G.=20
> 711 is precisely because ANYONE can implement it in basically ANY =20
> device.  You can buy a G.711 codec implementation or you can write =20
> your own or find some code on the Internet (google "G.711 source =20
> code"). You can implement G.711 in your big softswitch or=20
> IP-PBX... or =20
> you can implement it in some small embedded device.
>=20
> And it interoperates.
>=20
> We need that kind of codec for wideband.... which I would say is =20
> "royalty-free" and needs to have "open source" implementations =20
> available.   (And yes, I know Speex is out there - I don't=20
> have a view =20
> as to why it is not more widely implemented.  And yes, I know=20
> Skype is =20
> giving away SILK royalty-free, but it's not open source.)
>=20
> Can the IETF help with this?  I don't know... but I think=20
> it's worth a =20
> shot.  If there is a group that is passionate about this (and there =20
> seems to be) then lets get the mailing list set up and have=20
> it go the =20
> regular List -> BOF -> Working Group trajectory.
>=20
> In the best case perhaps we come out with the codec we=20
> want/need.  At =20
> the very worst I think we'd wind up with at least creating some good =20
> discussions and requirements for what we do need.
>=20
> My 2 cents,
> Dan
>=20
> --=20
> Dan York, Director of Conversations
> Voxeo Corporation   http://www.voxeo.com  dyork@voxeo.com
> Phone: +1-407-455-5859    Skype: danyork
>=20
> Join the Voxeo conversation:
> Blogs: http://blogs.voxeo.com  http://www.disruptivetelephony.com
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Hi, 

We have submitted a draft which proposes registration of an event
package, based on work ongoing in 3GPP (and TISPAN). 

(The most likely protocol output is going to be a new event package).

The document can also be found in: 

http://www.ietf.org/proceedings/staging/draft-avasarala-dispatch-comm-di
v-notification-00.txt

Updates since last version:

   Changes from draft-avasarala-sipping-comm-div-notification-01
 
   o  Changed contact details for co-author Subir Saha
 
   o  Moved from SIPPING to DISPATCH WG

Regards, 

John-Luc

 

 

--- 
John-Luc Bakker 
Standards Manager 
Research In Motion 
BlackBerry: +1 (908) 463 7321 
Office: +1 (972) 373-1761 
Internal: (820) 63761 
E-mail: jbakker@rim.com 

 


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<p><font size=3D2 face=3DArial><span style=3D'font-size:10.0pt;font-family:A=
rial'>Hi,</span></font>
<o:p></o:p></p>

<p><font size=3D2 face=3DArial><span style=3D'font-size:10.0pt;font-family:A=
rial'>We
have submitted a draft which proposes registration of an event package, base=
d
on work ongoing in 3GPP (and TISPAN).</span></font> <o:p></o:p></p>

<p><font size=3D2 face=3DArial><span style=3D'font-size:10.0pt;font-family:A=
rial'>(The
most likely protocol output is going to be a new event package).</span></fon=
t><o:p></o:p></p>

<p><font size=3D2 face=3DArial><span style=3D'font-size:10.0pt;font-family:A=
rial'>The
document can also be found in:</span></font> <o:p></o:p></p>

<p><font size=3D3 face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>=
<a
href=3D"http://www.ietf.org/proceedings/staging/draft-avasarala-dispatch-com=
m-div-notification-00.txt">http://www.ietf.org/proceedings/staging/draft-ava=
sarala-dispatch-comm-div-notification-00.txt</a><o:p></o:p></span></font></p=
>

<p><font size=3D2 face=3DArial><span style=3D'font-size:10.0pt;font-family:A=
rial'>Updates
since last version:<o:p></o:p></span></font></p>

<pre><font size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>&n=
bsp;&nbsp; Changes from draft-avasarala-sipping-comm-div-notification-01<o:p=
></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'><o:p>&nbsp;</=
o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>&nbsp;&nbsp;=
 o&nbsp; Changed contact details for co-author Subir Saha<o:p></o:p></span><=
/font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'><o:p>&nbsp;</=
o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>&nbsp;&nbsp;=
 o&nbsp; Moved from SIPPING to DISPATCH WG<o:p></o:p></span></font></pre>

<p><font size=3D2 face=3DArial><span style=3D'font-size:10.0pt;font-family:A=
rial'>Regards,</span></font>
<o:p></o:p></p>

<p><font size=3D2 face=3DArial><span style=3D'font-size:10.0pt;font-family:A=
rial'>John-Luc</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:10=
.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:10=
.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D1 color=3Dsilver face=3DArial><span style=
=3D'font-size:
7.5pt;font-family:Arial;color:silver'>--- <br>
John-Luc Bakker <br>
Standards Manager <br>
Research In Motion <br>
BlackBerry: +1 (908) 463 7321 <br>
Office: +1 (972) 373-1761 <br>
Internal: (820) 63761 <br>
E-mail: jbakker@rim.com </span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D'=
font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

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Subject: Re: [dispatch] comm-div-notification event package update
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John-Luc,

This sounds like a useful capability. Based on a quick scan, this 
proposal seems off to a good start.

For some of us that aren't actively involved with 3gpp, the references 
to 3gpp documents are somewhat inconvenient. Of particular note, I would 
like to see the xml schema for application/comm-div-info+xml included in 
this draft.

I'm also a bit confused, because you call for the same doc format to be 
used in both subscriptions and notifications. That at least seems odd, 
though perhaps it will become clear when I see the actual schema. Since 
it will perform a different role in these differing cases, it would be 
helpful if you could explain more about that.

Also I found the references to "user" somewhat ambiguous. I see a 
definition of Diverting User, and a note that "user" when not qualified 
should mean Diverting User. But I then also see "user" used in contexts 
discussing the subscriber.

For SUBSCRIBE, the R-URI identifies the "resource" to which a 
subscription is addressed. I would hope to see a very clear relationship 
spelled out between the subscription resource and the processing of 
requests that are "diverted". My assumption is that "diversion" is 
performed with respect to a particular R-URI in an INVITE request, and 
that to be notified of such a diversion one would need be subscribed to 
this event package addressed to the same URI. If that is so, it should 
be spelled out. And if that isn't quite what you mean then its even more 
important to spell out what you do mean.

	Thanks,
	Paul

John-Luc Bakker wrote:
> Hi,
> 
> We have submitted a draft which proposes registration of an event 
> package, based on work ongoing in 3GPP (and TISPAN).
> 
> (The most likely protocol output is going to be a new event package).
> 
> The document can also be found in:
> 
> http://www.ietf.org/proceedings/staging/draft-avasarala-dispatch-comm-div-notification-00.txt
> 
> Updates since last version:
> 
>    Changes from draft-avasarala-sipping-comm-div-notification-01
> 
>  
> 
>    o  Changed contact details for co-author Subir Saha
> 
>  
> 
>    o  Moved from SIPPING to DISPATCH WG
> 
> Regards,
> 
> John-Luc
> 
>  
> 
>  
> 
> ---
> John-Luc Bakker
> Standards Manager
> Research In Motion
> BlackBerry: +1 (908) 463 7321
> Office: +1 (972) 373-1761
> Internal: (820) 63761
> E-mail: jbakker@rim.com
> 
>  
> 
> ---------------------------------------------------------------------
> This transmission (including any attachments) may contain confidential 
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> transmission in error, please immediately reply to the sender and delete 
> this information from your system. Use, dissemination, distribution, or 
> reproduction of this transmission by unintended recipients is not 
> authorized and may be unlawful.
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

From mary.barnes@nortel.com  Thu Jun  4 13:16:11 2009
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Hi folks,

A reminder to folks that submitted proposals for the initial deadline
that the "charter proposals" are due next Monday, June 8th.  The
following summarizes the proposals (some already having detailed charter
proposals) that have been put forth:
1) 3rd party auth:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00052.html
2) CBUS:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00083.html
3) cc-uui:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00073.html
4) CLF:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00066.html
5) Codec:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00080.html
6) Sound level indicator:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00071.html
7) On demand media/app sharing:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00076.html

Also, if there are already drafts associated with your charter proposal,
please provide appropriate links in your email.=20

For WG participants, there are less than two weeks left (June 15th) to
provide input and feedback on the proposals in terms of whether there is
sufficient interest to warrant discussion during a f2f session in
Stockholm or whether the items can be dispatched prior to Stockholm.=20

Also, please let me know if I've missed something.=20

Regards,
Mary.=20

-----Original Message-----
From: Barnes, Mary (RICH2:AR00)=20
Sent: Friday, May 08, 2009 12:54 PM
To: dispatch@ietf.org
Cc: rai-ads@tools.ietf.org; 'Gonzalo Camarillo'
Subject: IETF-75 plans for DISPATCH=20

Hi all,

As mentioned previously, we are setting earlier deadlines for discussing
topics in DISPATCH to allow time for constructive discussions prior to
the WG session at IETF-75:=20
http://www.ietf.org/mail-archive/web/dispatch/current/msg00006.html

We are proposing to follow the deadlines for BoFs for IETF-75:
http://www.ietf.org/meetings/75/cutoff-dates.html
since we anticipate that many of the topic discussions will follow a
"mini-BoF" format, based on the DISPATCH charter:
http://www.ietf.org/html.charters/dispatch-charter.html

Thus, the following are the proposed deadlines:

May 25th - Cutoff date to notify the chairs and DISPATCH WG mailing list
of plans to submit a proposal, including the topic of the proposal. This
will help folks with similar topics to work together before the meeting.
If a preliminary charter proposal is available at this point, please
include a pointer to it.

June 8th - Cutoff for charter proposals for topics. A charter proposal
must consist of a minimum of a problem statement and at least one
milestone/deliverable. This deadline will allow time for consideration
of proposals that could potentially be "dispatched" prior to the WG
session.

June 15th - Topics that are to be the focus of IETF-75 are announced.
The idea here is to focus the discussion over the weeks preceding
IETF-75 on these main topics, noting that any updates to the documents
are due 4 weeks later.  This will ensure we have constructive
discussions before the meeting and are actually able to determine
consensus as to where work should be progressed (e.g., separate BoF, a
new WG, an existing WG, individual document, etc.) at IETF-75. Contrary
to the past approach in the SIPPING WG, we anticipate determining
consensus as to the disposition of a work item based on the level of
"commitment" by individuals to actually contribute to the work (e.g.,
conference calls, design team mailing lists, contributing text, etc.)
rather than based on "interest" in doing the work.  Note, that the exact
disposition for a topic may (per the usual process) require follow-up
and confirmation by the ADS and/or IESG (e.g., for a new WG, Bof,
individually sponsored document, etc.) or with another WG to ensure
agreement with the DISPATCH WG consensus for the topic.

Note the intention is not necessarily to preclude folks submitting
documents on other topics, however, it is very unlikely there would be
agenda time allocated to documents/topics submitted after the deadlines.
We can include one or two slides on these topics in the DISPATCH WG
chair charts, as we've done in the past in the SIPPING WG chair charts,
so that the authors can gather other interested parties to contribute to
the work.=20

We appreciate the patience of the WG as we evolve the model for bringing
in new work and welcome feedback.=20

Regards,
Mary and Gonzalo
DISPATCH WG co-chairs=20



From Even.roni@huawei.com  Thu Jun  4 14:10:04 2009
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Date: Fri, 05 Jun 2009 00:08:25 +0300
From: Roni Even <Even.roni@huawei.com>
In-reply-to: <1ECE0EB50388174790F9694F77522CCF1E590C4B@zrc2hxm0.corp.nortel.com>
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Subject: Re: [dispatch] Reminder: IETF-75 plans for DISPATCH
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Hi Mary,
Looking at the proposal I would like to point out that the last three items
were typically discussed before we had the dispatch group in AVT, XCON  or
MMUSIC. 
I think that the people who follow AVT and MMUSIC are not the typical
dispatch subscriber and if a meaningful discussion is solicited than those
message should have gone also to the AVT and MMUSIC mailing list. I did
forward the codec email to the AVT working group and urged people to join
the discussion on the codec topic but maybe the dispatch chairs should look
at the proposals and send them to current working groups that may provide
better feedback

Roni Even

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
Of Mary Barnes
Sent: Thursday, June 04, 2009 11:19 PM
To: dispatch@ietf.org
Cc: rai-ads@tools.ietf.org
Subject: [dispatch] Reminder: IETF-75 plans for DISPATCH

Hi folks,

A reminder to folks that submitted proposals for the initial deadline
that the "charter proposals" are due next Monday, June 8th.  The
following summarizes the proposals (some already having detailed charter
proposals) that have been put forth:
1) 3rd party auth:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00052.html
2) CBUS:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00083.html
3) cc-uui:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00073.html
4) CLF:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00066.html
5) Codec:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00080.html
6) Sound level indicator:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00071.html
7) On demand media/app sharing:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00076.html

Also, if there are already drafts associated with your charter proposal,
please provide appropriate links in your email. 

For WG participants, there are less than two weeks left (June 15th) to
provide input and feedback on the proposals in terms of whether there is
sufficient interest to warrant discussion during a f2f session in
Stockholm or whether the items can be dispatched prior to Stockholm. 

Also, please let me know if I've missed something. 

Regards,
Mary. 

-----Original Message-----
From: Barnes, Mary (RICH2:AR00) 
Sent: Friday, May 08, 2009 12:54 PM
To: dispatch@ietf.org
Cc: rai-ads@tools.ietf.org; 'Gonzalo Camarillo'
Subject: IETF-75 plans for DISPATCH 

Hi all,

As mentioned previously, we are setting earlier deadlines for discussing
topics in DISPATCH to allow time for constructive discussions prior to
the WG session at IETF-75: 
http://www.ietf.org/mail-archive/web/dispatch/current/msg00006.html

We are proposing to follow the deadlines for BoFs for IETF-75:
http://www.ietf.org/meetings/75/cutoff-dates.html
since we anticipate that many of the topic discussions will follow a
"mini-BoF" format, based on the DISPATCH charter:
http://www.ietf.org/html.charters/dispatch-charter.html

Thus, the following are the proposed deadlines:

May 25th - Cutoff date to notify the chairs and DISPATCH WG mailing list
of plans to submit a proposal, including the topic of the proposal. This
will help folks with similar topics to work together before the meeting.
If a preliminary charter proposal is available at this point, please
include a pointer to it.

June 8th - Cutoff for charter proposals for topics. A charter proposal
must consist of a minimum of a problem statement and at least one
milestone/deliverable. This deadline will allow time for consideration
of proposals that could potentially be "dispatched" prior to the WG
session.

June 15th - Topics that are to be the focus of IETF-75 are announced.
The idea here is to focus the discussion over the weeks preceding
IETF-75 on these main topics, noting that any updates to the documents
are due 4 weeks later.  This will ensure we have constructive
discussions before the meeting and are actually able to determine
consensus as to where work should be progressed (e.g., separate BoF, a
new WG, an existing WG, individual document, etc.) at IETF-75. Contrary
to the past approach in the SIPPING WG, we anticipate determining
consensus as to the disposition of a work item based on the level of
"commitment" by individuals to actually contribute to the work (e.g.,
conference calls, design team mailing lists, contributing text, etc.)
rather than based on "interest" in doing the work.  Note, that the exact
disposition for a topic may (per the usual process) require follow-up
and confirmation by the ADS and/or IESG (e.g., for a new WG, Bof,
individually sponsored document, etc.) or with another WG to ensure
agreement with the DISPATCH WG consensus for the topic.

Note the intention is not necessarily to preclude folks submitting
documents on other topics, however, it is very unlikely there would be
agenda time allocated to documents/topics submitted after the deadlines.
We can include one or two slides on these topics in the DISPATCH WG
chair charts, as we've done in the past in the SIPPING WG chair charts,
so that the authors can gather other interested parties to contribute to
the work. 

We appreciate the patience of the WG as we evolve the model for bringing
in new work and welcome feedback. 

Regards,
Mary and Gonzalo
DISPATCH WG co-chairs 


_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch


From mary.barnes@nortel.com  Thu Jun  4 14:39:03 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Roni Even" <Even.roni@huawei.com>, <dispatch@ietf.org>
Cc: rai-ads@tools.ietf.org
Subject: Re: [dispatch] Reminder: IETF-75 plans for DISPATCH
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Hi Roni,

I don't disagree that any technical work for the last 3 might fit into
the WGs you suggest. However, the point of the discussion is to agree
how to proceed with the work - i.e., should it go to WG x or does it
warrant a separate WG/BoF or is it something for which there is not
sufficient interest that the work should either be progressed as
individual or abandoned.  I agree that those last 3 categories are
beyond the scope (for the most part) of work that we discussed in the
SIPPING WG in the past. But, the role of DISPATCH is far broader than
SIPPING: http://www.ietf.org/html.charters/dispatch-charter.html
" The Dispatch working group is chartered to consider proposals for
new work in the RAI area and identify, or help create, an appropriate
venue for the work."=20

There is no mention that the proposals must be SIP specific nor is there
any restrictions as to what sort of work should be discussed in DISPATCH
WG - if it relates to RAI in anyway, then it's appropriate.  Also, keep
in mind that there may be new mailing lists setup for detailed
discussions of some of these topics. So, I don't think it's reasonable
to suggest that anything and everything that folks that follow AVT and
MMUSIC would be interested in will always be discussed in those WGs.  As
a WG chair, I think you can do a good job of letting your WG know when
relevant topics are being discussed for folks that might not want to be
on the DISPATCH WG mailing list. I have approved quite a few posts (and
added the folks as being able to post without subscribing to the ML)
over the past week to facilitate input from folks that are not
subscribed to the DISPATCH ML.=20

My understanding was that one of the reasons for the new structure was
to minimize new work having to shop for a WG - i.e., starting in one WG
and being told that the work belongs in another WG and then another,
etc.  There have been cases in the past where potential new work items
were shuttled between 3+ different WGs (with no one ever taking
ownership) and others presented in multiple WGs during a given IETF
meeting.  In general, this new approach should avoid both those
problems.=20

Now, this isn't to say that all potential new work items MUST first be
discussed in DISPATCH - there will be quite clear cases where the
appropriate WG is obvious from the outset.=20

And, again, we do appreciate patience as we work through this new model.


Regards,
Mary.
(as DISPATCH WG co-chair)

-----Original Message-----
From: Roni Even [mailto:Even.roni@huawei.com]=20
Sent: Thursday, June 04, 2009 4:08 PM
To: Barnes, Mary (RICH2:AR00); dispatch@ietf.org
Cc: rai-ads@tools.ietf.org
Subject: RE: [dispatch] Reminder: IETF-75 plans for DISPATCH

Hi Mary,
Looking at the proposal I would like to point out that the last three
items were typically discussed before we had the dispatch group in AVT,
XCON  or MMUSIC.=20
I think that the people who follow AVT and MMUSIC are not the typical
dispatch subscriber and if a meaningful discussion is solicited than
those message should have gone also to the AVT and MMUSIC mailing list.
I did forward the codec email to the AVT working group and urged people
to join the discussion on the codec topic but maybe the dispatch chairs
should look at the proposals and send them to current working groups
that may provide better feedback

Roni Even

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Mary Barnes
Sent: Thursday, June 04, 2009 11:19 PM
To: dispatch@ietf.org
Cc: rai-ads@tools.ietf.org
Subject: [dispatch] Reminder: IETF-75 plans for DISPATCH

Hi folks,

A reminder to folks that submitted proposals for the initial deadline
that the "charter proposals" are due next Monday, June 8th.  The
following summarizes the proposals (some already having detailed charter
proposals) that have been put forth:
1) 3rd party auth:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00052.html
2) CBUS:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00083.html
3) cc-uui:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00073.html
4) CLF:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00066.html
5) Codec:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00080.html
6) Sound level indicator:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00071.html
7) On demand media/app sharing:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00076.html

Also, if there are already drafts associated with your charter proposal,
please provide appropriate links in your email.=20

For WG participants, there are less than two weeks left (June 15th) to
provide input and feedback on the proposals in terms of whether there is
sufficient interest to warrant discussion during a f2f session in
Stockholm or whether the items can be dispatched prior to Stockholm.=20

Also, please let me know if I've missed something.=20

Regards,
Mary.=20

-----Original Message-----
From: Barnes, Mary (RICH2:AR00)
Sent: Friday, May 08, 2009 12:54 PM
To: dispatch@ietf.org
Cc: rai-ads@tools.ietf.org; 'Gonzalo Camarillo'
Subject: IETF-75 plans for DISPATCH=20

Hi all,

As mentioned previously, we are setting earlier deadlines for discussing
topics in DISPATCH to allow time for constructive discussions prior to
the WG session at IETF-75:=20
http://www.ietf.org/mail-archive/web/dispatch/current/msg00006.html

We are proposing to follow the deadlines for BoFs for IETF-75:
http://www.ietf.org/meetings/75/cutoff-dates.html
since we anticipate that many of the topic discussions will follow a
"mini-BoF" format, based on the DISPATCH charter:
http://www.ietf.org/html.charters/dispatch-charter.html

Thus, the following are the proposed deadlines:

May 25th - Cutoff date to notify the chairs and DISPATCH WG mailing list
of plans to submit a proposal, including the topic of the proposal. This
will help folks with similar topics to work together before the meeting.
If a preliminary charter proposal is available at this point, please
include a pointer to it.

June 8th - Cutoff for charter proposals for topics. A charter proposal
must consist of a minimum of a problem statement and at least one
milestone/deliverable. This deadline will allow time for consideration
of proposals that could potentially be "dispatched" prior to the WG
session.

June 15th - Topics that are to be the focus of IETF-75 are announced.
The idea here is to focus the discussion over the weeks preceding
IETF-75 on these main topics, noting that any updates to the documents
are due 4 weeks later.  This will ensure we have constructive
discussions before the meeting and are actually able to determine
consensus as to where work should be progressed (e.g., separate BoF, a
new WG, an existing WG, individual document, etc.) at IETF-75. Contrary
to the past approach in the SIPPING WG, we anticipate determining
consensus as to the disposition of a work item based on the level of
"commitment" by individuals to actually contribute to the work (e.g.,
conference calls, design team mailing lists, contributing text, etc.)
rather than based on "interest" in doing the work.  Note, that the exact
disposition for a topic may (per the usual process) require follow-up
and confirmation by the ADS and/or IESG (e.g., for a new WG, Bof,
individually sponsored document, etc.) or with another WG to ensure
agreement with the DISPATCH WG consensus for the topic.

Note the intention is not necessarily to preclude folks submitting
documents on other topics, however, it is very unlikely there would be
agenda time allocated to documents/topics submitted after the deadlines.
We can include one or two slides on these topics in the DISPATCH WG
chair charts, as we've done in the past in the SIPPING WG chair charts,
so that the authors can gather other interested parties to contribute to
the work.=20

We appreciate the patience of the WG as we evolve the model for bringing
in new work and welcome feedback.=20

Regards,
Mary and Gonzalo
DISPATCH WG co-chairs=20


_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch


From ron.even.tlv@gmail.com  Thu Jun  4 14:36:38 2009
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Emil Ivov'" <emcho@sip-communicator.org>, <dispatch@ietf.org>
References: <4A15D99F.9060100@sip-communicator.org>	<EA1CBEDB-319A-43BC-B874-909B8267F017@cisco.com>	<4A1D9C2C.8030108@telecomitalia.it>	<0D5F89FAC29E2C41B98A6A762007F5D001F17EE4@GBNTHT12009MSX.gb002.siemens.net>	<4A1E8B2E.1030906@sip-communicator.org>	<0D5F89FAC29E2C41B98A6A762007F5D001F18237@GBNTHT12009MSX.gb002.siemens.net> <4A1EBBAE.4090609@sip-communicator.org>
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Subject: Re: [dispatch] Yet another problem to dispatch
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Emil,
I think that John means that what you call loudest speaker is not
necessarily a speaker but may represent some noise which is high enough to
qualify as one of the N loudest speakers.
The conferencing bridge try to address this problem by defining some
thresholds for accepting a user to the mix. 

I also looked at the draft and would like to see a discussion about the
current options to receive the active speakers indications. I am not so sure
about the benefit of displaying sound level, note that the mixers usually do
some normalization on the volume of the mixed streams to enhance the loudest
speaker and to have a common level regardless of the source level in order
to prevent fluctuation of the voice that may be caused by different
microphones, encoders and background noise.  

Roni Even


-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
Of Emil Ivov
Sent: Thursday, May 28, 2009 7:29 PM
To: dispatch@ietf.org
Subject: Re: [dispatch] Yet another problem to dispatch

Hey John,

Elwell, John wrote:
> Emil,
> 
> But then you would need some sort of throttling, to prevent the UI
> constantly changing the list of names whose volumes are displayed. If
> the mixer were to do this throttling it would reduce the frequency of
> messages, which might allow a bigger choice of potential solutions.

I am not sure I understand exactly what you mean by throttling. Mixers
could be implemented to only include sound level information for the
loudest speakers thus having the possibility to ignore silent and almost
silent ones. It should be up to the mixer to determine the exact
threshold since that's where bandwidth would be most precious. Is this
what you are suggesting?

Then, once a client gets this information it can consider (and I guess
that's in line with 3550) that all CSRCs not present in a particular
packet have been silent for the corresponding period of time.

As to how exactly the client would render the sound level information,
whether or not it would display all of it, and whether it would somehow
rearrange participants, that should really be up to the UI implementors.

Does this make sense?

Cheers
Emil


> 
> These are just initial thoughts on my part, not firm opinions.
> 
> John
> 
> 
>> -----Original Message-----
>> From: Emil Ivov [mailto:emil@sip-communicator.org] On Behalf 
>> Of Emil Ivov
>> Sent: 28 May 2009 14:02
>> To: Elwell, John
>> Cc: Enrico Marocco; Cullen Jennings; dispatch@ietf.org
>> Subject: Re: [dispatch] Yet another problem to dispatch
>>
>> Hey John,
>>
>> Elwell, John wrote:
>>> Background noise in conferences is certainly a big issue. 
>> The web-type
>>> interface that Cullen mentions does in fact allow the moderator to
>>> momentarily mute each suspected noisy party in turn to 
>> discover who is
>>> the culprit. Clearly this is not scalable to large 
>> conferences. But also
>>> a simultaneous display of noise levels (e.g., in the form 
>> of bar graphs)
>>> is not scalable. 
>> One way to address this would be to allow mixer implementors to only
>> include sound levels for the N loudest participants. UI implementors
>> could then use a similar algorithm when presenting the information.
>>
>> How does this sound?
>>
>> Cheers
>> Emil
>>
>>
>>> Whilst it might be easy to find protocol solutions, we
>>> need to be confident that people will find ways of 
>> exploiting them in a
>>> reasonable way at the user interface. Certainly worth exploring.
>>>
>>> John
>>>
>>>
>>>> -----Original Message-----
>>>> From: dispatch-bounces@ietf.org 
>>>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Enrico Marocco
>>>> Sent: 27 May 2009 21:02
>>>> To: Cullen Jennings
>>>> Cc: dispatch@ietf.org
>>>> Subject: Re: [dispatch] Yet another problem to dispatch
>>>>
>>>> Cullen Jennings wrote:
>>>>> I find this "active speaker" problem very interesting. Often the  
>>>>> device that wants to display the active speaker information 
>>>> is not the  
>>>>> same as the device dealing with the RTP. For example, most the  
>>>>> conference systems I am using a phone but there is a web 
>> interface  
>>>>> that is dealing with active speaker information on my 
>>>> browser which is  
>>>>> no on my phone. Be interesting to look at all the existing 
>>>> approaches  
>>>>> to this.
>>>> Yes, I'm familiar with that kind of systems. However, the 
>>>> information we
>>>> are considering -- instantaneous sound levels of each 
>>>> participant -- is
>>>> definitely changing more rapidly than just active speaker 
>>>> indication and
>>>> also has more stringent synchronization requirements. 
>> That's why our
>>>> favorite approach would be to carry it in the media streams 
>>>> themselves,
>>>>  with some sort of a modulation of the RTP CSRC fields 
>> that could be
>>>> considered either an hack or an elegant solution. I would be very
>>>> interested in hearing any opinions about that.
>>>>
>>>> -- 
>>>> Ciao,
>>>> Enrico
>>>>
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>
>> -- 
>> Emil Ivov, Ph.D.                               30a rue de la Patrie
>> Project Lead                                   67300 Schiltigheim
>> SIP Communicator
>> emcho@sip-communicator.org                     FAX:   
>> +33.1.77.62.47.31
>> http://sip-communicator.org                    PHONE: 
>> +33.1.77.62.43.30
>>
> 

-- 
Emil Ivov, Ph.D.                               30a rue de la Patrie
Project Lead                                   67300 Schiltigheim
SIP Communicator
emcho@sip-communicator.org                     FAX:   +33.1.77.62.47.31
http://sip-communicator.org                    PHONE: +33.1.77.62.43.30
_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch


From mary.barnes@nortel.com  Thu Jun  4 14:44:33 2009
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Thread-Topic: [dispatch] Reminder: IETF-75 plans for DISPATCH
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Roni Even" <Even.roni@huawei.com>, <dispatch@ietf.org>
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Subject: Re: [dispatch] Reminder: IETF-75 plans for DISPATCH
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One last point - for item 6) for example, if you as chair of AVT can
gather consensus from your WG that this item is of interest, then we can
dispatch the item now (pending AD approval, of course) - i.e., there is
no requirement at all that we discuss something at a f2f meeting before
we dispatch it. =20

Thanks,
Mary.=20

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Barnes, Mary (RICH2:AR00)
Sent: Thursday, June 04, 2009 4:41 PM
To: Roni Even; dispatch@ietf.org
Cc: rai-ads@tools.ietf.org
Subject: Re: [dispatch] Reminder: IETF-75 plans for DISPATCH

Hi Roni,

I don't disagree that any technical work for the last 3 might fit into
the WGs you suggest. However, the point of the discussion is to agree
how to proceed with the work - i.e., should it go to WG x or does it
warrant a separate WG/BoF or is it something for which there is not
sufficient interest that the work should either be progressed as
individual or abandoned.  I agree that those last 3 categories are
beyond the scope (for the most part) of work that we discussed in the
SIPPING WG in the past. But, the role of DISPATCH is far broader than
SIPPING: http://www.ietf.org/html.charters/dispatch-charter.html
" The Dispatch working group is chartered to consider proposals for new
work in the RAI area and identify, or help create, an appropriate venue
for the work."=20

There is no mention that the proposals must be SIP specific nor is there
any restrictions as to what sort of work should be discussed in DISPATCH
WG - if it relates to RAI in anyway, then it's appropriate.  Also, keep
in mind that there may be new mailing lists setup for detailed
discussions of some of these topics. So, I don't think it's reasonable
to suggest that anything and everything that folks that follow AVT and
MMUSIC would be interested in will always be discussed in those WGs.  As
a WG chair, I think you can do a good job of letting your WG know when
relevant topics are being discussed for folks that might not want to be
on the DISPATCH WG mailing list. I have approved quite a few posts (and
added the folks as being able to post without subscribing to the ML)
over the past week to facilitate input from folks that are not
subscribed to the DISPATCH ML.=20

My understanding was that one of the reasons for the new structure was
to minimize new work having to shop for a WG - i.e., starting in one WG
and being told that the work belongs in another WG and then another,
etc.  There have been cases in the past where potential new work items
were shuttled between 3+ different WGs (with no one ever taking
ownership) and others presented in multiple WGs during a given IETF
meeting.  In general, this new approach should avoid both those
problems.=20

Now, this isn't to say that all potential new work items MUST first be
discussed in DISPATCH - there will be quite clear cases where the
appropriate WG is obvious from the outset.=20

And, again, we do appreciate patience as we work through this new model.


Regards,
Mary.
(as DISPATCH WG co-chair)

-----Original Message-----
From: Roni Even [mailto:Even.roni@huawei.com]
Sent: Thursday, June 04, 2009 4:08 PM
To: Barnes, Mary (RICH2:AR00); dispatch@ietf.org
Cc: rai-ads@tools.ietf.org
Subject: RE: [dispatch] Reminder: IETF-75 plans for DISPATCH

Hi Mary,
Looking at the proposal I would like to point out that the last three
items were typically discussed before we had the dispatch group in AVT,
XCON  or MMUSIC.=20
I think that the people who follow AVT and MMUSIC are not the typical
dispatch subscriber and if a meaningful discussion is solicited than
those message should have gone also to the AVT and MMUSIC mailing list.
I did forward the codec email to the AVT working group and urged people
to join the discussion on the codec topic but maybe the dispatch chairs
should look at the proposals and send them to current working groups
that may provide better feedback

Roni Even

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Mary Barnes
Sent: Thursday, June 04, 2009 11:19 PM
To: dispatch@ietf.org
Cc: rai-ads@tools.ietf.org
Subject: [dispatch] Reminder: IETF-75 plans for DISPATCH

Hi folks,

A reminder to folks that submitted proposals for the initial deadline
that the "charter proposals" are due next Monday, June 8th.  The
following summarizes the proposals (some already having detailed charter
proposals) that have been put forth:
1) 3rd party auth:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00052.html
2) CBUS:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00083.html
3) cc-uui:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00073.html
4) CLF:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00066.html
5) Codec:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00080.html
6) Sound level indicator:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00071.html
7) On demand media/app sharing:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00076.html

Also, if there are already drafts associated with your charter proposal,
please provide appropriate links in your email.=20

For WG participants, there are less than two weeks left (June 15th) to
provide input and feedback on the proposals in terms of whether there is
sufficient interest to warrant discussion during a f2f session in
Stockholm or whether the items can be dispatched prior to Stockholm.=20

Also, please let me know if I've missed something.=20

Regards,
Mary.=20

-----Original Message-----
From: Barnes, Mary (RICH2:AR00)
Sent: Friday, May 08, 2009 12:54 PM
To: dispatch@ietf.org
Cc: rai-ads@tools.ietf.org; 'Gonzalo Camarillo'
Subject: IETF-75 plans for DISPATCH=20

Hi all,

As mentioned previously, we are setting earlier deadlines for discussing
topics in DISPATCH to allow time for constructive discussions prior to
the WG session at IETF-75:=20
http://www.ietf.org/mail-archive/web/dispatch/current/msg00006.html

We are proposing to follow the deadlines for BoFs for IETF-75:
http://www.ietf.org/meetings/75/cutoff-dates.html
since we anticipate that many of the topic discussions will follow a
"mini-BoF" format, based on the DISPATCH charter:
http://www.ietf.org/html.charters/dispatch-charter.html

Thus, the following are the proposed deadlines:

May 25th - Cutoff date to notify the chairs and DISPATCH WG mailing list
of plans to submit a proposal, including the topic of the proposal. This
will help folks with similar topics to work together before the meeting.
If a preliminary charter proposal is available at this point, please
include a pointer to it.

June 8th - Cutoff for charter proposals for topics. A charter proposal
must consist of a minimum of a problem statement and at least one
milestone/deliverable. This deadline will allow time for consideration
of proposals that could potentially be "dispatched" prior to the WG
session.

June 15th - Topics that are to be the focus of IETF-75 are announced.
The idea here is to focus the discussion over the weeks preceding
IETF-75 on these main topics, noting that any updates to the documents
are due 4 weeks later.  This will ensure we have constructive
discussions before the meeting and are actually able to determine
consensus as to where work should be progressed (e.g., separate BoF, a
new WG, an existing WG, individual document, etc.) at IETF-75. Contrary
to the past approach in the SIPPING WG, we anticipate determining
consensus as to the disposition of a work item based on the level of
"commitment" by individuals to actually contribute to the work (e.g.,
conference calls, design team mailing lists, contributing text, etc.)
rather than based on "interest" in doing the work.  Note, that the exact
disposition for a topic may (per the usual process) require follow-up
and confirmation by the ADS and/or IESG (e.g., for a new WG, Bof,
individually sponsored document, etc.) or with another WG to ensure
agreement with the DISPATCH WG consensus for the topic.

Note the intention is not necessarily to preclude folks submitting
documents on other topics, however, it is very unlikely there would be
agenda time allocated to documents/topics submitted after the deadlines.
We can include one or two slides on these topics in the DISPATCH WG
chair charts, as we've done in the past in the SIPPING WG chair charts,
so that the authors can gather other interested parties to contribute to
the work.=20

We appreciate the patience of the WG as we evolve the model for bringing
in new work and welcome feedback.=20

Regards,
Mary and Gonzalo
DISPATCH WG co-chairs=20


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Date: Fri, 05 Jun 2009 07:21:29 +0300
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Mary,
I was not arguing the dispatch charter. I wanted to point out that since
dispatch charter is to start discussion on any RAI new topic it would make
sense from the dispatch chairs to forward these new topics to other relevant
RAI WGs mailing lists since these lists members may not follow the dispatch
list.

Roni Even

-----Original Message-----
From: Mary Barnes [mailto:mary.barnes@nortel.com] 
Sent: Friday, June 05, 2009 12:41 AM
To: Roni Even; dispatch@ietf.org
Cc: rai-ads@tools.ietf.org
Subject: RE: [dispatch] Reminder: IETF-75 plans for DISPATCH

Hi Roni,

I don't disagree that any technical work for the last 3 might fit into
the WGs you suggest. However, the point of the discussion is to agree
how to proceed with the work - i.e., should it go to WG x or does it
warrant a separate WG/BoF or is it something for which there is not
sufficient interest that the work should either be progressed as
individual or abandoned.  I agree that those last 3 categories are
beyond the scope (for the most part) of work that we discussed in the
SIPPING WG in the past. But, the role of DISPATCH is far broader than
SIPPING: http://www.ietf.org/html.charters/dispatch-charter.html
" The Dispatch working group is chartered to consider proposals for
new work in the RAI area and identify, or help create, an appropriate
venue for the work." 

There is no mention that the proposals must be SIP specific nor is there
any restrictions as to what sort of work should be discussed in DISPATCH
WG - if it relates to RAI in anyway, then it's appropriate.  Also, keep
in mind that there may be new mailing lists setup for detailed
discussions of some of these topics. So, I don't think it's reasonable
to suggest that anything and everything that folks that follow AVT and
MMUSIC would be interested in will always be discussed in those WGs.  As
a WG chair, I think you can do a good job of letting your WG know when
relevant topics are being discussed for folks that might not want to be
on the DISPATCH WG mailing list. I have approved quite a few posts (and
added the folks as being able to post without subscribing to the ML)
over the past week to facilitate input from folks that are not
subscribed to the DISPATCH ML. 

My understanding was that one of the reasons for the new structure was
to minimize new work having to shop for a WG - i.e., starting in one WG
and being told that the work belongs in another WG and then another,
etc.  There have been cases in the past where potential new work items
were shuttled between 3+ different WGs (with no one ever taking
ownership) and others presented in multiple WGs during a given IETF
meeting.  In general, this new approach should avoid both those
problems. 

Now, this isn't to say that all potential new work items MUST first be
discussed in DISPATCH - there will be quite clear cases where the
appropriate WG is obvious from the outset. 

And, again, we do appreciate patience as we work through this new model.


Regards,
Mary.
(as DISPATCH WG co-chair)

-----Original Message-----
From: Roni Even [mailto:Even.roni@huawei.com] 
Sent: Thursday, June 04, 2009 4:08 PM
To: Barnes, Mary (RICH2:AR00); dispatch@ietf.org
Cc: rai-ads@tools.ietf.org
Subject: RE: [dispatch] Reminder: IETF-75 plans for DISPATCH

Hi Mary,
Looking at the proposal I would like to point out that the last three
items were typically discussed before we had the dispatch group in AVT,
XCON  or MMUSIC. 
I think that the people who follow AVT and MMUSIC are not the typical
dispatch subscriber and if a meaningful discussion is solicited than
those message should have gone also to the AVT and MMUSIC mailing list.
I did forward the codec email to the AVT working group and urged people
to join the discussion on the codec topic but maybe the dispatch chairs
should look at the proposals and send them to current working groups
that may provide better feedback

Roni Even

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Mary Barnes
Sent: Thursday, June 04, 2009 11:19 PM
To: dispatch@ietf.org
Cc: rai-ads@tools.ietf.org
Subject: [dispatch] Reminder: IETF-75 plans for DISPATCH

Hi folks,

A reminder to folks that submitted proposals for the initial deadline
that the "charter proposals" are due next Monday, June 8th.  The
following summarizes the proposals (some already having detailed charter
proposals) that have been put forth:
1) 3rd party auth:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00052.html
2) CBUS:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00083.html
3) cc-uui:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00073.html
4) CLF:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00066.html
5) Codec:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00080.html
6) Sound level indicator:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00071.html
7) On demand media/app sharing:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00076.html

Also, if there are already drafts associated with your charter proposal,
please provide appropriate links in your email. 

For WG participants, there are less than two weeks left (June 15th) to
provide input and feedback on the proposals in terms of whether there is
sufficient interest to warrant discussion during a f2f session in
Stockholm or whether the items can be dispatched prior to Stockholm. 

Also, please let me know if I've missed something. 

Regards,
Mary. 

-----Original Message-----
From: Barnes, Mary (RICH2:AR00)
Sent: Friday, May 08, 2009 12:54 PM
To: dispatch@ietf.org
Cc: rai-ads@tools.ietf.org; 'Gonzalo Camarillo'
Subject: IETF-75 plans for DISPATCH 

Hi all,

As mentioned previously, we are setting earlier deadlines for discussing
topics in DISPATCH to allow time for constructive discussions prior to
the WG session at IETF-75: 
http://www.ietf.org/mail-archive/web/dispatch/current/msg00006.html

We are proposing to follow the deadlines for BoFs for IETF-75:
http://www.ietf.org/meetings/75/cutoff-dates.html
since we anticipate that many of the topic discussions will follow a
"mini-BoF" format, based on the DISPATCH charter:
http://www.ietf.org/html.charters/dispatch-charter.html

Thus, the following are the proposed deadlines:

May 25th - Cutoff date to notify the chairs and DISPATCH WG mailing list
of plans to submit a proposal, including the topic of the proposal. This
will help folks with similar topics to work together before the meeting.
If a preliminary charter proposal is available at this point, please
include a pointer to it.

June 8th - Cutoff for charter proposals for topics. A charter proposal
must consist of a minimum of a problem statement and at least one
milestone/deliverable. This deadline will allow time for consideration
of proposals that could potentially be "dispatched" prior to the WG
session.

June 15th - Topics that are to be the focus of IETF-75 are announced.
The idea here is to focus the discussion over the weeks preceding
IETF-75 on these main topics, noting that any updates to the documents
are due 4 weeks later.  This will ensure we have constructive
discussions before the meeting and are actually able to determine
consensus as to where work should be progressed (e.g., separate BoF, a
new WG, an existing WG, individual document, etc.) at IETF-75. Contrary
to the past approach in the SIPPING WG, we anticipate determining
consensus as to the disposition of a work item based on the level of
"commitment" by individuals to actually contribute to the work (e.g.,
conference calls, design team mailing lists, contributing text, etc.)
rather than based on "interest" in doing the work.  Note, that the exact
disposition for a topic may (per the usual process) require follow-up
and confirmation by the ADS and/or IESG (e.g., for a new WG, Bof,
individually sponsored document, etc.) or with another WG to ensure
agreement with the DISPATCH WG consensus for the topic.

Note the intention is not necessarily to preclude folks submitting
documents on other topics, however, it is very unlikely there would be
agenda time allocated to documents/topics submitted after the deadlines.
We can include one or two slides on these topics in the DISPATCH WG
chair charts, as we've done in the past in the SIPPING WG chair charts,
so that the authors can gather other interested parties to contribute to
the work. 

We appreciate the patience of the WG as we evolve the model for bringing
in new work and welcome feedback. 

Regards,
Mary and Gonzalo
DISPATCH WG co-chairs 


_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch


From Even.roni@huawei.com  Thu Jun  4 21:31:26 2009
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Date: Fri, 05 Jun 2009 07:29:56 +0300
From: Roni Even <Even.roni@huawei.com>
In-reply-to: <1ECE0EB50388174790F9694F77522CCF1E590E67@zrc2hxm0.corp.nortel.com>
To: 'Mary Barnes' <mary.barnes@nortel.com>, dispatch@ietf.org
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Cc: rai-ads@tools.ietf.org
Subject: Re: [dispatch] Reminder: IETF-75 plans for DISPATCH
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Mary,
On item 6 I am not sure if it is AVT or XCON. One of the problems in this
request is the synchronization between the application level user of the
conference (the display name) and his media streams. Today this is handled
in the conference event package. 
Having the audio level in the mixed RTP/RTCP stream will not provide enough
information to the client to synchronize the SSRC with the name the user is
suing in the conference. The CNAME does not supply this relation. 

I think one of the challenges will be what to discuss in dispatch. I hope
that the in the f2f meeting the discussion will be not on the technical
solution but more on why it came to dispatch, why did the authors of the
draft think that it does not fit the charter of any existing WG.


Roni


-----Original Message-----
From: Mary Barnes [mailto:mary.barnes@nortel.com] 
Sent: Friday, June 05, 2009 12:47 AM
To: Roni Even; dispatch@ietf.org
Cc: rai-ads@tools.ietf.org
Subject: RE: [dispatch] Reminder: IETF-75 plans for DISPATCH

One last point - for item 6) for example, if you as chair of AVT can
gather consensus from your WG that this item is of interest, then we can
dispatch the item now (pending AD approval, of course) - i.e., there is
no requirement at all that we discuss something at a f2f meeting before
we dispatch it.  

Thanks,
Mary. 

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Barnes, Mary (RICH2:AR00)
Sent: Thursday, June 04, 2009 4:41 PM
To: Roni Even; dispatch@ietf.org
Cc: rai-ads@tools.ietf.org
Subject: Re: [dispatch] Reminder: IETF-75 plans for DISPATCH

Hi Roni,

I don't disagree that any technical work for the last 3 might fit into
the WGs you suggest. However, the point of the discussion is to agree
how to proceed with the work - i.e., should it go to WG x or does it
warrant a separate WG/BoF or is it something for which there is not
sufficient interest that the work should either be progressed as
individual or abandoned.  I agree that those last 3 categories are
beyond the scope (for the most part) of work that we discussed in the
SIPPING WG in the past. But, the role of DISPATCH is far broader than
SIPPING: http://www.ietf.org/html.charters/dispatch-charter.html
" The Dispatch working group is chartered to consider proposals for new
work in the RAI area and identify, or help create, an appropriate venue
for the work." 

There is no mention that the proposals must be SIP specific nor is there
any restrictions as to what sort of work should be discussed in DISPATCH
WG - if it relates to RAI in anyway, then it's appropriate.  Also, keep
in mind that there may be new mailing lists setup for detailed
discussions of some of these topics. So, I don't think it's reasonable
to suggest that anything and everything that folks that follow AVT and
MMUSIC would be interested in will always be discussed in those WGs.  As
a WG chair, I think you can do a good job of letting your WG know when
relevant topics are being discussed for folks that might not want to be
on the DISPATCH WG mailing list. I have approved quite a few posts (and
added the folks as being able to post without subscribing to the ML)
over the past week to facilitate input from folks that are not
subscribed to the DISPATCH ML. 

My understanding was that one of the reasons for the new structure was
to minimize new work having to shop for a WG - i.e., starting in one WG
and being told that the work belongs in another WG and then another,
etc.  There have been cases in the past where potential new work items
were shuttled between 3+ different WGs (with no one ever taking
ownership) and others presented in multiple WGs during a given IETF
meeting.  In general, this new approach should avoid both those
problems. 

Now, this isn't to say that all potential new work items MUST first be
discussed in DISPATCH - there will be quite clear cases where the
appropriate WG is obvious from the outset. 

And, again, we do appreciate patience as we work through this new model.


Regards,
Mary.
(as DISPATCH WG co-chair)

-----Original Message-----
From: Roni Even [mailto:Even.roni@huawei.com]
Sent: Thursday, June 04, 2009 4:08 PM
To: Barnes, Mary (RICH2:AR00); dispatch@ietf.org
Cc: rai-ads@tools.ietf.org
Subject: RE: [dispatch] Reminder: IETF-75 plans for DISPATCH

Hi Mary,
Looking at the proposal I would like to point out that the last three
items were typically discussed before we had the dispatch group in AVT,
XCON  or MMUSIC. 
I think that the people who follow AVT and MMUSIC are not the typical
dispatch subscriber and if a meaningful discussion is solicited than
those message should have gone also to the AVT and MMUSIC mailing list.
I did forward the codec email to the AVT working group and urged people
to join the discussion on the codec topic but maybe the dispatch chairs
should look at the proposals and send them to current working groups
that may provide better feedback

Roni Even

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Mary Barnes
Sent: Thursday, June 04, 2009 11:19 PM
To: dispatch@ietf.org
Cc: rai-ads@tools.ietf.org
Subject: [dispatch] Reminder: IETF-75 plans for DISPATCH

Hi folks,

A reminder to folks that submitted proposals for the initial deadline
that the "charter proposals" are due next Monday, June 8th.  The
following summarizes the proposals (some already having detailed charter
proposals) that have been put forth:
1) 3rd party auth:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00052.html
2) CBUS:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00083.html
3) cc-uui:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00073.html
4) CLF:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00066.html
5) Codec:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00080.html
6) Sound level indicator:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00071.html
7) On demand media/app sharing:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00076.html

Also, if there are already drafts associated with your charter proposal,
please provide appropriate links in your email. 

For WG participants, there are less than two weeks left (June 15th) to
provide input and feedback on the proposals in terms of whether there is
sufficient interest to warrant discussion during a f2f session in
Stockholm or whether the items can be dispatched prior to Stockholm. 

Also, please let me know if I've missed something. 

Regards,
Mary. 

-----Original Message-----
From: Barnes, Mary (RICH2:AR00)
Sent: Friday, May 08, 2009 12:54 PM
To: dispatch@ietf.org
Cc: rai-ads@tools.ietf.org; 'Gonzalo Camarillo'
Subject: IETF-75 plans for DISPATCH 

Hi all,

As mentioned previously, we are setting earlier deadlines for discussing
topics in DISPATCH to allow time for constructive discussions prior to
the WG session at IETF-75: 
http://www.ietf.org/mail-archive/web/dispatch/current/msg00006.html

We are proposing to follow the deadlines for BoFs for IETF-75:
http://www.ietf.org/meetings/75/cutoff-dates.html
since we anticipate that many of the topic discussions will follow a
"mini-BoF" format, based on the DISPATCH charter:
http://www.ietf.org/html.charters/dispatch-charter.html

Thus, the following are the proposed deadlines:

May 25th - Cutoff date to notify the chairs and DISPATCH WG mailing list
of plans to submit a proposal, including the topic of the proposal. This
will help folks with similar topics to work together before the meeting.
If a preliminary charter proposal is available at this point, please
include a pointer to it.

June 8th - Cutoff for charter proposals for topics. A charter proposal
must consist of a minimum of a problem statement and at least one
milestone/deliverable. This deadline will allow time for consideration
of proposals that could potentially be "dispatched" prior to the WG
session.

June 15th - Topics that are to be the focus of IETF-75 are announced.
The idea here is to focus the discussion over the weeks preceding
IETF-75 on these main topics, noting that any updates to the documents
are due 4 weeks later.  This will ensure we have constructive
discussions before the meeting and are actually able to determine
consensus as to where work should be progressed (e.g., separate BoF, a
new WG, an existing WG, individual document, etc.) at IETF-75. Contrary
to the past approach in the SIPPING WG, we anticipate determining
consensus as to the disposition of a work item based on the level of
"commitment" by individuals to actually contribute to the work (e.g.,
conference calls, design team mailing lists, contributing text, etc.)
rather than based on "interest" in doing the work.  Note, that the exact
disposition for a topic may (per the usual process) require follow-up
and confirmation by the ADS and/or IESG (e.g., for a new WG, Bof,
individually sponsored document, etc.) or with another WG to ensure
agreement with the DISPATCH WG consensus for the topic.

Note the intention is not necessarily to preclude folks submitting
documents on other topics, however, it is very unlikely there would be
agenda time allocated to documents/topics submitted after the deadlines.
We can include one or two slides on these topics in the DISPATCH WG
chair charts, as we've done in the past in the SIPPING WG chair charts,
so that the authors can gather other interested parties to contribute to
the work. 

We appreciate the patience of the WG as we evolve the model for bringing
in new work and welcome feedback. 

Regards,
Mary and Gonzalo
DISPATCH WG co-chairs 


_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch

_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch


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Hey Roni,

Roni Even wrote:

> I think one of the challenges will be what to discuss in dispatch. I hope
> that the in the f2f meeting the discussion will be not on the technical
> solution but more on why it came to dispatch, why did the authors of the
> draft think that it does not fit the charter of any existing WG.

There's no mystery here, - we simply didn't know exactly where to go
with it. We are certainly not proposing a new WG. The solutions we were
considering back then involved both XCON and AVT. Dispatch therefore
seemed like the right place to go. Now, after some discussion, and the
pointers from Ingemar it seems like AVT would be the place to do this.

Cheers
Emil

> 
> 
> Roni
> 
> 
> -----Original Message-----
> From: Mary Barnes [mailto:mary.barnes@nortel.com] 
> Sent: Friday, June 05, 2009 12:47 AM
> To: Roni Even; dispatch@ietf.org
> Cc: rai-ads@tools.ietf.org
> Subject: RE: [dispatch] Reminder: IETF-75 plans for DISPATCH
> 
> One last point - for item 6) for example, if you as chair of AVT can
> gather consensus from your WG that this item is of interest, then we can
> dispatch the item now (pending AD approval, of course) - i.e., there is
> no requirement at all that we discuss something at a f2f meeting before
> we dispatch it.  
> 
> Thanks,
> Mary. 
> 
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Barnes, Mary (RICH2:AR00)
> Sent: Thursday, June 04, 2009 4:41 PM
> To: Roni Even; dispatch@ietf.org
> Cc: rai-ads@tools.ietf.org
> Subject: Re: [dispatch] Reminder: IETF-75 plans for DISPATCH
> 
> Hi Roni,
> 
> I don't disagree that any technical work for the last 3 might fit into
> the WGs you suggest. However, the point of the discussion is to agree
> how to proceed with the work - i.e., should it go to WG x or does it
> warrant a separate WG/BoF or is it something for which there is not
> sufficient interest that the work should either be progressed as
> individual or abandoned.  I agree that those last 3 categories are
> beyond the scope (for the most part) of work that we discussed in the
> SIPPING WG in the past. But, the role of DISPATCH is far broader than
> SIPPING: http://www.ietf.org/html.charters/dispatch-charter.html
> " The Dispatch working group is chartered to consider proposals for new
> work in the RAI area and identify, or help create, an appropriate venue
> for the work." 
> 
> There is no mention that the proposals must be SIP specific nor is there
> any restrictions as to what sort of work should be discussed in DISPATCH
> WG - if it relates to RAI in anyway, then it's appropriate.  Also, keep
> in mind that there may be new mailing lists setup for detailed
> discussions of some of these topics. So, I don't think it's reasonable
> to suggest that anything and everything that folks that follow AVT and
> MMUSIC would be interested in will always be discussed in those WGs.  As
> a WG chair, I think you can do a good job of letting your WG know when
> relevant topics are being discussed for folks that might not want to be
> on the DISPATCH WG mailing list. I have approved quite a few posts (and
> added the folks as being able to post without subscribing to the ML)
> over the past week to facilitate input from folks that are not
> subscribed to the DISPATCH ML. 
> 
> My understanding was that one of the reasons for the new structure was
> to minimize new work having to shop for a WG - i.e., starting in one WG
> and being told that the work belongs in another WG and then another,
> etc.  There have been cases in the past where potential new work items
> were shuttled between 3+ different WGs (with no one ever taking
> ownership) and others presented in multiple WGs during a given IETF
> meeting.  In general, this new approach should avoid both those
> problems. 
> 
> Now, this isn't to say that all potential new work items MUST first be
> discussed in DISPATCH - there will be quite clear cases where the
> appropriate WG is obvious from the outset. 
> 
> And, again, we do appreciate patience as we work through this new model.
> 
> 
> Regards,
> Mary.
> (as DISPATCH WG co-chair)
> 
> -----Original Message-----
> From: Roni Even [mailto:Even.roni@huawei.com]
> Sent: Thursday, June 04, 2009 4:08 PM
> To: Barnes, Mary (RICH2:AR00); dispatch@ietf.org
> Cc: rai-ads@tools.ietf.org
> Subject: RE: [dispatch] Reminder: IETF-75 plans for DISPATCH
> 
> Hi Mary,
> Looking at the proposal I would like to point out that the last three
> items were typically discussed before we had the dispatch group in AVT,
> XCON  or MMUSIC. 
> I think that the people who follow AVT and MMUSIC are not the typical
> dispatch subscriber and if a meaningful discussion is solicited than
> those message should have gone also to the AVT and MMUSIC mailing list.
> I did forward the codec email to the AVT working group and urged people
> to join the discussion on the codec topic but maybe the dispatch chairs
> should look at the proposals and send them to current working groups
> that may provide better feedback
> 
> Roni Even
> 
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Mary Barnes
> Sent: Thursday, June 04, 2009 11:19 PM
> To: dispatch@ietf.org
> Cc: rai-ads@tools.ietf.org
> Subject: [dispatch] Reminder: IETF-75 plans for DISPATCH
> 
> Hi folks,
> 
> A reminder to folks that submitted proposals for the initial deadline
> that the "charter proposals" are due next Monday, June 8th.  The
> following summarizes the proposals (some already having detailed charter
> proposals) that have been put forth:
> 1) 3rd party auth:
> http://www.ietf.org/mail-archive/web/dispatch/current/msg00052.html
> 2) CBUS:
> http://www.ietf.org/mail-archive/web/dispatch/current/msg00083.html
> 3) cc-uui:
> http://www.ietf.org/mail-archive/web/dispatch/current/msg00073.html
> 4) CLF:
> http://www.ietf.org/mail-archive/web/dispatch/current/msg00066.html
> 5) Codec:
> http://www.ietf.org/mail-archive/web/dispatch/current/msg00080.html
> 6) Sound level indicator:
> http://www.ietf.org/mail-archive/web/dispatch/current/msg00071.html
> 7) On demand media/app sharing:
> http://www.ietf.org/mail-archive/web/dispatch/current/msg00076.html
> 
> Also, if there are already drafts associated with your charter proposal,
> please provide appropriate links in your email. 
> 
> For WG participants, there are less than two weeks left (June 15th) to
> provide input and feedback on the proposals in terms of whether there is
> sufficient interest to warrant discussion during a f2f session in
> Stockholm or whether the items can be dispatched prior to Stockholm. 
> 
> Also, please let me know if I've missed something. 
> 
> Regards,
> Mary. 
> 
> -----Original Message-----
> From: Barnes, Mary (RICH2:AR00)
> Sent: Friday, May 08, 2009 12:54 PM
> To: dispatch@ietf.org
> Cc: rai-ads@tools.ietf.org; 'Gonzalo Camarillo'
> Subject: IETF-75 plans for DISPATCH 
> 
> Hi all,
> 
> As mentioned previously, we are setting earlier deadlines for discussing
> topics in DISPATCH to allow time for constructive discussions prior to
> the WG session at IETF-75: 
> http://www.ietf.org/mail-archive/web/dispatch/current/msg00006.html
> 
> We are proposing to follow the deadlines for BoFs for IETF-75:
> http://www.ietf.org/meetings/75/cutoff-dates.html
> since we anticipate that many of the topic discussions will follow a
> "mini-BoF" format, based on the DISPATCH charter:
> http://www.ietf.org/html.charters/dispatch-charter.html
> 
> Thus, the following are the proposed deadlines:
> 
> May 25th - Cutoff date to notify the chairs and DISPATCH WG mailing list
> of plans to submit a proposal, including the topic of the proposal. This
> will help folks with similar topics to work together before the meeting.
> If a preliminary charter proposal is available at this point, please
> include a pointer to it.
> 
> June 8th - Cutoff for charter proposals for topics. A charter proposal
> must consist of a minimum of a problem statement and at least one
> milestone/deliverable. This deadline will allow time for consideration
> of proposals that could potentially be "dispatched" prior to the WG
> session.
> 
> June 15th - Topics that are to be the focus of IETF-75 are announced.
> The idea here is to focus the discussion over the weeks preceding
> IETF-75 on these main topics, noting that any updates to the documents
> are due 4 weeks later.  This will ensure we have constructive
> discussions before the meeting and are actually able to determine
> consensus as to where work should be progressed (e.g., separate BoF, a
> new WG, an existing WG, individual document, etc.) at IETF-75. Contrary
> to the past approach in the SIPPING WG, we anticipate determining
> consensus as to the disposition of a work item based on the level of
> "commitment" by individuals to actually contribute to the work (e.g.,
> conference calls, design team mailing lists, contributing text, etc.)
> rather than based on "interest" in doing the work.  Note, that the exact
> disposition for a topic may (per the usual process) require follow-up
> and confirmation by the ADS and/or IESG (e.g., for a new WG, Bof,
> individually sponsored document, etc.) or with another WG to ensure
> agreement with the DISPATCH WG consensus for the topic.
> 
> Note the intention is not necessarily to preclude folks submitting
> documents on other topics, however, it is very unlikely there would be
> agenda time allocated to documents/topics submitted after the deadlines.
> We can include one or two slides on these topics in the DISPATCH WG
> chair charts, as we've done in the past in the SIPPING WG chair charts,
> so that the authors can gather other interested parties to contribute to
> the work. 
> 
> We appreciate the patience of the WG as we evolve the model for bringing
> in new work and welcome feedback. 
> 
> Regards,
> Mary and Gonzalo
> DISPATCH WG co-chairs 
> 
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

-- 
Emil Ivov, Ph.D.                               30a rue de la Patrie
Project Lead                                   67300 Schiltigheim, France
SIP Communicator
emcho@sip-communicator.org                     FAX:   +33.1.77.62.47.31
http://sip-communicator.org                    PHONE: +33.1.77.62.43.30

From emil@sip-communicator.org  Fri Jun  5 02:10:43 2009
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Subject: Re: [dispatch] Yet another problem to dispatch
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Hey Roni,

Roni Even wrote:
> Emil,
> I think that John means that what you call loudest speaker is not
> necessarily a speaker but may represent some noise which is high enough to
> qualify as one of the N loudest speakers.

Indeed, and that's one of the reasons why sound level indicators are so
useful. It is far easier for a human to determine what is background
noise and what is speech or other activity that is part of the conversation.

> The conferencing bridge try to address this problem by defining some
> thresholds for accepting a user to the mix. 

And they can continue doing so. Are you saying that there would be some
kind of conflict between these thresholds and the fact that mixers would
include sound level information for what they are sending?

> I also looked at the draft and would like to see a discussion about the
> current options to receive the active speakers indications. I am not so sure
> about the benefit of displaying sound level, 

The problem with active speaker designation comes from its binary
nature. Simply picking the one loudest media source (even after
normalization) and dubbing it "active speaker" does not always
correspond to what's actually happening in a call. Sound level
indicators would help identify sources of background noise that got
through the filtering on the mixer.

> note that the mixers usually do
> some normalization on the volume of the mixed streams to enhance the loudest
> speaker and to have a common level regardless of the source level in order
> to prevent fluctuation of the voice that may be caused by different
> microphones, encoders and background noise.  

I am not sure why this is a problem. After doing all of the above a
mixer would simply include sound level information for the data it is
about to send (i.e. the normalized streams).

Cheers
Emil


> 
> Roni Even
> 
> 
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
> Of Emil Ivov
> Sent: Thursday, May 28, 2009 7:29 PM
> To: dispatch@ietf.org
> Subject: Re: [dispatch] Yet another problem to dispatch
> 
> Hey John,
> 
> Elwell, John wrote:
>> Emil,
>>
>> But then you would need some sort of throttling, to prevent the UI
>> constantly changing the list of names whose volumes are displayed. If
>> the mixer were to do this throttling it would reduce the frequency of
>> messages, which might allow a bigger choice of potential solutions.
> 
> I am not sure I understand exactly what you mean by throttling. Mixers
> could be implemented to only include sound level information for the
> loudest speakers thus having the possibility to ignore silent and almost
> silent ones. It should be up to the mixer to determine the exact
> threshold since that's where bandwidth would be most precious. Is this
> what you are suggesting?
> 
> Then, once a client gets this information it can consider (and I guess
> that's in line with 3550) that all CSRCs not present in a particular
> packet have been silent for the corresponding period of time.
> 
> As to how exactly the client would render the sound level information,
> whether or not it would display all of it, and whether it would somehow
> rearrange participants, that should really be up to the UI implementors.
> 
> Does this make sense?
> 
> Cheers
> Emil
> 
> 
>> These are just initial thoughts on my part, not firm opinions.
>>
>> John
>>
>>
>>> -----Original Message-----
>>> From: Emil Ivov [mailto:emil@sip-communicator.org] On Behalf 
>>> Of Emil Ivov
>>> Sent: 28 May 2009 14:02
>>> To: Elwell, John
>>> Cc: Enrico Marocco; Cullen Jennings; dispatch@ietf.org
>>> Subject: Re: [dispatch] Yet another problem to dispatch
>>>
>>> Hey John,
>>>
>>> Elwell, John wrote:
>>>> Background noise in conferences is certainly a big issue. 
>>> The web-type
>>>> interface that Cullen mentions does in fact allow the moderator to
>>>> momentarily mute each suspected noisy party in turn to 
>>> discover who is
>>>> the culprit. Clearly this is not scalable to large 
>>> conferences. But also
>>>> a simultaneous display of noise levels (e.g., in the form 
>>> of bar graphs)
>>>> is not scalable. 
>>> One way to address this would be to allow mixer implementors to only
>>> include sound levels for the N loudest participants. UI implementors
>>> could then use a similar algorithm when presenting the information.
>>>
>>> How does this sound?
>>>
>>> Cheers
>>> Emil
>>>
>>>
>>>> Whilst it might be easy to find protocol solutions, we
>>>> need to be confident that people will find ways of 
>>> exploiting them in a
>>>> reasonable way at the user interface. Certainly worth exploring.
>>>>
>>>> John
>>>>
>>>>
>>>>> -----Original Message-----
>>>>> From: dispatch-bounces@ietf.org 
>>>>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Enrico Marocco
>>>>> Sent: 27 May 2009 21:02
>>>>> To: Cullen Jennings
>>>>> Cc: dispatch@ietf.org
>>>>> Subject: Re: [dispatch] Yet another problem to dispatch
>>>>>
>>>>> Cullen Jennings wrote:
>>>>>> I find this "active speaker" problem very interesting. Often the  
>>>>>> device that wants to display the active speaker information 
>>>>> is not the  
>>>>>> same as the device dealing with the RTP. For example, most the  
>>>>>> conference systems I am using a phone but there is a web 
>>> interface  
>>>>>> that is dealing with active speaker information on my 
>>>>> browser which is  
>>>>>> no on my phone. Be interesting to look at all the existing 
>>>>> approaches  
>>>>>> to this.
>>>>> Yes, I'm familiar with that kind of systems. However, the 
>>>>> information we
>>>>> are considering -- instantaneous sound levels of each 
>>>>> participant -- is
>>>>> definitely changing more rapidly than just active speaker 
>>>>> indication and
>>>>> also has more stringent synchronization requirements. 
>>> That's why our
>>>>> favorite approach would be to carry it in the media streams 
>>>>> themselves,
>>>>>  with some sort of a modulation of the RTP CSRC fields 
>>> that could be
>>>>> considered either an hack or an elegant solution. I would be very
>>>>> interested in hearing any opinions about that.
>>>>>
>>>>> -- 
>>>>> Ciao,
>>>>> Enrico
>>>>>
>>>> _______________________________________________
>>>> dispatch mailing list
>>>> dispatch@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>>
>>> -- 
>>> Emil Ivov, Ph.D.                               30a rue de la Patrie
>>> Project Lead                                   67300 Schiltigheim
>>> SIP Communicator
>>> emcho@sip-communicator.org                     FAX:   
>>> +33.1.77.62.47.31
>>> http://sip-communicator.org                    PHONE: 
>>> +33.1.77.62.43.30
>>>
> 

-- 
Emil Ivov, Ph.D.                               30a rue de la Patrie
Project Lead                                   67300 Schiltigheim, France
SIP Communicator
emcho@sip-communicator.org                     FAX:   +33.1.77.62.47.31
http://sip-communicator.org                    PHONE: +33.1.77.62.43.30

From ron.even.tlv@gmail.com  Fri Jun  5 02:26:42 2009
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Emil Ivov'" <emcho@sip-communicator.org>
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Subject: Re: [dispatch] Yet another problem to dispatch
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Hi Emil,
Inline

-----Original Message-----
From: Emil Ivov [mailto:emil@sip-communicator.org] On Behalf Of Emil Ivov
Sent: Friday, June 05, 2009 12:10 PM
To: Roni Even
Cc: dispatch@ietf.org
Subject: Re: [dispatch] Yet another problem to dispatch

Hey Roni,

Roni Even wrote:
> Emil,
> I think that John means that what you call loudest speaker is not
> necessarily a speaker but may represent some noise which is high enough to
> qualify as one of the N loudest speakers.

Indeed, and that's one of the reasons why sound level indicators are so
useful. It is far easier for a human to determine what is background
noise and what is speech or other activity that is part of the conversation.

RE: the information may not be relevant since it just describes the mix, the
user gets a mixed content and not the individual streams.

> The conferencing bridge try to address this problem by defining some
> thresholds for accepting a user to the mix. 

And they can continue doing so. Are you saying that there would be some
kind of conflict between these thresholds and the fact that mixers would
include sound level information for what they are sending?

RE: I am just saying that in my view it is an interesting information which
is useless.

> I also looked at the draft and would like to see a discussion about the
> current options to receive the active speakers indications. I am not so
sure
> about the benefit of displaying sound level, 

The problem with active speaker designation comes from its binary
nature. Simply picking the one loudest media source (even after
normalization) and dubbing it "active speaker" does not always
correspond to what's actually happening in a call. Sound level
indicators would help identify sources of background noise that got
through the filtering on the mixer.

RE: again the stream is already mixed.

> note that the mixers usually do
> some normalization on the volume of the mixed streams to enhance the
loudest
> speaker and to have a common level regardless of the source level in order
> to prevent fluctuation of the voice that may be caused by different
> microphones, encoders and background noise.  

I am not sure why this is a problem. After doing all of the above a
mixer would simply include sound level information for the data it is
about to send (i.e. the normalized streams).

RE: and what does it help???

Cheers
Emil


> 
> Roni Even
> 
> 
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf
> Of Emil Ivov
> Sent: Thursday, May 28, 2009 7:29 PM
> To: dispatch@ietf.org
> Subject: Re: [dispatch] Yet another problem to dispatch
> 
> Hey John,
> 
> Elwell, John wrote:
>> Emil,
>>
>> But then you would need some sort of throttling, to prevent the UI
>> constantly changing the list of names whose volumes are displayed. If
>> the mixer were to do this throttling it would reduce the frequency of
>> messages, which might allow a bigger choice of potential solutions.
> 
> I am not sure I understand exactly what you mean by throttling. Mixers
> could be implemented to only include sound level information for the
> loudest speakers thus having the possibility to ignore silent and almost
> silent ones. It should be up to the mixer to determine the exact
> threshold since that's where bandwidth would be most precious. Is this
> what you are suggesting?
> 
> Then, once a client gets this information it can consider (and I guess
> that's in line with 3550) that all CSRCs not present in a particular
> packet have been silent for the corresponding period of time.
> 
> As to how exactly the client would render the sound level information,
> whether or not it would display all of it, and whether it would somehow
> rearrange participants, that should really be up to the UI implementors.
> 
> Does this make sense?
> 
> Cheers
> Emil
> 
> 
>> These are just initial thoughts on my part, not firm opinions.
>>
>> John
>>
>>
>>> -----Original Message-----
>>> From: Emil Ivov [mailto:emil@sip-communicator.org] On Behalf 
>>> Of Emil Ivov
>>> Sent: 28 May 2009 14:02
>>> To: Elwell, John
>>> Cc: Enrico Marocco; Cullen Jennings; dispatch@ietf.org
>>> Subject: Re: [dispatch] Yet another problem to dispatch
>>>
>>> Hey John,
>>>
>>> Elwell, John wrote:
>>>> Background noise in conferences is certainly a big issue. 
>>> The web-type
>>>> interface that Cullen mentions does in fact allow the moderator to
>>>> momentarily mute each suspected noisy party in turn to 
>>> discover who is
>>>> the culprit. Clearly this is not scalable to large 
>>> conferences. But also
>>>> a simultaneous display of noise levels (e.g., in the form 
>>> of bar graphs)
>>>> is not scalable. 
>>> One way to address this would be to allow mixer implementors to only
>>> include sound levels for the N loudest participants. UI implementors
>>> could then use a similar algorithm when presenting the information.
>>>
>>> How does this sound?
>>>
>>> Cheers
>>> Emil
>>>
>>>
>>>> Whilst it might be easy to find protocol solutions, we
>>>> need to be confident that people will find ways of 
>>> exploiting them in a
>>>> reasonable way at the user interface. Certainly worth exploring.
>>>>
>>>> John
>>>>
>>>>
>>>>> -----Original Message-----
>>>>> From: dispatch-bounces@ietf.org 
>>>>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Enrico Marocco
>>>>> Sent: 27 May 2009 21:02
>>>>> To: Cullen Jennings
>>>>> Cc: dispatch@ietf.org
>>>>> Subject: Re: [dispatch] Yet another problem to dispatch
>>>>>
>>>>> Cullen Jennings wrote:
>>>>>> I find this "active speaker" problem very interesting. Often the  
>>>>>> device that wants to display the active speaker information 
>>>>> is not the  
>>>>>> same as the device dealing with the RTP. For example, most the  
>>>>>> conference systems I am using a phone but there is a web 
>>> interface  
>>>>>> that is dealing with active speaker information on my 
>>>>> browser which is  
>>>>>> no on my phone. Be interesting to look at all the existing 
>>>>> approaches  
>>>>>> to this.
>>>>> Yes, I'm familiar with that kind of systems. However, the 
>>>>> information we
>>>>> are considering -- instantaneous sound levels of each 
>>>>> participant -- is
>>>>> definitely changing more rapidly than just active speaker 
>>>>> indication and
>>>>> also has more stringent synchronization requirements. 
>>> That's why our
>>>>> favorite approach would be to carry it in the media streams 
>>>>> themselves,
>>>>>  with some sort of a modulation of the RTP CSRC fields 
>>> that could be
>>>>> considered either an hack or an elegant solution. I would be very
>>>>> interested in hearing any opinions about that.
>>>>>
>>>>> -- 
>>>>> Ciao,
>>>>> Enrico
>>>>>
>>>> _______________________________________________
>>>> dispatch mailing list
>>>> dispatch@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>>
>>> -- 
>>> Emil Ivov, Ph.D.                               30a rue de la Patrie
>>> Project Lead                                   67300 Schiltigheim
>>> SIP Communicator
>>> emcho@sip-communicator.org                     FAX:   
>>> +33.1.77.62.47.31
>>> http://sip-communicator.org                    PHONE: 
>>> +33.1.77.62.43.30
>>>
> 

-- 
Emil Ivov, Ph.D.                               30a rue de la Patrie
Project Lead                                   67300 Schiltigheim, France
SIP Communicator
emcho@sip-communicator.org                     FAX:   +33.1.77.62.47.31
http://sip-communicator.org                    PHONE: +33.1.77.62.43.30


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Subject: Re: [dispatch] Yet another problem to dispatch
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Roni Even wrote:
> I also looked at the draft and would like to see a discussion about the
> current options to receive the active speakers indications. I am not so sure
> about the benefit of displaying sound level

There are basically three reasons, apparently we should have done a
better job at listing them in the draft:

1. It's cool, Skype does it. This may sound like a stupid reason, but
you'll see it's not the case if you try to convince people in a service
provider marketing dept. to plan an investment for a service that will
have less features and a worse UI than what users can get for free on
the Internet.

2. It is very useful in conferences with a fairly high number of
participants to debug sources of background noise, like people typing or
breathing loudly in the mic.

3. It is useful in multiparty conversations where participant know each
other only by name and cannot distinguish individual voices.

-- 
Ciao,
Enrico

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From: Enrico Marocco <enrico.marocco@telecomitalia.it>
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To: Roni Even <ron.even.tlv@gmail.com>
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Subject: Re: [dispatch] Yet another problem to dispatch
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Roni Even wrote:
> Indeed, and that's one of the reasons why sound level indicators are so
> useful. It is far easier for a human to determine what is background
> noise and what is speech or other activity that is part of the conversation.
> 
> RE: the information may not be relevant since it just describes the mix, the
> user gets a mixed content and not the individual streams.

Uh? The information will represent the sound level of each individual
participant. If the streams are normalized by the mixer, the sound level
values will be normalized as well: the mixer can do that easily and the
resulting information will be consistent with the mixed stream.

>> The conferencing bridge try to address this problem by defining some
>> thresholds for accepting a user to the mix. 
> 
> And they can continue doing so. Are you saying that there would be some
> kind of conflict between these thresholds and the fact that mixers would
> include sound level information for what they are sending?
> 
> RE: I am just saying that in my view it is an interesting information which
> is useless.

Useless because you don't think displaying the sound levels of the
voices you hear in a multiparty conference has any value for the user,
or useless because you think a mixer can't do that reliably?

> The problem with active speaker designation comes from its binary
> nature. Simply picking the one loudest media source (even after
> normalization) and dubbing it "active speaker" does not always
> correspond to what's actually happening in a call. Sound level
> indicators would help identify sources of background noise that got
> through the filtering on the mixer.
> 
> RE: again the stream is already mixed.

Probably you use mixers way better than mine, but in my conferences I
keep hearing people typing and breathing all the time ;-)

-- 
Ciao,
Enrico

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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Enrico Marocco'" <enrico.marocco@telecomitalia.it>
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Subject: Re: [dispatch] Yet another problem to dispatch
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Enrico,

Skype also supports starting a multipoint audio conference from the UI. I
would consider this more practical than showing the sound level. XCON
defined this functionality in a standard draft enabling the creation of
audio and video conferences in an multivendor environment. Does your client
support this functionality. Do you see this as a mandatory requirement from
service providers.

My concern is that we develop solutions that are not deployed since they
have no high priority for developers and customers. Doing a "cool" solution
in a proprietary environment is simpler since you control the clients as
well as the servers and do not worry about interoperability with other
vendors. 
Roni Even

-----Original Message-----
From: Enrico Marocco [mailto:enrico.marocco@telecomitalia.it] 
Sent: Friday, June 05, 2009 12:31 PM
To: Roni Even
Cc: 'Emil Ivov'; dispatch@ietf.org
Subject: Re: [dispatch] Yet another problem to dispatch

Roni Even wrote:
> I also looked at the draft and would like to see a discussion about the
> current options to receive the active speakers indications. I am not so
sure
> about the benefit of displaying sound level

There are basically three reasons, apparently we should have done a
better job at listing them in the draft:

1. It's cool, Skype does it. This may sound like a stupid reason, but
you'll see it's not the case if you try to convince people in a service
provider marketing dept. to plan an investment for a service that will
have less features and a worse UI than what users can get for free on
the Internet.

2. It is very useful in conferences with a fairly high number of
participants to debug sources of background noise, like people typing or
breathing loudly in the mic.

3. It is useful in multiparty conversations where participant know each
other only by name and cannot distinguish individual voices.

-- 
Ciao,
Enrico


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Subject: Re: [dispatch] Yet another problem to dispatch
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Roni,

Roni Even wrote:
> Enrico,
> 
> Skype also supports starting a multipoint audio conference from the UI. I
> would consider this more practical than showing the sound level. XCON
> defined this functionality in a standard draft enabling the creation of
> audio and video conferences in an multivendor environment. Does your client
> support this functionality. 

I believe I already mentioned this in the beginning of this thread but
probably not clearly enough: this whole work has started because of the
ongoing effort in the SIP Communicator community to implement support
for conferences established and (in some cases) hosted by the SIP
Communicator client.

> Do you see this as a mandatory requirement from service providers.

I guess Enrico would better answer this one but I can certainly imagine
how this would become a trendy feature.

> My concern is that we develop solutions that are not deployed since they
> have no high priority for developers and customers. Doing a "cool" solution
> in a proprietary environment is simpler since you control the clients as
> well as the servers and do not worry about interoperability with other
> vendors. 

Well you can relax then. This is of very high priority to us ( a non
proprietary open source client with a significant community ) and it is
certainly not a mere whim for a coolish featury. We are also very
concerned about interoperability which is why we have started this whole
discussion.

Emil

> Roni Even
> 
> -----Original Message-----
> From: Enrico Marocco [mailto:enrico.marocco@telecomitalia.it] 
> Sent: Friday, June 05, 2009 12:31 PM
> To: Roni Even
> Cc: 'Emil Ivov'; dispatch@ietf.org
> Subject: Re: [dispatch] Yet another problem to dispatch
> 
> Roni Even wrote:
>> I also looked at the draft and would like to see a discussion about the
>> current options to receive the active speakers indications. I am not so
> sure
>> about the benefit of displaying sound level
> 
> There are basically three reasons, apparently we should have done a
> better job at listing them in the draft:
> 
> 1. It's cool, Skype does it. This may sound like a stupid reason, but
> you'll see it's not the case if you try to convince people in a service
> provider marketing dept. to plan an investment for a service that will
> have less features and a worse UI than what users can get for free on
> the Internet.
> 
> 2. It is very useful in conferences with a fairly high number of
> participants to debug sources of background noise, like people typing or
> breathing loudly in the mic.
> 
> 3. It is useful in multiparty conversations where participant know each
> other only by name and cannot distinguish individual voices.
> 

-- 
Emil Ivov, Ph.D.                               30a rue de la Patrie
Project Lead                                   67300 Schiltigheim, France
SIP Communicator
emcho@sip-communicator.org                     FAX:   +33.1.77.62.47.31
http://sip-communicator.org                    PHONE: +33.1.77.62.43.30

From vkg@alcatel-lucent.com  Fri Jun  5 07:13:05 2009
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Date: Fri, 05 Jun 2009 09:13:02 -0500
From: "Vijay K. Gurbani" <vkg@alcatel-lucent.com>
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Subject: Re: [dispatch] Reminder: IETF-75 plans for DISPATCH
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Mary Barnes wrote:
> Hi folks,
> 
> A reminder to folks that submitted proposals for the initial deadline
> that the "charter proposals" are due next Monday, June 8th.  
[...]

Folks: Here is the final "charter proposal" for the SIP CLF
work.  I received a couple of comments privately on typos
and such, but most of the matter is the same as the
one distributed on May 15th.

Unless anyone has any objections, I will submit this to the
dispatch list as per Mary's instructions above on or
before Monday, June 8.

Charter proposal for SIP Common Log File (CLF) format work
==========================================================
Vijay K. Gurbani and Eric Burger

Problem Statement
=================
Well-known web servers such as Apache and web proxies like Squid
support event logging using a common log format.  The logs produced
using these de-facto standard formats are invaluable to system
administrators for trouble-shooting a server, and to tool writers
for crafting tools that mine the log files to produce reports
and trends.  These log files also enable searches for a certain
SIP message or messages, a transaction or a related set of
transactions.

Furthermore, these log files can also be used to train anomaly
detection systems and feed events into a security event management
system.  The Session Initiation Protocol does not have a common log
format, and as a result, each server supports a distinct log format
that makes it unnecessarily complex to produce tools to do trend
analysis and security detection.

Ad ad-hoc meeting was sponsored by the SIPPING WG during the San
Francisco IETF where the participants expressed interest in undertaking
this work.  Minutes from the ad-hoc are available at:
http://www.ietf.org/mail-archive/web/sipping/current/msg17199.html.
Since then, various discussions on CLF file format and other
assorted discussions have occurred on the SIPPING mailing list,
the sip-ops mailing list and the newly formed sip-clf mailing list.

Milestones and deliverables
===========================

1) A document enunciating the problem statement, motivation,
the possible use cases of a SIP CLF, and the list of mandatory
fields that will allow identifying transactions, grouping
transactions into dialogs, and doing the latter with provisions
for allowing the systems administrator or an automata to
correlate forked branches.  Provisions must be made to
accommodate ad-hoc fields without adversely impacting the
parsing of the mandatory parameters.

A possible starting document for this deliverable is
http://tools.ietf.org/html/draft-gurbani-sipping-clf-01

2) A document that details the byte layout of the SIP CLF
record.

The participants have done preliminary work on writing encoders
and decoders for space-separated ASCII and binary format.  The
runtime complexity to produce the space-separated ASCII and
binary CLF is comparable, however, the binary CLF is appreciably
faster in locating random records from the binary CLF file.
On the other hand, a ASCII CLF format was preferable because
it allowed for a visual interpretation of the mandatory
fields to the benefit of a human user and allowed for
expedited operations on the data using text-based tools.

Based on subsequent deliberations, a text format has been
defined which lends itself well to fast searches while still
allowing the use of visual identification and interpretation
using text-based tools.  This format is documented in:
http://tools.ietf.org/html/draft-roach-sipping-clf-syntax-01
and can serve as a possible starting document for the
details of byte layout.

3) A document that provides reference implementation(s)
for decoding the byte layout of the CLF.

NOTE: It could very well be that three individual documents
are produced to meet the deliverables or a single document is
produced that merges all three aspects.  This can be decided
by the BoF/design team/mini working group.

Thanks,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/

From mary.barnes@nortel.com  Fri Jun  5 07:26:31 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Roni Even" <Even.roni@huawei.com>, <dispatch@ietf.org>
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Subject: Re: [dispatch] Reminder: IETF-75 plans for DISPATCH
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Yes, the f2f meeting should focus on the more general problem and not
the specifics of a technical solution - that's the reason we're asking
for input in the form of charters and proposed deliverables.=20

Your first comment is excellent feedback for and something that should
be considered by the folks working on that problem.=20

Thanks,
Mary.

-----Original Message-----
From: Roni Even [mailto:Even.roni@huawei.com]=20
Sent: Thursday, June 04, 2009 11:30 PM
To: Barnes, Mary (RICH2:AR00); dispatch@ietf.org
Cc: rai-ads@tools.ietf.org
Subject: RE: [dispatch] Reminder: IETF-75 plans for DISPATCH

Mary,
On item 6 I am not sure if it is AVT or XCON. One of the problems in
this request is the synchronization between the application level user
of the conference (the display name) and his media streams. Today this
is handled in the conference event package.=20
Having the audio level in the mixed RTP/RTCP stream will not provide
enough information to the client to synchronize the SSRC with the name
the user is suing in the conference. The CNAME does not supply this
relation.=20

I think one of the challenges will be what to discuss in dispatch. I
hope that the in the f2f meeting the discussion will be not on the
technical solution but more on why it came to dispatch, why did the
authors of the draft think that it does not fit the charter of any
existing WG.


Roni


-----Original Message-----
From: Mary Barnes [mailto:mary.barnes@nortel.com]
Sent: Friday, June 05, 2009 12:47 AM
To: Roni Even; dispatch@ietf.org
Cc: rai-ads@tools.ietf.org
Subject: RE: [dispatch] Reminder: IETF-75 plans for DISPATCH

One last point - for item 6) for example, if you as chair of AVT can
gather consensus from your WG that this item is of interest, then we can
dispatch the item now (pending AD approval, of course) - i.e., there is
no requirement at all that we discuss something at a f2f meeting before
we dispatch it. =20

Thanks,
Mary.=20

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Barnes, Mary (RICH2:AR00)
Sent: Thursday, June 04, 2009 4:41 PM
To: Roni Even; dispatch@ietf.org
Cc: rai-ads@tools.ietf.org
Subject: Re: [dispatch] Reminder: IETF-75 plans for DISPATCH

Hi Roni,

I don't disagree that any technical work for the last 3 might fit into
the WGs you suggest. However, the point of the discussion is to agree
how to proceed with the work - i.e., should it go to WG x or does it
warrant a separate WG/BoF or is it something for which there is not
sufficient interest that the work should either be progressed as
individual or abandoned.  I agree that those last 3 categories are
beyond the scope (for the most part) of work that we discussed in the
SIPPING WG in the past. But, the role of DISPATCH is far broader than
SIPPING: http://www.ietf.org/html.charters/dispatch-charter.html
" The Dispatch working group is chartered to consider proposals for new
work in the RAI area and identify, or help create, an appropriate venue
for the work."=20

There is no mention that the proposals must be SIP specific nor is there
any restrictions as to what sort of work should be discussed in DISPATCH
WG - if it relates to RAI in anyway, then it's appropriate.  Also, keep
in mind that there may be new mailing lists setup for detailed
discussions of some of these topics. So, I don't think it's reasonable
to suggest that anything and everything that folks that follow AVT and
MMUSIC would be interested in will always be discussed in those WGs.  As
a WG chair, I think you can do a good job of letting your WG know when
relevant topics are being discussed for folks that might not want to be
on the DISPATCH WG mailing list. I have approved quite a few posts (and
added the folks as being able to post without subscribing to the ML)
over the past week to facilitate input from folks that are not
subscribed to the DISPATCH ML.=20

My understanding was that one of the reasons for the new structure was
to minimize new work having to shop for a WG - i.e., starting in one WG
and being told that the work belongs in another WG and then another,
etc.  There have been cases in the past where potential new work items
were shuttled between 3+ different WGs (with no one ever taking
ownership) and others presented in multiple WGs during a given IETF
meeting.  In general, this new approach should avoid both those
problems.=20

Now, this isn't to say that all potential new work items MUST first be
discussed in DISPATCH - there will be quite clear cases where the
appropriate WG is obvious from the outset.=20

And, again, we do appreciate patience as we work through this new model.


Regards,
Mary.
(as DISPATCH WG co-chair)

-----Original Message-----
From: Roni Even [mailto:Even.roni@huawei.com]
Sent: Thursday, June 04, 2009 4:08 PM
To: Barnes, Mary (RICH2:AR00); dispatch@ietf.org
Cc: rai-ads@tools.ietf.org
Subject: RE: [dispatch] Reminder: IETF-75 plans for DISPATCH

Hi Mary,
Looking at the proposal I would like to point out that the last three
items were typically discussed before we had the dispatch group in AVT,
XCON  or MMUSIC.=20
I think that the people who follow AVT and MMUSIC are not the typical
dispatch subscriber and if a meaningful discussion is solicited than
those message should have gone also to the AVT and MMUSIC mailing list.
I did forward the codec email to the AVT working group and urged people
to join the discussion on the codec topic but maybe the dispatch chairs
should look at the proposals and send them to current working groups
that may provide better feedback

Roni Even

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Mary Barnes
Sent: Thursday, June 04, 2009 11:19 PM
To: dispatch@ietf.org
Cc: rai-ads@tools.ietf.org
Subject: [dispatch] Reminder: IETF-75 plans for DISPATCH

Hi folks,

A reminder to folks that submitted proposals for the initial deadline
that the "charter proposals" are due next Monday, June 8th.  The
following summarizes the proposals (some already having detailed charter
proposals) that have been put forth:
1) 3rd party auth:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00052.html
2) CBUS:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00083.html
3) cc-uui:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00073.html
4) CLF:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00066.html
5) Codec:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00080.html
6) Sound level indicator:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00071.html
7) On demand media/app sharing:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00076.html

Also, if there are already drafts associated with your charter proposal,
please provide appropriate links in your email.=20

For WG participants, there are less than two weeks left (June 15th) to
provide input and feedback on the proposals in terms of whether there is
sufficient interest to warrant discussion during a f2f session in
Stockholm or whether the items can be dispatched prior to Stockholm.=20

Also, please let me know if I've missed something.=20

Regards,
Mary.=20

-----Original Message-----
From: Barnes, Mary (RICH2:AR00)
Sent: Friday, May 08, 2009 12:54 PM
To: dispatch@ietf.org
Cc: rai-ads@tools.ietf.org; 'Gonzalo Camarillo'
Subject: IETF-75 plans for DISPATCH=20

Hi all,

As mentioned previously, we are setting earlier deadlines for discussing
topics in DISPATCH to allow time for constructive discussions prior to
the WG session at IETF-75:=20
http://www.ietf.org/mail-archive/web/dispatch/current/msg00006.html

We are proposing to follow the deadlines for BoFs for IETF-75:
http://www.ietf.org/meetings/75/cutoff-dates.html
since we anticipate that many of the topic discussions will follow a
"mini-BoF" format, based on the DISPATCH charter:
http://www.ietf.org/html.charters/dispatch-charter.html

Thus, the following are the proposed deadlines:

May 25th - Cutoff date to notify the chairs and DISPATCH WG mailing list
of plans to submit a proposal, including the topic of the proposal. This
will help folks with similar topics to work together before the meeting.
If a preliminary charter proposal is available at this point, please
include a pointer to it.

June 8th - Cutoff for charter proposals for topics. A charter proposal
must consist of a minimum of a problem statement and at least one
milestone/deliverable. This deadline will allow time for consideration
of proposals that could potentially be "dispatched" prior to the WG
session.

June 15th - Topics that are to be the focus of IETF-75 are announced.
The idea here is to focus the discussion over the weeks preceding
IETF-75 on these main topics, noting that any updates to the documents
are due 4 weeks later.  This will ensure we have constructive
discussions before the meeting and are actually able to determine
consensus as to where work should be progressed (e.g., separate BoF, a
new WG, an existing WG, individual document, etc.) at IETF-75. Contrary
to the past approach in the SIPPING WG, we anticipate determining
consensus as to the disposition of a work item based on the level of
"commitment" by individuals to actually contribute to the work (e.g.,
conference calls, design team mailing lists, contributing text, etc.)
rather than based on "interest" in doing the work.  Note, that the exact
disposition for a topic may (per the usual process) require follow-up
and confirmation by the ADS and/or IESG (e.g., for a new WG, Bof,
individually sponsored document, etc.) or with another WG to ensure
agreement with the DISPATCH WG consensus for the topic.

Note the intention is not necessarily to preclude folks submitting
documents on other topics, however, it is very unlikely there would be
agenda time allocated to documents/topics submitted after the deadlines.
We can include one or two slides on these topics in the DISPATCH WG
chair charts, as we've done in the past in the SIPPING WG chair charts,
so that the authors can gather other interested parties to contribute to
the work.=20

We appreciate the patience of the WG as we evolve the model for bringing
in new work and welcome feedback.=20

Regards,
Mary and Gonzalo
DISPATCH WG co-chairs=20


_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch

_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch


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Thread-Topic: [dispatch] Reminder: IETF-75 plans for DISPATCH
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Roni Even" <Even.roni@huawei.com>, <dispatch@ietf.org>
Cc: rai-ads@tools.ietf.org
Subject: Re: [dispatch] Reminder: IETF-75 plans for DISPATCH
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Roni,

I like the distributed model whereby the chairs of the specific working
group decide if the topic is of enough interest to alert their WG.  This
also removes the DISPATCH WG chairs from implying, for example, that a
particular work item should be bucketed in a specific working group
before we've had enough discussion. =20

Thanks for the feedback.

Mary.=20

-----Original Message-----
From: Roni Even [mailto:Even.roni@huawei.com]=20
Sent: Thursday, June 04, 2009 11:21 PM
To: Barnes, Mary (RICH2:AR00); dispatch@ietf.org
Cc: rai-ads@tools.ietf.org
Subject: RE: [dispatch] Reminder: IETF-75 plans for DISPATCH

Mary,
I was not arguing the dispatch charter. I wanted to point out that since
dispatch charter is to start discussion on any RAI new topic it would
make sense from the dispatch chairs to forward these new topics to other
relevant RAI WGs mailing lists since these lists members may not follow
the dispatch list.

Roni Even

-----Original Message-----
From: Mary Barnes [mailto:mary.barnes@nortel.com]
Sent: Friday, June 05, 2009 12:41 AM
To: Roni Even; dispatch@ietf.org
Cc: rai-ads@tools.ietf.org
Subject: RE: [dispatch] Reminder: IETF-75 plans for DISPATCH

Hi Roni,

I don't disagree that any technical work for the last 3 might fit into
the WGs you suggest. However, the point of the discussion is to agree
how to proceed with the work - i.e., should it go to WG x or does it
warrant a separate WG/BoF or is it something for which there is not
sufficient interest that the work should either be progressed as
individual or abandoned.  I agree that those last 3 categories are
beyond the scope (for the most part) of work that we discussed in the
SIPPING WG in the past. But, the role of DISPATCH is far broader than
SIPPING: http://www.ietf.org/html.charters/dispatch-charter.html
" The Dispatch working group is chartered to consider proposals for new
work in the RAI area and identify, or help create, an appropriate venue
for the work."=20

There is no mention that the proposals must be SIP specific nor is there
any restrictions as to what sort of work should be discussed in DISPATCH
WG - if it relates to RAI in anyway, then it's appropriate.  Also, keep
in mind that there may be new mailing lists setup for detailed
discussions of some of these topics. So, I don't think it's reasonable
to suggest that anything and everything that folks that follow AVT and
MMUSIC would be interested in will always be discussed in those WGs.  As
a WG chair, I think you can do a good job of letting your WG know when
relevant topics are being discussed for folks that might not want to be
on the DISPATCH WG mailing list. I have approved quite a few posts (and
added the folks as being able to post without subscribing to the ML)
over the past week to facilitate input from folks that are not
subscribed to the DISPATCH ML.=20

My understanding was that one of the reasons for the new structure was
to minimize new work having to shop for a WG - i.e., starting in one WG
and being told that the work belongs in another WG and then another,
etc.  There have been cases in the past where potential new work items
were shuttled between 3+ different WGs (with no one ever taking
ownership) and others presented in multiple WGs during a given IETF
meeting.  In general, this new approach should avoid both those
problems.=20

Now, this isn't to say that all potential new work items MUST first be
discussed in DISPATCH - there will be quite clear cases where the
appropriate WG is obvious from the outset.=20

And, again, we do appreciate patience as we work through this new model.


Regards,
Mary.
(as DISPATCH WG co-chair)

-----Original Message-----
From: Roni Even [mailto:Even.roni@huawei.com]
Sent: Thursday, June 04, 2009 4:08 PM
To: Barnes, Mary (RICH2:AR00); dispatch@ietf.org
Cc: rai-ads@tools.ietf.org
Subject: RE: [dispatch] Reminder: IETF-75 plans for DISPATCH

Hi Mary,
Looking at the proposal I would like to point out that the last three
items were typically discussed before we had the dispatch group in AVT,
XCON  or MMUSIC.=20
I think that the people who follow AVT and MMUSIC are not the typical
dispatch subscriber and if a meaningful discussion is solicited than
those message should have gone also to the AVT and MMUSIC mailing list.
I did forward the codec email to the AVT working group and urged people
to join the discussion on the codec topic but maybe the dispatch chairs
should look at the proposals and send them to current working groups
that may provide better feedback

Roni Even

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Mary Barnes
Sent: Thursday, June 04, 2009 11:19 PM
To: dispatch@ietf.org
Cc: rai-ads@tools.ietf.org
Subject: [dispatch] Reminder: IETF-75 plans for DISPATCH

Hi folks,

A reminder to folks that submitted proposals for the initial deadline
that the "charter proposals" are due next Monday, June 8th.  The
following summarizes the proposals (some already having detailed charter
proposals) that have been put forth:
1) 3rd party auth:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00052.html
2) CBUS:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00083.html
3) cc-uui:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00073.html
4) CLF:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00066.html
5) Codec:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00080.html
6) Sound level indicator:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00071.html
7) On demand media/app sharing:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00076.html

Also, if there are already drafts associated with your charter proposal,
please provide appropriate links in your email.=20

For WG participants, there are less than two weeks left (June 15th) to
provide input and feedback on the proposals in terms of whether there is
sufficient interest to warrant discussion during a f2f session in
Stockholm or whether the items can be dispatched prior to Stockholm.=20

Also, please let me know if I've missed something.=20

Regards,
Mary.=20

-----Original Message-----
From: Barnes, Mary (RICH2:AR00)
Sent: Friday, May 08, 2009 12:54 PM
To: dispatch@ietf.org
Cc: rai-ads@tools.ietf.org; 'Gonzalo Camarillo'
Subject: IETF-75 plans for DISPATCH=20

Hi all,

As mentioned previously, we are setting earlier deadlines for discussing
topics in DISPATCH to allow time for constructive discussions prior to
the WG session at IETF-75:=20
http://www.ietf.org/mail-archive/web/dispatch/current/msg00006.html

We are proposing to follow the deadlines for BoFs for IETF-75:
http://www.ietf.org/meetings/75/cutoff-dates.html
since we anticipate that many of the topic discussions will follow a
"mini-BoF" format, based on the DISPATCH charter:
http://www.ietf.org/html.charters/dispatch-charter.html

Thus, the following are the proposed deadlines:

May 25th - Cutoff date to notify the chairs and DISPATCH WG mailing list
of plans to submit a proposal, including the topic of the proposal. This
will help folks with similar topics to work together before the meeting.
If a preliminary charter proposal is available at this point, please
include a pointer to it.

June 8th - Cutoff for charter proposals for topics. A charter proposal
must consist of a minimum of a problem statement and at least one
milestone/deliverable. This deadline will allow time for consideration
of proposals that could potentially be "dispatched" prior to the WG
session.

June 15th - Topics that are to be the focus of IETF-75 are announced.
The idea here is to focus the discussion over the weeks preceding
IETF-75 on these main topics, noting that any updates to the documents
are due 4 weeks later.  This will ensure we have constructive
discussions before the meeting and are actually able to determine
consensus as to where work should be progressed (e.g., separate BoF, a
new WG, an existing WG, individual document, etc.) at IETF-75. Contrary
to the past approach in the SIPPING WG, we anticipate determining
consensus as to the disposition of a work item based on the level of
"commitment" by individuals to actually contribute to the work (e.g.,
conference calls, design team mailing lists, contributing text, etc.)
rather than based on "interest" in doing the work.  Note, that the exact
disposition for a topic may (per the usual process) require follow-up
and confirmation by the ADS and/or IESG (e.g., for a new WG, Bof,
individually sponsored document, etc.) or with another WG to ensure
agreement with the DISPATCH WG consensus for the topic.

Note the intention is not necessarily to preclude folks submitting
documents on other topics, however, it is very unlikely there would be
agenda time allocated to documents/topics submitted after the deadlines.
We can include one or two slides on these topics in the DISPATCH WG
chair charts, as we've done in the past in the SIPPING WG chair charts,
so that the authors can gather other interested parties to contribute to
the work.=20

We appreciate the patience of the WG as we evolve the model for bringing
in new work and welcome feedback.=20

Regards,
Mary and Gonzalo
DISPATCH WG co-chairs=20


_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch


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Roni Even wrote:
> Skype also supports starting a multipoint audio conference from the UI. I
> would consider this more practical than showing the sound level. XCON
> defined this functionality in a standard draft enabling the creation of
> audio and video conferences in an multivendor environment. Does your client
> support this functionality. Do you see this as a mandatory requirement from
> service providers.

Well, from my very limited perspective I actually see an increasing
interest in Internet conferencing -- the technology seems to be finally
usable if not mature, plus the shrinking of travel budgets pushes in
that direction -- and features available in free services/applications
are probably to be considered as a must-have for any commercial offering.

> My concern is that we develop solutions that are not deployed since they
> have no high priority for developers and customers. Doing a "cool" solution
> in a proprietary environment is simpler since you control the clients as
> well as the servers and do not worry about interoperability with other
> vendors. 

The extension needed to carry such information is trivial and folks at
SIP Communicator are actually going to implement that. Unfortunately
such an extension does not exist yet and the IETF community can provide
guidance to define and document it, rather than push them toward a
proprietary solution.

-- 
Ciao,
Enrico

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--------------ms080003000308040206000901--

From mizuno.shintaro@lab.ntt.co.jp  Mon Jun  8 02:20:10 2009
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Date: Mon, 8 Jun 2009 18:20:16 +0900
From: Shintaro Mizuno <mizuno.shintaro@lab.ntt.co.jp>
To: dispatch@ietf.org
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Subject: Re: [dispatch] Charter Proposal
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Dear DISPATCH Chair, ML members,

I've finalized the charter proposal for on-demand media/application
sharing.
Added milestone/deliverables section and some descriptions on the usage
of SIP, but most of the text remains the same.

--------------------------------------------------
"Charter Proposal for on-demand Media/Application Sharing between peers
over VPN"

Intent:
 The intent of this charter is to see interests in documenting what's
being deployed as an informational for on-demand Media/Application
sharing between peers over VPN using SIP and discuss what in the
overall effort may be standardized at IETF in the future.


Problem Statement: 
 Home servers or network-capable consumer electronic devices are
becoming widely deployed and people using such devices are wanting to
share contents or applications and are seeking a way to establish
multitudes of communication with each other.  

 In order to allow two people to share contents or application securely
using popular LAN based communication tool such as DLNA on devices that
are sitting at home, secure LAN (VPN) establishments between these user
devices are necessary. Applications that use proprietary or
multi-session protocols such as in some of LAN-based gaming will also
benifit from VPN connections when users attempt to communicate remotely.

 For a deployment that's underway and considered by several other
standard bodies, problem of establishing VPN between these user devices
were that current standards for establishing VPN didn't provide all the
toolset necessary to overcome some of the obstacles. 

 Namely following issues were found;
  (1) Name resolution mechanism under floating IP addresses and port
numbers
  (2) NAT traversal and hole-punching of firewall
  (3) Authentication/authorization of users.
  (4) Pre-sharing of key for establishing VPN isn't easy when VPN is to
be established on-demand between two user devices without any previous
affiliation.

As a result current deployment based on "draft-saito-mmusic-sdp-ike-04"
and other standard bodies are looking at the use of SIP for the
following reasons. 

 (1)Name resolution can be achieved by using REGISTER method and SDP
negotiation
 (2)NAT traversal and hole-punching of firewall can be achieved by
applying SIP+ICE.
 (3)Authentication/authorization can be achieved by having trusted SIP
proxy and user friendly SIP-URIs.

 SIP/SDP will be used to negotiate parameters necessary to establish
IKE/IPsec such as IP address, port numbers and authentication
information of peers. Current deployment uses SIP also to manage IPsec
connection, e.g. use BYE to shut down media sessions for IKE/IPsec,
however, further discussions can be made to find out if there are
intrests or use cases in using SIP only as a rendezvous protocol.

Use-cases that may see use for such standard effort;
  - Media sharing using DLNA or similar protocol over VPN between 2
users' devices.
  - Remote desktop sharing for Customer service over VPN initiated via
SIP call.
   * Similar to click to call, customer service agent can access user's
pc remotely to troubleshoot the problem customer is facing from the call.
  - Accessing medical equipments(medical robotics) remotely and
possibly controlling medical equipments to monitor elders in a rural
area that (remote care services). 
  - LAN based gaming protocol to be established peer to peer rather
than via gaming server.


Discussion Point:
 - To see interests in documenting what's being deployed based on
"draft-saito-mmusic-sdp-ike-04" as an informational document.
http://tools.ietf.org/html/draft-saito-mmusic-sdp-ike-04 
 (to be minor-updated to -05 to conform with this charter proposal)
 - To see interests in standardizing toolset for realizing use-cases
similar to what's mentioned above, even contemplating the possibility
of defining a guideline for SIP usage where SIP is used only as a
rendezvous protocol.


Milestone/Deliverables:
 - Aug 09 Issue Internet-Draft
    (by updating "draft-saito-mmusic-sdp-ike-05")
 - Nov 09 Submit I-D to IESG for publication as an RFC (Informational)

--------------------------------------------------
Best Regards,
Shintaro

----------------------------------------
Shintaro MIZUNO
mizuno.shintaro@lab.ntt.co.jp

NTT Information Sharing Platform Laboratories
NTT Corporation
----------------------------------------


From eburger@standardstrack.com  Mon Jun  8 03:38:18 2009
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From: Eric Burger <eburger@standardstrack.com>
To: Shintaro Mizuno <mizuno.shintaro@lab.ntt.co.jp>
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Subject: Re: [dispatch] Charter Proposal
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Given the proposal is not really for an Internet standard, and the  
product is not a protocol or standards track document, why are we  
working on it here?

On Jun 8, 2009, at 5:20 AM, Shintaro Mizuno wrote:

> Dear DISPATCH Chair, ML members,
>
> I've finalized the charter proposal for on-demand media/application
> sharing.
> Added milestone/deliverables section and some descriptions on the  
> usage
> of SIP, but most of the text remains the same.
>
> --------------------------------------------------
> "Charter Proposal for on-demand Media/Application Sharing between  
> peers
> over VPN"
>
> Intent:
> The intent of this charter is to see interests in documenting what's
> being deployed as an informational for on-demand Media/Application
> sharing between peers over VPN using SIP and discuss what in the
> overall effort may be standardized at IETF in the future.
>
>
> Problem Statement:
> Home servers or network-capable consumer electronic devices are
> becoming widely deployed and people using such devices are wanting to
> share contents or applications and are seeking a way to establish
> multitudes of communication with each other.
>
> In order to allow two people to share contents or application securely
> using popular LAN based communication tool such as DLNA on devices  
> that
> are sitting at home, secure LAN (VPN) establishments between these  
> user
> devices are necessary. Applications that use proprietary or
> multi-session protocols such as in some of LAN-based gaming will also
> benifit from VPN connections when users attempt to communicate  
> remotely.
>
> For a deployment that's underway and considered by several other
> standard bodies, problem of establishing VPN between these user  
> devices
> were that current standards for establishing VPN didn't provide all  
> the
> toolset necessary to overcome some of the obstacles.
>
> Namely following issues were found;
>  (1) Name resolution mechanism under floating IP addresses and port
> numbers
>  (2) NAT traversal and hole-punching of firewall
>  (3) Authentication/authorization of users.
>  (4) Pre-sharing of key for establishing VPN isn't easy when VPN is to
> be established on-demand between two user devices without any previous
> affiliation.
>
> As a result current deployment based on "draft-saito-mmusic-sdp- 
> ike-04"
> and other standard bodies are looking at the use of SIP for the
> following reasons.
>
> (1)Name resolution can be achieved by using REGISTER method and SDP
> negotiation
> (2)NAT traversal and hole-punching of firewall can be achieved by
> applying SIP+ICE.
> (3)Authentication/authorization can be achieved by having trusted SIP
> proxy and user friendly SIP-URIs.
>
> SIP/SDP will be used to negotiate parameters necessary to establish
> IKE/IPsec such as IP address, port numbers and authentication
> information of peers. Current deployment uses SIP also to manage IPsec
> connection, e.g. use BYE to shut down media sessions for IKE/IPsec,
> however, further discussions can be made to find out if there are
> intrests or use cases in using SIP only as a rendezvous protocol.
>
> Use-cases that may see use for such standard effort;
>  - Media sharing using DLNA or similar protocol over VPN between 2
> users' devices.
>  - Remote desktop sharing for Customer service over VPN initiated via
> SIP call.
>   * Similar to click to call, customer service agent can access user's
> pc remotely to troubleshoot the problem customer is facing from the  
> call.
>  - Accessing medical equipments(medical robotics) remotely and
> possibly controlling medical equipments to monitor elders in a rural
> area that (remote care services).
>  - LAN based gaming protocol to be established peer to peer rather
> than via gaming server.
>
>
> Discussion Point:
> - To see interests in documenting what's being deployed based on
> "draft-saito-mmusic-sdp-ike-04" as an informational document.
> http://tools.ietf.org/html/draft-saito-mmusic-sdp-ike-04
> (to be minor-updated to -05 to conform with this charter proposal)
> - To see interests in standardizing toolset for realizing use-cases
> similar to what's mentioned above, even contemplating the possibility
> of defining a guideline for SIP usage where SIP is used only as a
> rendezvous protocol.
>
>
> Milestone/Deliverables:
> - Aug 09 Issue Internet-Draft
>    (by updating "draft-saito-mmusic-sdp-ike-05")
> - Nov 09 Submit I-D to IESG for publication as an RFC (Informational)
>
> --------------------------------------------------
> Best Regards,
> Shintaro
>
> ----------------------------------------
> Shintaro MIZUNO
> mizuno.shintaro@lab.ntt.co.jp
>
> NTT Information Sharing Platform Laboratories
> NTT Corporation
> ----------------------------------------
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


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From vkg@alcatel-lucent.com  Mon Jun  8 08:45:56 2009
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Date: Mon, 08 Jun 2009 10:46:00 -0500
From: "Vijay K. Gurbani" <vkg@alcatel-lucent.com>
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Subject: [dispatch] sip-clf: final proposal
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Mary Barnes wrote:
 > Hi folks,
 > A reminder to folks that submitted proposals for the initial deadline
 > that the "charter proposals" are due next Monday, June 8th.
[...]

Mary: I did not receive any more follow-up comments on sip-clf
proposal description since my post last week (please see
http://www.ietf.org/mail-archive/web/dispatch/current/msg00176.html).

So, the description contained therein constitutes the charter
proposal for the sip-clf work.

Thanks,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/

From alan@sipstation.com  Mon Jun  8 09:17:03 2009
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From: Alan Johnston <alan@sipstation.com>
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Cc: Joanne McMillen <joanne@avaya.com>, dispatch@ietf.org
Subject: Re: [dispatch] Call Control UUI Problem Statement
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Dean,

Thanks for your comments - see mine below.

Thanks,
Alan

Dean Willis wrote:
> Alan Johnston wrote:
>   
>> Since the INFO method, RFC2976, was developed for ISUP interworking of
>> user-to-user information, it might seem to be the logical choice here. 
>> For non-call control user-to-user information, INFO can be utilized for
>> end to end transport.  However, for transport of call control
>> user-to-user information, INFO can not be used.  Since the contents
>> carry information that can impact call handling at the UAC, the
>> information must be conveyed with the INVITE transaction.  As the call
>> flows in the previous section show, the information is related to an
>> attempt to establish a session and must be passed with the session setup
>> request (INVITE), responses to that INVITE, or session termination
>> requests.  As a result, it is not possible to use INFO in these cases.
>>     
>
> I favor extending the approach of INFO packages to allow inclusion of
> informational bodies in other transactions, notably INVITE.
>
> This lets us re-use the negotiation and encoding mechanisms already
> painfully worked out for INFO, and keeps us from having to invent yet
> another method for semantic content negotiation, which I see as the
> weakness of the UUI-as-a-new-header-field approach that has been proposed.
>   

I think this is worth discussing more.  I also think we can add some 
additional semantics and tagging with UUI.  However, I don't see how the 
INFO approach, which does the negotiation via header fields in the 
initial dialog setup (INVITE/200/ACK) can work here, given the Call 
Control UUI must be present in the INVITE/200 exchange prior to dialog 
establishment.

> Further, I really think that the headers belong in the realm of the SIP
> protocol machine, whereas the bodies belong to the applications that sue
> SIP. Extending SIP with application-specific header fields feels like a
> layer violation; we already have enough of those and I'd like to avoid
> introducing more.
>   

The problem is that for call control UUI, the body approach does not 
meet the requirements we have discussed and agreed to in SIPPING.  Also, 
a body would require no standardization and work in the IETF - if this 
approach worked, none of us would be pushing a SIP extension for this.

> >From a DISPATCH perspective, this raises the question of where to do the
> work.
>
> I'm inclined to propose a new working group, which I'll tentatively call
> INFORM. The INFORM working group would assume responsibility for
> informational supplementation of SIP messages, including finishing the
> INFO packages work, and addressing the requirements of UUI as raised in
> Alan's draft. I'd also propose a third deliverable, which would be an
> informational document giving guidance on the usage of SIP's
> user-to-user informational extensions, perhaps a design-usage tutorial.
>
> So, three deliverables, a fairly mature problem statement for UUI and a
> very mature document for INFO combine to give us a manageable,
> achieveable scope. This is exactly the sort of short-lived,
> narrowly-defined working group I've been saying we need to use to
> replace the nigh-immortal behemoths like SIP and SIPPING and that we
> need to be able to spin up in order to keep SIPCORE and DISPATCH from
> turning into undead reanimations of SIP and SIPPING. Zombies may be
> fashionable, but they don't make for effective management models.
>
> Of course, we could call the new working group SINFORM for "Sip
> INFORMation". That has a nice ring to it...
>   

As it does seem silly to form a separate working group to standardize a 
single, trivial header field, I would not be opposed to grouping it with 
the INFO package work.  However, this would only be for admin reasons - 
the mechanism and requirements are, in reality, very very different.

> --
> Dean
>
>
>   


From enrico.marocco@telecomitalia.it  Mon Jun  8 12:33:46 2009
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Subject: [dispatch] Milestone(s) for the Sound Level Indicators problem
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--------------ms050103030304020504070505
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After some online and offline discussion about how to address the
problem described in draft-ivov-dispatch-slic-ps-00, i.e. how to enable
a mixer to pass the sound levels of the contributing streams that result
in a mixed stream back to the clients, it seems that a reasonable way
forward would be to have the AVT WG defining the missing piece (an RTP
extension to encode sound level values for the CSRCs). So, we would like
to propose the following milestone for AVT:

"Submit RTP header extension (as per RFC 5285) for multiple sound level
indicators."

For whoever has missed it, the problem was initially discussed here:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00071.html

As usual, comments are welcome.

-- 
Ciao,
Enrico

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From ron.even.tlv@gmail.com  Mon Jun  8 14:07:19 2009
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Cullen Jennings'" <fluffy@cisco.com>, "'Emil Ivov'" <emcho@sip-communicator.org>
References: <4A15D99F.9060100@sip-communicator.org> <EA1CBEDB-319A-43BC-B874-909B8267F017@cisco.com>
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Subject: Re: [dispatch] Yet another problem to dispatch
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Cullen,
In the PSTN integration I am aware of two methods of associating the PSTN
call with the web UI. The first is when the we UI is used to dial out to the
user using the PBX telephony API. The second is that when a user is defined
in the UI he gets a PIN number so when he dials in to the conference bridge
the user must enter this PIN number and using the telephony API it is
conveyed to the we UI.
There are other solutions
Roni

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
Of Cullen Jennings
Sent: Wednesday, May 27, 2009 8:15 PM
To: Emil Ivov
Cc: dispatch@ietf.org
Subject: Re: [dispatch] Yet another problem to dispatch


I find this "active speaker" problem very interesting. Often the  
device that wants to display the active speaker information is not the  
same as the device dealing with the RTP. For example, most the  
conference systems I am using a phone but there is a web interface  
that is dealing with active speaker information on my browser which is  
no on my phone. Be interesting to look at all the existing approaches  
to this.


Cullen <in my individual contributor role>

On May 21, 2009, at 4:45 PM, Emil Ivov wrote:

> Hey folks,
>
> We have been working on client side conference calls  for SIP
> Communicator lately and were thinking about implementing participant
> sound level indicators. The feature was inspired by Skype's Mac OS X  
> GUI
> which you can have a look at over here:
>
> http://sip-communicator.org/skypeconf/skypeconf.png
>
> Since we couldn't find an existing IETF mechanism for a mixer to
> dispatch information on sound level to conference participants, Enrico
> and I thought that this may be an issue that is worth addressing  
> through
> dispatch.
>
> After talking to some of you privately and discussing it among  
> ourselves
> we have decided to come up with a document presenting the issue in  
> some
> more detail. We are also including short descriptions of what we
> consider to be possible approaches for resolving it.
>
> http://www.ietf.org/internet-drafts/draft-ivov-dispatch-slic-ps-00.txt
>
> Comments are most welcome, as well as any show of interest or
> indications of what you think would be the right place(s) to work on  
> the
> subject.
>
> Thanks,
> Emil
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch


From Even.roni@huawei.com  Mon Jun  8 14:29:28 2009
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Date: Tue, 09 Jun 2009 00:28:05 +0300
From: Roni Even <Even.roni@huawei.com>
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Subject: Re: [dispatch] Milestone(s) for the Sound Level Indicators problem
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Enrico
I read draft-ivov-dispatch-slic-ps-00 and what is still bothering me is that
since you do not want to use the conference event package to map between the
ssrc and the display name for the user (like Bob, Alice), I am not sure how
this will work. The mixer will just give the SSRCs of each original stream.
It can not supply the mapping since the mixer is not supposed to know it.

Are you going to address this part or will it be left to each specific
implementation and is out of scope to this work. I am not sure that the
client can do it by itself without any information from a central entity.

Roni Even


-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
Of Enrico Marocco
Sent: Monday, June 08, 2009 10:34 PM
To: dispatch@ietf.org
Subject: [dispatch] Milestone(s) for the Sound Level Indicators problem

After some online and offline discussion about how to address the problem
described in draft-ivov-dispatch-slic-ps-00, i.e. how to enable a mixer to
pass the sound levels of the contributing streams that result in a mixed
stream back to the clients, it seems that a reasonable way forward would be
to have the AVT WG defining the missing piece (an RTP extension to encode
sound level values for the CSRCs). So, we would like to propose the
following milestone for AVT:

"Submit RTP header extension (as per RFC 5285) for multiple sound level
indicators."

For whoever has missed it, the problem was initially discussed here:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00071.html

As usual, comments are welcome.

--
Ciao,
Enrico


From emil@sip-communicator.org  Mon Jun  8 17:22:23 2009
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Sender: Emil Ivov <emil@sip-communicator.org>
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Date: Tue, 09 Jun 2009 02:22:22 +0200
From: Emil Ivov <emcho@sip-communicator.org>
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To: Roni Even <Even.roni@huawei.com>
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Subject: Re: [dispatch] Milestone(s) for the Sound Level Indicators problem
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Hey Roni,


We definitely want to use the conference package for the mapping.
There's absolutely no doubt about this. I believe we mention it once or
twice in the draft but probably not clearly enough. Apologies.

What we don't want to do is use the mechanisms in the conference package
to transport the sound level information itself. In other words we want
the actual sound levels in RTP packets and not in SIP messages.

Hope this makes it clearer.

Emil

Roni Even wrote:
> Enrico
> I read draft-ivov-dispatch-slic-ps-00 and what is still bothering me is that
> since you do not want to use the conference event package to map between the
> ssrc and the display name for the user (like Bob, Alice), I am not sure how
> this will work. The mixer will just give the SSRCs of each original stream.
> It can not supply the mapping since the mixer is not supposed to know it.
> 
> Are you going to address this part or will it be left to each specific
> implementation and is out of scope to this work. I am not sure that the
> client can do it by itself without any information from a central entity.
> 
> Roni Even
> 
> 
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
> Of Enrico Marocco
> Sent: Monday, June 08, 2009 10:34 PM
> To: dispatch@ietf.org
> Subject: [dispatch] Milestone(s) for the Sound Level Indicators problem
> 
> After some online and offline discussion about how to address the problem
> described in draft-ivov-dispatch-slic-ps-00, i.e. how to enable a mixer to
> pass the sound levels of the contributing streams that result in a mixed
> stream back to the clients, it seems that a reasonable way forward would be
> to have the AVT WG defining the missing piece (an RTP extension to encode
> sound level values for the CSRCs). So, we would like to propose the
> following milestone for AVT:
> 
> "Submit RTP header extension (as per RFC 5285) for multiple sound level
> indicators."
> 
> For whoever has missed it, the problem was initially discussed here:
> http://www.ietf.org/mail-archive/web/dispatch/current/msg00071.html
> 
> As usual, comments are welcome.
> 
> --
> Ciao,
> Enrico
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

-- 
Emil Ivov, Ph.D.                               30a rue de la Patrie
Project Lead                                   67300 Schiltigheim, France
SIP Communicator
emcho@sip-communicator.org                     FAX:   +33.1.77.62.47.31
http://sip-communicator.org                    PHONE: +33.1.77.62.43.30

From Even.roni@huawei.com  Mon Jun  8 22:21:33 2009
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From: Roni Even <Even.roni@huawei.com>
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Emil,
Sorry if I did not understand but the draft mentions that SIP based solution
like conference event package will not work in XMPP environment (section 2.1
of the draft)

" It is probably also worth mentioning that the use of RFC 4575
   [RFC4575] for such a feature would make the mechanism incompatible
   with non-SIP signaling protocols like, for example, XMPP [RFC3920]
   and its Jingle extensions."

Roni

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
Of Emil Ivov
Sent: Tuesday, June 09, 2009 3:22 AM
To: Roni Even
Cc: dispatch@ietf.org
Subject: Re: [dispatch] Milestone(s) for the Sound Level Indicators problem

Hey Roni,


We definitely want to use the conference package for the mapping.
There's absolutely no doubt about this. I believe we mention it once or
twice in the draft but probably not clearly enough. Apologies.

What we don't want to do is use the mechanisms in the conference package
to transport the sound level information itself. In other words we want
the actual sound levels in RTP packets and not in SIP messages.

Hope this makes it clearer.

Emil

Roni Even wrote:
> Enrico
> I read draft-ivov-dispatch-slic-ps-00 and what is still bothering me is
that
> since you do not want to use the conference event package to map between
the
> ssrc and the display name for the user (like Bob, Alice), I am not sure
how
> this will work. The mixer will just give the SSRCs of each original
stream.
> It can not supply the mapping since the mixer is not supposed to know it.
> 
> Are you going to address this part or will it be left to each specific
> implementation and is out of scope to this work. I am not sure that the
> client can do it by itself without any information from a central entity.
> 
> Roni Even
> 
> 
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf
> Of Enrico Marocco
> Sent: Monday, June 08, 2009 10:34 PM
> To: dispatch@ietf.org
> Subject: [dispatch] Milestone(s) for the Sound Level Indicators problem
> 
> After some online and offline discussion about how to address the problem
> described in draft-ivov-dispatch-slic-ps-00, i.e. how to enable a mixer to
> pass the sound levels of the contributing streams that result in a mixed
> stream back to the clients, it seems that a reasonable way forward would
be
> to have the AVT WG defining the missing piece (an RTP extension to encode
> sound level values for the CSRCs). So, we would like to propose the
> following milestone for AVT:
> 
> "Submit RTP header extension (as per RFC 5285) for multiple sound level
> indicators."
> 
> For whoever has missed it, the problem was initially discussed here:
> http://www.ietf.org/mail-archive/web/dispatch/current/msg00071.html
> 
> As usual, comments are welcome.
> 
> --
> Ciao,
> Enrico
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

-- 
Emil Ivov, Ph.D.                               30a rue de la Patrie
Project Lead                                   67300 Schiltigheim, France
SIP Communicator
emcho@sip-communicator.org                     FAX:   +33.1.77.62.47.31
http://sip-communicator.org                    PHONE: +33.1.77.62.43.30
_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch


From emil@sip-communicator.org  Tue Jun  9 00:38:44 2009
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Cc: dispatch@ietf.org
Subject: Re: [dispatch] Milestone(s) for the Sound Level Indicators problem
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Hey Roni,

Roni Even wrote:
> Emil,
> Sorry if I did not understand but the draft mentions that SIP based solution
> like conference event package will not work in XMPP environment (section 2.1
> of the draft)
> 
> " It is probably also worth mentioning that the use of RFC 4575
>    [RFC4575] for such a feature would make the mechanism incompatible
>    with non-SIP signaling protocols like, for example, XMPP [RFC3920]
>    and its Jingle extensions."

Yes and again, this is about using 4575 to _transport_ the sound levels.

We'll soon submit a draft proposing a possible solution. It will be
explaining our approach in more detail and would hopefully clarify
issues that were left unclear by the problem statement.

Cheers
Emil

> 
> Roni
> 
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
> Of Emil Ivov
> Sent: Tuesday, June 09, 2009 3:22 AM
> To: Roni Even
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] Milestone(s) for the Sound Level Indicators problem
> 
> Hey Roni,
> 
> 
> We definitely want to use the conference package for the mapping.
> There's absolutely no doubt about this. I believe we mention it once or
> twice in the draft but probably not clearly enough. Apologies.
> 
> What we don't want to do is use the mechanisms in the conference package
> to transport the sound level information itself. In other words we want
> the actual sound levels in RTP packets and not in SIP messages.
> 
> Hope this makes it clearer.
> 
> Emil
> 
> Roni Even wrote:
>> Enrico
>> I read draft-ivov-dispatch-slic-ps-00 and what is still bothering me is
> that
>> since you do not want to use the conference event package to map between
> the
>> ssrc and the display name for the user (like Bob, Alice), I am not sure
> how
>> this will work. The mixer will just give the SSRCs of each original
> stream.
>> It can not supply the mapping since the mixer is not supposed to know it.
>>
>> Are you going to address this part or will it be left to each specific
>> implementation and is out of scope to this work. I am not sure that the
>> client can do it by itself without any information from a central entity.
>>
>> Roni Even
>>
>>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf
>> Of Enrico Marocco
>> Sent: Monday, June 08, 2009 10:34 PM
>> To: dispatch@ietf.org
>> Subject: [dispatch] Milestone(s) for the Sound Level Indicators problem
>>
>> After some online and offline discussion about how to address the problem
>> described in draft-ivov-dispatch-slic-ps-00, i.e. how to enable a mixer to
>> pass the sound levels of the contributing streams that result in a mixed
>> stream back to the clients, it seems that a reasonable way forward would
> be
>> to have the AVT WG defining the missing piece (an RTP extension to encode
>> sound level values for the CSRCs). So, we would like to propose the
>> following milestone for AVT:
>>
>> "Submit RTP header extension (as per RFC 5285) for multiple sound level
>> indicators."
>>
>> For whoever has missed it, the problem was initially discussed here:
>> http://www.ietf.org/mail-archive/web/dispatch/current/msg00071.html
>>
>> As usual, comments are welcome.
>>
>> --
>> Ciao,
>> Enrico
>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
> 

-- 
Emil Ivov, Ph.D.                               30a rue de la Patrie
Project Lead                                   67300 Schiltigheim, France
SIP Communicator
emcho@sip-communicator.org                     FAX:   +33.1.77.62.47.31
http://sip-communicator.org                    PHONE: +33.1.77.62.43.30

From dean.willis@softarmor.com  Tue Jun  9 01:15:01 2009
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Date: Tue, 09 Jun 2009 03:14:53 -0500
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Cc: Joanne McMillen <joanne@avaya.com>, dispatch@ietf.org
Subject: Re: [dispatch] Call Control UUI Problem Statement
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Alan Johnston wrote:

> I think this is worth discussing more.  I also think we can add some 
> additional semantics and tagging with UUI.  However, I don't see how the 
> INFO approach, which does the negotiation via header fields in the 
> initial dialog setup (INVITE/200/ACK) can work here, given the Call 
> Control UUI must be present in the INVITE/200 exchange prior to dialog 
> establishment.

I'm not saying use INFO per se, but use the "do you understand this 
usage of INFO" style of negotiation to solve the "do you understand this 
type of UUI" problem.

What happens before INVITE? OPTIONS can, of course.

How do you reject an INVITE that carries an informational body? By 
rejecting the option tag that says "Understanding this sort of UUI is 
required".


>> Further, I really think that the headers belong in the realm of the SIP
>> protocol machine, whereas the bodies belong to the applications that sue
>> SIP. Extending SIP with application-specific header fields feels like a
>> layer violation; we already have enough of those and I'd like to avoid
>> introducing more.
>>   
> 
> 
> The problem is that for call control UUI, the body approach does not 
> meet the requirements we have discussed and agreed to in SIPPING.  Also, 
> a body would require no standardization and work in the IETF - if this 
> approach worked, none of us would be pushing a SIP extension for this.

I'm not sure I believe that. Scratch that; I'm sure I don't believe it, 
and I don't think SIPPING had the right to say "This MUST be a header" 
anyhow.

Further, defining an option tag for "Understands this body type in this 
context" DOES require standardization.

> As it does seem silly to form a separate working group to standardize a 
> single, trivial header field, I would not be opposed to grouping it with 
> the INFO package work.  However, this would only be for admin reasons - 
> the mechanism and requirements are, in reality, very very different.

It's not a trivial header field. As worded, it's a 
cross-layer-violating, bad-usage-encouraging very dangerous hack that is 
going to put us into a must-fix-the-unnegotiated-usages problem in a few 
years if we don't fix it the first time.

--
Dean

From mizuno.shintaro@lab.ntt.co.jp  Tue Jun  9 06:02:43 2009
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Date: Tue, 9 Jun 2009 22:02:42 +0900
From: Shintaro Mizuno <mizuno.shintaro@lab.ntt.co.jp>
To: Eric Burger <eburger@standardstrack.com>
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Subject: Re: [dispatch] Charter Proposal
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Mr. Burger,
 Thank you for the comment.

 We don't necessary see a need for a WG to be formed to document the  
informational document we have listed, the document is there to see if
the proposed way forward (informational document possibly
as an independent submission) for the document listed is on track or  
whether something should be done at the WG level.

 On the other hand, we believe that use of SIP similar to what we  
have done for VPN will likely to happen more. And on that point, we are
wanting to see whether IETF is interested in documenting a guideline
for dealing with application that's not voice or video. 
As written in the discussion point of the charter, the question is how
much should SIP be involved.

 1. Should SIP just be used as a rendezvous protocol and leave the  
rest of the negotiation to whatever application?
 2. Should SIP involve in management of the established session as it
does for audio/video?
 3. Should use of SIP not be standardized for peer to peer application 
beside audio/video?


Regards,
Shintaro


On Mon, 8 Jun 2009 06:38:20 -0400
Eric Burger <eburger@standardstrack.com> wrote:

> Given the proposal is not really for an Internet standard, and the  
> product is not a protocol or standards track document, why are we  
> working on it here?
> 
> On Jun 8, 2009, at 5:20 AM, Shintaro Mizuno wrote:
> 
> > Dear DISPATCH Chair, ML members,
> >
> > I've finalized the charter proposal for on-demand media/application
> > sharing.
> > Added milestone/deliverables section and some descriptions on the  
> > usage
> > of SIP, but most of the text remains the same.
> >
> > --------------------------------------------------
> > "Charter Proposal for on-demand Media/Application Sharing between  
> > peers
> > over VPN"
> >
> > Intent:
> > The intent of this charter is to see interests in documenting what's
> > being deployed as an informational for on-demand Media/Application
> > sharing between peers over VPN using SIP and discuss what in the
> > overall effort may be standardized at IETF in the future.
> >
> >
> > Problem Statement:
> > Home servers or network-capable consumer electronic devices are
> > becoming widely deployed and people using such devices are wanting to
> > share contents or applications and are seeking a way to establish
> > multitudes of communication with each other.
> >
> > In order to allow two people to share contents or application securely
> > using popular LAN based communication tool such as DLNA on devices  
> > that
> > are sitting at home, secure LAN (VPN) establishments between these  
> > user
> > devices are necessary. Applications that use proprietary or
> > multi-session protocols such as in some of LAN-based gaming will also
> > benifit from VPN connections when users attempt to communicate  
> > remotely.
> >
> > For a deployment that's underway and considered by several other
> > standard bodies, problem of establishing VPN between these user  
> > devices
> > were that current standards for establishing VPN didn't provide all  
> > the
> > toolset necessary to overcome some of the obstacles.
> >
> > Namely following issues were found;
> >  (1) Name resolution mechanism under floating IP addresses and port
> > numbers
> >  (2) NAT traversal and hole-punching of firewall
> >  (3) Authentication/authorization of users.
> >  (4) Pre-sharing of key for establishing VPN isn't easy when VPN is to
> > be established on-demand between two user devices without any previous
> > affiliation.
> >
> > As a result current deployment based on "draft-saito-mmusic-sdp- 
> > ike-04"
> > and other standard bodies are looking at the use of SIP for the
> > following reasons.
> >
> > (1)Name resolution can be achieved by using REGISTER method and SDP
> > negotiation
> > (2)NAT traversal and hole-punching of firewall can be achieved by
> > applying SIP+ICE.
> > (3)Authentication/authorization can be achieved by having trusted SIP
> > proxy and user friendly SIP-URIs.
> >
> > SIP/SDP will be used to negotiate parameters necessary to establish
> > IKE/IPsec such as IP address, port numbers and authentication
> > information of peers. Current deployment uses SIP also to manage IPsec
> > connection, e.g. use BYE to shut down media sessions for IKE/IPsec,
> > however, further discussions can be made to find out if there are
> > intrests or use cases in using SIP only as a rendezvous protocol.
> >
> > Use-cases that may see use for such standard effort;
> >  - Media sharing using DLNA or similar protocol over VPN between 2
> > users' devices.
> >  - Remote desktop sharing for Customer service over VPN initiated via
> > SIP call.
> >   * Similar to click to call, customer service agent can access user's
> > pc remotely to troubleshoot the problem customer is facing from the  
> > call.
> >  - Accessing medical equipments(medical robotics) remotely and
> > possibly controlling medical equipments to monitor elders in a rural
> > area that (remote care services).
> >  - LAN based gaming protocol to be established peer to peer rather
> > than via gaming server.
> >
> >
> > Discussion Point:
> > - To see interests in documenting what's being deployed based on
> > "draft-saito-mmusic-sdp-ike-04" as an informational document.
> > http://tools.ietf.org/html/draft-saito-mmusic-sdp-ike-04
> > (to be minor-updated to -05 to conform with this charter proposal)
> > - To see interests in standardizing toolset for realizing use-cases
> > similar to what's mentioned above, even contemplating the possibility
> > of defining a guideline for SIP usage where SIP is used only as a
> > rendezvous protocol.
> >
> >
> > Milestone/Deliverables:
> > - Aug 09 Issue Internet-Draft
> >    (by updating "draft-saito-mmusic-sdp-ike-05")
> > - Nov 09 Submit I-D to IESG for publication as an RFC (Informational)
> >
> > --------------------------------------------------
> > Best Regards,
> > Shintaro
> >
> > ----------------------------------------
> > Shintaro MIZUNO
> > mizuno.shintaro@lab.ntt.co.jp
> >
> > NTT Information Sharing Platform Laboratories
> > NTT Corporation
> > ----------------------------------------
> >
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> 
> 


----------------------------------------
Shintaro MIZUNO
mizuno.shintaro@lab.ntt.co.jp

NTT Information Sharing Platform Laboratories
NTT Corporation
----------------------------------------


From alan@sipstation.com  Tue Jun  9 06:24:23 2009
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Date: Tue, 09 Jun 2009 08:24:24 -0500
From: Alan Johnston <alan@sipstation.com>
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To: Dean Willis <dean.willis@softarmor.com>
References: <4A16DEF2.1010505@sipstation.com> <4A17104A.80706@softarmor.com> <4A2D397F.5080408@sipstation.com> <4A2E19FD.80504@softarmor.com>
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Cc: Joanne McMillen <joanne@avaya.com>, dispatch@ietf.org
Subject: Re: [dispatch] Call Control UUI Problem Statement
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Dean,

SIPPING didn't say it had to be a header, but one of the most important 
requirements is that UUI be passable in Refer-To and redirection URIs.  
While most simple UAs support escaping header fields into URIs for 
transfer, none support escaping bodies into URIs, and for good reason.

However, these are discussions we can have post-DISPATCH.  I don't think 
there really are any open issues around the strong interest in this 
mechanism, and as I said, I'm not opposed to bundling the work with INFO.

Thanks,
Alan

Dean Willis wrote:
> Alan Johnston wrote:
>
>> I think this is worth discussing more.  I also think we can add some 
>> additional semantics and tagging with UUI.  However, I don't see how 
>> the INFO approach, which does the negotiation via header fields in 
>> the initial dialog setup (INVITE/200/ACK) can work here, given the 
>> Call Control UUI must be present in the INVITE/200 exchange prior to 
>> dialog establishment.
>
> I'm not saying use INFO per se, but use the "do you understand this 
> usage of INFO" style of negotiation to solve the "do you understand 
> this type of UUI" problem.
>
> What happens before INVITE? OPTIONS can, of course.
>
> How do you reject an INVITE that carries an informational body? By 
> rejecting the option tag that says "Understanding this sort of UUI is 
> required".
>
>
>>> Further, I really think that the headers belong in the realm of the SIP
>>> protocol machine, whereas the bodies belong to the applications that 
>>> sue
>>> SIP. Extending SIP with application-specific header fields feels like a
>>> layer violation; we already have enough of those and I'd like to avoid
>>> introducing more.
>>>   
>>
>>
>> The problem is that for call control UUI, the body approach does not 
>> meet the requirements we have discussed and agreed to in SIPPING.  
>> Also, a body would require no standardization and work in the IETF - 
>> if this approach worked, none of us would be pushing a SIP extension 
>> for this.
>
> I'm not sure I believe that. Scratch that; I'm sure I don't believe 
> it, and I don't think SIPPING had the right to say "This MUST be a 
> header" anyhow.
>
> Further, defining an option tag for "Understands this body type in 
> this context" DOES require standardization.
>
>> As it does seem silly to form a separate working group to standardize 
>> a single, trivial header field, I would not be opposed to grouping it 
>> with the INFO package work.  However, this would only be for admin 
>> reasons - the mechanism and requirements are, in reality, very very 
>> different.
>
> It's not a trivial header field. As worded, it's a 
> cross-layer-violating, bad-usage-encouraging very dangerous hack that 
> is going to put us into a must-fix-the-unnegotiated-usages problem in 
> a few years if we don't fix it the first time.
>
> -- 
> Dean
>


From hsinnrei@adobe.com  Tue Jun  9 06:32:01 2009
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From: Henry Sinnreich <hsinnrei@adobe.com>
To: Shintaro Mizuno <mizuno.shintaro@lab.ntt.co.jp>, Eric Burger <eburger@standardstrack.com>
Date: Tue, 9 Jun 2009 06:31:52 -0700
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--_000_C653CE783FF1hsinnreiadobecom_
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You may also want to consider HIP, since it solves NAT traversal for all ap=
plication layer protocols, including SIP.

http://www.ietf.org/internet-drafts/draft-ietf-hip-nat-traversal-07.txt

It also takes care of the IPv4 to IPv6 transition for endpoints, mobility a=
nd multi-homing.
Open source code is plentifully available for reference implementations.

http://www.ietf.org/html.charters/hip-charter.html

Henry



On 6/9/09 8:02 AM, "Shintaro Mizuno" <mizuno.shintaro@lab.ntt.co.jp> wrote:

Mr. Burger,
 Thank you for the comment.

 We don't necessary see a need for a WG to be formed to document the
informational document we have listed, the document is there to see if
the proposed way forward (informational document possibly
as an independent submission) for the document listed is on track or
whether something should be done at the WG level.

 On the other hand, we believe that use of SIP similar to what we
have done for VPN will likely to happen more. And on that point, we are
wanting to see whether IETF is interested in documenting a guideline
for dealing with application that's not voice or video.
As written in the discussion point of the charter, the question is how
much should SIP be involved.

 1. Should SIP just be used as a rendezvous protocol and leave the
rest of the negotiation to whatever application?
 2. Should SIP involve in management of the established session as it
does for audio/video?
 3. Should use of SIP not be standardized for peer to peer application
beside audio/video?


Regards,
Shintaro


On Mon, 8 Jun 2009 06:38:20 -0400
Eric Burger <eburger@standardstrack.com> wrote:

> Given the proposal is not really for an Internet standard, and the
> product is not a protocol or standards track document, why are we
> working on it here?
>
> On Jun 8, 2009, at 5:20 AM, Shintaro Mizuno wrote:
>
> > Dear DISPATCH Chair, ML members,
> >
> > I've finalized the charter proposal for on-demand media/application
> > sharing.
> > Added milestone/deliverables section and some descriptions on the
> > usage
> > of SIP, but most of the text remains the same.
> >
> > --------------------------------------------------
> > "Charter Proposal for on-demand Media/Application Sharing between
> > peers
> > over VPN"
> >
> > Intent:
> > The intent of this charter is to see interests in documenting what's
> > being deployed as an informational for on-demand Media/Application
> > sharing between peers over VPN using SIP and discuss what in the
> > overall effort may be standardized at IETF in the future.
> >
> >
> > Problem Statement:
> > Home servers or network-capable consumer electronic devices are
> > becoming widely deployed and people using such devices are wanting to
> > share contents or applications and are seeking a way to establish
> > multitudes of communication with each other.
> >
> > In order to allow two people to share contents or application securely
> > using popular LAN based communication tool such as DLNA on devices
> > that
> > are sitting at home, secure LAN (VPN) establishments between these
> > user
> > devices are necessary. Applications that use proprietary or
> > multi-session protocols such as in some of LAN-based gaming will also
> > benifit from VPN connections when users attempt to communicate
> > remotely.
> >
> > For a deployment that's underway and considered by several other
> > standard bodies, problem of establishing VPN between these user
> > devices
> > were that current standards for establishing VPN didn't provide all
> > the
> > toolset necessary to overcome some of the obstacles.
> >
> > Namely following issues were found;
> >  (1) Name resolution mechanism under floating IP addresses and port
> > numbers
> >  (2) NAT traversal and hole-punching of firewall
> >  (3) Authentication/authorization of users.
> >  (4) Pre-sharing of key for establishing VPN isn't easy when VPN is to
> > be established on-demand between two user devices without any previous
> > affiliation.
> >
> > As a result current deployment based on "draft-saito-mmusic-sdp-
> > ike-04"
> > and other standard bodies are looking at the use of SIP for the
> > following reasons.
> >
> > (1)Name resolution can be achieved by using REGISTER method and SDP
> > negotiation
> > (2)NAT traversal and hole-punching of firewall can be achieved by
> > applying SIP+ICE.
> > (3)Authentication/authorization can be achieved by having trusted SIP
> > proxy and user friendly SIP-URIs.
> >
> > SIP/SDP will be used to negotiate parameters necessary to establish
> > IKE/IPsec such as IP address, port numbers and authentication
> > information of peers. Current deployment uses SIP also to manage IPsec
> > connection, e.g. use BYE to shut down media sessions for IKE/IPsec,
> > however, further discussions can be made to find out if there are
> > intrests or use cases in using SIP only as a rendezvous protocol.
> >
> > Use-cases that may see use for such standard effort;
> >  - Media sharing using DLNA or similar protocol over VPN between 2
> > users' devices.
> >  - Remote desktop sharing for Customer service over VPN initiated via
> > SIP call.
> >   * Similar to click to call, customer service agent can access user's
> > pc remotely to troubleshoot the problem customer is facing from the
> > call.
> >  - Accessing medical equipments(medical robotics) remotely and
> > possibly controlling medical equipments to monitor elders in a rural
> > area that (remote care services).
> >  - LAN based gaming protocol to be established peer to peer rather
> > than via gaming server.
> >
> >
> > Discussion Point:
> > - To see interests in documenting what's being deployed based on
> > "draft-saito-mmusic-sdp-ike-04" as an informational document.
> > http://tools.ietf.org/html/draft-saito-mmusic-sdp-ike-04
> > (to be minor-updated to -05 to conform with this charter proposal)
> > - To see interests in standardizing toolset for realizing use-cases
> > similar to what's mentioned above, even contemplating the possibility
> > of defining a guideline for SIP usage where SIP is used only as a
> > rendezvous protocol.
> >
> >
> > Milestone/Deliverables:
> > - Aug 09 Issue Internet-Draft
> >    (by updating "draft-saito-mmusic-sdp-ike-05")
> > - Nov 09 Submit I-D to IESG for publication as an RFC (Informational)
> >
> > --------------------------------------------------
> > Best Regards,
> > Shintaro
> >
> > ----------------------------------------
> > Shintaro MIZUNO
> > mizuno.shintaro@lab.ntt.co.jp
> >
> > NTT Information Sharing Platform Laboratories
> > NTT Corporation
> > ----------------------------------------
> >
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
>
>


----------------------------------------
Shintaro MIZUNO
mizuno.shintaro@lab.ntt.co.jp

NTT Information Sharing Platform Laboratories
NTT Corporation
----------------------------------------

_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch


--_000_C653CE783FF1hsinnreiadobecom_
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<HTML>
<HEAD>
<TITLE>Re: [dispatch] Charter Proposal</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
18pt'>You may also want to consider HIP, since it solves NAT traversal for =
<I>all</I> application layer protocols, including SIP.<BR>
<BR>
<a href=3D"http://www.ietf.org/internet-drafts/draft-ietf-hip-nat-traversal=
-07.txt">http://www.ietf.org/internet-drafts/draft-ietf-hip-nat-traversal-0=
7.txt</a><BR>
<BR>
It also takes care of the IPv4 to IPv6 transition for endpoints, mobility a=
nd multi-homing.<BR>
Open source code is plentifully available for reference implementations.<BR=
>
<BR>
<a href=3D"http://www.ietf.org/html.charters/hip-charter.html">http://www.i=
etf.org/html.charters/hip-charter.html</a><BR>
<BR>
Henry<BR>
<BR>
<BR>
<BR>
On 6/9/09 8:02 AM, &quot;Shintaro Mizuno&quot; &lt;<a href=3D"mizuno.shinta=
ro@lab.ntt.co.jp">mizuno.shintaro@lab.ntt.co.jp</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:18pt'>Mr. Burger,<BR>
&nbsp;Thank you for the comment.<BR>
<BR>
&nbsp;We don't necessary see a need for a WG to be formed to document the <=
BR>
informational document we have listed, the document is there to see if<BR>
the proposed way forward (informational document possibly<BR>
as an independent submission) for the document listed is on track or <BR>
whether something should be done at the WG level.<BR>
<BR>
&nbsp;On the other hand, we believe that use of SIP similar to what we <BR>
have done for VPN will likely to happen more. And on that point, we are<BR>
wanting to see whether IETF is interested in documenting a guideline<BR>
for dealing with application that's not voice or video.<BR>
As written in the discussion point of the charter, the question is how<BR>
much should SIP be involved.<BR>
<BR>
&nbsp;1. Should SIP just be used as a rendezvous protocol and leave the <BR=
>
rest of the negotiation to whatever application?<BR>
&nbsp;2. Should SIP involve in management of the established session as it<=
BR>
does for audio/video?<BR>
&nbsp;3. Should use of SIP not be standardized for peer to peer application=
<BR>
beside audio/video?<BR>
<BR>
<BR>
Regards,<BR>
Shintaro<BR>
<BR>
<BR>
On Mon, 8 Jun 2009 06:38:20 -0400<BR>
Eric Burger &lt;<a href=3D"eburger@standardstrack.com">eburger@standardstra=
ck.com</a>&gt; wrote:<BR>
<BR>
&gt; Given the proposal is not really for an Internet standard, and the <BR=
>
&gt; product is not a protocol or standards track document, why are we <BR>
&gt; working on it here?<BR>
&gt;<BR>
&gt; On Jun 8, 2009, at 5:20 AM, Shintaro Mizuno wrote:<BR>
&gt;<BR>
&gt; &gt; Dear DISPATCH Chair, ML members,<BR>
&gt; &gt;<BR>
&gt; &gt; I've finalized the charter proposal for on-demand media/applicati=
on<BR>
&gt; &gt; sharing.<BR>
&gt; &gt; Added milestone/deliverables section and some descriptions on the=
 <BR>
&gt; &gt; usage<BR>
&gt; &gt; of SIP, but most of the text remains the same.<BR>
&gt; &gt;<BR>
&gt; &gt; --------------------------------------------------<BR>
&gt; &gt; &quot;Charter Proposal for on-demand Media/Application Sharing be=
tween <BR>
&gt; &gt; peers<BR>
&gt; &gt; over VPN&quot;<BR>
&gt; &gt;<BR>
&gt; &gt; Intent:<BR>
&gt; &gt; The intent of this charter is to see interests in documenting wha=
t's<BR>
&gt; &gt; being deployed as an informational for on-demand Media/Applicatio=
n<BR>
&gt; &gt; sharing between peers over VPN using SIP and discuss what in the<=
BR>
&gt; &gt; overall effort may be standardized at IETF in the future.<BR>
&gt; &gt;<BR>
&gt; &gt;<BR>
&gt; &gt; Problem Statement:<BR>
&gt; &gt; Home servers or network-capable consumer electronic devices are<B=
R>
&gt; &gt; becoming widely deployed and people using such devices are wantin=
g to<BR>
&gt; &gt; share contents or applications and are seeking a way to establish=
<BR>
&gt; &gt; multitudes of communication with each other.<BR>
&gt; &gt;<BR>
&gt; &gt; In order to allow two people to share contents or application sec=
urely<BR>
&gt; &gt; using popular LAN based communication tool such as DLNA on device=
s <BR>
&gt; &gt; that<BR>
&gt; &gt; are sitting at home, secure LAN (VPN) establishments between thes=
e <BR>
&gt; &gt; user<BR>
&gt; &gt; devices are necessary. Applications that use proprietary or<BR>
&gt; &gt; multi-session protocols such as in some of LAN-based gaming will =
also<BR>
&gt; &gt; benifit from VPN connections when users attempt to communicate <B=
R>
&gt; &gt; remotely.<BR>
&gt; &gt;<BR>
&gt; &gt; For a deployment that's underway and considered by several other<=
BR>
&gt; &gt; standard bodies, problem of establishing VPN between these user <=
BR>
&gt; &gt; devices<BR>
&gt; &gt; were that current standards for establishing VPN didn't provide a=
ll <BR>
&gt; &gt; the<BR>
&gt; &gt; toolset necessary to overcome some of the obstacles.<BR>
&gt; &gt;<BR>
&gt; &gt; Namely following issues were found;<BR>
&gt; &gt; &nbsp;(1) Name resolution mechanism under floating IP addresses a=
nd port<BR>
&gt; &gt; numbers<BR>
&gt; &gt; &nbsp;(2) NAT traversal and hole-punching of firewall<BR>
&gt; &gt; &nbsp;(3) Authentication/authorization of users.<BR>
&gt; &gt; &nbsp;(4) Pre-sharing of key for establishing VPN isn't easy when=
 VPN is to<BR>
&gt; &gt; be established on-demand between two user devices without any pre=
vious<BR>
&gt; &gt; affiliation.<BR>
&gt; &gt;<BR>
&gt; &gt; As a result current deployment based on &quot;draft-saito-mmusic-=
sdp-<BR>
&gt; &gt; ike-04&quot;<BR>
&gt; &gt; and other standard bodies are looking at the use of SIP for the<B=
R>
&gt; &gt; following reasons.<BR>
&gt; &gt;<BR>
&gt; &gt; (1)Name resolution can be achieved by using REGISTER method and S=
DP<BR>
&gt; &gt; negotiation<BR>
&gt; &gt; (2)NAT traversal and hole-punching of firewall can be achieved by=
<BR>
&gt; &gt; applying SIP+ICE.<BR>
&gt; &gt; (3)Authentication/authorization can be achieved by having trusted=
 SIP<BR>
&gt; &gt; proxy and user friendly SIP-URIs.<BR>
&gt; &gt;<BR>
&gt; &gt; SIP/SDP will be used to negotiate parameters necessary to establi=
sh<BR>
&gt; &gt; IKE/IPsec such as IP address, port numbers and authentication<BR>
&gt; &gt; information of peers. Current deployment uses SIP also to manage =
IPsec<BR>
&gt; &gt; connection, e.g. use BYE to shut down media sessions for IKE/IPse=
c,<BR>
&gt; &gt; however, further discussions can be made to find out if there are=
<BR>
&gt; &gt; intrests or use cases in using SIP only as a rendezvous protocol.=
<BR>
&gt; &gt;<BR>
&gt; &gt; Use-cases that may see use for such standard effort;<BR>
&gt; &gt; &nbsp;- Media sharing using DLNA or similar protocol over VPN bet=
ween 2<BR>
&gt; &gt; users' devices.<BR>
&gt; &gt; &nbsp;- Remote desktop sharing for Customer service over VPN init=
iated via<BR>
&gt; &gt; SIP call.<BR>
&gt; &gt; &nbsp;&nbsp;* Similar to click to call, customer service agent ca=
n access user's<BR>
&gt; &gt; pc remotely to troubleshoot the problem customer is facing from t=
he <BR>
&gt; &gt; call.<BR>
&gt; &gt; &nbsp;- Accessing medical equipments(medical robotics) remotely a=
nd<BR>
&gt; &gt; possibly controlling medical equipments to monitor elders in a ru=
ral<BR>
&gt; &gt; area that (remote care services).<BR>
&gt; &gt; &nbsp;- LAN based gaming protocol to be established peer to peer =
rather<BR>
&gt; &gt; than via gaming server.<BR>
&gt; &gt;<BR>
&gt; &gt;<BR>
&gt; &gt; Discussion Point:<BR>
&gt; &gt; - To see interests in documenting what's being deployed based on<=
BR>
&gt; &gt; &quot;draft-saito-mmusic-sdp-ike-04&quot; as an informational doc=
ument.<BR>
&gt; &gt; <a href=3D"http://tools.ietf.org/html/draft-saito-mmusic-sdp-ike-=
04">http://tools.ietf.org/html/draft-saito-mmusic-sdp-ike-04</a><BR>
&gt; &gt; (to be minor-updated to -05 to conform with this charter proposal=
)<BR>
&gt; &gt; - To see interests in standardizing toolset for realizing use-cas=
es<BR>
&gt; &gt; similar to what's mentioned above, even contemplating the possibi=
lity<BR>
&gt; &gt; of defining a guideline for SIP usage where SIP is used only as a=
<BR>
&gt; &gt; rendezvous protocol.<BR>
&gt; &gt;<BR>
&gt; &gt;<BR>
&gt; &gt; Milestone/Deliverables:<BR>
&gt; &gt; - Aug 09 Issue Internet-Draft<BR>
&gt; &gt; &nbsp;&nbsp;&nbsp;(by updating &quot;draft-saito-mmusic-sdp-ike-0=
5&quot;)<BR>
&gt; &gt; - Nov 09 Submit I-D to IESG for publication as an RFC (Informatio=
nal)<BR>
&gt; &gt;<BR>
&gt; &gt; --------------------------------------------------<BR>
&gt; &gt; Best Regards,<BR>
&gt; &gt; Shintaro<BR>
&gt; &gt;<BR>
&gt; &gt; ----------------------------------------<BR>
&gt; &gt; Shintaro MIZUNO<BR>
&gt; &gt; <a href=3D"mizuno.shintaro@lab.ntt.co.jp">mizuno.shintaro@lab.ntt=
.co.jp</a><BR>
&gt; &gt;<BR>
&gt; &gt; NTT Information Sharing Platform Laboratories<BR>
&gt; &gt; NTT Corporation<BR>
&gt; &gt; ----------------------------------------<BR>
&gt; &gt;<BR>
&gt; &gt; _______________________________________________<BR>
&gt; &gt; dispatch mailing list<BR>
&gt; &gt; <a href=3D"dispatch@ietf.org">dispatch@ietf.org</a><BR>
&gt; &gt; <a href=3D"https://www.ietf.org/mailman/listinfo/dispatch">https:=
//www.ietf.org/mailman/listinfo/dispatch</a><BR>
&gt;<BR>
&gt;<BR>
<BR>
<BR>
----------------------------------------<BR>
Shintaro MIZUNO<BR>
<a href=3D"mizuno.shintaro@lab.ntt.co.jp">mizuno.shintaro@lab.ntt.co.jp</a>=
<BR>
<BR>
NTT Information Sharing Platform Laboratories<BR>
NTT Corporation<BR>
----------------------------------------<BR>
<BR>
_______________________________________________<BR>
dispatch mailing list<BR>
<a href=3D"dispatch@ietf.org">dispatch@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf=
.org/mailman/listinfo/dispatch</a><BR>
<BR>
</SPAN></FONT></BLOCKQUOTE>
</BODY>
</HTML>


--_000_C653CE783FF1hsinnreiadobecom_--

From vkg@alcatel-lucent.com  Tue Jun  9 14:01:41 2009
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Date: Tue, 09 Jun 2009 16:01:38 -0500
From: "Vijay K. Gurbani" <vkg@alcatel-lucent.com>
Organization: Bell Labs Security Technology Research Group
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Cc: Leon Portman <Leon.Portman@nice.com>, "Jain, Rajnish" <Rajnish.Jain@ipc.com>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: [dispatch] Session recording in SIP
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Hi: I realize that the deadline for charter proposals
was yesterday, but I hope that it is not too late to submit
one more.

A few interested people (Hadriel Kaplan, Dan Wing, Rajnish Jain,
Leon Portman, Andrew Hutton and I) have been interested in
RTP session recording in SIP.  The requirement draft will
be released shortly.

We would like to request agenda time in dispatch to propose
the formation of a new working group to define protocol
extensions and an architecture for RTP recording.

Session recording in SIP
Mailing Lists: TBD
Chairs: TBD
Area Directorate: Real Time Applications (RAI)

Purpose:

Session recording is a critical operational requirement in many
businesses, especially where voice is used as a medium for
commerce and customer support. A prime example where voice is used
for trade is the financial industry. The call recording requirements
in this industry are quite stringent. The recorded calls are used
for dispute resolution and regulatory compliance. Other businesses
such as customer support call centers typically employ call
recording for quality control or business analytics.

Depending on the country and its regulatory requirements, financial
trading floors typically must record all calls. The recorded media
content must be an exact copy of the actual conversation (i.e.
clipping and loss of media are unacceptable).  Some deployments
and regulations require that calls be aborted or rejected if
the recording device is unavailable.

This group will specify requirements for a SIP based protocol
interface between a communications system and a recorder. The
Communications System is responsible for establishing media sessions
where the actual business is conducted. The Recorder is the sink of
the recorded media.

The recorded sessions can be of any kind such as voice, video and
instant messaging. A recorded session is typically comprised of
actual media content and the call metadata. The call metadata allows
recording archives to be searched and filtered at a later time.
The conveyance of call metadata from the communications system
to the recorder is outside the scope of this document.

This group will only looks into active recording, where the recorded
system purposefully streams media to a recording device. Passive
recording, where a recording device detects media directly from
the network, is outside the scope of this document. In addition,
lawful intercept is outside the scope of the group.

Proposed deliverables:

1) Requirements document;
2) Solutions document, including reference architecture.

Thanks,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/

From dromasca@avaya.com  Wed Jun 10 02:23:42 2009
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Thread-Topic: [dispatch] Session recording in SIP
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References: <4A2ECDB2.7000601@alcatel-lucent.com>
From: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
To: "Vijay K. Gurbani" <vkg@alcatel-lucent.com>, <dispatch@ietf.org>
Cc: Leon Portman <Leon.Portman@nice.com>, "Jain, Rajnish" <Rajnish.Jain@ipc.com>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [dispatch] Session recording in SIP
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I believe that this is an important application, and I support doing
work in this direction.=20

Three comments on the preliminary words:=20

1. The requirement is for both signaling and media recording, right?
Probably good to say it.=20
2. Security and regulations on how the information is accessed and
protected are of high importance. They would probably be reflected in
the requirements, but explicit wording in the future charter can help.
3. I suggest that we look at the IPFIX protocol as a possible technology
to re-use

Dan
=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Vijay K. Gurbani
> Sent: Wednesday, June 10, 2009 12:02 AM
> To: dispatch@ietf.org
> Cc: Leon Portman; Jain,Rajnish; Hadriel Kaplan
> Subject: [dispatch] Session recording in SIP
>=20
> Hi: I realize that the deadline for charter proposals was=20
> yesterday, but I hope that it is not too late to submit one more.
>=20
> A few interested people (Hadriel Kaplan, Dan Wing, Rajnish=20
> Jain, Leon Portman, Andrew Hutton and I) have been interested=20
> in RTP session recording in SIP.  The requirement draft will=20
> be released shortly.
>=20
> We would like to request agenda time in dispatch to propose=20
> the formation of a new working group to define protocol=20
> extensions and an architecture for RTP recording.
>=20
> Session recording in SIP
> Mailing Lists: TBD
> Chairs: TBD
> Area Directorate: Real Time Applications (RAI)
>=20
> Purpose:
>=20
> Session recording is a critical operational requirement in=20
> many businesses, especially where voice is used as a medium=20
> for commerce and customer support. A prime example where=20
> voice is used for trade is the financial industry. The call=20
> recording requirements in this industry are quite stringent.=20
> The recorded calls are used for dispute resolution and=20
> regulatory compliance. Other businesses such as customer=20
> support call centers typically employ call recording for=20
> quality control or business analytics.
>=20
> Depending on the country and its regulatory requirements,=20
> financial trading floors typically must record all calls. The=20
> recorded media content must be an exact copy of the actual=20
> conversation (i.e.
> clipping and loss of media are unacceptable).  Some=20
> deployments and regulations require that calls be aborted or=20
> rejected if the recording device is unavailable.
>=20
> This group will specify requirements for a SIP based protocol=20
> interface between a communications system and a recorder. The=20
> Communications System is responsible for establishing media=20
> sessions where the actual business is conducted. The Recorder=20
> is the sink of the recorded media.
>=20
> The recorded sessions can be of any kind such as voice, video=20
> and instant messaging. A recorded session is typically=20
> comprised of actual media content and the call metadata. The=20
> call metadata allows recording archives to be searched and=20
> filtered at a later time.
> The conveyance of call metadata from the communications=20
> system to the recorder is outside the scope of this document.
>=20
> This group will only looks into active recording, where the=20
> recorded system purposefully streams media to a recording=20
> device. Passive recording, where a recording device detects=20
> media directly from the network, is outside the scope of this=20
> document. In addition, lawful intercept is outside the scope=20
> of the group.
>=20
> Proposed deliverables:
>=20
> 1) Requirements document;
> 2) Solutions document, including reference architecture.
>=20
> Thanks,
>=20
> - vijay
> --
> Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent 1960=20
> Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
> Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
> Web:   http://ect.bell-labs.com/who/vkg/
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

From Leon.Portman@nice.com  Wed Jun 10 05:42:07 2009
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From: Leon Portman <Leon.Portman@nice.com>
To: "Romascanu, Dan (Dan)" <dromasca@avaya.com>, "Vijay K. Gurbani" <vkg@alcatel-lucent.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Wed, 10 Jun 2009 15:42:07 +0300
Thread-Topic: [dispatch] Session recording in SIP
Thread-Index: AcnpRYT1lltupJoVRoye5fl5+wRurwAZfn1gAAcXUbA=
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Cc: "Jain, Rajnish" <Rajnish.Jain@ipc.com>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [dispatch] Session recording in SIP
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Dan, Hi

Thank you for the comments.

Please see below

Leon

-----Original Message-----
From: Romascanu, Dan (Dan) [mailto:dromasca@avaya.com]=20
Sent: Wednesday, June 10, 2009 12:24 PM
To: Vijay K. Gurbani; dispatch@ietf.org
Cc: Leon Portman; Jain,Rajnish; Hadriel Kaplan
Subject: RE: [dispatch] Session recording in SIP

I believe that this is an important application, and I support doing
work in this direction.=20

Three comments on the preliminary words:=20

1. The requirement is for both signaling and media recording, right?
Probably good to say it.=20
[LeonP] Actually we want to separate into two separate documents: Media for=
king invocation (via SIP invite ) and SIP Event package for signaling repor=
ting

2. Security and regulations on how the information is accessed and
protected are of high importance. They would probably be reflected in
the requirements, but explicit wording in the future charter can help.
[LeonP] Currently it was out of scope however we can discuss it
3. I suggest that we look at the IPFIX protocol as a possible technology
to re-use
[LeonP] OK, we will

Dan
=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Vijay K. Gurbani
> Sent: Wednesday, June 10, 2009 12:02 AM
> To: dispatch@ietf.org
> Cc: Leon Portman; Jain,Rajnish; Hadriel Kaplan
> Subject: [dispatch] Session recording in SIP
>=20
> Hi: I realize that the deadline for charter proposals was=20
> yesterday, but I hope that it is not too late to submit one more.
>=20
> A few interested people (Hadriel Kaplan, Dan Wing, Rajnish=20
> Jain, Leon Portman, Andrew Hutton and I) have been interested=20
> in RTP session recording in SIP.  The requirement draft will=20
> be released shortly.
>=20
> We would like to request agenda time in dispatch to propose=20
> the formation of a new working group to define protocol=20
> extensions and an architecture for RTP recording.
>=20
> Session recording in SIP
> Mailing Lists: TBD
> Chairs: TBD
> Area Directorate: Real Time Applications (RAI)
>=20
> Purpose:
>=20
> Session recording is a critical operational requirement in=20
> many businesses, especially where voice is used as a medium=20
> for commerce and customer support. A prime example where=20
> voice is used for trade is the financial industry. The call=20
> recording requirements in this industry are quite stringent.=20
> The recorded calls are used for dispute resolution and=20
> regulatory compliance. Other businesses such as customer=20
> support call centers typically employ call recording for=20
> quality control or business analytics.
>=20
> Depending on the country and its regulatory requirements,=20
> financial trading floors typically must record all calls. The=20
> recorded media content must be an exact copy of the actual=20
> conversation (i.e.
> clipping and loss of media are unacceptable).  Some=20
> deployments and regulations require that calls be aborted or=20
> rejected if the recording device is unavailable.
>=20
> This group will specify requirements for a SIP based protocol=20
> interface between a communications system and a recorder. The=20
> Communications System is responsible for establishing media=20
> sessions where the actual business is conducted. The Recorder=20
> is the sink of the recorded media.
>=20
> The recorded sessions can be of any kind such as voice, video=20
> and instant messaging. A recorded session is typically=20
> comprised of actual media content and the call metadata. The=20
> call metadata allows recording archives to be searched and=20
> filtered at a later time.
> The conveyance of call metadata from the communications=20
> system to the recorder is outside the scope of this document.
>=20
> This group will only looks into active recording, where the=20
> recorded system purposefully streams media to a recording=20
> device. Passive recording, where a recording device detects=20
> media directly from the network, is outside the scope of this=20
> document. In addition, lawful intercept is outside the scope=20
> of the group.
>=20
> Proposed deliverables:
>=20
> 1) Requirements document;
> 2) Solutions document, including reference architecture.
>=20
> Thanks,
>=20
> - vijay
> --
> Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent 1960=20
> Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
> Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
> Web:   http://ect.bell-labs.com/who/vkg/
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

From alan@sipstation.com  Wed Jun 10 06:42:29 2009
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Cc: Leon Portman <Leon.Portman@nice.com>, dispatch@ietf.org, "Jain, Rajnish" <Rajnish.Jain@ipc.com>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [dispatch] Session recording in SIP
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This is important work to be done in RAI.  Key management for secure 
recording is an important component that needs to be addressed.

draft-wing-sipping-srtp-key-04 addresses many of these topics.

Thanks,
Alan


Romascanu, Dan (Dan) wrote:
> I believe that this is an important application, and I support doing
> work in this direction. 
>
> Three comments on the preliminary words: 
>
> 1. The requirement is for both signaling and media recording, right?
> Probably good to say it. 
> 2. Security and regulations on how the information is accessed and
> protected are of high importance. They would probably be reflected in
> the requirements, but explicit wording in the future charter can help.
> 3. I suggest that we look at the IPFIX protocol as a possible technology
> to re-use
>
> Dan
>  
>
>   
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org 
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Vijay K. Gurbani
>> Sent: Wednesday, June 10, 2009 12:02 AM
>> To: dispatch@ietf.org
>> Cc: Leon Portman; Jain,Rajnish; Hadriel Kaplan
>> Subject: [dispatch] Session recording in SIP
>>
>> Hi: I realize that the deadline for charter proposals was 
>> yesterday, but I hope that it is not too late to submit one more.
>>
>> A few interested people (Hadriel Kaplan, Dan Wing, Rajnish 
>> Jain, Leon Portman, Andrew Hutton and I) have been interested 
>> in RTP session recording in SIP.  The requirement draft will 
>> be released shortly.
>>
>> We would like to request agenda time in dispatch to propose 
>> the formation of a new working group to define protocol 
>> extensions and an architecture for RTP recording.
>>
>> Session recording in SIP
>> Mailing Lists: TBD
>> Chairs: TBD
>> Area Directorate: Real Time Applications (RAI)
>>
>> Purpose:
>>
>> Session recording is a critical operational requirement in 
>> many businesses, especially where voice is used as a medium 
>> for commerce and customer support. A prime example where 
>> voice is used for trade is the financial industry. The call 
>> recording requirements in this industry are quite stringent. 
>> The recorded calls are used for dispute resolution and 
>> regulatory compliance. Other businesses such as customer 
>> support call centers typically employ call recording for 
>> quality control or business analytics.
>>
>> Depending on the country and its regulatory requirements, 
>> financial trading floors typically must record all calls. The 
>> recorded media content must be an exact copy of the actual 
>> conversation (i.e.
>> clipping and loss of media are unacceptable).  Some 
>> deployments and regulations require that calls be aborted or 
>> rejected if the recording device is unavailable.
>>
>> This group will specify requirements for a SIP based protocol 
>> interface between a communications system and a recorder. The 
>> Communications System is responsible for establishing media 
>> sessions where the actual business is conducted. The Recorder 
>> is the sink of the recorded media.
>>
>> The recorded sessions can be of any kind such as voice, video 
>> and instant messaging. A recorded session is typically 
>> comprised of actual media content and the call metadata. The 
>> call metadata allows recording archives to be searched and 
>> filtered at a later time.
>> The conveyance of call metadata from the communications 
>> system to the recorder is outside the scope of this document.
>>
>> This group will only looks into active recording, where the 
>> recorded system purposefully streams media to a recording 
>> device. Passive recording, where a recording device detects 
>> media directly from the network, is outside the scope of this 
>> document. In addition, lawful intercept is outside the scope 
>> of the group.
>>
>> Proposed deliverables:
>>
>> 1) Requirements document;
>> 2) Solutions document, including reference architecture.
>>
>> Thanks,
>>
>> - vijay
>> --
>> Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent 1960 
>> Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
>> Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
>> Web:   http://ect.bell-labs.com/who/vkg/
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
>>     
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>
>   


From Even.roni@huawei.com  Wed Jun 10 09:19:10 2009
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Date: Wed, 10 Jun 2009 19:18:20 +0300
From: Roni Even <Even.roni@huawei.com>
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To: 'Alan Johnston' <alan@sipstation.com>, "'Romascanu, Dan (Dan)'" <dromasca@avaya.com>
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Cc: 'Leon Portman' <Leon.Portman@nice.com>, dispatch@ietf.org, "'Jain, Rajnish'" <Rajnish.Jain@ipc.com>, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [dispatch] Session recording in SIP
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Hi,
I agree that this is an important work but I am not sure about a WG. This
may be done in XCON and AVT.
Note that the recording is not only for point to point call but at least for
three parties like in call center scenarios

I also did not notice any mention of the playback part but it should also be
addressed

Roni Even

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
Of Alan Johnston
Sent: Wednesday, June 10, 2009 4:42 PM
To: Romascanu, Dan (Dan)
Cc: Leon Portman; dispatch@ietf.org; Jain, Rajnish; Hadriel Kaplan
Subject: Re: [dispatch] Session recording in SIP

This is important work to be done in RAI.  Key management for secure 
recording is an important component that needs to be addressed.

draft-wing-sipping-srtp-key-04 addresses many of these topics.

Thanks,
Alan


Romascanu, Dan (Dan) wrote:
> I believe that this is an important application, and I support doing
> work in this direction. 
>
> Three comments on the preliminary words: 
>
> 1. The requirement is for both signaling and media recording, right?
> Probably good to say it. 
> 2. Security and regulations on how the information is accessed and
> protected are of high importance. They would probably be reflected in
> the requirements, but explicit wording in the future charter can help.
> 3. I suggest that we look at the IPFIX protocol as a possible technology
> to re-use
>
> Dan
>  
>
>   
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org 
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Vijay K. Gurbani
>> Sent: Wednesday, June 10, 2009 12:02 AM
>> To: dispatch@ietf.org
>> Cc: Leon Portman; Jain,Rajnish; Hadriel Kaplan
>> Subject: [dispatch] Session recording in SIP
>>
>> Hi: I realize that the deadline for charter proposals was 
>> yesterday, but I hope that it is not too late to submit one more.
>>
>> A few interested people (Hadriel Kaplan, Dan Wing, Rajnish 
>> Jain, Leon Portman, Andrew Hutton and I) have been interested 
>> in RTP session recording in SIP.  The requirement draft will 
>> be released shortly.
>>
>> We would like to request agenda time in dispatch to propose 
>> the formation of a new working group to define protocol 
>> extensions and an architecture for RTP recording.
>>
>> Session recording in SIP
>> Mailing Lists: TBD
>> Chairs: TBD
>> Area Directorate: Real Time Applications (RAI)
>>
>> Purpose:
>>
>> Session recording is a critical operational requirement in 
>> many businesses, especially where voice is used as a medium 
>> for commerce and customer support. A prime example where 
>> voice is used for trade is the financial industry. The call 
>> recording requirements in this industry are quite stringent. 
>> The recorded calls are used for dispute resolution and 
>> regulatory compliance. Other businesses such as customer 
>> support call centers typically employ call recording for 
>> quality control or business analytics.
>>
>> Depending on the country and its regulatory requirements, 
>> financial trading floors typically must record all calls. The 
>> recorded media content must be an exact copy of the actual 
>> conversation (i.e.
>> clipping and loss of media are unacceptable).  Some 
>> deployments and regulations require that calls be aborted or 
>> rejected if the recording device is unavailable.
>>
>> This group will specify requirements for a SIP based protocol 
>> interface between a communications system and a recorder. The 
>> Communications System is responsible for establishing media 
>> sessions where the actual business is conducted. The Recorder 
>> is the sink of the recorded media.
>>
>> The recorded sessions can be of any kind such as voice, video 
>> and instant messaging. A recorded session is typically 
>> comprised of actual media content and the call metadata. The 
>> call metadata allows recording archives to be searched and 
>> filtered at a later time.
>> The conveyance of call metadata from the communications 
>> system to the recorder is outside the scope of this document.
>>
>> This group will only looks into active recording, where the 
>> recorded system purposefully streams media to a recording 
>> device. Passive recording, where a recording device detects 
>> media directly from the network, is outside the scope of this 
>> document. In addition, lawful intercept is outside the scope 
>> of the group.
>>
>> Proposed deliverables:
>>
>> 1) Requirements document;
>> 2) Solutions document, including reference architecture.
>>
>> Thanks,
>>
>> - vijay
>> --
>> Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent 1960 
>> Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
>> Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
>> Web:   http://ect.bell-labs.com/who/vkg/
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
>>     
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>
>   

_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch


From dworley@nortel.com  Wed Jun 10 13:36:43 2009
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From: "Dale Worley" <dworley@nortel.com>
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Subject: [dispatch] Getting discussion going on draft-ietf-sipping-profile-datasets-03
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This is the list of discussion points that I've assembled.  The first 5
are from the last IETF, and the last 3 are ones that have come to mind
when I reviewed the draft:

DP 1:  Merging rules are specified in the schema.

Everyone agrees that the merging rules for each property are to be
specified in the profile schema.  The only interesting merge rule now
used is "intersection", for combining values that are sets of an
underlying data type.

DP 2:  Devices containing sub-devices that are separately configured

The device will obtain one device profile and one local network
profile.  The device profile will specify the set of users, each user
implying one sub-device.  Each sub-device obtains its own user
profile.

DP 3:  Relax NG schema

Everyone prefers Relax NG schema language to XML schema language.

DP 4:  Versioning

A schema is extended by defining a new schema whose properties augment
or supersede the properties of the original schema (in a way defined
in the new schema).  This method avoids extending the set of values of
an existing property.  It also makes clear how the policy server
intends to configure a device that does not support the extension:
Such a device uses the dataset of the original schema.

DP 5:  Visibility

Each property in a dataset can specify its visibility separately.  We
need to merge visibility values to determine the visibility in the
merged "working profile".  Should the working visibility be copied
from the visibility of the value that was selected, or should the
working visibility be the AND of the visibility valuses?

DP 6:  The policy attributes

sipping-profile-datasets contains a rather messy mechanism for
specifying sets of values, tagging each value as "allowed" or
"disallowed", and then providing a policy for values that are not
mentioned.  It seems to me that this could be made much simpler by
having a mechanism that specifies either sets of allowed values or
sets of disallowed values, along these lines:

   <datum-name>
      <set>
         <value>1</value>
         <value>2</value>
         <value>3</value>
      </set>
   </datum-name>

   <datum-name2>
      <exclude>
         <value>1</value>
         <value>2</value>
         <value>3</value>
      </exclude>
   </datum-name2>

As long as we have an equality test for base type (that is, the
strings that can appear in the <value> elements), <set> and <exclude>
can be intersected and only generate <set> and <exclude> results.
(That is important because of the "intersection" merge rule.)  And we
assume that the base type is specified via Relax NG, which provides
such an equality test.

DP 7:  The direction attribute

sipping-profile-datasets provides for tagging values with a
"direction" attribute, which has values recvonly, sendonly, and
sendrecv.  The implication is that each property has two values, one
for incoming traffic and one for outgoing traffic.  Is this a good
model, since some values aren't specific for traffic direction?

DP 8:  The q attribute

sipping-profile-datasets provides a "q" attribute that can be
specified for elements of sets, so that the choices from a set can be
prioritized.  Is this facility needed?  And if we keep it, how are q
values to be combined when values are merged?

Dale



From wwwrun@core3.amsl.com  Wed Jun 10 14:13:37 2009
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Cc: dispatch mailing list <dispatch@ietf.org>, dispatch chair <dispatch-chairs@tools.ietf.org>, Internet Architecture Board <iab@iab.org>, RFC Editor <rfc-editor@rfc-editor.org>
Subject: [dispatch] Protocol Action: 'Managing Client Initiated Connections in the Session Initiation Protocol (SIP)' to Proposed Standard
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The IESG has approved the following document:

- 'Managing Client Initiated Connections in the Session Initiation 
   Protocol (SIP) '
   <draft-ietf-sip-outbound-20.txt> as a Proposed Standard

This document is the product of the Dispatch Working Group. 

The IESG contact persons are Robert Sparks and Cullen Jennings.

A URL of this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-20.txt

Technical Summary

   This document defines an extension to the Session Initiation Protocol
   (SIP) that provides for persistent and reusable connections between
   SIP User Agents and SIP Proxy Servers. In particular, this allows
   proxy servers to initiate TCP connections or to send asynchronous UDP
   datagrams to User Agents in order to deliver requests.  However, in a
   large number of real deployments, many practical considerations, such
   as the existence of firewalls and Network Address Translators (NATs)
   or the use of TLS with server-provided certificates, prevent servers
   from connecting to User Agents in this way.  This specification
   defines behaviors for User Agents, registrars and proxy servers that
   allow requests to be delivered on existing connections established by
   the User Agent.  It also defines keep alive behaviors needed to keep
   NAT bindings open and specifies the usage of multiple connections from
   the User Agent to its Registrar.

Working Group Summary

   The working group process on this document was exceptionally long. The


   first WG version of the draft appeared in the summer of 2005. Working
   group last call initiated in the summer of 2006 and extended until the
   summer of 2008, requiring several iterations of the draft and the
   assignment of Francois Audet as a "process champion" for the draft
   within the working group. Most delays seem to have been related to
   slow cycle time on the part of the authors, but the process was also
   delayed by a number of changes occurring during the review cycle.

   Particular sticking points included the keepalive mechanism and a
   requirement for binding to multiple outbound proxies if so
   configured. Both were eventually resolved by a widely-accepted
   compromises. Additional work is being pursued in SIPCORE to allow
   the use of the keepalive mechanism independently of the outbound
   mechanism. 

Document Quality

  There have been implementations of previous versions of this 
  draft testing at SIPit events since 2005.

Personnel

   Dean Willis is the document shepherd with significant assistance
   from Francois Audet. Robert Sparks is the responsible area director.


From Even.roni@huawei.com  Wed Jun 10 14:28:57 2009
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Date: Thu, 11 Jun 2009 00:28:07 +0300
From: Roni Even <Even.roni@huawei.com>
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Cc: 'Leon Portman' <Leon.Portman@nice.com>, dispatch@ietf.org, "'Jain, Rajnish'" <Rajnish.Jain@ipc.com>, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [dispatch] Session recording in SIP
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Hi,
I agree that this is an important work but I am not sure about a WG. This
may be done in XCON and AVT.
Note that the recording is not only for point to point call but at least for
three parties like in call center scenarios

I also did not notice any mention of the playback part but it should also be
addressed

Roni Even

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
Of Alan Johnston
Sent: Wednesday, June 10, 2009 4:42 PM
To: Romascanu, Dan (Dan)
Cc: Leon Portman; dispatch@ietf.org; Jain, Rajnish; Hadriel Kaplan
Subject: Re: [dispatch] Session recording in SIP

This is important work to be done in RAI.  Key management for secure 
recording is an important component that needs to be addressed.

draft-wing-sipping-srtp-key-04 addresses many of these topics.

Thanks,
Alan


Romascanu, Dan (Dan) wrote:
> I believe that this is an important application, and I support doing
> work in this direction. 
>
> Three comments on the preliminary words: 
>
> 1. The requirement is for both signaling and media recording, right?
> Probably good to say it. 
> 2. Security and regulations on how the information is accessed and
> protected are of high importance. They would probably be reflected in
> the requirements, but explicit wording in the future charter can help.
> 3. I suggest that we look at the IPFIX protocol as a possible technology
> to re-use
>
> Dan
>  
>
>   
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org 
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Vijay K. Gurbani
>> Sent: Wednesday, June 10, 2009 12:02 AM
>> To: dispatch@ietf.org
>> Cc: Leon Portman; Jain,Rajnish; Hadriel Kaplan
>> Subject: [dispatch] Session recording in SIP
>>
>> Hi: I realize that the deadline for charter proposals was 
>> yesterday, but I hope that it is not too late to submit one more.
>>
>> A few interested people (Hadriel Kaplan, Dan Wing, Rajnish 
>> Jain, Leon Portman, Andrew Hutton and I) have been interested 
>> in RTP session recording in SIP.  The requirement draft will 
>> be released shortly.
>>
>> We would like to request agenda time in dispatch to propose 
>> the formation of a new working group to define protocol 
>> extensions and an architecture for RTP recording.
>>
>> Session recording in SIP
>> Mailing Lists: TBD
>> Chairs: TBD
>> Area Directorate: Real Time Applications (RAI)
>>
>> Purpose:
>>
>> Session recording is a critical operational requirement in 
>> many businesses, especially where voice is used as a medium 
>> for commerce and customer support. A prime example where 
>> voice is used for trade is the financial industry. The call 
>> recording requirements in this industry are quite stringent. 
>> The recorded calls are used for dispute resolution and 
>> regulatory compliance. Other businesses such as customer 
>> support call centers typically employ call recording for 
>> quality control or business analytics.
>>
>> Depending on the country and its regulatory requirements, 
>> financial trading floors typically must record all calls. The 
>> recorded media content must be an exact copy of the actual 
>> conversation (i.e.
>> clipping and loss of media are unacceptable).  Some 
>> deployments and regulations require that calls be aborted or 
>> rejected if the recording device is unavailable.
>>
>> This group will specify requirements for a SIP based protocol 
>> interface between a communications system and a recorder. The 
>> Communications System is responsible for establishing media 
>> sessions where the actual business is conducted. The Recorder 
>> is the sink of the recorded media.
>>
>> The recorded sessions can be of any kind such as voice, video 
>> and instant messaging. A recorded session is typically 
>> comprised of actual media content and the call metadata. The 
>> call metadata allows recording archives to be searched and 
>> filtered at a later time.
>> The conveyance of call metadata from the communications 
>> system to the recorder is outside the scope of this document.
>>
>> This group will only looks into active recording, where the 
>> recorded system purposefully streams media to a recording 
>> device. Passive recording, where a recording device detects 
>> media directly from the network, is outside the scope of this 
>> document. In addition, lawful intercept is outside the scope 
>> of the group.
>>
>> Proposed deliverables:
>>
>> 1) Requirements document;
>> 2) Solutions document, including reference architecture.
>>
>> Thanks,
>>
>> - vijay
>> --
>> Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent 1960 
>> Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
>> Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
>> Web:   http://ect.bell-labs.com/who/vkg/
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
>>     
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>
>   

_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch



From rjsparks@nostrum.com  Wed Jun 10 14:32:56 2009
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Subject: Re: [dispatch] Protocol Action: 'Managing Client Initiated Connections in the Session Initiation Protocol (SIP)' to Proposed Standard
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This is, of course, a product of the SIP working group.
I've asked the secretariat to issue a correction.

RjS

On Jun 10, 2009, at 4:13 PM, The IESG wrote:

> The IESG has approved the following document:
>
> - 'Managing Client Initiated Connections in the Session Initiation
>   Protocol (SIP) '
>   <draft-ietf-sip-outbound-20.txt> as a Proposed Standard
>
> This document is the product of the Dispatch Working Group.
>
> The IESG contact persons are Robert Sparks and Cullen Jennings.
>
> A URL of this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-20.txt
>
> Technical Summary
>
>   This document defines an extension to the Session Initiation  
> Protocol
>   (SIP) that provides for persistent and reusable connections between
>   SIP User Agents and SIP Proxy Servers. In particular, this allows
>   proxy servers to initiate TCP connections or to send asynchronous  
> UDP
>   datagrams to User Agents in order to deliver requests.  However,  
> in a
>   large number of real deployments, many practical considerations,  
> such
>   as the existence of firewalls and Network Address Translators (NATs)
>   or the use of TLS with server-provided certificates, prevent servers
>   from connecting to User Agents in this way.  This specification
>   defines behaviors for User Agents, registrars and proxy servers that
>   allow requests to be delivered on existing connections established  
> by
>   the User Agent.  It also defines keep alive behaviors needed to keep
>   NAT bindings open and specifies the usage of multiple connections  
> from
>   the User Agent to its Registrar.
>
> Working Group Summary
>
>   The working group process on this document was exceptionally long.  
> The
>
>
>   first WG version of the draft appeared in the summer of 2005.  
> Working
>   group last call initiated in the summer of 2006 and extended until  
> the
>   summer of 2008, requiring several iterations of the draft and the
>   assignment of Francois Audet as a "process champion" for the draft
>   within the working group. Most delays seem to have been related to
>   slow cycle time on the part of the authors, but the process was also
>   delayed by a number of changes occurring during the review cycle.
>
>   Particular sticking points included the keepalive mechanism and a
>   requirement for binding to multiple outbound proxies if so
>   configured. Both were eventually resolved by a widely-accepted
>   compromises. Additional work is being pursued in SIPCORE to allow
>   the use of the keepalive mechanism independently of the outbound
>   mechanism.
>
> Document Quality
>
>  There have been implementations of previous versions of this
>  draft testing at SIPit events since 2005.
>
> Personnel
>
>   Dean Willis is the document shepherd with significant assistance
>   from Francois Audet. Robert Sparks is the responsible area director.
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


From jmpolk@cisco.com  Wed Jun 10 14:59:52 2009
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To: Roni Even <Even.roni@huawei.com>, "'Alan Johnston'" <alan@sipstation.com>,  "'Romascanu, Dan (Dan)'" <dromasca@avaya.com>
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Subject: Re: [dispatch] Session recording in SIP
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Hey

I agree with Roni about this probably being best suited for XCON, 
after all, any point to point recording is likely (merely) adding a 
3rd party (the recorder) to the session.

This brings up probably the most glaring whole (in my mind) that is 
not mentioned below - the ability to have a recorder without at least 
one side knowing it's there.  Not every time, and in fact probably 
most times, you don't want the other person knowing you are recording 
them. It's likely more of a case of not explicitly asking the called 
party "can I record this conversation", it just happens.

James

At 11:18 AM 6/10/2009, Roni Even wrote:
>Hi,
>I agree that this is an important work but I am not sure about a WG. This
>may be done in XCON and AVT.
>Note that the recording is not only for point to point call but at least for
>three parties like in call center scenarios
>
>I also did not notice any mention of the playback part but it should also be
>addressed
>
>Roni Even
>
>-----Original Message-----
>From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
>Of Alan Johnston
>Sent: Wednesday, June 10, 2009 4:42 PM
>To: Romascanu, Dan (Dan)
>Cc: Leon Portman; dispatch@ietf.org; Jain, Rajnish; Hadriel Kaplan
>Subject: Re: [dispatch] Session recording in SIP
>
>This is important work to be done in RAI.  Key management for secure
>recording is an important component that needs to be addressed.
>
>draft-wing-sipping-srtp-key-04 addresses many of these topics.
>
>Thanks,
>Alan
>
>
>Romascanu, Dan (Dan) wrote:
> > I believe that this is an important application, and I support doing
> > work in this direction.
> >
> > Three comments on the preliminary words:
> >
> > 1. The requirement is for both signaling and media recording, right?
> > Probably good to say it.
> > 2. Security and regulations on how the information is accessed and
> > protected are of high importance. They would probably be reflected in
> > the requirements, but explicit wording in the future charter can help.
> > 3. I suggest that we look at the IPFIX protocol as a possible technology
> > to re-use
> >
> > Dan
> >
> >
> >
> >> -----Original Message-----
> >> From: dispatch-bounces@ietf.org
> >> [mailto:dispatch-bounces@ietf.org] On Behalf Of Vijay K. Gurbani
> >> Sent: Wednesday, June 10, 2009 12:02 AM
> >> To: dispatch@ietf.org
> >> Cc: Leon Portman; Jain,Rajnish; Hadriel Kaplan
> >> Subject: [dispatch] Session recording in SIP
> >>
> >> Hi: I realize that the deadline for charter proposals was
> >> yesterday, but I hope that it is not too late to submit one more.
> >>
> >> A few interested people (Hadriel Kaplan, Dan Wing, Rajnish
> >> Jain, Leon Portman, Andrew Hutton and I) have been interested
> >> in RTP session recording in SIP.  The requirement draft will
> >> be released shortly.
> >>
> >> We would like to request agenda time in dispatch to propose
> >> the formation of a new working group to define protocol
> >> extensions and an architecture for RTP recording.
> >>
> >> Session recording in SIP
> >> Mailing Lists: TBD
> >> Chairs: TBD
> >> Area Directorate: Real Time Applications (RAI)
> >>
> >> Purpose:
> >>
> >> Session recording is a critical operational requirement in
> >> many businesses, especially where voice is used as a medium
> >> for commerce and customer support. A prime example where
> >> voice is used for trade is the financial industry. The call
> >> recording requirements in this industry are quite stringent.
> >> The recorded calls are used for dispute resolution and
> >> regulatory compliance. Other businesses such as customer
> >> support call centers typically employ call recording for
> >> quality control or business analytics.
> >>
> >> Depending on the country and its regulatory requirements,
> >> financial trading floors typically must record all calls. The
> >> recorded media content must be an exact copy of the actual
> >> conversation (i.e.
> >> clipping and loss of media are unacceptable).  Some
> >> deployments and regulations require that calls be aborted or
> >> rejected if the recording device is unavailable.
> >>
> >> This group will specify requirements for a SIP based protocol
> >> interface between a communications system and a recorder. The
> >> Communications System is responsible for establishing media
> >> sessions where the actual business is conducted. The Recorder
> >> is the sink of the recorded media.
> >>
> >> The recorded sessions can be of any kind such as voice, video
> >> and instant messaging. A recorded session is typically
> >> comprised of actual media content and the call metadata. The
> >> call metadata allows recording archives to be searched and
> >> filtered at a later time.
> >> The conveyance of call metadata from the communications
> >> system to the recorder is outside the scope of this document.
> >>
> >> This group will only looks into active recording, where the
> >> recorded system purposefully streams media to a recording
> >> device. Passive recording, where a recording device detects
> >> media directly from the network, is outside the scope of this
> >> document. In addition, lawful intercept is outside the scope
> >> of the group.
> >>
> >> Proposed deliverables:
> >>
> >> 1) Requirements document;
> >> 2) Solutions document, including reference architecture.
> >>
> >> Thanks,
> >>
> >> - vijay
> >> --
> >> Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent 1960
> >> Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
> >> Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
> >> Web:   http://ect.bell-labs.com/who/vkg/
> >> _______________________________________________
> >> dispatch mailing list
> >> dispatch@ietf.org
> >> https://www.ietf.org/mailman/listinfo/dispatch
> >>
> >>
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> >
> >
>
>_______________________________________________
>dispatch mailing list
>dispatch@ietf.org
>https://www.ietf.org/mailman/listinfo/dispatch
>
>_______________________________________________
>dispatch mailing list
>dispatch@ietf.org
>https://www.ietf.org/mailman/listinfo/dispatch


From vkg@alcatel-lucent.com  Wed Jun 10 15:48:25 2009
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Subject: Re: [dispatch] Session recording in SIP
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James M. Polk wrote:
> I agree with Roni about this probably being best suited for XCON, after 
> all, any point to point recording is likely (merely) adding a 3rd party 
> (the recorder) to the session.

Could the dispatch chairs/RAI ADs please weigh in on where to
do this work?  This will help target the requirements
draft accordingly.

Yesterday, when we had a quick email exchange between
authors on where to submit it, sipcore and dispatch were
the candidates, and between them, dispatch was more appropriate.
And given that to get work going in dispatch required a "mini-
charter", we went ahead and submitted one.

Between xcon and avt, probably xcon will be more appropriate
since the bulk of this work revolves more around SIP than it
does on RTP-specific issues.  But then, this work does not
quite fit in the current xcon charter and deliverables (at least
as far as I can see it; I may be wrong, of course.)

Thanks,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/

From jdrosen@cisco.com  Wed Jun 10 18:02:27 2009
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Cc: Leon Portman <Leon.Portman@nice.com>, "dispatch@ietf.org" <dispatch@ietf.org>, "Jain, Rajnish" <Rajnish.Jain@ipc.com>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [dispatch] Session recording in SIP
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I think this would be interesting to look at, and I'd also support work 
in this direction.

I also think its pretty fertile ground. Though it seems simple at first 
blush, recording today is actually a really complicated space. Some of 
the issues:

  * where does the 'media forking' happen
  * interfaces from recording applications to find and request calls to 
record
  * storage of recordings
  * protocols for access to recordings
  * meta-data about recorded calls
  * multimedia recording - e.g., record the PPT share in conjunction 
with the audio

I think this is pretty chunky and could merit its own group.

I also don't think this is about conferencing. Conferencing and 
recording are not the same thing, though they both involve sending media 
streams to multiple places.

-Jonathan R.

Vijay K. Gurbani wrote:
> Hi: I realize that the deadline for charter proposals
> was yesterday, but I hope that it is not too late to submit
> one more.
> 
> A few interested people (Hadriel Kaplan, Dan Wing, Rajnish Jain,
> Leon Portman, Andrew Hutton and I) have been interested in
> RTP session recording in SIP.  The requirement draft will
> be released shortly.
> 
> We would like to request agenda time in dispatch to propose
> the formation of a new working group to define protocol
> extensions and an architecture for RTP recording.
> 
> Session recording in SIP
> Mailing Lists: TBD
> Chairs: TBD
> Area Directorate: Real Time Applications (RAI)
> 
> Purpose:
> 
> Session recording is a critical operational requirement in many
> businesses, especially where voice is used as a medium for
> commerce and customer support. A prime example where voice is used
> for trade is the financial industry. The call recording requirements
> in this industry are quite stringent. The recorded calls are used
> for dispute resolution and regulatory compliance. Other businesses
> such as customer support call centers typically employ call
> recording for quality control or business analytics.
> 
> Depending on the country and its regulatory requirements, financial
> trading floors typically must record all calls. The recorded media
> content must be an exact copy of the actual conversation (i.e.
> clipping and loss of media are unacceptable).  Some deployments
> and regulations require that calls be aborted or rejected if
> the recording device is unavailable.
> 
> This group will specify requirements for a SIP based protocol
> interface between a communications system and a recorder. The
> Communications System is responsible for establishing media sessions
> where the actual business is conducted. The Recorder is the sink of
> the recorded media.
> 
> The recorded sessions can be of any kind such as voice, video and
> instant messaging. A recorded session is typically comprised of
> actual media content and the call metadata. The call metadata allows
> recording archives to be searched and filtered at a later time.
> The conveyance of call metadata from the communications system
> to the recorder is outside the scope of this document.
> 
> This group will only looks into active recording, where the recorded
> system purposefully streams media to a recording device. Passive
> recording, where a recording device detects media directly from
> the network, is outside the scope of this document. In addition,
> lawful intercept is outside the scope of the group.
> 
> Proposed deliverables:
> 
> 1) Requirements document;
> 2) Solutions document, including reference architecture.
> 
> Thanks,
> 
> - vijay

-- 
Jonathan D. Rosenberg, Ph.D.                   111 Wood Avenue South
Cisco Fellow                                   Iselin, NJ 08830
Cisco, Voice Technology Group
jdrosen@cisco.com
http://www.jdrosen.net                         PHONE: (408) 902-3084
http://www.cisco.com

From mary.barnes@nortel.com  Wed Jun 10 19:05:52 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Vijay K. Gurbani" <vkg@alcatel-lucent.com>, <dispatch@ietf.org>
Subject: Re: [dispatch] Session recording in SIP
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At this stage, DISPATCH is the appropriate place to be having these
discussions. Since this topic was just introduced to the WG yesterday,
it's a bit too early, and not obvious based on the various opinions
expressed thus far, to decide that it might belong elsewhere, need a new
WG/BoF, etc. So, let's continue the discussion on DISPATCH and refine
the problem space and solution scope.=20

Thanks,
Mary.
DISPATCH WG co-chair

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Vijay K. Gurbani
Sent: Wednesday, June 10, 2009 5:48 PM
To: dispatch@ietf.org
Subject: Re: [dispatch] Session recording in SIP

James M. Polk wrote:
> I agree with Roni about this probably being best suited for XCON,=20
> after all, any point to point recording is likely (merely) adding a=20
> 3rd party (the recorder) to the session.

Could the dispatch chairs/RAI ADs please weigh in on where to do this
work?  This will help target the requirements draft accordingly.

Yesterday, when we had a quick email exchange between authors on where
to submit it, sipcore and dispatch were the candidates, and between
them, dispatch was more appropriate.
And given that to get work going in dispatch required a "mini- charter",
we went ahead and submitted one.

Between xcon and avt, probably xcon will be more appropriate since the
bulk of this work revolves more around SIP than it does on RTP-specific
issues.  But then, this work does not quite fit in the current xcon
charter and deliverables (at least as far as I can see it; I may be
wrong, of course.)

Thanks,

- vijay
--
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent 1960 Lucent Lane,
Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/
_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch

From Even.roni@huawei.com  Wed Jun 10 22:08:13 2009
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Date: Thu, 11 Jun 2009 08:07:34 +0300
From: Roni Even <Even.roni@huawei.com>
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Subject: Re: [dispatch] Session recording in SIP
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Mary,
This brings another question about dispatch, what does it mean that dispatch
decides that this should be for example a specific WG work. (let's take XCON
for example) 

Does this qualify it as an XCON WG item
Does this make the topic a charter item for the XCON WG.

I assume that the answer is no for both questions.

If the assumption is correct, how do the chairs/Ads decide if to have a long
discussion in dispatch about the problem space and solution scope  or to
move it to a working group and finish this discussion there in order to
expedite the work and not repeat the discussion.

Roni Even

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
Of Mary Barnes
Sent: Thursday, June 11, 2009 5:08 AM
To: Vijay K. Gurbani; dispatch@ietf.org
Subject: Re: [dispatch] Session recording in SIP

At this stage, DISPATCH is the appropriate place to be having these
discussions. Since this topic was just introduced to the WG yesterday,
it's a bit too early, and not obvious based on the various opinions
expressed thus far, to decide that it might belong elsewhere, need a new
WG/BoF, etc. So, let's continue the discussion on DISPATCH and refine
the problem space and solution scope. 

Thanks,
Mary.
DISPATCH WG co-chair

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Vijay K. Gurbani
Sent: Wednesday, June 10, 2009 5:48 PM
To: dispatch@ietf.org
Subject: Re: [dispatch] Session recording in SIP

James M. Polk wrote:
> I agree with Roni about this probably being best suited for XCON, 
> after all, any point to point recording is likely (merely) adding a 
> 3rd party (the recorder) to the session.

Could the dispatch chairs/RAI ADs please weigh in on where to do this
work?  This will help target the requirements draft accordingly.

Yesterday, when we had a quick email exchange between authors on where
to submit it, sipcore and dispatch were the candidates, and between
them, dispatch was more appropriate.
And given that to get work going in dispatch required a "mini- charter",
we went ahead and submitted one.

Between xcon and avt, probably xcon will be more appropriate since the
bulk of this work revolves more around SIP than it does on RTP-specific
issues.  But then, this work does not quite fit in the current xcon
charter and deliverables (at least as far as I can see it; I may be
wrong, of course.)

Thanks,

- vijay
--
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent 1960 Lucent Lane,
Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/
_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch
_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch


From christer.holmberg@ericsson.com  Wed Jun 10 22:45:58 2009
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Thread-Topic: [dispatch] Session recording in SIP
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References: <4A2ECDB2.7000601@alcatel-lucent.com><EDC652A26FB23C4EB6384A4584434A04017901E7@307622ANEX5.global.avaya.com><4A2FB842.7050302@sipstation.com><023d01c9e9e7$1b015880$51040980$%roni@huawei.com><XFE-SJC-211EaJswFCH00000178@xfe-sjc-211.amer.cisco.com><4A30383D.7060902@alcatel-lucent.com><1ECE0EB50388174790F9694F77522CCF1E727D56@zrc2hxm0.corp.nortel.com> <02c001c9ea52$8bb543a0$a31fcae0$%roni@huawei.com>
From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Roni Even" <Even.roni@huawei.com>, "Mary Barnes" <mary.barnes@nortel.com>, "Vijay K. Gurbani" <vkg@alcatel-lucent.com>, <dispatch@ietf.org>
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Hi,

A question on the use-case behind this:

Is the idea that users can record sessions, and then listen to them =
later? Ie it is some kind of "network based PVR for SIP sessions" :)

OR, is the idea that sessions can be recorded, because there may be some =
authority which requires it? In that case, playback functionality etc =
may not be needed on the interface between the user and the recorder.

Regards,

Christer








=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Roni Even
> Sent: 11. kes=E4kuuta 2009 8:08
> To: 'Mary Barnes'; 'Vijay K. Gurbani'; dispatch@ietf.org
> Subject: Re: [dispatch] Session recording in SIP
>=20
> Mary,
> This brings another question about dispatch, what does it=20
> mean that dispatch decides that this should be for example a=20
> specific WG work. (let's take XCON for example)=20
>=20
> Does this qualify it as an XCON WG item
> Does this make the topic a charter item for the XCON WG.
>=20
> I assume that the answer is no for both questions.
>=20
> If the assumption is correct, how do the chairs/Ads decide if=20
> to have a long discussion in dispatch about the problem space=20
> and solution scope  or to move it to a working group and=20
> finish this discussion there in order to expedite the work=20
> and not repeat the discussion.
>=20
> Roni Even
>=20
> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Mary Barnes
> Sent: Thursday, June 11, 2009 5:08 AM
> To: Vijay K. Gurbani; dispatch@ietf.org
> Subject: Re: [dispatch] Session recording in SIP
>=20
> At this stage, DISPATCH is the appropriate place to be having=20
> these discussions. Since this topic was just introduced to=20
> the WG yesterday, it's a bit too early, and not obvious based=20
> on the various opinions expressed thus far, to decide that it=20
> might belong elsewhere, need a new WG/BoF, etc. So, let's=20
> continue the discussion on DISPATCH and refine the problem=20
> space and solution scope.=20
>=20
> Thanks,
> Mary.
> DISPATCH WG co-chair
>=20
> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Vijay K. Gurbani
> Sent: Wednesday, June 10, 2009 5:48 PM
> To: dispatch@ietf.org
> Subject: Re: [dispatch] Session recording in SIP
>=20
> James M. Polk wrote:
> > I agree with Roni about this probably being best suited for XCON,=20
> > after all, any point to point recording is likely (merely) adding a=20
> > 3rd party (the recorder) to the session.
>=20
> Could the dispatch chairs/RAI ADs please weigh in on where to=20
> do this work?  This will help target the requirements draft=20
> accordingly.
>=20
> Yesterday, when we had a quick email exchange between authors=20
> on where to submit it, sipcore and dispatch were the=20
> candidates, and between them, dispatch was more appropriate.
> And given that to get work going in dispatch required a=20
> "mini- charter", we went ahead and submitted one.
>=20
> Between xcon and avt, probably xcon will be more appropriate=20
> since the bulk of this work revolves more around SIP than it=20
> does on RTP-specific issues.  But then, this work does not=20
> quite fit in the current xcon charter and deliverables (at=20
> least as far as I can see it; I may be wrong, of course.)
>=20
> Thanks,
>=20
> - vijay
> --
> Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent 1960=20
> Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
> Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
> Web:   http://ect.bell-labs.com/who/vkg/
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

From gonzalo.camarillo@ericsson.com  Wed Jun 10 23:14:57 2009
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Folks,

there have been significant list discussions on whether or not the IETF 
should be working on the topics proposed in the email below:

http://www.ietf.org/mail-archive/web/dispatch/current/msg00080.html

As a way forward, we have decided to organize the face-to-face 
discussions around this topic in Stockholm as a regular BOF (i.e., 
outside DISPATCH). We believe that, given the nature of the questions 
that need to be answered, this topic will benefit from discussions from 
a wide range of people. Organizing the meeting as a regular BOF ensures 
such discussions. A regular BOF also provides more time for discussions.

Therefore, from now on, please use the following mailing list (instead 
of the DISPATCH list) for discussions on this topic:

   codec@ietf.org

Thanks,

Gonzalo
DISPATCH co-chair

From andrew.hutton@siemens-enterprise.com  Thu Jun 11 02:07:34 2009
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Date: Thu, 11 Jun 2009 10:07:30 +0100
From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
In-reply-to: <028a01c9ea12$5ec96e60$1c5c4b20$%roni@huawei.com>
To: Roni Even <Even.roni@huawei.com>, Alan Johnston <alan@sipstation.com>, "Romascanu, Dan (Dan)" <dromasca@avaya.com>
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Cc: Leon Portman <Leon.Portman@nice.com>, dispatch@ietf.org, "Jain, Rajnish" <Rajnish.Jain@ipc.com>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [dispatch] Session recording in SIP
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Hi,

I think that currently this discussion should continue to be in the
dispatch WG as the current status seems to fit exactly the Dispatch
charter.

In my opinion it does not belong in AVT as it does not require any
changes to media formats and I also don't think it belongs in XCON as it
really is not conferencing although I would agree it has some
similarity.

Regards
Andy
=20

>-----Original Message-----
>From: dispatch-bounces@ietf.org=20
>[mailto:dispatch-bounces@ietf.org] On Behalf Of Roni Even
>Sent: 10 June 2009 22:28
>To: 'Alan Johnston'; 'Romascanu, Dan (Dan)'
>Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain,Rajnish'; 'Hadriel Kaplan'
>Subject: Re: [dispatch] Session recording in SIP
>
>Hi,
>I agree that this is an important work but I am not sure about=20
>a WG. This
>may be done in XCON and AVT.
>Note that the recording is not only for point to point call=20
>but at least for
>three parties like in call center scenarios
>
>I also did not notice any mention of the playback part but it=20
>should also be
>addressed
>
>Roni Even
>
>-----Original Message-----
>From: dispatch-bounces@ietf.org=20
>[mailto:dispatch-bounces@ietf.org] On Behalf
>Of Alan Johnston
>Sent: Wednesday, June 10, 2009 4:42 PM
>To: Romascanu, Dan (Dan)
>Cc: Leon Portman; dispatch@ietf.org; Jain, Rajnish; Hadriel Kaplan
>Subject: Re: [dispatch] Session recording in SIP
>
>This is important work to be done in RAI.  Key management for secure=20
>recording is an important component that needs to be addressed.
>
>draft-wing-sipping-srtp-key-04 addresses many of these topics.
>
>Thanks,
>Alan
>
>
>Romascanu, Dan (Dan) wrote:
>> I believe that this is an important application, and I support doing
>> work in this direction.=20
>>
>> Three comments on the preliminary words:=20
>>
>> 1. The requirement is for both signaling and media recording, right?
>> Probably good to say it.=20
>> 2. Security and regulations on how the information is accessed and
>> protected are of high importance. They would probably be reflected in
>> the requirements, but explicit wording in the future charter=20
>can help.
>> 3. I suggest that we look at the IPFIX protocol as a=20
>possible technology
>> to re-use
>>
>> Dan
>> =20
>>
>>  =20
>>> -----Original Message-----
>>> From: dispatch-bounces@ietf.org=20
>>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Vijay K. Gurbani
>>> Sent: Wednesday, June 10, 2009 12:02 AM
>>> To: dispatch@ietf.org
>>> Cc: Leon Portman; Jain,Rajnish; Hadriel Kaplan
>>> Subject: [dispatch] Session recording in SIP
>>>
>>> Hi: I realize that the deadline for charter proposals was=20
>>> yesterday, but I hope that it is not too late to submit one more.
>>>
>>> A few interested people (Hadriel Kaplan, Dan Wing, Rajnish=20
>>> Jain, Leon Portman, Andrew Hutton and I) have been interested=20
>>> in RTP session recording in SIP.  The requirement draft will=20
>>> be released shortly.
>>>
>>> We would like to request agenda time in dispatch to propose=20
>>> the formation of a new working group to define protocol=20
>>> extensions and an architecture for RTP recording.
>>>
>>> Session recording in SIP
>>> Mailing Lists: TBD
>>> Chairs: TBD
>>> Area Directorate: Real Time Applications (RAI)
>>>
>>> Purpose:
>>>
>>> Session recording is a critical operational requirement in=20
>>> many businesses, especially where voice is used as a medium=20
>>> for commerce and customer support. A prime example where=20
>>> voice is used for trade is the financial industry. The call=20
>>> recording requirements in this industry are quite stringent.=20
>>> The recorded calls are used for dispute resolution and=20
>>> regulatory compliance. Other businesses such as customer=20
>>> support call centers typically employ call recording for=20
>>> quality control or business analytics.
>>>
>>> Depending on the country and its regulatory requirements,=20
>>> financial trading floors typically must record all calls. The=20
>>> recorded media content must be an exact copy of the actual=20
>>> conversation (i.e.
>>> clipping and loss of media are unacceptable).  Some=20
>>> deployments and regulations require that calls be aborted or=20
>>> rejected if the recording device is unavailable.
>>>
>>> This group will specify requirements for a SIP based protocol=20
>>> interface between a communications system and a recorder. The=20
>>> Communications System is responsible for establishing media=20
>>> sessions where the actual business is conducted. The Recorder=20
>>> is the sink of the recorded media.
>>>
>>> The recorded sessions can be of any kind such as voice, video=20
>>> and instant messaging. A recorded session is typically=20
>>> comprised of actual media content and the call metadata. The=20
>>> call metadata allows recording archives to be searched and=20
>>> filtered at a later time.
>>> The conveyance of call metadata from the communications=20
>>> system to the recorder is outside the scope of this document.
>>>
>>> This group will only looks into active recording, where the=20
>>> recorded system purposefully streams media to a recording=20
>>> device. Passive recording, where a recording device detects=20
>>> media directly from the network, is outside the scope of this=20
>>> document. In addition, lawful intercept is outside the scope=20
>>> of the group.
>>>
>>> Proposed deliverables:
>>>
>>> 1) Requirements document;
>>> 2) Solutions document, including reference architecture.
>>>
>>> Thanks,
>>>
>>> - vijay
>>> --
>>> Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent 1960=20
>>> Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
>>> Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
>>> Web:   http://ect.bell-labs.com/who/vkg/
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>
>>>    =20
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
>>  =20
>
>_______________________________________________
>dispatch mailing list
>dispatch@ietf.org
>https://www.ietf.org/mailman/listinfo/dispatch
>
>
>_______________________________________________
>dispatch mailing list
>dispatch@ietf.org
>https://www.ietf.org/mailman/listinfo/dispatch
>

From Even.roni@huawei.com  Thu Jun 11 03:49:33 2009
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Date: Thu, 11 Jun 2009 13:47:34 +0300
From: Roni Even <Even.roni@huawei.com>
In-reply-to: <719F9BBB3E7E71428A2D13F3D76C768F01C7BBE8@GBNTHT12009MSX.gb002.siemens.net>
To: "'Hutton, Andrew'" <andrew.hutton@siemens-enterprise.com>, 'Alan Johnston' <alan@sipstation.com>, "'Romascanu, Dan (Dan)'" <dromasca@avaya.com>
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Cc: 'Leon Portman' <Leon.Portman@nice.com>, dispatch@ietf.org, "'Jain, Rajnish'" <Rajnish.Jain@ipc.com>, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [dispatch] Session recording in SIP
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Hi,
I looked at the suggested text and this is why I think it can be done in
XCON and that there are AVT aspects

The solution wants to record not only point to point calls since call center
scenarios include multipoint so it can be done in XCON instead of forming a
new WG . XCON is finishing its current charter and may get this as a new
charter.

As for the media, if you want to record multimedia RTP streams you need to
be able to synchronize between the separate streams (done in RTCP now), I
think Jonathan made this comment. You may also need to address the SRTP keys
if you want to keep the information secure. There is some drafts on the
topics currently in AVT. If not RTP is going to stored, you need to address
the storage format and require some transcoding capabilities on the record
server for audio and video unless you only want a G.711 solution. So if some
other packetization is needed maybe this is also AVT work


Roni Even

-----Original Message-----
From: Hutton, Andrew [mailto:andrew.hutton@siemens-enterprise.com] 
Sent: Thursday, June 11, 2009 12:08 PM
To: Roni Even; Alan Johnston; Romascanu, Dan (Dan)
Cc: Leon Portman; dispatch@ietf.org; Jain,Rajnish; Hadriel Kaplan
Subject: RE: [dispatch] Session recording in SIP

Hi,

I think that currently this discussion should continue to be in the
dispatch WG as the current status seems to fit exactly the Dispatch
charter.

In my opinion it does not belong in AVT as it does not require any
changes to media formats and I also don't think it belongs in XCON as it
really is not conferencing although I would agree it has some
similarity.

Regards
Andy
 

>-----Original Message-----
>From: dispatch-bounces@ietf.org 
>[mailto:dispatch-bounces@ietf.org] On Behalf Of Roni Even
>Sent: 10 June 2009 22:28
>To: 'Alan Johnston'; 'Romascanu, Dan (Dan)'
>Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain,Rajnish'; 'Hadriel Kaplan'
>Subject: Re: [dispatch] Session recording in SIP
>
>Hi,
>I agree that this is an important work but I am not sure about 
>a WG. This
>may be done in XCON and AVT.
>Note that the recording is not only for point to point call 
>but at least for
>three parties like in call center scenarios
>
>I also did not notice any mention of the playback part but it 
>should also be
>addressed
>
>Roni Even
>
>-----Original Message-----
>From: dispatch-bounces@ietf.org 
>[mailto:dispatch-bounces@ietf.org] On Behalf
>Of Alan Johnston
>Sent: Wednesday, June 10, 2009 4:42 PM
>To: Romascanu, Dan (Dan)
>Cc: Leon Portman; dispatch@ietf.org; Jain, Rajnish; Hadriel Kaplan
>Subject: Re: [dispatch] Session recording in SIP
>
>This is important work to be done in RAI.  Key management for secure 
>recording is an important component that needs to be addressed.
>
>draft-wing-sipping-srtp-key-04 addresses many of these topics.
>
>Thanks,
>Alan
>
>
>Romascanu, Dan (Dan) wrote:
>> I believe that this is an important application, and I support doing
>> work in this direction. 
>>
>> Three comments on the preliminary words: 
>>
>> 1. The requirement is for both signaling and media recording, right?
>> Probably good to say it. 
>> 2. Security and regulations on how the information is accessed and
>> protected are of high importance. They would probably be reflected in
>> the requirements, but explicit wording in the future charter 
>can help.
>> 3. I suggest that we look at the IPFIX protocol as a 
>possible technology
>> to re-use
>>
>> Dan
>>  
>>
>>   
>>> -----Original Message-----
>>> From: dispatch-bounces@ietf.org 
>>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Vijay K. Gurbani
>>> Sent: Wednesday, June 10, 2009 12:02 AM
>>> To: dispatch@ietf.org
>>> Cc: Leon Portman; Jain,Rajnish; Hadriel Kaplan
>>> Subject: [dispatch] Session recording in SIP
>>>
>>> Hi: I realize that the deadline for charter proposals was 
>>> yesterday, but I hope that it is not too late to submit one more.
>>>
>>> A few interested people (Hadriel Kaplan, Dan Wing, Rajnish 
>>> Jain, Leon Portman, Andrew Hutton and I) have been interested 
>>> in RTP session recording in SIP.  The requirement draft will 
>>> be released shortly.
>>>
>>> We would like to request agenda time in dispatch to propose 
>>> the formation of a new working group to define protocol 
>>> extensions and an architecture for RTP recording.
>>>
>>> Session recording in SIP
>>> Mailing Lists: TBD
>>> Chairs: TBD
>>> Area Directorate: Real Time Applications (RAI)
>>>
>>> Purpose:
>>>
>>> Session recording is a critical operational requirement in 
>>> many businesses, especially where voice is used as a medium 
>>> for commerce and customer support. A prime example where 
>>> voice is used for trade is the financial industry. The call 
>>> recording requirements in this industry are quite stringent. 
>>> The recorded calls are used for dispute resolution and 
>>> regulatory compliance. Other businesses such as customer 
>>> support call centers typically employ call recording for 
>>> quality control or business analytics.
>>>
>>> Depending on the country and its regulatory requirements, 
>>> financial trading floors typically must record all calls. The 
>>> recorded media content must be an exact copy of the actual 
>>> conversation (i.e.
>>> clipping and loss of media are unacceptable).  Some 
>>> deployments and regulations require that calls be aborted or 
>>> rejected if the recording device is unavailable.
>>>
>>> This group will specify requirements for a SIP based protocol 
>>> interface between a communications system and a recorder. The 
>>> Communications System is responsible for establishing media 
>>> sessions where the actual business is conducted. The Recorder 
>>> is the sink of the recorded media.
>>>
>>> The recorded sessions can be of any kind such as voice, video 
>>> and instant messaging. A recorded session is typically 
>>> comprised of actual media content and the call metadata. The 
>>> call metadata allows recording archives to be searched and 
>>> filtered at a later time.
>>> The conveyance of call metadata from the communications 
>>> system to the recorder is outside the scope of this document.
>>>
>>> This group will only looks into active recording, where the 
>>> recorded system purposefully streams media to a recording 
>>> device. Passive recording, where a recording device detects 
>>> media directly from the network, is outside the scope of this 
>>> document. In addition, lawful intercept is outside the scope 
>>> of the group.
>>>
>>> Proposed deliverables:
>>>
>>> 1) Requirements document;
>>> 2) Solutions document, including reference architecture.
>>>
>>> Thanks,
>>>
>>> - vijay
>>> --
>>> Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent 1960 
>>> Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
>>> Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
>>> Web:   http://ect.bell-labs.com/who/vkg/
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>
>>>     
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
>>   
>
>_______________________________________________
>dispatch mailing list
>dispatch@ietf.org
>https://www.ietf.org/mailman/listinfo/dispatch
>
>
>_______________________________________________
>dispatch mailing list
>dispatch@ietf.org
>https://www.ietf.org/mailman/listinfo/dispatch
>


From mary.barnes@nortel.com  Thu Jun 11 06:03:02 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Roni Even" <Even.roni@huawei.com>, "Vijay K. Gurbani" <vkg@alcatel-lucent.com>, <dispatch@ietf.org>
Subject: Re: [dispatch] Session recording in SIP
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Roni,

DISPATCH WG can tentatively agree that a work item is appropriate and
scoped for a particular WG. That's certainly non-binding as are any
decisions in any WG in terms of chartering new work items, until the ADs
approve. But, we can ask the ADs to move the work to the appropriate WG.


As far as how we decide how long to discuss a topic, that certainly
depends upon the topic. But, one of the reasons we set the early
deadlines for Stockholm was to get focused discussion on specific
topics, so that we could fairly quickly decide if and where work should
be done.  We do have the milestone of June 15th to finalize the topics
that are the focus for Stockholm and we are already in the process of
dispatching the items that were submitted on the appropriate deadlines.
We're doing those item by item so they're in the appropriate thread of
discussion and we'll provide a summary.  But, again, this topic has not
had enough discussion to make that decision yet. If the group comes to a
concensus on scope etc., then it may be possible to dispatch this item
prior to Stockholm or use f2f time if needed.=20

Mary.=20

-----Original Message-----
From: Roni Even [mailto:Even.roni@huawei.com]=20
Sent: Thursday, June 11, 2009 12:08 AM
To: Barnes, Mary (RICH2:AR00); 'Vijay K. Gurbani'; dispatch@ietf.org
Subject: RE: [dispatch] Session recording in SIP

Mary,
This brings another question about dispatch, what does it mean that
dispatch decides that this should be for example a specific WG work.
(let's take XCON for example)=20

Does this qualify it as an XCON WG item
Does this make the topic a charter item for the XCON WG.

I assume that the answer is no for both questions.

If the assumption is correct, how do the chairs/Ads decide if to have a
long discussion in dispatch about the problem space and solution scope
or to move it to a working group and finish this discussion there in
order to expedite the work and not repeat the discussion.

Roni Even

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Mary Barnes
Sent: Thursday, June 11, 2009 5:08 AM
To: Vijay K. Gurbani; dispatch@ietf.org
Subject: Re: [dispatch] Session recording in SIP

At this stage, DISPATCH is the appropriate place to be having these
discussions. Since this topic was just introduced to the WG yesterday,
it's a bit too early, and not obvious based on the various opinions
expressed thus far, to decide that it might belong elsewhere, need a new
WG/BoF, etc. So, let's continue the discussion on DISPATCH and refine
the problem space and solution scope.=20

Thanks,
Mary.
DISPATCH WG co-chair

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Vijay K. Gurbani
Sent: Wednesday, June 10, 2009 5:48 PM
To: dispatch@ietf.org
Subject: Re: [dispatch] Session recording in SIP

James M. Polk wrote:
> I agree with Roni about this probably being best suited for XCON,=20
> after all, any point to point recording is likely (merely) adding a=20
> 3rd party (the recorder) to the session.

Could the dispatch chairs/RAI ADs please weigh in on where to do this
work?  This will help target the requirements draft accordingly.

Yesterday, when we had a quick email exchange between authors on where
to submit it, sipcore and dispatch were the candidates, and between
them, dispatch was more appropriate.
And given that to get work going in dispatch required a "mini- charter",
we went ahead and submitted one.

Between xcon and avt, probably xcon will be more appropriate since the
bulk of this work revolves more around SIP than it does on RTP-specific
issues.  But then, this work does not quite fit in the current xcon
charter and deliverables (at least as far as I can see it; I may be
wrong, of course.)

Thanks,

- vijay
--
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent 1960 Lucent Lane,
Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/
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From mary.barnes@nortel.com  Thu Jun 11 06:41:43 2009
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Subject: Re: [dispatch] Milestone(s) for the Sound Level Indicators problem
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Hi folks,

The DISPATCH WG chairs have asked the ADs to move this work item to the
AVT WG.  Obviously, the work begins with an individual document in the
AVT WG and the ongoing discussion in that context would be entirely on
the AVT WG mailing list. =20

Thanks,
Mary.=20

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Enrico Marocco
Sent: Monday, June 08, 2009 2:34 PM
To: dispatch@ietf.org
Subject: [dispatch] Milestone(s) for the Sound Level Indicators problem

After some online and offline discussion about how to address the
problem described in draft-ivov-dispatch-slic-ps-00, i.e. how to enable
a mixer to pass the sound levels of the contributing streams that result
in a mixed stream back to the clients, it seems that a reasonable way
forward would be to have the AVT WG defining the missing piece (an RTP
extension to encode sound level values for the CSRCs). So, we would like
to propose the following milestone for AVT:

"Submit RTP header extension (as per RFC 5285) for multiple sound level
indicators."

For whoever has missed it, the problem was initially discussed here:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00071.html

As usual, comments are welcome.

--
Ciao,
Enrico

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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "RAI DISPATCH" <dispatch@ietf.org>
Cc: SIP Operations <sip-ops@ietf.org>
Subject: Re: [dispatch] draft-lawrence-sip-3rd-party-authorization-00
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Hi folks,

There seems to be sufficient interest in this topic to warrant f2f
agenda time in Stockholm. However, additional feedback prior to IETF-75
is necessary to ensure that progress can be made in Stockholm.=20

Thanks,
Mary.=20
DISPATCH WG co-chair

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Lawrence, Scott (BL60:9D30)
Sent: Sunday, May 03, 2009 2:37 PM
To: RAI DISPATCH
Cc: SIP Operations
Subject: [dispatch] draft-lawrence-sip-3rd-party-authorization-00

draft-lawrence-sip-3rd-party-authorization-00.txt has been submitted by
Scott Lawrence and posted to the IETF repository.

Filename:	 draft-lawrence-sip-3rd-party-authorization
Revision:	 00
Title:		 Third Party Authorization in the Session Initiation
Protocol
Creation_date:	 2009-05-03
WG ID:		 Independent Submission
Number_of_pages: 15

Abstract:
        This draft describes some circumstances that are common in SIP
        deployments which lack a rigorous authorization model, and
points out
        some ways in which this has resulted in poor security
        characteristics.
       =20
        The purpose of this document is to stimulate discussion of the
        identified problem and proposed requirements for any solution.

In this draft, I lay out a case for why SIP would benefit from an
authorization model that allows for one party to send a request to a
second party who must decide whether or not to allow the request, but
have a third party provide an explicit authorization in the request.
The goal is to allow separation of the evaluation of a request with
respect to a security policy from other parts of responding to the
request.

I _do_not_ include any discussion in this draft of _how_ the
requirements it lists could or should be met.  While I'm very interested
in that discussion, I think it's important to first discover whether or
not there is agreement in the SIP community that there really is a
problem and what part or parts of that problem need to be solved; this
draft is focused on that discussion.



_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch

From enrico.marocco@telecomitalia.it  Thu Jun 11 07:00:47 2009
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Date: Thu, 11 Jun 2009 16:00:58 +0200
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Subject: Re: [dispatch] Milestone(s) for the Sound Level Indicators problem
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Thanks Mary, we are about to submit a new draft in AVT and will try to
move the work forward there.

As an unasked, gratuitous comment, I would like to say that we have
quite appreciated the new process: in our particular case, a few weeks
of discussion in DISPATCH have probably saved several iterations in
different WGs.

Enrico

Mary Barnes wrote:
> Hi folks,
> 
> The DISPATCH WG chairs have asked the ADs to move this work item to the
> AVT WG.  Obviously, the work begins with an individual document in the
> AVT WG and the ongoing discussion in that context would be entirely on
> the AVT WG mailing list.  
> 
> Thanks,
> Mary. 
> 
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Enrico Marocco
> Sent: Monday, June 08, 2009 2:34 PM
> To: dispatch@ietf.org
> Subject: [dispatch] Milestone(s) for the Sound Level Indicators problem
> 
> After some online and offline discussion about how to address the
> problem described in draft-ivov-dispatch-slic-ps-00, i.e. how to enable
> a mixer to pass the sound levels of the contributing streams that result
> in a mixed stream back to the clients, it seems that a reasonable way
> forward would be to have the AVT WG defining the missing piece (an RTP
> extension to encode sound level values for the CSRCs). So, we would like
> to propose the following milestone for AVT:
> 
> "Submit RTP header extension (as per RFC 5285) for multiple sound level
> indicators."
> 
> For whoever has missed it, the problem was initially discussed here:
> http://www.ietf.org/mail-archive/web/dispatch/current/msg00071.html
> 
> As usual, comments are welcome.
> 
> --
> Ciao,
> Enrico

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From andrew.hutton@siemens-enterprise.com  Thu Jun 11 07:39:15 2009
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Date: Thu, 11 Jun 2009 15:36:11 +0100
From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
In-reply-to: <002801c9ea82$0da51320$28ef3960$%roni@huawei.com>
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Subject: Re: [dispatch] Session recording in SIP
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Hi,

I am sure that there are aspects of session recording that could require
work in AVT but I don't see that this is the focus of this proposed work
item which I think is to define the SIP requirements and possible SIP
protocol extensions that might be needed.=20

I find it harder to see how this relates to XCON.

Regards
Andy
=20

>-----Original Message-----
>From: Roni Even [mailto:Even.roni@huawei.com]=20
>Sent: 11 June 2009 11:48
>To: Hutton, Andrew; 'Alan Johnston'; 'Romascanu, Dan (Dan)'
>Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain,Rajnish'; 'Hadriel Kaplan'
>Subject: RE: [dispatch] Session recording in SIP
>
>Hi,
>I looked at the suggested text and this is why I think it can=20
>be done in
>XCON and that there are AVT aspects
>
>The solution wants to record not only point to point calls=20
>since call center
>scenarios include multipoint so it can be done in XCON instead=20
>of forming a
>new WG . XCON is finishing its current charter and may get=20
>this as a new
>charter.
>
>As for the media, if you want to record multimedia RTP streams=20
>you need to
>be able to synchronize between the separate streams (done in=20
>RTCP now), I
>think Jonathan made this comment. You may also need to address=20
>the SRTP keys
>if you want to keep the information secure. There is some drafts on the
>topics currently in AVT. If not RTP is going to stored, you=20
>need to address
>the storage format and require some transcoding capabilities=20
>on the record
>server for audio and video unless you only want a G.711=20
>solution. So if some
>other packetization is needed maybe this is also AVT work
>
>
>Roni Even
>
>-----Original Message-----
>From: Hutton, Andrew [mailto:andrew.hutton@siemens-enterprise.com]=20
>Sent: Thursday, June 11, 2009 12:08 PM
>To: Roni Even; Alan Johnston; Romascanu, Dan (Dan)
>Cc: Leon Portman; dispatch@ietf.org; Jain,Rajnish; Hadriel Kaplan
>Subject: RE: [dispatch] Session recording in SIP
>
>Hi,
>
>I think that currently this discussion should continue to be in the
>dispatch WG as the current status seems to fit exactly the Dispatch
>charter.
>
>In my opinion it does not belong in AVT as it does not require any
>changes to media formats and I also don't think it belongs in=20
>XCON as it
>really is not conferencing although I would agree it has some
>similarity.
>
>Regards
>Andy
>=20
>
>>-----Original Message-----
>>From: dispatch-bounces@ietf.org=20
>>[mailto:dispatch-bounces@ietf.org] On Behalf Of Roni Even
>>Sent: 10 June 2009 22:28
>>To: 'Alan Johnston'; 'Romascanu, Dan (Dan)'
>>Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain,Rajnish';=20
>'Hadriel Kaplan'
>>Subject: Re: [dispatch] Session recording in SIP
>>
>>Hi,
>>I agree that this is an important work but I am not sure about=20
>>a WG. This
>>may be done in XCON and AVT.
>>Note that the recording is not only for point to point call=20
>>but at least for
>>three parties like in call center scenarios
>>
>>I also did not notice any mention of the playback part but it=20
>>should also be
>>addressed
>>
>>Roni Even
>>
>>-----Original Message-----
>>From: dispatch-bounces@ietf.org=20
>>[mailto:dispatch-bounces@ietf.org] On Behalf
>>Of Alan Johnston
>>Sent: Wednesday, June 10, 2009 4:42 PM
>>To: Romascanu, Dan (Dan)
>>Cc: Leon Portman; dispatch@ietf.org; Jain, Rajnish; Hadriel Kaplan
>>Subject: Re: [dispatch] Session recording in SIP
>>
>>This is important work to be done in RAI.  Key management for secure=20
>>recording is an important component that needs to be addressed.
>>
>>draft-wing-sipping-srtp-key-04 addresses many of these topics.
>>
>>Thanks,
>>Alan
>>
>>
>>Romascanu, Dan (Dan) wrote:
>>> I believe that this is an important application, and I support doing
>>> work in this direction.=20
>>>
>>> Three comments on the preliminary words:=20
>>>
>>> 1. The requirement is for both signaling and media recording, right?
>>> Probably good to say it.=20
>>> 2. Security and regulations on how the information is accessed and
>>> protected are of high importance. They would probably be=20
>reflected in
>>> the requirements, but explicit wording in the future charter=20
>>can help.
>>> 3. I suggest that we look at the IPFIX protocol as a=20
>>possible technology
>>> to re-use
>>>
>>> Dan
>>> =20
>>>
>>>  =20
>>>> -----Original Message-----
>>>> From: dispatch-bounces@ietf.org=20
>>>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Vijay K. Gurbani
>>>> Sent: Wednesday, June 10, 2009 12:02 AM
>>>> To: dispatch@ietf.org
>>>> Cc: Leon Portman; Jain,Rajnish; Hadriel Kaplan
>>>> Subject: [dispatch] Session recording in SIP
>>>>
>>>> Hi: I realize that the deadline for charter proposals was=20
>>>> yesterday, but I hope that it is not too late to submit one more.
>>>>
>>>> A few interested people (Hadriel Kaplan, Dan Wing, Rajnish=20
>>>> Jain, Leon Portman, Andrew Hutton and I) have been interested=20
>>>> in RTP session recording in SIP.  The requirement draft will=20
>>>> be released shortly.
>>>>
>>>> We would like to request agenda time in dispatch to propose=20
>>>> the formation of a new working group to define protocol=20
>>>> extensions and an architecture for RTP recording.
>>>>
>>>> Session recording in SIP
>>>> Mailing Lists: TBD
>>>> Chairs: TBD
>>>> Area Directorate: Real Time Applications (RAI)
>>>>
>>>> Purpose:
>>>>
>>>> Session recording is a critical operational requirement in=20
>>>> many businesses, especially where voice is used as a medium=20
>>>> for commerce and customer support. A prime example where=20
>>>> voice is used for trade is the financial industry. The call=20
>>>> recording requirements in this industry are quite stringent.=20
>>>> The recorded calls are used for dispute resolution and=20
>>>> regulatory compliance. Other businesses such as customer=20
>>>> support call centers typically employ call recording for=20
>>>> quality control or business analytics.
>>>>
>>>> Depending on the country and its regulatory requirements,=20
>>>> financial trading floors typically must record all calls. The=20
>>>> recorded media content must be an exact copy of the actual=20
>>>> conversation (i.e.
>>>> clipping and loss of media are unacceptable).  Some=20
>>>> deployments and regulations require that calls be aborted or=20
>>>> rejected if the recording device is unavailable.
>>>>
>>>> This group will specify requirements for a SIP based protocol=20
>>>> interface between a communications system and a recorder. The=20
>>>> Communications System is responsible for establishing media=20
>>>> sessions where the actual business is conducted. The Recorder=20
>>>> is the sink of the recorded media.
>>>>
>>>> The recorded sessions can be of any kind such as voice, video=20
>>>> and instant messaging. A recorded session is typically=20
>>>> comprised of actual media content and the call metadata. The=20
>>>> call metadata allows recording archives to be searched and=20
>>>> filtered at a later time.
>>>> The conveyance of call metadata from the communications=20
>>>> system to the recorder is outside the scope of this document.
>>>>
>>>> This group will only looks into active recording, where the=20
>>>> recorded system purposefully streams media to a recording=20
>>>> device. Passive recording, where a recording device detects=20
>>>> media directly from the network, is outside the scope of this=20
>>>> document. In addition, lawful intercept is outside the scope=20
>>>> of the group.
>>>>
>>>> Proposed deliverables:
>>>>
>>>> 1) Requirements document;
>>>> 2) Solutions document, including reference architecture.
>>>>
>>>> Thanks,
>>>>
>>>> - vijay
>>>> --
>>>> Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent 1960=20
>>>> Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
>>>> Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
>>>> Web:   http://ect.bell-labs.com/who/vkg/
>>>> _______________________________________________
>>>> dispatch mailing list
>>>> dispatch@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>>
>>>>    =20
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>
>>>  =20
>>
>>_______________________________________________
>>dispatch mailing list
>>dispatch@ietf.org
>>https://www.ietf.org/mailman/listinfo/dispatch
>>
>>
>>_______________________________________________
>>dispatch mailing list
>>dispatch@ietf.org
>>https://www.ietf.org/mailman/listinfo/dispatch
>>
>
>

From mary.barnes@nortel.com  Thu Jun 11 07:39:22 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
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Subject: Re: [dispatch] Call Control UUI Problem Statement
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Hi folks,=20

Based on the amount of feedback received, as well as the level of
commitment to complete the work, we (ADs/chairs) have decided to propose
a new WG (a "mini-WG", if you will) for this topic.  It doesn't fit
SIPCORE, nor any other current WG and there is enough interest that this
shouldn't be individual/AD sponsored. =20

The proposed WG would have a single deliverable and would provide a
focused venue for discussions. The idea would be that it would finish
very quickly.=20

We will be allocating 30 minutes of agenda time for this topic, with the
focus being on the charter for the WG.=20

Regards,
Mary.
DISPATCH WG co-chair

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Alan Johnston
Sent: Friday, May 22, 2009 12:21 PM
To: dispatch@ietf.org
Cc: Joanne McMillen
Subject: [dispatch] Call Control UUI Problem Statement

Several approaches to transporting the ITU-T Q.931 User to User
Information Element (UU IE) data in SIP have been proposed.  As networks
move to SIP it is important that applications requiring this data can
continue to function in SIP networks as well as the ability to interwork
with this ISDN service for end-to-end transparency.  While the
information is carried in ISUP, the contents can be in a number of
protocols that are not ISUP, and as such it is architecturally
unreasonable to interwork these protocols at the SIP / ISUP gateway.=20
This work area involves defining a new SIP mechanism to transport this
information.=20

Since the INFO method, RFC2976, was developed for ISUP interworking of
user-to-user information, it might seem to be the logical choice here. =20
For non-call control user-to-user information, INFO can be utilized for
end to end transport.  However, for transport of call control
user-to-user information, INFO can not be used.  Since the contents
carry information that can impact call handling at the UAC, the
information must be conveyed with the INVITE transaction.  As the call
flows in the previous section show, the information is related to an
attempt to establish a session and must be passed with the session setup
request (INVITE), responses to that INVITE, or session termination
requests.  As a result, it is not possible to use INFO in these cases.

Also, the use of message body is another obvious mechanism, but this is
not suitable for a number of reasons including difficulty with multipart
MIME INVITEs and 200 OKs, and in escaping UUI information into redirects
and REFERs.

The I-D draft-johnston-sipping-cc-uui will be used as the basis for this
work.  This draft was first submitted 3 years ago and has had plenty of
discussion in the old SIPPING Working Group.  The requirements in this
document completed a WGLC in SIPPING in December 2008 with strong
support from the group.  This draft proposes a new SIP header field
User-to-User and parameters for transporting this information.

This extension will also be used for native SIP endpoints implementing
similar services and interworking with ISDN services.  Example use cases
include an exchange between two user agents, retargeting by a proxy, and
redirection.  An example application is an Automatic Call Distributor
(ACD) in a contact center.

The coordinators for this work will be Alan Johnston
<alan@sipstation.com> and Keith Drage <drage@alcatel-lucent.com>

Thanks,
Alan


_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch

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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Dispatch Working Group of the IETF.


	Title           : Managing Client Initiated Connections in the Session Initiation Protocol (SIP)
	Author(s)       : C. Jennings
	Filename        : draft-ietf-sip-outbound-20.txt
	Pages           : 49
	Date            : 2009-06-10

The Session Initiation Protocol (SIP) allows proxy servers to
initiate TCP connections or to send asynchronous UDP datagrams to
User Agents in order to deliver requests.  However, in a large number
of real deployments, many practical considerations, such as the
existence of firewalls and Network Address Translators (NATs) or the
use of TLS with server-provided certificates, prevent servers from
connecting to User Agents in this way.  This specification defines
behaviors for User Agents, registrars and proxy servers that allow
requests to be delivered on existing connections established by the
User Agent.  It also defines keep alive behaviors needed to keep NAT
bindings open and specifies the usage of multiple connections from
the User Agent to its Registrar.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-20.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

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From tme@americafree.tv  Thu Jun 11 08:13:30 2009
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On Jun 11, 2009, at 2:15 AM, Gonzalo Camarillo wrote:

> Folks,
>
> there have been significant list discussions on whether or not the  
> IETF should be working on the topics proposed in the email below:
>
> http://www.ietf.org/mail-archive/web/dispatch/current/msg00080.html
>
> As a way forward, we have decided to organize the face-to-face  
> discussions around this topic in Stockholm as a regular BOF (i.e.,  
> outside DISPATCH). We believe that, given the nature of the  
> questions that need to be answered, this topic will benefit from  
> discussions from a wide range of people. Organizing the meeting as a  
> regular BOF ensures such discussions. A regular BOF also provides  
> more time for discussions.
>
> Therefore, from now on, please use the following mailing list  
> (instead of the DISPATCH list) for discussions on this topic:
>
>  codec@ietf.org
>
> Thanks,
>
> Gonzalo
> DISPATCH co-chair
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>


From Even.roni@huawei.com  Thu Jun 11 08:18:03 2009
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Subject: Re: [dispatch] Session recording in SIP
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Andy,
You have a term called communication system in the proposal
Can you clarify what it is? Is this the end points in the call is this some
type of a proxy, application aware middle box?

There is also a recorder in the proposal, is this a SIP UA or is it just a
RTP terminating device like a "media server" in which case this is similar
to mediactrl WG work


Roni Even

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
Of Hutton, Andrew
Sent: Thursday, June 11, 2009 5:36 PM
To: Roni Even; Alan Johnston; Romascanu, Dan (Dan)
Cc: Leon Portman; dispatch@ietf.org; Jain, Rajnish; Hadriel Kaplan
Subject: Re: [dispatch] Session recording in SIP

Hi,

I am sure that there are aspects of session recording that could require
work in AVT but I don't see that this is the focus of this proposed work
item which I think is to define the SIP requirements and possible SIP
protocol extensions that might be needed. 

I find it harder to see how this relates to XCON.

Regards
Andy
 

>-----Original Message-----
>From: Roni Even [mailto:Even.roni@huawei.com] 
>Sent: 11 June 2009 11:48
>To: Hutton, Andrew; 'Alan Johnston'; 'Romascanu, Dan (Dan)'
>Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain,Rajnish'; 'Hadriel Kaplan'
>Subject: RE: [dispatch] Session recording in SIP
>
>Hi,
>I looked at the suggested text and this is why I think it can 
>be done in
>XCON and that there are AVT aspects
>
>The solution wants to record not only point to point calls 
>since call center
>scenarios include multipoint so it can be done in XCON instead 
>of forming a
>new WG . XCON is finishing its current charter and may get 
>this as a new
>charter.
>
>As for the media, if you want to record multimedia RTP streams 
>you need to
>be able to synchronize between the separate streams (done in 
>RTCP now), I
>think Jonathan made this comment. You may also need to address 
>the SRTP keys
>if you want to keep the information secure. There is some drafts on the
>topics currently in AVT. If not RTP is going to stored, you 
>need to address
>the storage format and require some transcoding capabilities 
>on the record
>server for audio and video unless you only want a G.711 
>solution. So if some
>other packetization is needed maybe this is also AVT work
>
>
>Roni Even
>
>-----Original Message-----
>From: Hutton, Andrew [mailto:andrew.hutton@siemens-enterprise.com] 
>Sent: Thursday, June 11, 2009 12:08 PM
>To: Roni Even; Alan Johnston; Romascanu, Dan (Dan)
>Cc: Leon Portman; dispatch@ietf.org; Jain,Rajnish; Hadriel Kaplan
>Subject: RE: [dispatch] Session recording in SIP
>
>Hi,
>
>I think that currently this discussion should continue to be in the
>dispatch WG as the current status seems to fit exactly the Dispatch
>charter.
>
>In my opinion it does not belong in AVT as it does not require any
>changes to media formats and I also don't think it belongs in 
>XCON as it
>really is not conferencing although I would agree it has some
>similarity.
>
>Regards
>Andy
> 
>
>>-----Original Message-----
>>From: dispatch-bounces@ietf.org 
>>[mailto:dispatch-bounces@ietf.org] On Behalf Of Roni Even
>>Sent: 10 June 2009 22:28
>>To: 'Alan Johnston'; 'Romascanu, Dan (Dan)'
>>Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain,Rajnish'; 
>'Hadriel Kaplan'
>>Subject: Re: [dispatch] Session recording in SIP
>>
>>Hi,
>>I agree that this is an important work but I am not sure about 
>>a WG. This
>>may be done in XCON and AVT.
>>Note that the recording is not only for point to point call 
>>but at least for
>>three parties like in call center scenarios
>>
>>I also did not notice any mention of the playback part but it 
>>should also be
>>addressed
>>
>>Roni Even
>>
>>-----Original Message-----
>>From: dispatch-bounces@ietf.org 
>>[mailto:dispatch-bounces@ietf.org] On Behalf
>>Of Alan Johnston
>>Sent: Wednesday, June 10, 2009 4:42 PM
>>To: Romascanu, Dan (Dan)
>>Cc: Leon Portman; dispatch@ietf.org; Jain, Rajnish; Hadriel Kaplan
>>Subject: Re: [dispatch] Session recording in SIP
>>
>>This is important work to be done in RAI.  Key management for secure 
>>recording is an important component that needs to be addressed.
>>
>>draft-wing-sipping-srtp-key-04 addresses many of these topics.
>>
>>Thanks,
>>Alan
>>
>>
>>Romascanu, Dan (Dan) wrote:
>>> I believe that this is an important application, and I support doing
>>> work in this direction. 
>>>
>>> Three comments on the preliminary words: 
>>>
>>> 1. The requirement is for both signaling and media recording, right?
>>> Probably good to say it. 
>>> 2. Security and regulations on how the information is accessed and
>>> protected are of high importance. They would probably be 
>reflected in
>>> the requirements, but explicit wording in the future charter 
>>can help.
>>> 3. I suggest that we look at the IPFIX protocol as a 
>>possible technology
>>> to re-use
>>>
>>> Dan
>>>  
>>>
>>>   
>>>> -----Original Message-----
>>>> From: dispatch-bounces@ietf.org 
>>>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Vijay K. Gurbani
>>>> Sent: Wednesday, June 10, 2009 12:02 AM
>>>> To: dispatch@ietf.org
>>>> Cc: Leon Portman; Jain,Rajnish; Hadriel Kaplan
>>>> Subject: [dispatch] Session recording in SIP
>>>>
>>>> Hi: I realize that the deadline for charter proposals was 
>>>> yesterday, but I hope that it is not too late to submit one more.
>>>>
>>>> A few interested people (Hadriel Kaplan, Dan Wing, Rajnish 
>>>> Jain, Leon Portman, Andrew Hutton and I) have been interested 
>>>> in RTP session recording in SIP.  The requirement draft will 
>>>> be released shortly.
>>>>
>>>> We would like to request agenda time in dispatch to propose 
>>>> the formation of a new working group to define protocol 
>>>> extensions and an architecture for RTP recording.
>>>>
>>>> Session recording in SIP
>>>> Mailing Lists: TBD
>>>> Chairs: TBD
>>>> Area Directorate: Real Time Applications (RAI)
>>>>
>>>> Purpose:
>>>>
>>>> Session recording is a critical operational requirement in 
>>>> many businesses, especially where voice is used as a medium 
>>>> for commerce and customer support. A prime example where 
>>>> voice is used for trade is the financial industry. The call 
>>>> recording requirements in this industry are quite stringent. 
>>>> The recorded calls are used for dispute resolution and 
>>>> regulatory compliance. Other businesses such as customer 
>>>> support call centers typically employ call recording for 
>>>> quality control or business analytics.
>>>>
>>>> Depending on the country and its regulatory requirements, 
>>>> financial trading floors typically must record all calls. The 
>>>> recorded media content must be an exact copy of the actual 
>>>> conversation (i.e.
>>>> clipping and loss of media are unacceptable).  Some 
>>>> deployments and regulations require that calls be aborted or 
>>>> rejected if the recording device is unavailable.
>>>>
>>>> This group will specify requirements for a SIP based protocol 
>>>> interface between a communications system and a recorder. The 
>>>> Communications System is responsible for establishing media 
>>>> sessions where the actual business is conducted. The Recorder 
>>>> is the sink of the recorded media.
>>>>
>>>> The recorded sessions can be of any kind such as voice, video 
>>>> and instant messaging. A recorded session is typically 
>>>> comprised of actual media content and the call metadata. The 
>>>> call metadata allows recording archives to be searched and 
>>>> filtered at a later time.
>>>> The conveyance of call metadata from the communications 
>>>> system to the recorder is outside the scope of this document.
>>>>
>>>> This group will only looks into active recording, where the 
>>>> recorded system purposefully streams media to a recording 
>>>> device. Passive recording, where a recording device detects 
>>>> media directly from the network, is outside the scope of this 
>>>> document. In addition, lawful intercept is outside the scope 
>>>> of the group.
>>>>
>>>> Proposed deliverables:
>>>>
>>>> 1) Requirements document;
>>>> 2) Solutions document, including reference architecture.
>>>>
>>>> Thanks,
>>>>
>>>> - vijay
>>>> --
>>>> Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent 1960 
>>>> Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
>>>> Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
>>>> Web:   http://ect.bell-labs.com/who/vkg/
>>>> _______________________________________________
>>>> dispatch mailing list
>>>> dispatch@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>>
>>>>     
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>
>>>   
>>
>>_______________________________________________
>>dispatch mailing list
>>dispatch@ietf.org
>>https://www.ietf.org/mailman/listinfo/dispatch
>>
>>
>>_______________________________________________
>>dispatch mailing list
>>dispatch@ietf.org
>>https://www.ietf.org/mailman/listinfo/dispatch
>>
>
>
_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch


From tme@americafree.tv  Thu Jun 11 08:46:27 2009
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Subject: Re: [dispatch] Wideband audio codec proposal dispatched as a regular BOF
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On Jun 11, 2009, at 11:13 AM, Marshall Eubanks wrote:

>
> On Jun 11, 2009, at 2:15 AM, Gonzalo Camarillo wrote:
>
>> Folks,
>>
>> there have been significant list discussions on whether or not the  
>> IETF should be working on the topics proposed in the email below:
>>
>> http://www.ietf.org/mail-archive/web/dispatch/current/msg00080.html
>>
>> As a way forward, we have decided to organize the face-to-face  
>> discussions around this topic in Stockholm as a regular BOF (i.e.,  
>> outside DISPATCH). We believe that, given the nature of the  
>> questions that need to be answered, this topic will benefit from  
>> discussions from a wide range of people. Organizing the meeting as  
>> a regular BOF ensures such discussions. A regular BOF also provides  
>> more time for discussions.
>>
>> Therefore, from now on, please use the following mailing list  
>> (instead of the DISPATCH list) for discussions on this topic:
>>
>> codec@ietf.org

For those who are slow, like me, you can go to

https://www.ietf.org/mailman/listinfo/codec

to subscribe.

Regards
Marshall

>>
>>
>> Thanks,
>>
>> Gonzalo
>> DISPATCH co-chair
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>


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Subject: Re: [dispatch] Charter Proposal
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> -----Original Message-----
> From: dispatch-bounces@ietf.org 
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Henry Sinnreich
> Sent: Tuesday, June 09, 2009 6:32 AM
> To: Shintaro Mizuno; Eric Burger
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] Charter Proposal
> 
> You may also want to consider HIP, since it solves NAT 
> traversal for all application layer protocols, including SIP.
> 
> http://www.ietf.org/internet-drafts/draft-ietf-hip-nat-travers
> al-07.txt

Yes, I am aware of HIP's NAT traversal capabilities, as I 
encouraged its adoption of ICE methodologies.

> It also takes care of the IPv4 to IPv6 transition for 
> endpoints, mobility and multi-homing.
> Open source code is plentifully available for reference 
> implementations.
> 
> http://www.ietf.org/html.charters/hip-charter.html

There are always new protocols.  But it is often pragmatic to leverage the
investment of deployed protocols -- much like TCP which continues to evolve
even though SCTP and DCCP provide superior functionality, much like how IPsec
was originally intended for IPv6 but was then supported on IPv4, etc.

We are interested in using SIP for establishing VPN sessions in much the same
way that SIP has embraced TLS (RFC4572) and embraced file transfer (RFC5547).

-d


> Henry
> 
> 
> 
> On 6/9/09 8:02 AM, "Shintaro Mizuno" 
> <mizuno.shintaro@lab.ntt.co.jp> wrote:
> 
> 
> 
> 	Mr. Burger,
> 	 Thank you for the comment.
> 	
> 	 We don't necessary see a need for a WG to be formed to 
> document the 
> 	informational document we have listed, the document is 
> there to see if
> 	the proposed way forward (informational document possibly
> 	as an independent submission) for the document listed 
> is on track or 
> 	whether something should be done at the WG level.
> 	
> 	 On the other hand, we believe that use of SIP similar 
> to what we 
> 	have done for VPN will likely to happen more. And on 
> that point, we are
> 	wanting to see whether IETF is interested in 
> documenting a guideline
> 	for dealing with application that's not voice or video.
> 	As written in the discussion point of the charter, the 
> question is how
> 	much should SIP be involved.
> 	
> 	 1. Should SIP just be used as a rendezvous protocol 
> and leave the 
> 	rest of the negotiation to whatever application?
> 	 2. Should SIP involve in management of the established 
> session as it
> 	does for audio/video?
> 	 3. Should use of SIP not be standardized for peer to 
> peer application
> 	beside audio/video?
> 	
> 	
> 	Regards,
> 	Shintaro
> 	
> 	
> 	On Mon, 8 Jun 2009 06:38:20 -0400
> 	Eric Burger <eburger@standardstrack.com> wrote:
> 	
> 	> Given the proposal is not really for an Internet 
> standard, and the 
> 	> product is not a protocol or standards track 
> document, why are we 
> 	> working on it here?
> 	>
> 	> On Jun 8, 2009, at 5:20 AM, Shintaro Mizuno wrote:
> 	>
> 	> > Dear DISPATCH Chair, ML members,
> 	> >
> 	> > I've finalized the charter proposal for on-demand 
> media/application
> 	> > sharing.
> 	> > Added milestone/deliverables section and some 
> descriptions on the 
> 	> > usage
> 	> > of SIP, but most of the text remains the same.
> 	> >
> 	> > --------------------------------------------------
> 	> > "Charter Proposal for on-demand Media/Application 
> Sharing between 
> 	> > peers
> 	> > over VPN"
> 	> >
> 	> > Intent:
> 	> > The intent of this charter is to see interests in 
> documenting what's
> 	> > being deployed as an informational for on-demand 
> Media/Application
> 	> > sharing between peers over VPN using SIP and 
> discuss what in the
> 	> > overall effort may be standardized at IETF in the future.
> 	> >
> 	> >
> 	> > Problem Statement:
> 	> > Home servers or network-capable consumer electronic 
> devices are
> 	> > becoming widely deployed and people using such 
> devices are wanting to
> 	> > share contents or applications and are seeking a 
> way to establish
> 	> > multitudes of communication with each other.
> 	> >
> 	> > In order to allow two people to share contents or 
> application securely
> 	> > using popular LAN based communication tool such as 
> DLNA on devices 
> 	> > that
> 	> > are sitting at home, secure LAN (VPN) 
> establishments between these 
> 	> > user
> 	> > devices are necessary. Applications that use proprietary or
> 	> > multi-session protocols such as in some of 
> LAN-based gaming will also
> 	> > benifit from VPN connections when users attempt to 
> communicate 
> 	> > remotely.
> 	> >
> 	> > For a deployment that's underway and considered by 
> several other
> 	> > standard bodies, problem of establishing VPN 
> between these user 
> 	> > devices
> 	> > were that current standards for establishing VPN 
> didn't provide all 
> 	> > the
> 	> > toolset necessary to overcome some of the obstacles.
> 	> >
> 	> > Namely following issues were found;
> 	> >  (1) Name resolution mechanism under floating IP 
> addresses and port
> 	> > numbers
> 	> >  (2) NAT traversal and hole-punching of firewall
> 	> >  (3) Authentication/authorization of users.
> 	> >  (4) Pre-sharing of key for establishing VPN isn't 
> easy when VPN is to
> 	> > be established on-demand between two user devices 
> without any previous
> 	> > affiliation.
> 	> >
> 	> > As a result current deployment based on 
> "draft-saito-mmusic-sdp-
> 	> > ike-04"
> 	> > and other standard bodies are looking at the use of 
> SIP for the
> 	> > following reasons.
> 	> >
> 	> > (1)Name resolution can be achieved by using 
> REGISTER method and SDP
> 	> > negotiation
> 	> > (2)NAT traversal and hole-punching of firewall can 
> be achieved by
> 	> > applying SIP+ICE.
> 	> > (3)Authentication/authorization can be achieved by 
> having trusted SIP
> 	> > proxy and user friendly SIP-URIs.
> 	> >
> 	> > SIP/SDP will be used to negotiate parameters 
> necessary to establish
> 	> > IKE/IPsec such as IP address, port numbers and 
> authentication
> 	> > information of peers. Current deployment uses SIP 
> also to manage IPsec
> 	> > connection, e.g. use BYE to shut down media 
> sessions for IKE/IPsec,
> 	> > however, further discussions can be made to find 
> out if there are
> 	> > intrests or use cases in using SIP only as a 
> rendezvous protocol.
> 	> >
> 	> > Use-cases that may see use for such standard effort;
> 	> >  - Media sharing using DLNA or similar protocol 
> over VPN between 2
> 	> > users' devices.
> 	> >  - Remote desktop sharing for Customer service over 
> VPN initiated via
> 	> > SIP call.
> 	> >   * Similar to click to call, customer service 
> agent can access user's
> 	> > pc remotely to troubleshoot the problem customer is 
> facing from the 
> 	> > call.
> 	> >  - Accessing medical equipments(medical robotics) 
> remotely and
> 	> > possibly controlling medical equipments to monitor 
> elders in a rural
> 	> > area that (remote care services).
> 	> >  - LAN based gaming protocol to be established peer 
> to peer rather
> 	> > than via gaming server.
> 	> >
> 	> >
> 	> > Discussion Point:
> 	> > - To see interests in documenting what's being 
> deployed based on
> 	> > "draft-saito-mmusic-sdp-ike-04" as an informational 
> document.
> 	> > http://tools.ietf.org/html/draft-saito-mmusic-sdp-ike-04
> 	> > (to be minor-updated to -05 to conform with this 
> charter proposal)
> 	> > - To see interests in standardizing toolset for 
> realizing use-cases
> 	> > similar to what's mentioned above, even 
> contemplating the possibility
> 	> > of defining a guideline for SIP usage where SIP is 
> used only as a
> 	> > rendezvous protocol.
> 	> >
> 	> >
> 	> > Milestone/Deliverables:
> 	> > - Aug 09 Issue Internet-Draft
> 	> >    (by updating "draft-saito-mmusic-sdp-ike-05")
> 	> > - Nov 09 Submit I-D to IESG for publication as an 
> RFC (Informational)
> 	> >
> 	> > --------------------------------------------------
> 	> > Best Regards,
> 	> > Shintaro
> 	> >
> 	> > ----------------------------------------
> 	> > Shintaro MIZUNO
> 	> > mizuno.shintaro@lab.ntt.co.jp
> 	> >
> 	> > NTT Information Sharing Platform Laboratories
> 	> > NTT Corporation
> 	> > ----------------------------------------
> 	> >
> 	> > _______________________________________________
> 	> > dispatch mailing list
> 	> > dispatch@ietf.org
> 	> > https://www.ietf.org/mailman/listinfo/dispatch
> 	>
> 	>
> 	
> 	
> 	----------------------------------------
> 	Shintaro MIZUNO
> 	mizuno.shintaro@lab.ntt.co.jp
> 	
> 	NTT Information Sharing Platform Laboratories
> 	NTT Corporation
> 	----------------------------------------
> 	
> 	_______________________________________________
> 	dispatch mailing list
> 	dispatch@ietf.org
> 	https://www.ietf.org/mailman/listinfo/dispatch
> 	
> 	
> 
> 


From HKaplan@acmepacket.com  Thu Jun 11 11:42:42 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Roni Even <Even.roni@huawei.com>, "'Hutton, Andrew'" <andrew.hutton@siemens-enterprise.com>, 'Alan Johnston' <alan@sipstation.com>, "'Romascanu, Dan (Dan)'" <dromasca@avaya.com>
Date: Thu, 11 Jun 2009 14:42:45 -0400
Thread-Topic: [dispatch] Session recording in SIP
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Cc: 'Leon Portman' <Leon.Portman@nice.com>, "dispatch@ietf.org" <dispatch@ietf.org>, "'Jain, Rajnish'" <Rajnish.Jain@ipc.com>
Subject: Re: [dispatch] Session recording in SIP
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> -----Original Message-----
> From: Roni Even [mailto:Even.roni@huawei.com]
> Sent: Thursday, June 11, 2009 11:16 AM
>=20
> Andy,
> You have a term called communication system in the proposal
> Can you clarify what it is? Is this the end points in the call is this
> some
> type of a proxy, application aware middle box?

Currently the proposed requirements would be to support a SIP UA be the com=
munication system, i.e., the device which performs the replication from an =
active UA-to-UA call to a remote recording system may be either a phone, ga=
teway, or a B2BUA-type middlebox.  Any SIP UA could logically perform that =
role.  I believe today some vendors do it in phones, others in middleboxes.
=20
> There is also a recorder in the proposal, is this a SIP UA or is it just =
a
> RTP terminating device like a "media server" in which case this is simila=
r
> to mediactrl WG work

Well, one possible solution to address the requirements would be to make th=
e recorder in fact a SIP UA. (for example there is a requirement to possibl=
y be able to inject media back into the real UA-to-UA session)

I don't debate that it has obvious parallels with conferencing - I've even =
argued to others that it's almost like a 3-party conf call, but where one o=
r two of the parties don't know it - but it has some other requirements/nee=
ds as well which really don't make it a traditional conference call scenari=
o.

-hadriel


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To: Hadriel Kaplan <HKaplan@acmepacket.com>, Roni Even <Even.roni@huawei.com>,  "'Hutton, Andrew'" <andrew.hutton@siemens-enterprise.com>, "'Alan Johnston'" <alan@sipstation.com>, "'Romascanu, Dan (Dan)'" <dromasca@avaya.com>
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Cc: 'Leon Portman' <Leon.Portman@nice.com>, "dispatch@ietf.org" <dispatch@ietf.org>, "'Jain, Rajnish'" <Rajnish.Jain@ipc.com>
Subject: Re: [dispatch] Session recording in SIP
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At 01:42 PM 6/11/2009, Hadriel Kaplan wrote:
>I don't debate that it has obvious parallels with conferencing - 
>I've even argued to others that it's almost like a 3-party conf 
>call, but where one or two of the parties don't know it -

Hadriel -- this is how I mostly see it also, at least as a foundation 
to work from.

>but it has some other requirements/needs as well which really don't 
>make it a traditional conference call scenario.
>
>-hadriel


From Even.roni@huawei.com  Thu Jun 11 12:55:31 2009
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Date: Thu, 11 Jun 2009 22:53:46 +0300
From: Roni Even <Even.roni@huawei.com>
In-reply-to: <E6C2E8958BA59A4FB960963D475F7AC3193F96D920@mail>
To: 'Hadriel Kaplan' <HKaplan@acmepacket.com>, "'Hutton, Andrew'" <andrew.hutton@siemens-enterprise.com>, 'Alan Johnston' <alan@sipstation.com>, "'Romascanu, Dan (Dan)'" <dromasca@avaya.com>
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Subject: Re: [dispatch] Session recording in SIP
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Hadriel,
I am still confused by the following text which was the reason for the
previous question

"This group will specify requirements for a SIP based protocol interface
between a communications system and a recorder. The Communications System is
responsible for establishing media sessions where the actual business is
conducted. The Recorder is the sink of the recorded media."


Can someone draw a figure showing the talking parties the Communication
system and the recorder and explain where is this interface. According to
your response I can think that there are at least two separate architectures
your are proposing one where the parties (or one of them) in the call know
the recorder and somehow forwards the media to it. The other architecture
has some B2BUA in the call path who is either forking the media to the
recorder and to the parties or the recorder is in the middle of the media
path.

My humble opinion is that a preferred solution will use a B2BUA in the call
path who will initiate the recording and this is why I was looking at XCON
or MEDIACTRL work in order to not develop a new protocol but leverage
existing ones.
Thanks
Roni 



-----Original Message-----
From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com] 
Sent: Thursday, June 11, 2009 9:43 PM
To: Roni Even; 'Hutton, Andrew'; 'Alan Johnston'; 'Romascanu, Dan (Dan)'
Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'
Subject: RE: [dispatch] Session recording in SIP



> -----Original Message-----
> From: Roni Even [mailto:Even.roni@huawei.com]
> Sent: Thursday, June 11, 2009 11:16 AM
> 
> Andy,
> You have a term called communication system in the proposal
> Can you clarify what it is? Is this the end points in the call is this
> some
> type of a proxy, application aware middle box?

Currently the proposed requirements would be to support a SIP UA be the
communication system, i.e., the device which performs the replication from
an active UA-to-UA call to a remote recording system may be either a phone,
gateway, or a B2BUA-type middlebox.  Any SIP UA could logically perform that
role.  I believe today some vendors do it in phones, others in middleboxes.
 
> There is also a recorder in the proposal, is this a SIP UA or is it just a
> RTP terminating device like a "media server" in which case this is similar
> to mediactrl WG work

Well, one possible solution to address the requirements would be to make the
recorder in fact a SIP UA. (for example there is a requirement to possibly
be able to inject media back into the real UA-to-UA session)

I don't debate that it has obvious parallels with conferencing - I've even
argued to others that it's almost like a 3-party conf call, but where one or
two of the parties don't know it - but it has some other requirements/needs
as well which really don't make it a traditional conference call scenario.

-hadriel


From HKaplan@acmepacket.com  Thu Jun 11 15:51:24 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Roni Even <Even.roni@huawei.com>, "'Hutton, Andrew'" <andrew.hutton@siemens-enterprise.com>, 'Alan Johnston' <alan@sipstation.com>, "'Romascanu, Dan (Dan)'" <dromasca@avaya.com>
Date: Thu, 11 Jun 2009 18:51:28 -0400
Thread-Topic: [dispatch] Session recording in SIP
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Cc: 'Leon Portman' <Leon.Portman@nice.com>, "dispatch@ietf.org" <dispatch@ietf.org>, "'Jain, Rajnish'" <Rajnish.Jain@ipc.com>
Subject: Re: [dispatch] Session recording in SIP
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Hey Roni,
Being a middlebox man myself, a B2BUA doing the media forking makes the mos=
t sense to me too, but some other folks don't do it/want-it that way.  So t=
he proposed work is to cover both potential models.

The statement you cite reflects the belief that the solution to the require=
ments will be to use SIP between the UA doing the media forking, and the re=
corder doing the collection; and that the work of specification will be to =
define that.

So one diagram could be this, where "TBD" is the To-Be-Defined work:

   +-----+              +-----+
   |     |   Call       |     |
   |UA-A +<------------>+UA-B |
   |     |              |     |
   +-++--+              +-----+
      \\
       \\
    TBD \\
         \\+----------+
          \+          |
           + Recorder |
           |          |
           +----------+

Or this:



+-----+              +-----+              +-----+
|     |     Call     |     |   Call       |     |
|UA-A +<------------>+B2BUA+<------------>+UA-B |
|     |              |     |              |     |
+-----+              +-++--+              +-----+
                        \\
                         \\
                      TBD \\
                           \\+----------+
                            \+          |
                             + Recorder |
                             |          |
                             +----------+

-hadriel

> -----Original Message-----
> From: Roni Even [mailto:Even.roni@huawei.com]
> Sent: Thursday, June 11, 2009 3:54 PM
> To: Hadriel Kaplan; 'Hutton, Andrew'; 'Alan Johnston'; 'Romascanu, Dan
> (Dan)'
> Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'
> Subject: RE: [dispatch] Session recording in SIP
>=20
> Hadriel,
> I am still confused by the following text which was the reason for the
> previous question
>=20
> "This group will specify requirements for a SIP based protocol interface
> between a communications system and a recorder. The Communications System
> is
> responsible for establishing media sessions where the actual business is
> conducted. The Recorder is the sink of the recorded media."
>=20
>=20
> Can someone draw a figure showing the talking parties the Communication
> system and the recorder and explain where is this interface. According to
> your response I can think that there are at least two separate
> architectures
> your are proposing one where the parties (or one of them) in the call kno=
w
> the recorder and somehow forwards the media to it. The other architecture
> has some B2BUA in the call path who is either forking the media to the
> recorder and to the parties or the recorder is in the middle of the media
> path.
>=20


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Subject: Re: [dispatch] Session recording in SIP
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Hadriel Kaplan wrote:
> Hey Roni,
> Being a middlebox man myself, a B2BUA doing the media forking makes the most sense to me too, but some other folks don't do it/want-it that way.  So the proposed work is to cover both potential models.
> 
> The statement you cite reflects the belief that the solution to the requirements will be to use SIP between the UA doing the media forking, and the recorder doing the collection; and that the work of specification will be to define that.
> 
> So one diagram could be this, where "TBD" is the To-Be-Defined work:
> 
>    +-----+              +-----+
>    |     |   Call       |     |
>    |UA-A +<------------>+UA-B |
>    |     |              |     |
>    +-++--+              +-----+
>       \\
>        \\
>     TBD \\
>          \\+----------+
>           \+          |
>            + Recorder |
>            |          |
>            +----------+

This model actually can make a lot of sense in *some* circumstances.
For instance, in a call center, where you may have control over what 
components are used throughout, it could make sense to do the media 
forking for recording and monitoring right at the agent phone.

But in those sorts of monitoring/recording we don't talk about in IETF, 
its essential that the recorded parties not be able to distinguish calls 
that are monitored from those that are not. Anything that requires 
cooperation by one of the endpoints is a no-go. Even something as simple 
as a middlebox replacing one media port with another in order to 
initiate monitoring might be enough to clue in the monitored parties. So 
  in those cases you had better have a way that looks exactly the same 
whether recording is happening or not, which generally requires 
something like your picture below.

	Thanks,
	Paul

> Or this:
> 
> 
> 
> +-----+              +-----+              +-----+
> |     |     Call     |     |   Call       |     |
> |UA-A +<------------>+B2BUA+<------------>+UA-B |
> |     |              |     |              |     |
> +-----+              +-++--+              +-----+
>                         \\
>                          \\
>                       TBD \\
>                            \\+----------+
>                             \+          |
>                              + Recorder |
>                              |          |
>                              +----------+
> 
> -hadriel
> 
>> -----Original Message-----
>> From: Roni Even [mailto:Even.roni@huawei.com]
>> Sent: Thursday, June 11, 2009 3:54 PM
>> To: Hadriel Kaplan; 'Hutton, Andrew'; 'Alan Johnston'; 'Romascanu, Dan
>> (Dan)'
>> Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'
>> Subject: RE: [dispatch] Session recording in SIP
>>
>> Hadriel,
>> I am still confused by the following text which was the reason for the
>> previous question
>>
>> "This group will specify requirements for a SIP based protocol interface
>> between a communications system and a recorder. The Communications System
>> is
>> responsible for establishing media sessions where the actual business is
>> conducted. The Recorder is the sink of the recorded media."
>>
>>
>> Can someone draw a figure showing the talking parties the Communication
>> system and the recorder and explain where is this interface. According to
>> your response I can think that there are at least two separate
>> architectures
>> your are proposing one where the parties (or one of them) in the call know
>> the recorder and somehow forwards the media to it. The other architecture
>> has some B2BUA in the call path who is either forking the media to the
>> recorder and to the parties or the recorder is in the middle of the media
>> path.
>>
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

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Subject: Re: [dispatch] Session recording in SIP
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At 06:11 PM 6/11/2009, Paul Kyzivat wrote:


>Hadriel Kaplan wrote:
>>Hey Roni,
>>Being a middlebox man myself, a B2BUA doing the media forking makes 
>>the most sense to me too, but some other folks don't do it/want-it 
>>that way.  So the proposed work is to cover both potential models.
>>The statement you cite reflects the belief that the solution to the 
>>requirements will be to use SIP between the UA doing the media 
>>forking, and the recorder doing the collection; and that the work 
>>of specification will be to define that.
>>So one diagram could be this, where "TBD" is the To-Be-Defined work:
>>    +-----+              +-----+
>>    |     |   Call       |     |
>>    |UA-A +<------------>+UA-B |
>>    |     |              |     |
>>    +-++--+              +-----+
>>       \\
>>        \\
>>     TBD \\
>>          \\+----------+
>>           \+          |
>>            + Recorder |
>>            |          |
>>            +----------+
>
>This model actually can make a lot of sense in *some* circumstances.
>For instance, in a call center, where you may have control over what 
>components are used throughout, it could make sense to do the media 
>forking for recording and monitoring right at the agent phone.

Paul - how is your call center example *not* better accomplished with 
the figure below instead of the figure above?

It seems to me that any enterprise based call center would have to 
have a SIP server of some form (i.e., the caller is not going to have 
a pt-to-pt SIP established session directly with a call center call 
taker). It seems much easier if that call center wants to record 
calls (even some of the calls, that to the caller, the B2BUA is the 
UAS, meaning the sRTP keys only have to be between the caller and the 
B2BUA, and not all the way to the call taker.  In this scenario, the 
B2BUA merely replicates the RTP stream out towards the recording 
device with RTCP in tact, an can allow the B2BUA provide any meta 
data about the specifics of the call signaling it wants to also store.

This scenario also means the caller is never aware the call is being 
recorded, creating anonymity wrt that facet of the solution.

I'm not advocating B2BUAs here, I just see this as a better solution 
for call center type recording than pt-to-pt between caller and call taker.


>But in those sorts of monitoring/recording we don't talk about in 
>IETF, its essential that the recorded parties not be able to 
>distinguish calls that are monitored from those that are not. 
>Anything that requires cooperation by one of the endpoints is a 
>no-go. Even something as simple as a middlebox replacing one media 
>port with another in order to initiate monitoring might be enough to 
>clue in the monitored parties. So  in those cases you had better 
>have a way that looks exactly the same whether recording is 
>happening or not, which generally requires something like your picture below.
>
>         Thanks,
>         Paul
>
>>Or this:
>>
>>+-----+              +-----+              +-----+
>>|     |     Call     |     |   Call       |     |
>>|UA-A +<------------>+B2BUA+<------------>+UA-B |
>>|     |              |     |              |     |
>>+-----+              +-++--+              +-----+
>>                         \\
>>                          \\
>>                       TBD \\
>>                            \\+----------+
>>                             \+          |
>>                              + Recorder |
>>                              |          |
>>                              +----------+
>>-hadriel
>>
>>>-----Original Message-----
>>>From: Roni Even [mailto:Even.roni@huawei.com]
>>>Sent: Thursday, June 11, 2009 3:54 PM
>>>To: Hadriel Kaplan; 'Hutton, Andrew'; 'Alan Johnston'; 'Romascanu, Dan
>>>(Dan)'
>>>Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'
>>>Subject: RE: [dispatch] Session recording in SIP
>>>
>>>Hadriel,
>>>I am still confused by the following text which was the reason for the
>>>previous question
>>>
>>>"This group will specify requirements for a SIP based protocol interface
>>>between a communications system and a recorder. The Communications System
>>>is
>>>responsible for establishing media sessions where the actual business is
>>>conducted. The Recorder is the sink of the recorded media."
>>>
>>>
>>>Can someone draw a figure showing the talking parties the Communication
>>>system and the recorder and explain where is this interface. According to
>>>your response I can think that there are at least two separate
>>>architectures
>>>your are proposing one where the parties (or one of them) in the call know
>>>the recorder and somehow forwards the media to it. The other architecture
>>>has some B2BUA in the call path who is either forking the media to the
>>>recorder and to the parties or the recorder is in the middle of the media
>>>path.
>>_______________________________________________
>>dispatch mailing list
>>dispatch@ietf.org
>>https://www.ietf.org/mailman/listinfo/dispatch
>_______________________________________________
>dispatch mailing list
>dispatch@ietf.org
>https://www.ietf.org/mailman/listinfo/dispatch


From alan@sipstation.com  Thu Jun 11 20:39:17 2009
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From: Alan Johnston <alan@sipstation.com>
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References: <4A2ECDB2.7000601@alcatel-lucent.com>	<EDC652A26FB23C4EB6384A4584434A04017901E7@307622ANEX5.global.avaya.com>	<4A2FB842.7050302@sipstation.com>	<028a01c9ea12$5ec96e60$1c5c4b20$%roni@huawei.com>	<719F9BBB3E7E71428A2D13F3D76C768F01C7BBE8@GBNTHT12009MSX.gb002.siemens.net>	<002801c9ea82$0da51320$28ef3960$%roni@huawei.com>	<719F9BBB3E7E71428A2D13F3D76C768F01C7BE97@GBNTHT12009MSX.gb002.siemens.net>	<004a01c9eaa7$94ef7990$bece6cb0$%roni@huawei.com>	<E6C2E8958BA59A4FB960963D475F7AC3193F96D920@mail>	<00ab01c9eace$58b7c190$0a2744b0$%roni@huawei.com>	<E6C2E8958BA59A4FB960963D475F7AC31940E5668D@mail>	<4A318F1E.9080308@cisco.com> <XFE-SJC-21271mrvoGd000002ea@xfe-sjc-212.amer.cisco.com>
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Cc: 'Leon Portman' <Leon.Portman@nice.com>, "dispatch@ietf.org" <dispatch@ietf.org>, "'Jain, Rajnish'" <Rajnish.Jain@ipc.com>, Roni Even <Even.roni@huawei.com>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [dispatch] Session recording in SIP
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These various models and modes of recording are described in

      http://tools.ietf.org/html/draft-wing-sipping-srtp-key-04#section-5

To summarize some of the arguments for forking the media in the middle 
as opposed to doing it in an endpoint:

1.  Requirements to use basic UAs without any extensions
2.  Not always having double the media bandwidth in endpoints, 
especially when they are remote
3.  The recording mode (always on, on demand, required, pause and 
resume) and policy is usually controlled centrally, and so another 
protocol would be needed to tell the UA to start and stop recording.

Thanks,
Alan

James M. Polk wrote:
> At 06:11 PM 6/11/2009, Paul Kyzivat wrote:
>
>
>> Hadriel Kaplan wrote:
>>> Hey Roni,
>>> Being a middlebox man myself, a B2BUA doing the media forking makes 
>>> the most sense to me too, but some other folks don't do it/want-it 
>>> that way.  So the proposed work is to cover both potential models.
>>> The statement you cite reflects the belief that the solution to the 
>>> requirements will be to use SIP between the UA doing the media 
>>> forking, and the recorder doing the collection; and that the work of 
>>> specification will be to define that.
>>> So one diagram could be this, where "TBD" is the To-Be-Defined work:
>>>    +-----+              +-----+
>>>    |     |   Call       |     |
>>>    |UA-A +<------------>+UA-B |
>>>    |     |              |     |
>>>    +-++--+              +-----+
>>>       \\
>>>        \\
>>>     TBD \\
>>>          \\+----------+
>>>           \+          |
>>>            + Recorder |
>>>            |          |
>>>            +----------+
>>
>> This model actually can make a lot of sense in *some* circumstances.
>> For instance, in a call center, where you may have control over what 
>> components are used throughout, it could make sense to do the media 
>> forking for recording and monitoring right at the agent phone.
>
> Paul - how is your call center example *not* better accomplished with 
> the figure below instead of the figure above?
>
> It seems to me that any enterprise based call center would have to 
> have a SIP server of some form (i.e., the caller is not going to have 
> a pt-to-pt SIP established session directly with a call center call 
> taker). It seems much easier if that call center wants to record calls 
> (even some of the calls, that to the caller, the B2BUA is the UAS, 
> meaning the sRTP keys only have to be between the caller and the 
> B2BUA, and not all the way to the call taker.  In this scenario, the 
> B2BUA merely replicates the RTP stream out towards the recording 
> device with RTCP in tact, an can allow the B2BUA provide any meta data 
> about the specifics of the call signaling it wants to also store.
>
> This scenario also means the caller is never aware the call is being 
> recorded, creating anonymity wrt that facet of the solution.
>
> I'm not advocating B2BUAs here, I just see this as a better solution 
> for call center type recording than pt-to-pt between caller and call 
> taker.
>
>
>> But in those sorts of monitoring/recording we don't talk about in 
>> IETF, its essential that the recorded parties not be able to 
>> distinguish calls that are monitored from those that are not. 
>> Anything that requires cooperation by one of the endpoints is a 
>> no-go. Even something as simple as a middlebox replacing one media 
>> port with another in order to initiate monitoring might be enough to 
>> clue in the monitored parties. So  in those cases you had better have 
>> a way that looks exactly the same whether recording is happening or 
>> not, which generally requires something like your picture below.
>>
>>         Thanks,
>>         Paul
>>
>>> Or this:
>>>
>>> +-----+              +-----+              +-----+
>>> |     |     Call     |     |   Call       |     |
>>> |UA-A +<------------>+B2BUA+<------------>+UA-B |
>>> |     |              |     |              |     |
>>> +-----+              +-++--+              +-----+
>>>                         \\
>>>                          \\
>>>                       TBD \\
>>>                            \\+----------+
>>>                             \+          |
>>>                              + Recorder |
>>>                              |          |
>>>                              +----------+
>>> -hadriel
>>>
>>>> -----Original Message-----
>>>> From: Roni Even [mailto:Even.roni@huawei.com]
>>>> Sent: Thursday, June 11, 2009 3:54 PM
>>>> To: Hadriel Kaplan; 'Hutton, Andrew'; 'Alan Johnston'; 'Romascanu, Dan
>>>> (Dan)'
>>>> Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'
>>>> Subject: RE: [dispatch] Session recording in SIP
>>>>
>>>> Hadriel,
>>>> I am still confused by the following text which was the reason for the
>>>> previous question
>>>>
>>>> "This group will specify requirements for a SIP based protocol 
>>>> interface
>>>> between a communications system and a recorder. The Communications 
>>>> System
>>>> is
>>>> responsible for establishing media sessions where the actual 
>>>> business is
>>>> conducted. The Recorder is the sink of the recorded media."
>>>>
>>>>
>>>> Can someone draw a figure showing the talking parties the 
>>>> Communication
>>>> system and the recorder and explain where is this interface. 
>>>> According to
>>>> your response I can think that there are at least two separate
>>>> architectures
>>>> your are proposing one where the parties (or one of them) in the 
>>>> call know
>>>> the recorder and somehow forwards the media to it. The other 
>>>> architecture
>>>> has some B2BUA in the call path who is either forking the media to the
>>>> recorder and to the parties or the recorder is in the middle of the 
>>>> media
>>>> path.
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>


From AUDET@nortel.com  Thu Jun 11 20:42:51 2009
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Subject: Re: [dispatch] Session recording in SIP
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I agree with James.

There are *some* scenarios where the UA-initiated recording will work, but
they are very limited. They have the following disadvantages:
1- It makes the client "aware" that it is being recorded
2- It makes it very easy to avoid being recorded by interrupting the media
stream
3 - It makes it much easier to "dispute" (i.e., lawyers) the authenticity of
a recorded conversation if there is potential for significant differences in
the streams as seen by different ends
4 - It doubles the bandwidth requirements on the UA, which can be a problem
in some scenarios
5 - It doubles the processing power required in the client (think SRTP/key
negotiation & al). This can be a problem with many phones.
6 - They require phones that actually support this. Which means, you can't
use "normal" SIP phones. Which means,  if you get the $30 phone, you don't
get tapped into, but if you get the $300 phone, you don't.


But UA-initiated recording has very few advantages:
1 - Less work on servers. This is actually a big advantage because lots of
servers are so complex, it is difficult to do anything new on them.
2 - Looks much more abstract and cool in an IETF RFC :-)


On Jun11 2009 19:13 , "James M. Polk" <jmpolk@cisco.com> wrote:

> At 06:11 PM 6/11/2009, Paul Kyzivat wrote:
> 
> 
>> Hadriel Kaplan wrote:
>>> Hey Roni,
>>> Being a middlebox man myself, a B2BUA doing the media forking makes
>>> the most sense to me too, but some other folks don't do it/want-it
>>> that way.  So the proposed work is to cover both potential models.
>>> The statement you cite reflects the belief that the solution to the
>>> requirements will be to use SIP between the UA doing the media
>>> forking, and the recorder doing the collection; and that the work
>>> of specification will be to define that.
>>> So one diagram could be this, where "TBD" is the To-Be-Defined work:
>>>    +-----+              +-----+
>>>    |     |   Call       |     |
>>>    |UA-A +<------------>+UA-B |
>>>    |     |              |     |
>>>    +-++--+              +-----+
>>>       \\
>>>        \\
>>>     TBD \\
>>>          \\+----------+
>>>           \+          |
>>>            + Recorder |
>>>            |          |
>>>            +----------+
>> 
>> This model actually can make a lot of sense in *some* circumstances.
>> For instance, in a call center, where you may have control over what
>> components are used throughout, it could make sense to do the media
>> forking for recording and monitoring right at the agent phone.
> 
> Paul - how is your call center example *not* better accomplished with
> the figure below instead of the figure above?
> 
> It seems to me that any enterprise based call center would have to
> have a SIP server of some form (i.e., the caller is not going to have
> a pt-to-pt SIP established session directly with a call center call
> taker). It seems much easier if that call center wants to record
> calls (even some of the calls, that to the caller, the B2BUA is the
> UAS, meaning the sRTP keys only have to be between the caller and the
> B2BUA, and not all the way to the call taker.  In this scenario, the
> B2BUA merely replicates the RTP stream out towards the recording
> device with RTCP in tact, an can allow the B2BUA provide any meta
> data about the specifics of the call signaling it wants to also store.
> 
> This scenario also means the caller is never aware the call is being
> recorded, creating anonymity wrt that facet of the solution.
> 
> I'm not advocating B2BUAs here, I just see this as a better solution
> for call center type recording than pt-to-pt between caller and call taker.
> 
> 
>> But in those sorts of monitoring/recording we don't talk about in
>> IETF, its essential that the recorded parties not be able to
>> distinguish calls that are monitored from those that are not.
>> Anything that requires cooperation by one of the endpoints is a
>> no-go. Even something as simple as a middlebox replacing one media
>> port with another in order to initiate monitoring might be enough to
>> clue in the monitored parties. So  in those cases you had better
>> have a way that looks exactly the same whether recording is
>> happening or not, which generally requires something like your picture below.
>> 
>>         Thanks,
>>         Paul
>> 
>>> Or this:
>>> 
>>> +-----+              +-----+              +-----+
>>> |     |     Call     |     |   Call       |     |
>>> |UA-A +<------------>+B2BUA+<------------>+UA-B |
>>> |     |              |     |              |     |
>>> +-----+              +-++--+              +-----+
>>>                         \\
>>>                          \\
>>>                       TBD \\
>>>                            \\+----------+
>>>                             \+          |
>>>                              + Recorder |
>>>                              |          |
>>>                              +----------+
>>> -hadriel
>>> 
>>>> -----Original Message-----
>>>> From: Roni Even [mailto:Even.roni@huawei.com]
>>>> Sent: Thursday, June 11, 2009 3:54 PM
>>>> To: Hadriel Kaplan; 'Hutton, Andrew'; 'Alan Johnston'; 'Romascanu, Dan
>>>> (Dan)'
>>>> Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'
>>>> Subject: RE: [dispatch] Session recording in SIP
>>>> 
>>>> Hadriel,
>>>> I am still confused by the following text which was the reason for the
>>>> previous question
>>>> 
>>>> "This group will specify requirements for a SIP based protocol interface
>>>> between a communications system and a recorder. The Communications System
>>>> is
>>>> responsible for establishing media sessions where the actual business is
>>>> conducted. The Recorder is the sink of the recorded media."
>>>> 
>>>> 
>>>> Can someone draw a figure showing the talking parties the Communication
>>>> system and the recorder and explain where is this interface. According to
>>>> your response I can think that there are at least two separate
>>>> architectures
>>>> your are proposing one where the parties (or one of them) in the call know
>>>> the recorder and somehow forwards the media to it. The other architecture
>>>> has some B2BUA in the call path who is either forking the media to the
>>>> recorder and to the parties or the recorder is in the middle of the media
>>>> path.
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


From HKaplan@acmepacket.com  Thu Jun 11 20:55:11 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Francois Audet <audet@nortel.com>, "James M. Polk" <jmpolk@cisco.com>, Paul Kyzivat <pkyzivat@cisco.com>
Date: Thu, 11 Jun 2009 23:54:26 -0400
Thread-Topic: [dispatch] Session recording in SIP
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Subject: Re: [dispatch] Session recording in SIP
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Sure there are advantages/disadvantages to each model, but why bother argui=
ng over it?  I think we probably all agree we need to support at least a B2=
BUA model, so once you have to standardize that doesn't the UA model come f=
or virtually free?  (i.e., I would presume the details of a B2BUA model are=
 either a superset or the same as a single UA model, not less)

-hadriel

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Cc: 'Leon Portman' <Leon.Portman@nice.com>, "dispatch@ietf.org" <dispatch@ietf.org>, "'Jain, Rajnish'" <Rajnish.Jain@ipc.com>, Roni Even <Even.roni@huawei.com>
Subject: Re: [dispatch] Session recording in SIP
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I'm not sure the UA model does "come for free", give there needs to 
be new code telling the UA to all of a sudden replicate the media 
stream (if it can) back out (which may or may not have the BW or the 
reachability to the sink device).  If the original media is sRTP, 
does this (or not) mean sRTP needs to be used to the media sink? That 
takes more negotiation (and time). Does the sink always trust what 
comes at it from any UA? It can be configured much more easily to 
trust certain (limited number of) servers that can send it media to record.

I could be wet behind the ears, but I don't believe the UA model is 
low hanging fruit if the server model is specified. I (think I) see 
each as completely separate solutions, requiring however much work to 
do regardless of whether the other is done too (other than having the 
recording sink specified/developed -- that part is free once it is 
done for either model).

This is an interesting problem space, and I think we should look into 
it. I think there will be more than we expect once we peel away the layers.

James

At 10:54 PM 6/11/2009, Hadriel Kaplan wrote:

>Sure there are advantages/disadvantages to each model, but why 
>bother arguing over it?  I think we probably all agree we need to 
>support at least a B2BUA model, so once you have to standardize that 
>doesn't the UA model come for virtually free?  (i.e., I would 
>presume the details of a B2BUA model are either a superset or the 
>same as a single UA model, not less)
>
>-hadriel


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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "James M. Polk" <jmpolk@cisco.com>, Francois Audet <audet@nortel.com>, Paul Kyzivat <pkyzivat@cisco.com>
Date: Fri, 12 Jun 2009 01:35:34 -0400
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Subject: Re: [dispatch] Session recording in SIP
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> -----Original Message-----
> From: James M. Polk [mailto:jmpolk@cisco.com]
> Sent: Friday, June 12, 2009 1:06 AM
>=20
> I'm not sure the UA model does "come for free", give there needs to
> be new code telling the UA to all of a sudden replicate the media
> stream (if it can) back out (which may or may not have the BW or the
> reachability to the sink device).  If the original media is sRTP,
> does this (or not) mean sRTP needs to be used to the media sink? That
> takes more negotiation (and time). Does the sink always trust what
> comes at it from any UA? It can be configured much more easily to
> trust certain (limited number of) servers that can send it media to
> record.

When I say "for free" I sure don't mean code-wise. :)
I meant from a protocol specification perspective.  Any protocol issue I ca=
n imagine of a UA would be likewise true of a B2BUA in any spec we define. =
 Can you think of any that aren't?

For example, in some recording applications, a B2BUA has to tear down the c=
all (or not allow it to begin with) if it does not have sufficient bandwidt=
h or reachability to the recorder.  Though I'm not sure the spec has to say=
 much about that - seems more of a local policy decision not affecting inte=
roperability.

Likewise any authentication of devices and protection of replicated media w=
ill undoubtedly need to be covered in any RFC we try to do, no matter the m=
odel, if nothing else than to get past the IESG. (and in practice I've been=
 told there are some cases where the recorders are not in the same domain o=
f the servers either, so having such measures is appropriate)  I have no do=
ubt that in real-world practice many folks who deploy a server model would =
just skip most of that, but I doubt we'll be able to avoid it in an RFC.

-hadriel

From steffen.fries@siemens.com  Thu Jun 11 23:48:19 2009
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From: "Fries, Steffen" <steffen.fries@siemens.com>
To: "James M. Polk" <jmpolk@cisco.com>, "Hadriel Kaplan" <HKaplan@acmepacket.com>, "Francois Audet" <audet@nortel.com>, "Paul Kyzivat" <pkyzivat@cisco.com>
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Subject: Re: [dispatch] Session recording in SIP
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Hi,=20

> If the original media is sRTP, does this (or not) mean sRTP=20
> needs to be used to the media sink? That takes more=20
> negotiation (and time). Does the sink always trust what comes=20
> at it from any UA? It can be configured much more easily to=20
> trust certain (limited number of) servers that can send it=20
> media to record.
That is one point we adressed in
http://tools.ietf.org/html/draft-wing-sipping-srtp-key-04#page-15 by
using a setup, were the call recording is done on the media way and the
client just discloses the session key. This saves an extra encryption
between the client and the recorder.

Steffen

>=20
> I could be wet behind the ears, but I don't believe the UA=20
> model is low hanging fruit if the server model is specified.=20
> I (think I) see each as completely separate solutions,=20
> requiring however much work to do regardless of whether the=20
> other is done too (other than having the recording sink=20
> specified/developed -- that part is free once it is done for=20
> either model).
>=20
> This is an interesting problem space, and I think we should=20
> look into it. I think there will be more than we expect once=20
> we peel away the layers.
>=20
> James
>=20
> At 10:54 PM 6/11/2009, Hadriel Kaplan wrote:
>=20
> >Sure there are advantages/disadvantages to each model, but=20
> why bother=20
> >arguing over it?  I think we probably all agree we need to=20
> support at=20
> >least a B2BUA model, so once you have to standardize that=20
> doesn't the=20
> >UA model come for virtually free?  (i.e., I would presume=20
> the details=20
> >of a B2BUA model are either a superset or the same as a single UA=20
> >model, not less)
> >
> >-hadriel
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

From Even.roni@huawei.com  Fri Jun 12 00:23:09 2009
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Subject: Re: [dispatch] Session recording in SIP
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Hadriel,
I think that there may be a different protocol if you use the B2BUA
solution. In this case it will make more sense to use a "mediactrl"
direction where SIP is used to open a control channel to the recorder but
the recording protocol itself runs in the  control channel. This control
channel will be also used to deliver the meta data for the recording. So
maybe the sentence  " SIP based protocol interface between a communications
system and a recorder" is limiting the solution space.


Roni Even


-----Original Message-----
From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com] 
Sent: Friday, June 12, 2009 8:36 AM
To: James M. Polk; Francois Audet; Paul Kyzivat
Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'; Roni Even
Subject: RE: [dispatch] Session recording in SIP



> -----Original Message-----
> From: James M. Polk [mailto:jmpolk@cisco.com]
> Sent: Friday, June 12, 2009 1:06 AM
> 
> I'm not sure the UA model does "come for free", give there needs to
> be new code telling the UA to all of a sudden replicate the media
> stream (if it can) back out (which may or may not have the BW or the
> reachability to the sink device).  If the original media is sRTP,
> does this (or not) mean sRTP needs to be used to the media sink? That
> takes more negotiation (and time). Does the sink always trust what
> comes at it from any UA? It can be configured much more easily to
> trust certain (limited number of) servers that can send it media to
> record.

When I say "for free" I sure don't mean code-wise. :)
I meant from a protocol specification perspective.  Any protocol issue I can
imagine of a UA would be likewise true of a B2BUA in any spec we define.
Can you think of any that aren't?

For example, in some recording applications, a B2BUA has to tear down the
call (or not allow it to begin with) if it does not have sufficient
bandwidth or reachability to the recorder.  Though I'm not sure the spec has
to say much about that - seems more of a local policy decision not affecting
interoperability.

Likewise any authentication of devices and protection of replicated media
will undoubtedly need to be covered in any RFC we try to do, no matter the
model, if nothing else than to get past the IESG. (and in practice I've been
told there are some cases where the recorders are not in the same domain of
the servers either, so having such measures is appropriate)  I have no doubt
that in real-world practice many folks who deploy a server model would just
skip most of that, but I doubt we'll be able to avoid it in an RFC.

-hadriel


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Date: Fri, 12 Jun 2009 10:27:16 +0300
From: Roni Even <Even.roni@huawei.com>
In-reply-to: <XFE-SJC-21271mrvoGd000002ea@xfe-sjc-212.amer.cisco.com>
To: "'James M. Polk'" <jmpolk@cisco.com>, 'Paul Kyzivat' <pkyzivat@cisco.com>,  'Hadriel Kaplan' <HKaplan@acmepacket.com>
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Cc: 'Leon Portman' <Leon.Portman@nice.com>, dispatch@ietf.org, "'Jain, Rajnish'" <Rajnish.Jain@ipc.com>
Subject: Re: [dispatch] Session recording in SIP
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James,
The B2BUA does not need to see the RTP/RTCP stream they can go directly from
the source to the recorder 




+-----+              +-----+              +-----+
|     |     Call     |     |   Call       |     |
|UA-A +<------------>+B2BUA+<------------>+UA-B |
|     |              |     |              |     |
+-----+              +-++--+              +-----+
    .                   \\                   .
      .                  \\ control          .
         .  RTP       TBD \\                 .RTP
             .             \\+----------+    .
                .           \+          |..... 
                      .      + Recorder |
                             |          |
                             +----------+

Roni



-----Original Message-----
From: James M. Polk [mailto:jmpolk@cisco.com] 
Sent: Friday, June 12, 2009 5:13 AM
To: Paul Kyzivat; Hadriel Kaplan
Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'; Roni Even
Subject: Re: [dispatch] Session recording in SIP

At 06:11 PM 6/11/2009, Paul Kyzivat wrote:


>Hadriel Kaplan wrote:
>>Hey Roni,
>>Being a middlebox man myself, a B2BUA doing the media forking makes 
>>the most sense to me too, but some other folks don't do it/want-it 
>>that way.  So the proposed work is to cover both potential models.
>>The statement you cite reflects the belief that the solution to the 
>>requirements will be to use SIP between the UA doing the media 
>>forking, and the recorder doing the collection; and that the work 
>>of specification will be to define that.
>>So one diagram could be this, where "TBD" is the To-Be-Defined work:
>>    +-----+              +-----+
>>    |     |   Call       |     |
>>    |UA-A +<------------>+UA-B |
>>    |     |              |     |
>>    +-++--+              +-----+
>>       \\
>>        \\
>>     TBD \\
>>          \\+----------+
>>           \+          |
>>            + Recorder |
>>            |          |
>>            +----------+
>
>This model actually can make a lot of sense in *some* circumstances.
>For instance, in a call center, where you may have control over what 
>components are used throughout, it could make sense to do the media 
>forking for recording and monitoring right at the agent phone.

Paul - how is your call center example *not* better accomplished with 
the figure below instead of the figure above?

It seems to me that any enterprise based call center would have to 
have a SIP server of some form (i.e., the caller is not going to have 
a pt-to-pt SIP established session directly with a call center call 
taker). It seems much easier if that call center wants to record 
calls (even some of the calls, that to the caller, the B2BUA is the 
UAS, meaning the sRTP keys only have to be between the caller and the 
B2BUA, and not all the way to the call taker.  In this scenario, the 
B2BUA merely replicates the RTP stream out towards the recording 
device with RTCP in tact, an can allow the B2BUA provide any meta 
data about the specifics of the call signaling it wants to also store.

This scenario also means the caller is never aware the call is being 
recorded, creating anonymity wrt that facet of the solution.

I'm not advocating B2BUAs here, I just see this as a better solution 
for call center type recording than pt-to-pt between caller and call taker.


>But in those sorts of monitoring/recording we don't talk about in 
>IETF, its essential that the recorded parties not be able to 
>distinguish calls that are monitored from those that are not. 
>Anything that requires cooperation by one of the endpoints is a 
>no-go. Even something as simple as a middlebox replacing one media 
>port with another in order to initiate monitoring might be enough to 
>clue in the monitored parties. So  in those cases you had better 
>have a way that looks exactly the same whether recording is 
>happening or not, which generally requires something like your picture
below.
>
>         Thanks,
>         Paul
>
>>Or this:
>>
>>+-----+              +-----+              +-----+
>>|     |     Call     |     |   Call       |     |
>>|UA-A +<------------>+B2BUA+<------------>+UA-B |
>>|     |              |     |              |     |
>>+-----+              +-++--+              +-----+
>>                         \\
>>                          \\
>>                       TBD \\
>>                            \\+----------+
>>                             \+          |
>>                              + Recorder |
>>                              |          |
>>                              +----------+
>>-hadriel
>>
>>>-----Original Message-----
>>>From: Roni Even [mailto:Even.roni@huawei.com]
>>>Sent: Thursday, June 11, 2009 3:54 PM
>>>To: Hadriel Kaplan; 'Hutton, Andrew'; 'Alan Johnston'; 'Romascanu, Dan
>>>(Dan)'
>>>Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'
>>>Subject: RE: [dispatch] Session recording in SIP
>>>
>>>Hadriel,
>>>I am still confused by the following text which was the reason for the
>>>previous question
>>>
>>>"This group will specify requirements for a SIP based protocol interface
>>>between a communications system and a recorder. The Communications System
>>>is
>>>responsible for establishing media sessions where the actual business is
>>>conducted. The Recorder is the sink of the recorded media."
>>>
>>>
>>>Can someone draw a figure showing the talking parties the Communication
>>>system and the recorder and explain where is this interface. According to
>>>your response I can think that there are at least two separate
>>>architectures
>>>your are proposing one where the parties (or one of them) in the call
know
>>>the recorder and somehow forwards the media to it. The other architecture
>>>has some B2BUA in the call path who is either forking the media to the
>>>recorder and to the parties or the recorder is in the middle of the media
>>>path.
>>_______________________________________________
>>dispatch mailing list
>>dispatch@ietf.org
>>https://www.ietf.org/mailman/listinfo/dispatch
>_______________________________________________
>dispatch mailing list
>dispatch@ietf.org
>https://www.ietf.org/mailman/listinfo/dispatch


From drage@alcatel-lucent.com  Fri Jun 12 05:00:12 2009
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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: Francois Audet <audet@nortel.com>, "James M. Polk" <jmpolk@cisco.com>, Paul Kyzivat <pkyzivat@cisco.com>, Hadriel Kaplan <HKaplan@acmepacket.com>
Date: Fri, 12 Jun 2009 13:59:50 +0200
Thread-Topic: [dispatch] Session recording in SIP
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Cc: 'Leon Portman' <Leon.Portman@nice.com>, "dispatch@ietf.org" <dispatch@ietf.org>, "'Jain, Rajnish'" <Rajnish.Jain@ipc.com>, Roni Even <Even.roni@huawei.com>
Subject: Re: [dispatch] Session recording in SIP
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I have no response to any of the other points, but please note that for 1),=
 in the scenarios of call centres, some jurisdictions do require the client=
 to be informed that the call is being recorded. Normally this is done by a=
 verbal announcement.

So in those use cases, 1) is a non-argument.

regards

Keith=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Francois Audet
> Sent: Friday, June 12, 2009 4:43 AM
> To: James M. Polk; Paul Kyzivat; Hadriel Kaplan
> Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'; Roni Even
> Subject: Re: [dispatch] Session recording in SIP
>=20
> I agree with James.
>=20
> There are *some* scenarios where the UA-initiated recording=20
> will work, but they are very limited. They have the following=20
> disadvantages:
> 1- It makes the client "aware" that it is being recorded
> 2- It makes it very easy to avoid being recorded by=20
> interrupting the media stream
> 3 - It makes it much easier to "dispute" (i.e., lawyers) the=20
> authenticity of a recorded conversation if there is potential=20
> for significant differences in the streams as seen by different ends
> 4 - It doubles the bandwidth requirements on the UA, which=20
> can be a problem in some scenarios
> 5 - It doubles the processing power required in the client=20
> (think SRTP/key negotiation & al). This can be a problem with=20
> many phones.
> 6 - They require phones that actually support this. Which=20
> means, you can't use "normal" SIP phones. Which means,  if=20
> you get the $30 phone, you don't get tapped into, but if you=20
> get the $300 phone, you don't.
>=20
>=20
> But UA-initiated recording has very few advantages:
> 1 - Less work on servers. This is actually a big advantage=20
> because lots of servers are so complex, it is difficult to do=20
> anything new on them.
> 2 - Looks much more abstract and cool in an IETF RFC :-)
>=20
>=20
> On Jun11 2009 19:13 , "James M. Polk" <jmpolk@cisco.com> wrote:
>=20
> > At 06:11 PM 6/11/2009, Paul Kyzivat wrote:
> >=20
> >=20
> >> Hadriel Kaplan wrote:
> >>> Hey Roni,
> >>> Being a middlebox man myself, a B2BUA doing the media=20
> forking makes=20
> >>> the most sense to me too, but some other folks don't do=20
> it/want-it=20
> >>> that way.  So the proposed work is to cover both potential models.
> >>> The statement you cite reflects the belief that the=20
> solution to the=20
> >>> requirements will be to use SIP between the UA doing the media=20
> >>> forking, and the recorder doing the collection; and that=20
> the work of=20
> >>> specification will be to define that.
> >>> So one diagram could be this, where "TBD" is the=20
> To-Be-Defined work:
> >>>    +-----+              +-----+
> >>>    |     |   Call       |     |
> >>>    |UA-A +<------------>+UA-B |
> >>>    |     |              |     |
> >>>    +-++--+              +-----+
> >>>       \\
> >>>        \\
> >>>     TBD \\
> >>>          \\+----------+
> >>>           \+          |
> >>>            + Recorder |
> >>>            |          |
> >>>            +----------+
> >>=20
> >> This model actually can make a lot of sense in *some*=20
> circumstances.
> >> For instance, in a call center, where you may have control=20
> over what=20
> >> components are used throughout, it could make sense to do=20
> the media=20
> >> forking for recording and monitoring right at the agent phone.
> >=20
> > Paul - how is your call center example *not* better=20
> accomplished with=20
> > the figure below instead of the figure above?
> >=20
> > It seems to me that any enterprise based call center would have to=20
> > have a SIP server of some form (i.e., the caller is not=20
> going to have=20
> > a pt-to-pt SIP established session directly with a call center call=20
> > taker). It seems much easier if that call center wants to=20
> record calls=20
> > (even some of the calls, that to the caller, the B2BUA is the UAS,=20
> > meaning the sRTP keys only have to be between the caller and the=20
> > B2BUA, and not all the way to the call taker.  In this=20
> scenario, the=20
> > B2BUA merely replicates the RTP stream out towards the recording=20
> > device with RTCP in tact, an can allow the B2BUA provide=20
> any meta data=20
> > about the specifics of the call signaling it wants to also store.
> >=20
> > This scenario also means the caller is never aware the call=20
> is being=20
> > recorded, creating anonymity wrt that facet of the solution.
> >=20
> > I'm not advocating B2BUAs here, I just see this as a better=20
> solution=20
> > for call center type recording than pt-to-pt between caller=20
> and call taker.
> >=20
> >=20
> >> But in those sorts of monitoring/recording we don't talk about in=20
> >> IETF, its essential that the recorded parties not be able to=20
> >> distinguish calls that are monitored from those that are not.
> >> Anything that requires cooperation by one of the endpoints is a=20
> >> no-go. Even something as simple as a middlebox replacing one media=20
> >> port with another in order to initiate monitoring might be=20
> enough to=20
> >> clue in the monitored parties. So  in those cases you had=20
> better have=20
> >> a way that looks exactly the same whether recording is=20
> happening or=20
> >> not, which generally requires something like your picture below.
> >>=20
> >>         Thanks,
> >>         Paul
> >>=20
> >>> Or this:
> >>>=20
> >>> +-----+              +-----+              +-----+
> >>> |     |     Call     |     |   Call       |     |
> >>> |UA-A +<------------>+B2BUA+<------------>+UA-B |
> >>> |     |              |     |              |     |
> >>> +-----+              +-++--+              +-----+
> >>>                         \\
> >>>                          \\
> >>>                       TBD \\
> >>>                            \\+----------+
> >>>                             \+          |
> >>>                              + Recorder |
> >>>                              |          |
> >>>                              +----------+ -hadriel
> >>>=20
> >>>> -----Original Message-----
> >>>> From: Roni Even [mailto:Even.roni@huawei.com]
> >>>> Sent: Thursday, June 11, 2009 3:54 PM
> >>>> To: Hadriel Kaplan; 'Hutton, Andrew'; 'Alan Johnston';=20
> 'Romascanu,=20
> >>>> Dan (Dan)'
> >>>> Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'
> >>>> Subject: RE: [dispatch] Session recording in SIP
> >>>>=20
> >>>> Hadriel,
> >>>> I am still confused by the following text which was the=20
> reason for=20
> >>>> the previous question
> >>>>=20
> >>>> "This group will specify requirements for a SIP based protocol=20
> >>>> interface between a communications system and a recorder. The=20
> >>>> Communications System is responsible for establishing media=20
> >>>> sessions where the actual business is conducted. The Recorder is=20
> >>>> the sink of the recorded media."
> >>>>=20
> >>>>=20
> >>>> Can someone draw a figure showing the talking parties the=20
> >>>> Communication system and the recorder and explain where is this=20
> >>>> interface. According to your response I can think that=20
> there are at=20
> >>>> least two separate architectures your are proposing one=20
> where the=20
> >>>> parties (or one of them) in the call know the recorder=20
> and somehow=20
> >>>> forwards the media to it. The other architecture has=20
> some B2BUA in=20
> >>>> the call path who is either forking the media to the=20
> recorder and=20
> >>>> to the parties or the recorder is in the middle of the=20
> media path.
> >>> _______________________________________________
> >>> dispatch mailing list
> >>> dispatch@ietf.org
> >>> https://www.ietf.org/mailman/listinfo/dispatch
> >> _______________________________________________
> >> dispatch mailing list
> >> dispatch@ietf.org
> >> https://www.ietf.org/mailman/listinfo/dispatch
> >=20
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> =

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Folks,

we have had significant email discussions on the SIP CLF proposal:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00176.html

We also set up a dedicated mailing list for this work:
https://www.ietf.org/mailman/listinfo/sip-clf

Based on the discussions up to now, there seems to be enough interest 
and energy to work on this. Our proposal is to form a new WG with a 
charter based on the proposal above. If there are no objections, we will 
  proceed with the formation of the new WG and will not allocate any 
agenda time for this in DISPATCH in Stockholm. Only if we could not 
reach consensus on the list, we would discuss it face to face in 
Stockholm. As we have pointed out a few times, the model we want to 
follow in DISPATCH is to only use face-to-face time for things that 
cannot be agreed on the list.

Comments?

Thanks,

Gonzalo
DISPATCH co-chair

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From: Paul Kyzivat <pkyzivat@cisco.com>
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References: <4A2ECDB2.7000601@alcatel-lucent.com> <EDC652A26FB23C4EB6384A4584434A04017901E7@307622ANEX5.global.avaya.com> <4A2FB842.7050302@sipstation.com> <028a01c9ea12$5ec96e60$1c5c4b20$%roni@huawei.com> <719F9BBB3E7E71428A2D13F3D76C768F01C7BBE8@GBNTHT12009MSX.gb002.siemens.net> <002801c9ea82$0da51320$28ef3960$%roni@huawei.com> <719F9BBB3E7E71428A2D13F3D76C768F01C7BE97@GBNTHT12009MSX.gb002.siemens.net> <004a01c9eaa7$94ef7990$bece6cb0$%roni@huawei.com> <E6C2E8958BA59A4FB960963D475F7AC3193F96D920@mail> <00ab01c9eace$58b7c190$0a2744b0$%roni@huawei.com> <E6C2E8958BA59A4FB960963D475F7AC31940E5668D@mail> <4A318F1E.9080308@cisco.com> <XFE-SJC-21271mrvoGd000002ea@xfe-sjc-212.amer.cisco.com>
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Cc: 'Leon Portman' <Leon.Portman@nice.com>, "dispatch@ietf.org" <dispatch@ietf.org>, "'Jain, Rajnish'" <Rajnish.Jain@ipc.com>, Roni Even <Even.roni@huawei.com>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [dispatch] Session recording in SIP
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James M. Polk wrote:
> At 06:11 PM 6/11/2009, Paul Kyzivat wrote:
> 
> 
>> Hadriel Kaplan wrote:
>>> Hey Roni,
>>> Being a middlebox man myself, a B2BUA doing the media forking makes 
>>> the most sense to me too, but some other folks don't do it/want-it 
>>> that way.  So the proposed work is to cover both potential models.
>>> The statement you cite reflects the belief that the solution to the 
>>> requirements will be to use SIP between the UA doing the media 
>>> forking, and the recorder doing the collection; and that the work of 
>>> specification will be to define that.
>>> So one diagram could be this, where "TBD" is the To-Be-Defined work:
>>>    +-----+              +-----+
>>>    |     |   Call       |     |
>>>    |UA-A +<------------>+UA-B |
>>>    |     |              |     |
>>>    +-++--+              +-----+
>>>       \\
>>>        \\
>>>     TBD \\
>>>          \\+----------+
>>>           \+          |
>>>            + Recorder |
>>>            |          |
>>>            +----------+
>>
>> This model actually can make a lot of sense in *some* circumstances.
>> For instance, in a call center, where you may have control over what 
>> components are used throughout, it could make sense to do the media 
>> forking for recording and monitoring right at the agent phone.
> 
> Paul - how is your call center example *not* better accomplished with 
> the figure below instead of the figure above?
> 
> It seems to me that any enterprise based call center would have to have 
> a SIP server of some form (i.e., the caller is not going to have a 
> pt-to-pt SIP established session directly with a call center call 
> taker). It seems much easier if that call center wants to record calls 
> (even some of the calls, that to the caller, the B2BUA is the UAS, 
> meaning the sRTP keys only have to be between the caller and the B2BUA, 
> and not all the way to the call taker.  In this scenario, the B2BUA 
> merely replicates the RTP stream out towards the recording device with 
> RTCP in tact, an can allow the B2BUA provide any meta data about the 
> specifics of the call signaling it wants to also store.
> 
> This scenario also means the caller is never aware the call is being 
> recorded, creating anonymity wrt that facet of the solution.
> 
> I'm not advocating B2BUAs here, I just see this as a better solution for 
> call center type recording than pt-to-pt between caller and call taker.

There are pros and cons.
I agree that many implemenations *will* have a central server in any 
case. Routing *all* the media through a central server has resource 
costs, especially if you then require redundancy as well. Often forking 
media at the phone can be "free" in that it is well within the 
capabilities of even a cheap sip phone.

The point is that there are complex architectural tradeoffs here based 
on individual priorities, so it makes sense to consider both as viable.

>> But in those sorts of monitoring/recording we don't talk about in 
>> IETF, its essential that the recorded parties not be able to 
>> distinguish calls that are monitored from those that are not. Anything 
>> that requires cooperation by one of the endpoints is a no-go. Even 
>> something as simple as a middlebox replacing one media port with 
>> another in order to initiate monitoring might be enough to clue in the 
>> monitored parties. So  in those cases you had better have a way that 
>> looks exactly the same whether recording is happening or not, which 
>> generally requires something like your picture below.
>>
>>         Thanks,
>>         Paul
>>
>>> Or this:
>>>
>>> +-----+              +-----+              +-----+
>>> |     |     Call     |     |   Call       |     |
>>> |UA-A +<------------>+B2BUA+<------------>+UA-B |
>>> |     |              |     |              |     |
>>> +-----+              +-++--+              +-----+
>>>                         \\
>>>                          \\
>>>                       TBD \\
>>>                            \\+----------+
>>>                             \+          |
>>>                              + Recorder |
>>>                              |          |
>>>                              +----------+
>>> -hadriel
>>>
>>>> -----Original Message-----
>>>> From: Roni Even [mailto:Even.roni@huawei.com]
>>>> Sent: Thursday, June 11, 2009 3:54 PM
>>>> To: Hadriel Kaplan; 'Hutton, Andrew'; 'Alan Johnston'; 'Romascanu, Dan
>>>> (Dan)'
>>>> Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'
>>>> Subject: RE: [dispatch] Session recording in SIP
>>>>
>>>> Hadriel,
>>>> I am still confused by the following text which was the reason for the
>>>> previous question
>>>>
>>>> "This group will specify requirements for a SIP based protocol 
>>>> interface
>>>> between a communications system and a recorder. The Communications 
>>>> System
>>>> is
>>>> responsible for establishing media sessions where the actual 
>>>> business is
>>>> conducted. The Recorder is the sink of the recorded media."
>>>>
>>>>
>>>> Can someone draw a figure showing the talking parties the Communication
>>>> system and the recorder and explain where is this interface. 
>>>> According to
>>>> your response I can think that there are at least two separate
>>>> architectures
>>>> your are proposing one where the parties (or one of them) in the 
>>>> call know
>>>> the recorder and somehow forwards the media to it. The other 
>>>> architecture
>>>> has some B2BUA in the call path who is either forking the media to the
>>>> recorder and to the parties or the recorder is in the middle of the 
>>>> media
>>>> path.
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
> 
> 

From christer.holmberg@ericsson.com  Fri Jun 12 06:22:02 2009
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References: <4A2ECDB2.7000601@alcatel-lucent.com><EDC652A26FB23C4EB6384A4584434A04017901E7@307622ANEX5.global.avaya.com><4A2FB842.7050302@sipstation.com><028a01c9ea12$5ec96e60$1c5c4b20$%roni@huawei.com><719F9BBB3E7E71428A2D13F3D76C768F01C7BBE8@GBNTHT12009MSX.gb002.siemens.net><002801c9ea82$0da51320$28ef3960$%roni@huawei.com><719F9BBB3E7E71428A2D13F3D76C768F01C7BE97@GBNTHT12009MSX.gb002.siemens.net><004a01c9eaa7$94ef7990$bece6cb0$%roni@huawei.com><E6C2E8958BA59A4FB960963D475F7AC3193F96D920@mail><00ab01c9eace$58b7c190$0a2744b0$%roni@huawei.com><E6C2E8958BA59A4FB960963D475F7AC31940E5668D@mail><4A318F1E.9080308@cisco.com><XFE-SJC-21271mrvoGd000002ea@xfe-sjc-212.amer.cisco.com> <4A325424.9020901@cisco.com>
From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>, "James M. Polk" <jmpolk@cisco.com>
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Cc: Leon Portman <Leon.Portman@nice.com>, dispatch@ietf.org, "Jain, Rajnish" <Rajnish.Jain@ipc.com>, Hadriel Kaplan <HKaplan@acmepacket.com>, Roni Even <Even.roni@huawei.com>
Subject: Re: [dispatch] Session recording in SIP
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Hi,=20

>There are pros and cons.
>I agree that many implemenations *will* have a central server=20
>in any case. Routing *all* the media through a central server=20
>has resource costs, especially if you then require redundancy=20
>as well. Often forking media at the phone can be "free" in=20
>that it is well within the capabilities of even a cheap sip phone.

It may have impacts on battery- and/or CPU limited devices.

In addtion media forking at the phone may have resource cost on the
access network, if you have limited resources on your access network and
both the UA-to-UA and UA-to-recorder media has to be sent over that
access network.

Regards,

Christer




>=20
> The point is that there are complex architectural tradeoffs=20
> here based on individual priorities, so it makes sense to=20
> consider both as viable.
>=20
> >> But in those sorts of monitoring/recording we don't talk about in=20
> >> IETF, its essential that the recorded parties not be able to=20
> >> distinguish calls that are monitored from those that are not.=20
> >> Anything that requires cooperation by one of the endpoints is a=20
> >> no-go. Even something as simple as a middlebox replacing one media=20
> >> port with another in order to initiate monitoring might be=20
> enough to=20
> >> clue in the monitored parties. So  in those cases you had=20
> better have=20
> >> a way that looks exactly the same whether recording is=20
> happening or=20
> >> not, which generally requires something like your picture below.
> >>
> >>         Thanks,
> >>         Paul
> >>
> >>> Or this:
> >>>
> >>> +-----+              +-----+              +-----+
> >>> |     |     Call     |     |   Call       |     |
> >>> |UA-A +<------------>+B2BUA+<------------>+UA-B |
> >>> |     |              |     |              |     |
> >>> +-----+              +-++--+              +-----+
> >>>                         \\
> >>>                          \\
> >>>                       TBD \\
> >>>                            \\+----------+
> >>>                             \+          |
> >>>                              + Recorder |
> >>>                              |          |
> >>>                              +----------+ -hadriel
> >>>
> >>>> -----Original Message-----
> >>>> From: Roni Even [mailto:Even.roni@huawei.com]
> >>>> Sent: Thursday, June 11, 2009 3:54 PM
> >>>> To: Hadriel Kaplan; 'Hutton, Andrew'; 'Alan Johnston';=20
> 'Romascanu,=20
> >>>> Dan (Dan)'
> >>>> Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'
> >>>> Subject: RE: [dispatch] Session recording in SIP
> >>>>
> >>>> Hadriel,
> >>>> I am still confused by the following text which was the=20
> reason for=20
> >>>> the previous question
> >>>>
> >>>> "This group will specify requirements for a SIP based protocol=20
> >>>> interface between a communications system and a recorder. The=20
> >>>> Communications System is responsible for establishing media=20
> >>>> sessions where the actual business is conducted. The Recorder is=20
> >>>> the sink of the recorded media."
> >>>>
> >>>>
> >>>> Can someone draw a figure showing the talking parties the=20
> >>>> Communication system and the recorder and explain where=20
> is this interface.
> >>>> According to
> >>>> your response I can think that there are at least two separate=20
> >>>> architectures your are proposing one where the parties=20
> (or one of=20
> >>>> them) in the call know the recorder and somehow forwards=20
> the media=20
> >>>> to it. The other architecture has some B2BUA in the call=20
> path who=20
> >>>> is either forking the media to the recorder and to the=20
> parties or=20
> >>>> the recorder is in the middle of the media path.
> >>> _______________________________________________
> >>> dispatch mailing list
> >>> dispatch@ietf.org
> >>> https://www.ietf.org/mailman/listinfo/dispatch
> >> _______________________________________________
> >> dispatch mailing list
> >> dispatch@ietf.org
> >> https://www.ietf.org/mailman/listinfo/dispatch
> >=20
> >=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

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From: "Francois Audet" <audet@nortel.com>
To: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>, "James M. Polk" <jmpolk@cisco.com>, "Paul Kyzivat" <pkyzivat@cisco.com>, "Hadriel Kaplan" <HKaplan@acmepacket.com>
Cc: Leon Portman <Leon.Portman@nice.com>, dispatch@ietf.org, "Jain, Rajnish" <Rajnish.Jain@ipc.com>, Roni Even <Even.roni@huawei.com>
Subject: Re: [dispatch] Session recording in SIP
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You are correct.=20

> -----Original Message-----
> From: DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]=20
> Sent: Friday, June 12, 2009 05:00
> To: Audet, Francois (SC100:3055); James M. Polk; Paul=20
> Kyzivat; Hadriel Kaplan
> Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'; Roni Even
> Subject: RE: [dispatch] Session recording in SIP
>=20
> I have no response to any of the other points, but please=20
> note that for 1), in the scenarios of call centres, some=20
> jurisdictions do require the client to be informed that the=20
> call is being recorded. Normally this is done by a verbal=20
> announcement.
>=20
> So in those use cases, 1) is a non-argument.
>=20
> regards
>=20
> Keith=20
>=20
> > -----Original Message-----
> > From: dispatch-bounces@ietf.org
> > [mailto:dispatch-bounces@ietf.org] On Behalf Of Francois Audet
> > Sent: Friday, June 12, 2009 4:43 AM
> > To: James M. Polk; Paul Kyzivat; Hadriel Kaplan
> > Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'; Roni Even
> > Subject: Re: [dispatch] Session recording in SIP
> >=20
> > I agree with James.
> >=20
> > There are *some* scenarios where the UA-initiated recording=20
> will work,=20
> > but they are very limited. They have the following
> > disadvantages:
> > 1- It makes the client "aware" that it is being recorded
> > 2- It makes it very easy to avoid being recorded by=20
> interrupting the=20
> > media stream
> > 3 - It makes it much easier to "dispute" (i.e., lawyers) the=20
> > authenticity of a recorded conversation if there is potential for=20
> > significant differences in the streams as seen by different ends
> > 4 - It doubles the bandwidth requirements on the UA, which can be a=20
> > problem in some scenarios
> > 5 - It doubles the processing power required in the client (think=20
> > SRTP/key negotiation & al). This can be a problem with many phones.
> > 6 - They require phones that actually support this. Which=20
> means, you=20
> > can't use "normal" SIP phones. Which means,  if you get the=20
> $30 phone,=20
> > you don't get tapped into, but if you get the $300 phone, you don't.
> >=20
> >=20
> > But UA-initiated recording has very few advantages:
> > 1 - Less work on servers. This is actually a big advantage because=20
> > lots of servers are so complex, it is difficult to do=20
> anything new on=20
> > them.
> > 2 - Looks much more abstract and cool in an IETF RFC :-)
> >=20
> >=20
> > On Jun11 2009 19:13 , "James M. Polk" <jmpolk@cisco.com> wrote:
> >=20
> > > At 06:11 PM 6/11/2009, Paul Kyzivat wrote:
> > >=20
> > >=20
> > >> Hadriel Kaplan wrote:
> > >>> Hey Roni,
> > >>> Being a middlebox man myself, a B2BUA doing the media
> > forking makes
> > >>> the most sense to me too, but some other folks don't do
> > it/want-it
> > >>> that way.  So the proposed work is to cover both=20
> potential models.
> > >>> The statement you cite reflects the belief that the
> > solution to the
> > >>> requirements will be to use SIP between the UA doing the media=20
> > >>> forking, and the recorder doing the collection; and that
> > the work of
> > >>> specification will be to define that.
> > >>> So one diagram could be this, where "TBD" is the
> > To-Be-Defined work:
> > >>>    +-----+              +-----+
> > >>>    |     |   Call       |     |
> > >>>    |UA-A +<------------>+UA-B |
> > >>>    |     |              |     |
> > >>>    +-++--+              +-----+
> > >>>       \\
> > >>>        \\
> > >>>     TBD \\
> > >>>          \\+----------+
> > >>>           \+          |
> > >>>            + Recorder |
> > >>>            |          |
> > >>>            +----------+
> > >>=20
> > >> This model actually can make a lot of sense in *some*
> > circumstances.
> > >> For instance, in a call center, where you may have control
> > over what
> > >> components are used throughout, it could make sense to do
> > the media
> > >> forking for recording and monitoring right at the agent phone.
> > >=20
> > > Paul - how is your call center example *not* better
> > accomplished with
> > > the figure below instead of the figure above?
> > >=20
> > > It seems to me that any enterprise based call center=20
> would have to=20
> > > have a SIP server of some form (i.e., the caller is not
> > going to have
> > > a pt-to-pt SIP established session directly with a call=20
> center call=20
> > > taker). It seems much easier if that call center wants to
> > record calls
> > > (even some of the calls, that to the caller, the B2BUA is=20
> the UAS,=20
> > > meaning the sRTP keys only have to be between the caller and the=20
> > > B2BUA, and not all the way to the call taker.  In this
> > scenario, the
> > > B2BUA merely replicates the RTP stream out towards the recording=20
> > > device with RTCP in tact, an can allow the B2BUA provide
> > any meta data
> > > about the specifics of the call signaling it wants to also store.
> > >=20
> > > This scenario also means the caller is never aware the call
> > is being
> > > recorded, creating anonymity wrt that facet of the solution.
> > >=20
> > > I'm not advocating B2BUAs here, I just see this as a better
> > solution
> > > for call center type recording than pt-to-pt between caller
> > and call taker.
> > >=20
> > >=20
> > >> But in those sorts of monitoring/recording we don't talk=20
> about in=20
> > >> IETF, its essential that the recorded parties not be able to=20
> > >> distinguish calls that are monitored from those that are not.
> > >> Anything that requires cooperation by one of the endpoints is a=20
> > >> no-go. Even something as simple as a middlebox replacing=20
> one media=20
> > >> port with another in order to initiate monitoring might be
> > enough to
> > >> clue in the monitored parties. So  in those cases you had
> > better have
> > >> a way that looks exactly the same whether recording is
> > happening or
> > >> not, which generally requires something like your picture below.
> > >>=20
> > >>         Thanks,
> > >>         Paul
> > >>=20
> > >>> Or this:
> > >>>=20
> > >>> +-----+              +-----+              +-----+
> > >>> |     |     Call     |     |   Call       |     |
> > >>> |UA-A +<------------>+B2BUA+<------------>+UA-B |
> > >>> |     |              |     |              |     |
> > >>> +-----+              +-++--+              +-----+
> > >>>                         \\
> > >>>                          \\
> > >>>                       TBD \\
> > >>>                            \\+----------+
> > >>>                             \+          |
> > >>>                              + Recorder |
> > >>>                              |          |
> > >>>                              +----------+ -hadriel
> > >>>=20
> > >>>> -----Original Message-----
> > >>>> From: Roni Even [mailto:Even.roni@huawei.com]
> > >>>> Sent: Thursday, June 11, 2009 3:54 PM
> > >>>> To: Hadriel Kaplan; 'Hutton, Andrew'; 'Alan Johnston';
> > 'Romascanu,
> > >>>> Dan (Dan)'
> > >>>> Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'
> > >>>> Subject: RE: [dispatch] Session recording in SIP
> > >>>>=20
> > >>>> Hadriel,
> > >>>> I am still confused by the following text which was the
> > reason for
> > >>>> the previous question
> > >>>>=20
> > >>>> "This group will specify requirements for a SIP based protocol=20
> > >>>> interface between a communications system and a recorder. The=20
> > >>>> Communications System is responsible for establishing media=20
> > >>>> sessions where the actual business is conducted. The=20
> Recorder is=20
> > >>>> the sink of the recorded media."
> > >>>>=20
> > >>>>=20
> > >>>> Can someone draw a figure showing the talking parties the=20
> > >>>> Communication system and the recorder and explain=20
> where is this=20
> > >>>> interface. According to your response I can think that
> > there are at
> > >>>> least two separate architectures your are proposing one
> > where the
> > >>>> parties (or one of them) in the call know the recorder
> > and somehow
> > >>>> forwards the media to it. The other architecture has
> > some B2BUA in
> > >>>> the call path who is either forking the media to the
> > recorder and
> > >>>> to the parties or the recorder is in the middle of the
> > media path.
> > >>> _______________________________________________
> > >>> dispatch mailing list
> > >>> dispatch@ietf.org
> > >>> https://www.ietf.org/mailman/listinfo/dispatch
> > >> _______________________________________________
> > >> dispatch mailing list
> > >> dispatch@ietf.org
> > >> https://www.ietf.org/mailman/listinfo/dispatch
> > >=20
> > > _______________________________________________
> > > dispatch mailing list
> > > dispatch@ietf.org
> > > https://www.ietf.org/mailman/listinfo/dispatch
> >=20
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> >=20
>=20

From dworley@nortel.com  Fri Jun 12 10:06:41 2009
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From: "Dale Worley" <dworley@nortel.com>
To: DISPATCH <dispatch@ietf.org>
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Date: Fri, 12 Jun 2009 13:06:28 -0400
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Subject: [dispatch] Incompatibility between policy dataset framework and	media-policy-dataset
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Currently, the only defined policy schema within the
draft-ietf-sipping-profile-datasets framework is
draft-ietf-sipping-media-policy-dataset.  Unfortunately, the
media-policy-dataset includes elements that cannot be modeled with the
profile-datasets framework.

Consider the example in section 7.1 of
draft-ietf-sipping-media-policy-dataset-07:

  <property-set>
    <session-policy>
      <context>
        <policy-server-URI>policy@biloxi.example.com</policy-server-URI>
        <contact>sip:policy_manager@example.com</contact>
        <info>Access network policies</info>
      </context>
      <media-types excluded-policy="disallow">
        <media-type policy="allow">audio</media-type>
        <media-type policy="allow">video</media-type>
      </media-types>
      <codecs excluded-policy="allow">
        <codec policy="disallow">
          <mime-type>audio/G729</mime-type>
        </codec>
        <codec policy="disallow">
          <mime-type>audio/G723</mime-type>
        </codec>
      </codecs>
    </session-policy>
  </property-set>

This dataset contains bindings of the properties "media-types" and
"codecs".  But it also contains a <context> element that does not fit
within the system of the policy-dataset framework.

One way to remove this inconsistency is to remove the <context> from
within the dataset, and rather have the <context> and the dataset be
siblings within the XML:  (This example includes correcting the
top-level element of the dataset to be propertySet, so we can use
<session-policy> for the higher-level element.)

  <session-policy>
    <context>
      <policy-server-URI>policy@biloxi.example.com</policy-server-URI>
      <contact>sip:policy_manager@example.com</contact>
      <info>Access network policies</info>
    </context>
    <propertySet>
      <media-types excluded-policy="disallow">
        <media-type policy="allow">audio</media-type>
        <media-type policy="allow">video</media-type>
      </media-types>
      <codecs excluded-policy="allow">
        <codec policy="disallow">
          <mime-type>audio/G729</mime-type>
        </codec>
        <codec policy="disallow">
          <mime-type>audio/G723</mime-type>
        </codec>
      </codecs>
    </propertySet>
  </session-policy>

Worse, the <streams> element cannot be put within the policy-dataset
framework at all, as it consists of an array-of-structures, which we
have earlier noted is not supported within the framework.

Dale



From fluffy@cisco.com  Fri Jun 12 10:28:03 2009
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From: Cullen Jennings <fluffy@cisco.com>
To: dispatch@ietf.org, IETF AVT WG <avt@ietf.org>, rai@ietf.org
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Subject: [dispatch] CODEC BOF
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The CODEC proposal to form a WG on Internet Wideband Audio Codecs has  
been raised many issues - I think it is better to consider this in the  
context of a formal BOF instead of discussing it in the DISPATCH WG. I  
have approved this BOF for the next meeting but there are many issues  
that have been raised and need discussion before this would be  
approved as a WG.  I encourage people to take those to the codec@ietf.org 
  list.

Please to not reply to this email (particularly since I cross posted  
to three WG), but instead go join the threads on the codec@ietf.org  
mailing list to discuss this.

Thanks,
Cullen <RAI AD>



From fluffy@cisco.com  Fri Jun 12 10:59:35 2009
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From: Cullen Jennings <fluffy@cisco.com>
To: Cullen Jennings <fluffy@cisco.com>
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Cc: dispatch@ietf.org, rai@ietf.org, IETF AVT WG <avt@ietf.org>
Subject: Re: [dispatch] CODEC BOF
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I forgot to add, you can join the codec@ietf list at
https://www.ietf.org/mailman/listinfo/codec


On Jun 12, 2009, at 11:28 , Cullen Jennings wrote:

>
> The CODEC proposal to form a WG on Internet Wideband Audio Codecs  
> has been raised many issues - I think it is better to consider this  
> in the context of a formal BOF instead of discussing it in the  
> DISPATCH WG. I have approved this BOF for the next meeting but there  
> are many issues that have been raised and need discussion before  
> this would be approved as a WG.  I encourage people to take those to  
> the codec@ietf.org list.
>
> Please to not reply to this email (particularly since I cross posted  
> to three WG), but instead go join the threads on the codec@ietf.org  
> mailing list to discuss this.
>
> Thanks,
> Cullen <RAI AD>
>
>


From dworley@nortel.com  Fri Jun 12 12:36:29 2009
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Cc: SIP Operations <sip-ops@ietf.org>, RAI DISPATCH <dispatch@ietf.org>
Subject: Re: [dispatch] draft-lawrence-sip-3rd-party-authorization-00
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On Thu, 2009-06-11 at 08:47 -0500, Mary Barnes wrote:
> There seems to be sufficient interest in this topic to warrant f2f
> agenda time in Stockholm. However, additional feedback prior to IETF-75
> is necessary to ensure that progress can be made in Stockholm. 

It looks like a good description of the problem.  Since the I-D is not
to discuss solutions, I'm not sure what there is to discuss until we
open the discussion to solutions.

Dale



From HKaplan@acmepacket.com  Fri Jun 12 12:48:46 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Roni Even <Even.roni@huawei.com>, "'James M. Polk'" <jmpolk@cisco.com>, 'Francois Audet' <audet@nortel.com>, 'Paul Kyzivat' <pkyzivat@cisco.com>
Date: Fri, 12 Jun 2009 15:48:50 -0400
Thread-Topic: [dispatch] Session recording in SIP
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Cc: 'Leon Portman' <Leon.Portman@nice.com>, "dispatch@ietf.org" <dispatch@ietf.org>, "'Jain, Rajnish'" <Rajnish.Jain@ipc.com>
Subject: Re: [dispatch] Session recording in SIP
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I agree that assuming SIP is the protocol between the recordee and the reco=
rder is probably premature, but I still don't see it as making us have to c=
hoose whether to support a UA vs. a B2BUA. =20

I mean even from a logical perspective, a UA could just pretend to be an in=
tegrated B2BUA to itself and who would know?  I.e., this:

            UA-A
   +----------------------+
   | +-----+      +-----+ |            +-----+
   | |     |      |     | | Call       |     |
   | | UA  +<---->+B2BUA+<+----------->+UA-B |
   | |     |      |     | |            |     |
   | +-----+      +-++--+ |            +-----+
   +-----------------\\---+
                      \\
                   TBD \\
                        \\+----------+
                         \+          |
                          + Recorder |
                          |          |
                          +----------+

-hadriel


> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Roni Even
> Sent: Friday, June 12, 2009 3:21 AM
>=20
> I think that there may be a different protocol if you use the B2BUA
> solution. In this case it will make more sense to use a "mediactrl"
> direction where SIP is used to open a control channel to the recorder but
> the recording protocol itself runs in the  control channel. This control
> channel will be also used to deliver the meta data for the recording. So
> maybe the sentence  " SIP based protocol interface between a
> communications
> system and a recorder" is limiting the solution space.
>=20
> Roni Even

From jdrosen@cisco.com  Fri Jun 12 12:50:33 2009
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Cc: 'Leon Portman' <Leon.Portman@nice.com>, "dispatch@ietf.org" <dispatch@ietf.org>, "'Jain, Rajnish'" <Rajnish.Jain@ipc.com>, Hadriel Kaplan <HKaplan@acmepacket.com>, Roni Even <Even.roni@huawei.com>
Subject: Re: [dispatch] Session recording in SIP
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I'll add - from a dispatch perspective - that the liveliness of this 
discussion, and the breadth of opinions/options on architecture - show 
pretty clearly that this is big enough to warrant its own group, rather 
than just another thing to do in AVT or XCON.

-Jonathan R.

DRAGE, Keith (Keith) wrote:
> I have no response to any of the other points, but please note that for 1), in the scenarios of call centres, some jurisdictions do require the client to be informed that the call is being recorded. Normally this is done by a verbal announcement.
> 
> So in those use cases, 1) is a non-argument.
> 
> regards
> 
> Keith 
> 
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org 
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Francois Audet
>> Sent: Friday, June 12, 2009 4:43 AM
>> To: James M. Polk; Paul Kyzivat; Hadriel Kaplan
>> Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'; Roni Even
>> Subject: Re: [dispatch] Session recording in SIP
>>
>> I agree with James.
>>
>> There are *some* scenarios where the UA-initiated recording 
>> will work, but they are very limited. They have the following 
>> disadvantages:
>> 1- It makes the client "aware" that it is being recorded
>> 2- It makes it very easy to avoid being recorded by 
>> interrupting the media stream
>> 3 - It makes it much easier to "dispute" (i.e., lawyers) the 
>> authenticity of a recorded conversation if there is potential 
>> for significant differences in the streams as seen by different ends
>> 4 - It doubles the bandwidth requirements on the UA, which 
>> can be a problem in some scenarios
>> 5 - It doubles the processing power required in the client 
>> (think SRTP/key negotiation & al). This can be a problem with 
>> many phones.
>> 6 - They require phones that actually support this. Which 
>> means, you can't use "normal" SIP phones. Which means,  if 
>> you get the $30 phone, you don't get tapped into, but if you 
>> get the $300 phone, you don't.
>>
>>
>> But UA-initiated recording has very few advantages:
>> 1 - Less work on servers. This is actually a big advantage 
>> because lots of servers are so complex, it is difficult to do 
>> anything new on them.
>> 2 - Looks much more abstract and cool in an IETF RFC :-)
>>
>>
>> On Jun11 2009 19:13 , "James M. Polk" <jmpolk@cisco.com> wrote:
>>
>>> At 06:11 PM 6/11/2009, Paul Kyzivat wrote:
>>>
>>>
>>>> Hadriel Kaplan wrote:
>>>>> Hey Roni,
>>>>> Being a middlebox man myself, a B2BUA doing the media 
>> forking makes 
>>>>> the most sense to me too, but some other folks don't do 
>> it/want-it 
>>>>> that way.  So the proposed work is to cover both potential models.
>>>>> The statement you cite reflects the belief that the 
>> solution to the 
>>>>> requirements will be to use SIP between the UA doing the media 
>>>>> forking, and the recorder doing the collection; and that 
>> the work of 
>>>>> specification will be to define that.
>>>>> So one diagram could be this, where "TBD" is the 
>> To-Be-Defined work:
>>>>>    +-----+              +-----+
>>>>>    |     |   Call       |     |
>>>>>    |UA-A +<------------>+UA-B |
>>>>>    |     |              |     |
>>>>>    +-++--+              +-----+
>>>>>       \\
>>>>>        \\
>>>>>     TBD \\
>>>>>          \\+----------+
>>>>>           \+          |
>>>>>            + Recorder |
>>>>>            |          |
>>>>>            +----------+
>>>> This model actually can make a lot of sense in *some* 
>> circumstances.
>>>> For instance, in a call center, where you may have control 
>> over what 
>>>> components are used throughout, it could make sense to do 
>> the media 
>>>> forking for recording and monitoring right at the agent phone.
>>> Paul - how is your call center example *not* better 
>> accomplished with 
>>> the figure below instead of the figure above?
>>>
>>> It seems to me that any enterprise based call center would have to 
>>> have a SIP server of some form (i.e., the caller is not 
>> going to have 
>>> a pt-to-pt SIP established session directly with a call center call 
>>> taker). It seems much easier if that call center wants to 
>> record calls 
>>> (even some of the calls, that to the caller, the B2BUA is the UAS, 
>>> meaning the sRTP keys only have to be between the caller and the 
>>> B2BUA, and not all the way to the call taker.  In this 
>> scenario, the 
>>> B2BUA merely replicates the RTP stream out towards the recording 
>>> device with RTCP in tact, an can allow the B2BUA provide 
>> any meta data 
>>> about the specifics of the call signaling it wants to also store.
>>>
>>> This scenario also means the caller is never aware the call 
>> is being 
>>> recorded, creating anonymity wrt that facet of the solution.
>>>
>>> I'm not advocating B2BUAs here, I just see this as a better 
>> solution 
>>> for call center type recording than pt-to-pt between caller 
>> and call taker.
>>>
>>>> But in those sorts of monitoring/recording we don't talk about in 
>>>> IETF, its essential that the recorded parties not be able to 
>>>> distinguish calls that are monitored from those that are not.
>>>> Anything that requires cooperation by one of the endpoints is a 
>>>> no-go. Even something as simple as a middlebox replacing one media 
>>>> port with another in order to initiate monitoring might be 
>> enough to 
>>>> clue in the monitored parties. So  in those cases you had 
>> better have 
>>>> a way that looks exactly the same whether recording is 
>> happening or 
>>>> not, which generally requires something like your picture below.
>>>>
>>>>         Thanks,
>>>>         Paul
>>>>
>>>>> Or this:
>>>>>
>>>>> +-----+              +-----+              +-----+
>>>>> |     |     Call     |     |   Call       |     |
>>>>> |UA-A +<------------>+B2BUA+<------------>+UA-B |
>>>>> |     |              |     |              |     |
>>>>> +-----+              +-++--+              +-----+
>>>>>                         \\
>>>>>                          \\
>>>>>                       TBD \\
>>>>>                            \\+----------+
>>>>>                             \+          |
>>>>>                              + Recorder |
>>>>>                              |          |
>>>>>                              +----------+ -hadriel
>>>>>
>>>>>> -----Original Message-----
>>>>>> From: Roni Even [mailto:Even.roni@huawei.com]
>>>>>> Sent: Thursday, June 11, 2009 3:54 PM
>>>>>> To: Hadriel Kaplan; 'Hutton, Andrew'; 'Alan Johnston'; 
>> 'Romascanu, 
>>>>>> Dan (Dan)'
>>>>>> Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'
>>>>>> Subject: RE: [dispatch] Session recording in SIP
>>>>>>
>>>>>> Hadriel,
>>>>>> I am still confused by the following text which was the 
>> reason for 
>>>>>> the previous question
>>>>>>
>>>>>> "This group will specify requirements for a SIP based protocol 
>>>>>> interface between a communications system and a recorder. The 
>>>>>> Communications System is responsible for establishing media 
>>>>>> sessions where the actual business is conducted. The Recorder is 
>>>>>> the sink of the recorded media."
>>>>>>
>>>>>>
>>>>>> Can someone draw a figure showing the talking parties the 
>>>>>> Communication system and the recorder and explain where is this 
>>>>>> interface. According to your response I can think that 
>> there are at 
>>>>>> least two separate architectures your are proposing one 
>> where the 
>>>>>> parties (or one of them) in the call know the recorder 
>> and somehow 
>>>>>> forwards the media to it. The other architecture has 
>> some B2BUA in 
>>>>>> the call path who is either forking the media to the 
>> recorder and 
>>>>>> to the parties or the recorder is in the middle of the 
>> media path.
>>>>> _______________________________________________
>>>>> dispatch mailing list
>>>>> dispatch@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>> _______________________________________________
>>>> dispatch mailing list
>>>> dispatch@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   111 Wood Avenue South
Cisco Fellow                                   Iselin, NJ 08830
Cisco, Voice Technology Group
jdrosen@cisco.com
http://www.jdrosen.net                         PHONE: (408) 902-3084
http://www.cisco.com

From HKaplan@acmepacket.com  Fri Jun 12 13:08:10 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Roni Even <Even.roni@huawei.com>, "'James M. Polk'" <jmpolk@cisco.com>, 'Paul Kyzivat' <pkyzivat@cisco.com>
Date: Fri, 12 Jun 2009 16:08:04 -0400
Thread-Topic: [dispatch] Session recording in SIP
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Cc: 'Leon Portman' <Leon.Portman@nice.com>, "dispatch@ietf.org" <dispatch@ietf.org>, "'Jain, Rajnish'" <Rajnish.Jain@ipc.com>
Subject: Re: [dispatch] Session recording in SIP
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Huh, interesting idea.  I don't see how that could easily work though.  The=
 recorder needs to be able to associate specific audio streams with specifi=
c calls, and from which called/calling party they came.  Unless there's a c=
ontrol channel between each UA and the recorder, you'd have to somehow get =
the B2BUA to know what IP:port each UA can send the forked media from, and =
provide each UA with the addr:port of the Recorder to send the forked media=
 to.  Not to mention if we decide to allow a different codec to be used on =
the forked-media connection vs. the in-call side (which I personally don't =
like the idea of but others do).  At the end of the day, I'd bet you'd have=
 one complicated protocol for all that to occur, with a bunch of re-INVITEs=
/UPDATEs and all kinds of whacked corner-cases with pre-conditions, and IPv=
4/v6, and god-knows-what else.  :)

-hadriel

> -----Original Message-----
> From: Roni Even [mailto:Even.roni@huawei.com]
> Sent: Friday, June 12, 2009 3:27 AM
>=20
> James,
> The B2BUA does not need to see the RTP/RTCP stream they can go directly
> from
> the source to the recorder
>=20
>=20
> +-----+              +-----+              +-----+
> |     |     Call     |     |   Call       |     |
> |UA-A +<------------>+B2BUA+<------------>+UA-B |
> |     |              |     |              |     |
> +-----+              +-++--+              +-----+
>     .                   \\                   .
>       .                  \\ control          .
>          .  RTP       TBD \\                 .RTP
>              .             \\+----------+    .
>                 .           \+          |.....
>                       .      + Recorder |
>                              |          |
>                              +----------+
>=20
> Roni
=20

From mary.barnes@nortel.com  Fri Jun 12 13:52:25 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Dale Worley" <dworley@nortel.com>
Cc: SIP Operations <sip-ops@ietf.org>, RAI DISPATCH <dispatch@ietf.org>
Subject: Re: [dispatch] draft-lawrence-sip-3rd-party-authorization-00
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Correct - we are not discussing solutions because we don't develop
solutions in DISPATCH. =20
The idea in DISPATCH if folks agree that there is a problem, is to then
determine the scope of the problem to be solved - that's not been
entirely concluded for this topic. If that can happen on the ML prior to
Stockholm, then certainly no f2f time is needed - really the ideal
operation of DISPATCH is that we don't need f2f time - i.e., work comes
in, we figure out the scope of the problem, whether there is critical
mass in the WG in terms of contributing to the work and then figure out
whether we need a new WG, a BoF, an individual/AD doc, etc. =20

At this point, for this topic, there has not yet been a critical mass in
terms of feedback on the proposal - that's what we're asking for now.
Basically, we are still dealing with the "...what part or parts of that
problem need to be solved" aspect of this topic.  If we don't get more
input it is highly unlikely that any progress could be made on this
topic in Stockholm, which would put us at the point that you are
suggesting we are now in terms of having nothing to discuss. That was
the implication of my "However...".  I'll be more blunt next time.=20

Thanks,
Mary.=20

-----Original Message-----
From: Worley, Dale (BL60:9D30)=20
Sent: Friday, June 12, 2009 2:37 PM
To: Barnes, Mary (RICH2:AR00)
Cc: RAI DISPATCH; SIP Operations
Subject: Re: [dispatch] draft-lawrence-sip-3rd-party-authorization-00

On Thu, 2009-06-11 at 08:47 -0500, Mary Barnes wrote:
> There seems to be sufficient interest in this topic to warrant f2f=20
> agenda time in Stockholm. However, additional feedback prior to=20
> IETF-75 is necessary to ensure that progress can be made in Stockholm.

It looks like a good description of the problem.  Since the I-D is not
to discuss solutions, I'm not sure what there is to discuss until we
open the discussion to solutions.

Dale



From scott.lawrence@nortel.com  Fri Jun 12 14:04:32 2009
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Subject: Re: [dispatch] Session recording in SIP
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On Fri, 2009-06-12 at 15:50 -0400, Jonathan Rosenberg wrote:
> I'll add - from a dispatch perspective - that the liveliness of this 
> discussion, and the breadth of opinions/options on architecture - show 
> pretty clearly that this is big enough to warrant its own group, rather 
> than just another thing to do in AVT or XCON.

I certainly support getting some clarity around how to do this cleanly.

I'm not crazy about some of the ideas thrown around so far, but the need
is clear.  I also agree with previous posters that while this overlaps
with XCON and some other things, it's really a distinct problem and
would probably be solvable more quickly in a more focused setting.  One
of the major ideas behind our recent reorg was that smaller, more
time-limited WGs will work better than the perpetual ones we've been
using.



From Even.roni@huawei.com  Fri Jun 12 14:09:01 2009
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Date: Sat, 13 Jun 2009 00:07:17 +0300
From: Roni Even <Even.roni@huawei.com>
In-reply-to: <E6C2E8958BA59A4FB960963D475F7AC3194146DE35@mail>
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Subject: Re: [dispatch] Session recording in SIP
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Hadriel,
This is what mediactrl is about. Look in http://tools.ietf.org/html/rfc5567 

Roni

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
Of Hadriel Kaplan
Sent: Friday, June 12, 2009 11:08 PM
To: Roni Even; 'James M. Polk'; 'Paul Kyzivat'
Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'
Subject: Re: [dispatch] Session recording in SIP


Huh, interesting idea.  I don't see how that could easily work though.  The
recorder needs to be able to associate specific audio streams with specific
calls, and from which called/calling party they came.  Unless there's a
control channel between each UA and the recorder, you'd have to somehow get
the B2BUA to know what IP:port each UA can send the forked media from, and
provide each UA with the addr:port of the Recorder to send the forked media
to.  Not to mention if we decide to allow a different codec to be used on
the forked-media connection vs. the in-call side (which I personally don't
like the idea of but others do).  At the end of the day, I'd bet you'd have
one complicated protocol for all that to occur, with a bunch of
re-INVITEs/UPDATEs and all kinds of whacked corner-cases with
pre-conditions, and IPv4/v6, and god-knows-what else.  :)

-hadriel

> -----Original Message-----
> From: Roni Even [mailto:Even.roni@huawei.com]
> Sent: Friday, June 12, 2009 3:27 AM
> 
> James,
> The B2BUA does not need to see the RTP/RTCP stream they can go directly
> from
> the source to the recorder
> 
> 
> +-----+              +-----+              +-----+
> |     |     Call     |     |   Call       |     |
> |UA-A +<------------>+B2BUA+<------------>+UA-B |
> |     |              |     |              |     |
> +-----+              +-++--+              +-----+
>     .                   \\                   .
>       .                  \\ control          .
>          .  RTP       TBD \\                 .RTP
>              .             \\+----------+    .
>                 .           \+          |.....
>                       .      + Recorder |
>                              |          |
>                              +----------+
> 
> Roni
 
_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch


From Even.roni@huawei.com  Fri Jun 12 14:16:35 2009
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Date: Sat, 13 Jun 2009 00:14:53 +0300
From: Roni Even <Even.roni@huawei.com>
In-reply-to: <1244840653.21162.349.camel@scott>
To: 'Scott Lawrence' <scott.lawrence@nortel.com>, dispatch@ietf.org
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Subject: Re: [dispatch] Session recording in SIP
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Hi,
I think that RAI have two wgs working on different aspects of a central
server. One is XCON and the other is Mediactrl. I think that once an
architecture for the recorder is specified it may be just another
application (recording application) that can be based on one of these two
WGs work. 
Recording is just another application for a central server or media server
like conferencing, IVR, and others. Recording was even discussed in
mediactrl but there were no requirement at the time.

Roni Even 

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
Of Scott Lawrence
Sent: Saturday, June 13, 2009 12:04 AM
To: dispatch@ietf.org
Subject: Re: [dispatch] Session recording in SIP

On Fri, 2009-06-12 at 15:50 -0400, Jonathan Rosenberg wrote:
> I'll add - from a dispatch perspective - that the liveliness of this 
> discussion, and the breadth of opinions/options on architecture - show 
> pretty clearly that this is big enough to warrant its own group, rather 
> than just another thing to do in AVT or XCON.

I certainly support getting some clarity around how to do this cleanly.

I'm not crazy about some of the ideas thrown around so far, but the need
is clear.  I also agree with previous posters that while this overlaps
with XCON and some other things, it's really a distinct problem and
would probably be solvable more quickly in a more focused setting.  One
of the major ideas behind our recent reorg was that smaller, more
time-limited WGs will work better than the perpetual ones we've been
using.


_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch


From HKaplan@acmepacket.com  Fri Jun 12 14:54:39 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Roni Even <Even.roni@huawei.com>, "'James M. Polk'" <jmpolk@cisco.com>, 'Paul Kyzivat' <pkyzivat@cisco.com>
Date: Fri, 12 Jun 2009 17:54:19 -0400
Thread-Topic: [dispatch] Session recording in SIP
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Cc: 'Leon Portman' <Leon.Portman@nice.com>, "dispatch@ietf.org" <dispatch@ietf.org>, "'Jain, Rajnish'" <Rajnish.Jain@ipc.com>
Subject: Re: [dispatch] Session recording in SIP
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I was about to write a long-winded reply on how your example was not the sa=
me as mediactrl, but I just realized your drawing was subtly different from=
 what I had thought it implied. :)  The example you're describing is one wh=
ere the recorder is not being sent forked/replicated media content by the U=
A's, but is just put in the media path itself for the real media, so the or=
iginal media itself is going from UA to recorder to UA.=20
Right?

If so, good point - I assume that's one possible deployment model to use.  =
I don't think it's how some (most?) people do it, because they can't or don=
't want to move the media path from where it began, especially for the UA s=
ide that's outside their domain.  (though I dunno much about why they do wh=
at they do)  But it is one interesting solution.

-hadriel


> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Roni Even
> Sent: Friday, June 12, 2009 5:07 PM
> To: Hadriel Kaplan; 'James M. Polk'; 'Paul Kyzivat'
> Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'
> Subject: Re: [dispatch] Session recording in SIP
>=20
> Hadriel,
> This is what mediactrl is about. Look in
> http://tools.ietf.org/html/rfc5567
>=20
> Roni
>=20
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf
> Of Hadriel Kaplan
> Sent: Friday, June 12, 2009 11:08 PM
> To: Roni Even; 'James M. Polk'; 'Paul Kyzivat'
> Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'
> Subject: Re: [dispatch] Session recording in SIP
>=20
>=20
> Huh, interesting idea.  I don't see how that could easily work though.
> The
> recorder needs to be able to associate specific audio streams with
> specific
> calls, and from which called/calling party they came.  Unless there's a
> control channel between each UA and the recorder, you'd have to somehow
> get
> the B2BUA to know what IP:port each UA can send the forked media from, an=
d
> provide each UA with the addr:port of the Recorder to send the forked
> media
> to.  Not to mention if we decide to allow a different codec to be used on
> the forked-media connection vs. the in-call side (which I personally don'=
t
> like the idea of but others do).  At the end of the day, I'd bet you'd
> have
> one complicated protocol for all that to occur, with a bunch of
> re-INVITEs/UPDATEs and all kinds of whacked corner-cases with
> pre-conditions, and IPv4/v6, and god-knows-what else.  :)
>=20
> -hadriel
>=20
> > -----Original Message-----
> > From: Roni Even [mailto:Even.roni@huawei.com]
> > Sent: Friday, June 12, 2009 3:27 AM
> >
> > James,
> > The B2BUA does not need to see the RTP/RTCP stream they can go directly
> > from
> > the source to the recorder
> >
> >
> > +-----+              +-----+              +-----+
> > |     |     Call     |     |   Call       |     |
> > |UA-A +<------------>+B2BUA+<------------>+UA-B |
> > |     |              |     |              |     |
> > +-----+              +-++--+              +-----+
> >     .                   \\                   .
> >       .                  \\ control          .
> >          .  RTP       TBD \\                 .RTP
> >              .             \\+----------+    .
> >                 .           \+          |.....
> >                       .      + Recorder |
> >                              |          |
> >                              +----------+
> >
> > Roni
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

From Even.roni@huawei.com  Sat Jun 13 23:49:30 2009
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Date: Sun, 14 Jun 2009 09:47:44 +0300
From: Roni Even <Even.roni@huawei.com>
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Cc: 'Leon Portman' <Leon.Portman@nice.com>, dispatch@ietf.org
Subject: Re: [dispatch] Session recording in SIP
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Hi,
This is a good observation but in the conferencing case you can have the
media coming not from the UAs but from the media mixer or have the mixer
functionality as part of the recorder. 

I understand the UA forking scenario for the call center application where
the UAs in the call center will be required to support this functionality.
Note that in this case if you have a multipoint video call you are recording
just what the UA receives (you record the view point of the recording UA)
and not the whole conference. If the handling UA is recording you may also
miss in the call center scenario any side discussion between the customer
and a supervisor if the handling UA is excluded from this part, in some
cases he will not even know about being kept out of some side call.
 
The general case is where you want to record a call but do not have the
media forking and recording application on your UA. In this case my view is
that you will need a recording application server in the network. It will
need to record both point to point and multipoint calls. It can be like
suggested by Hadriel where the RTP streams are forked by the Application
Server (note that I am not calling it a general B2BUA) or using a decomposed
Application Server like the mediactrl architecture.  

Roni Even

-----Original Message-----
From: Jain, Rajnish [mailto:Rajnish.Jain@ipc.com] 
Sent: Sunday, June 14, 2009 4:54 AM
To: Hadriel Kaplan; Roni Even; 'James M. Polk'; 'Paul Kyzivat'
Cc: 'Leon Portman'; dispatch@ietf.org
Subject: RE: [dispatch] Session recording in SIP

I think the UA-UA RTP line was missing in the diagram. Otherwise, the
recorder is being expected to do conferencing (assuming more than 2 recordee
UAs) and transcoding. That's really not the purpose of the recorders.

--
Raj Jain


-----Original Message-----
From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]
Sent: Friday, June 12, 2009 5:54 PM
To: Roni Even; 'James M. Polk'; 'Paul Kyzivat'
Cc: 'Leon Portman'; dispatch@ietf.org; Jain, Rajnish
Subject: RE: [dispatch] Session recording in SIP


I was about to write a long-winded reply on how your example was not the
same as mediactrl, but I just realized your drawing was subtly different
from what I had thought it implied. :)  The example you're describing is one
where the recorder is not being sent forked/replicated media content by the
UA's, but is just put in the media path itself for the real media, so the
original media itself is going from UA to recorder to UA.
Right?

If so, good point - I assume that's one possible deployment model to use.  I
don't think it's how some (most?) people do it, because they can't or don't
want to move the media path from where it began, especially for the UA side
that's outside their domain.  (though I dunno much about why they do what
they do)  But it is one interesting solution.

-hadriel


> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Roni Even
> Sent: Friday, June 12, 2009 5:07 PM
> To: Hadriel Kaplan; 'James M. Polk'; 'Paul Kyzivat'
> Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'
> Subject: Re: [dispatch] Session recording in SIP
>
> Hadriel,
> This is what mediactrl is about. Look in
> http://tools.ietf.org/html/rfc5567
>
> Roni
>
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf
> Of Hadriel Kaplan
> Sent: Friday, June 12, 2009 11:08 PM
> To: Roni Even; 'James M. Polk'; 'Paul Kyzivat'
> Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'
> Subject: Re: [dispatch] Session recording in SIP
>
>
> Huh, interesting idea.  I don't see how that could easily work though.
> The
> recorder needs to be able to associate specific audio streams with
> specific
> calls, and from which called/calling party they came.  Unless there's a
> control channel between each UA and the recorder, you'd have to somehow
> get
> the B2BUA to know what IP:port each UA can send the forked media from, and
> provide each UA with the addr:port of the Recorder to send the forked
> media
> to.  Not to mention if we decide to allow a different codec to be used on
> the forked-media connection vs. the in-call side (which I personally don't
> like the idea of but others do).  At the end of the day, I'd bet you'd
> have
> one complicated protocol for all that to occur, with a bunch of
> re-INVITEs/UPDATEs and all kinds of whacked corner-cases with
> pre-conditions, and IPv4/v6, and god-knows-what else.  :)
>
> -hadriel
>
> > -----Original Message-----
> > From: Roni Even [mailto:Even.roni@huawei.com]
> > Sent: Friday, June 12, 2009 3:27 AM
> >
> > James,
> > The B2BUA does not need to see the RTP/RTCP stream they can go directly
> > from
> > the source to the recorder
> >
> >
> > +-----+              +-----+              +-----+
> > |     |     Call     |     |   Call       |     |
> > |UA-A +<------------>+B2BUA+<------------>+UA-B |
> > |     |              |     |              |     |
> > +-----+              +-++--+              +-----+
> >     .                   \\                   .
> >       .                  \\ control          .
> >          .  RTP       TBD \\                 .RTP
> >              .             \\+----------+    .
> >                 .           \+          |.....
> >                       .      + Recorder |
> >                              |          |
> >                              +----------+
> >
> > Roni
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

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From: "Jain, Rajnish" <Rajnish.Jain@ipc.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, Roni Even <Even.roni@huawei.com>,  "'James M. Polk'" <jmpolk@cisco.com>, 'Paul Kyzivat' <pkyzivat@cisco.com>
Date: Sat, 13 Jun 2009 21:54:25 -0400
Thread-Topic: [dispatch] Session recording in SIP
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Cc: 'Leon Portman' <Leon.Portman@nice.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Subject: Re: [dispatch] Session recording in SIP
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I think the UA-UA RTP line was missing in the diagram. Otherwise, the recor=
der is being expected to do conferencing (assuming more than 2 recordee UAs=
) and transcoding. That's really not the purpose of the recorders.

--
Raj Jain


-----Original Message-----
From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]
Sent: Friday, June 12, 2009 5:54 PM
To: Roni Even; 'James M. Polk'; 'Paul Kyzivat'
Cc: 'Leon Portman'; dispatch@ietf.org; Jain, Rajnish
Subject: RE: [dispatch] Session recording in SIP


I was about to write a long-winded reply on how your example was not the sa=
me as mediactrl, but I just realized your drawing was subtly different from=
 what I had thought it implied. :)  The example you're describing is one wh=
ere the recorder is not being sent forked/replicated media content by the U=
A's, but is just put in the media path itself for the real media, so the or=
iginal media itself is going from UA to recorder to UA.
Right?

If so, good point - I assume that's one possible deployment model to use.  =
I don't think it's how some (most?) people do it, because they can't or don=
't want to move the media path from where it began, especially for the UA s=
ide that's outside their domain.  (though I dunno much about why they do wh=
at they do)  But it is one interesting solution.

-hadriel


> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Roni Even
> Sent: Friday, June 12, 2009 5:07 PM
> To: Hadriel Kaplan; 'James M. Polk'; 'Paul Kyzivat'
> Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'
> Subject: Re: [dispatch] Session recording in SIP
>
> Hadriel,
> This is what mediactrl is about. Look in
> http://tools.ietf.org/html/rfc5567
>
> Roni
>
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf
> Of Hadriel Kaplan
> Sent: Friday, June 12, 2009 11:08 PM
> To: Roni Even; 'James M. Polk'; 'Paul Kyzivat'
> Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'
> Subject: Re: [dispatch] Session recording in SIP
>
>
> Huh, interesting idea.  I don't see how that could easily work though.
> The
> recorder needs to be able to associate specific audio streams with
> specific
> calls, and from which called/calling party they came.  Unless there's a
> control channel between each UA and the recorder, you'd have to somehow
> get
> the B2BUA to know what IP:port each UA can send the forked media from, an=
d
> provide each UA with the addr:port of the Recorder to send the forked
> media
> to.  Not to mention if we decide to allow a different codec to be used on
> the forked-media connection vs. the in-call side (which I personally don'=
t
> like the idea of but others do).  At the end of the day, I'd bet you'd
> have
> one complicated protocol for all that to occur, with a bunch of
> re-INVITEs/UPDATEs and all kinds of whacked corner-cases with
> pre-conditions, and IPv4/v6, and god-knows-what else.  :)
>
> -hadriel
>
> > -----Original Message-----
> > From: Roni Even [mailto:Even.roni@huawei.com]
> > Sent: Friday, June 12, 2009 3:27 AM
> >
> > James,
> > The B2BUA does not need to see the RTP/RTCP stream they can go directly
> > from
> > the source to the recorder
> >
> >
> > +-----+              +-----+              +-----+
> > |     |     Call     |     |   Call       |     |
> > |UA-A +<------------>+B2BUA+<------------>+UA-B |
> > |     |              |     |              |     |
> > +-----+              +-++--+              +-----+
> >     .                   \\                   .
> >       .                  \\ control          .
> >          .  RTP       TBD \\                 .RTP
> >              .             \\+----------+    .
> >                 .           \+          |.....
> >                       .      + Recorder |
> >                              |          |
> >                              +----------+
> >
> > Roni
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

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From rajnish.jain@ipc.com  Sun Jun 14 18:42:37 2009
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From: "Jain, Rajnish" <Rajnish.Jain@ipc.com>
To: Roni Even <Even.roni@huawei.com>, 'Hadriel Kaplan' <HKaplan@acmepacket.com>, "'James M. Polk'" <jmpolk@cisco.com>, 'Paul Kyzivat' <pkyzivat@cisco.com>
Date: Sun, 14 Jun 2009 21:42:31 -0400
Thread-Topic: [dispatch] Session recording in SIP
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Cc: 'Leon Portman' <Leon.Portman@nice.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Subject: Re: [dispatch] Session recording in SIP
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Hi,

Inline:

> -----Original Message-----
> From: Roni Even [mailto:Even.roni@huawei.com]
> Sent: Sunday, June 14, 2009 2:48 AM
> To: Jain, Rajnish; 'Hadriel Kaplan'; 'James M. Polk'; 'Paul Kyzivat'
> Cc: 'Leon Portman'; dispatch@ietf.org
> Subject: RE: [dispatch] Session recording in SIP
>
> Hi,
> This is a good observation but in the conferencing case you can have the
> media coming not from the UAs but from the media mixer or have the mixer
> functionality as part of the recorder.

I think it'll help to model the mixer and the recorder as separate logical =
entities. The mixer can feed the recorder if a conference call needs to be =
recorded. Or the mixer can be used as a tapping device in case of non-forki=
ng UAs. Below is how I'd draw it (control path from app server to recorder =
not shown for brevity):


           SIP   +------------+ SIP
      /----------| App Server |--------
     /           +------------+        \
    /                   |SIP            \
   /                    |                \
+-----+              +-----+              +-----+
|     |     RTP      |     |   RTP        |     |
|UA-A +<------------>+Mixer+<------------>+UA-B |
|     |              |     |              |     |
+-----+              +-++--+              +-----+
                      |   |
           RTP UA-A   |   | RTP UA-B (Rx+Tx)
           (Rx+Tx)    V   V
                   +----------+
                   |          |
                   | Recorder |
                   |          |
                   +----------+

> I understand the UA forking scenario for the call center application wher=
e
> the UAs in the call center will be required to support this functionality=
.
> Note that in this case if you have a multipoint video call you are record=
ing
> just what the UA receives (you record the view point of the recording UA)
> and not the whole conference. If the handling UA is recording you may als=
o
> miss in the call center scenario any side discussion between the customer
> and a supervisor if the handling UA is excluded from this part, in some
> cases he will not even know about being kept out of some side call.

Yes, you typically record the view point of the recording UA. But, there ma=
y be requirement to record a conference call as opposed to individual parti=
cipants to save disk space.

> The general case is where you want to record a call but do not have the
> media forking and recording application on your UA. In this case my view =
is
> that you will need a recording application server in the network. It will
> need to record both point to point and multipoint calls. It can be like
> suggested by Hadriel where the RTP streams are forked by the Application
> Server (note that I am not calling it a general B2BUA) or using a decompo=
sed
> Application Server like the mediactrl architecture.

Yes. I agree that an application server will compensate for non-forking UAs=
, but I think the RTP streams will be forked by the media server and not th=
e application server. The application server will dictate the media server =
to do so. The diagram above reflects this arrangement.

--
Raj Jain


> -----Original Message-----
> From: Jain, Rajnish [mailto:Rajnish.Jain@ipc.com]
> Sent: Sunday, June 14, 2009 4:54 AM
> To: Hadriel Kaplan; Roni Even; 'James M. Polk'; 'Paul Kyzivat'
> Cc: 'Leon Portman'; dispatch@ietf.org
> Subject: RE: [dispatch] Session recording in SIP
>
> I think the UA-UA RTP line was missing in the diagram. Otherwise, the
> recorder is being expected to do conferencing (assuming more than 2 recor=
dee
> UAs) and transcoding. That's really not the purpose of the recorders.
>
> --
> Raj Jain
>
>
> -----Original Message-----
> From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]
> Sent: Friday, June 12, 2009 5:54 PM
> To: Roni Even; 'James M. Polk'; 'Paul Kyzivat'
> Cc: 'Leon Portman'; dispatch@ietf.org; Jain, Rajnish
> Subject: RE: [dispatch] Session recording in SIP
>
>
> I was about to write a long-winded reply on how your example was not the
> same as mediactrl, but I just realized your drawing was subtly different
> from what I had thought it implied. :)  The example you're describing is =
one
> where the recorder is not being sent forked/replicated media content by t=
he
> UA's, but is just put in the media path itself for the real media, so the
> original media itself is going from UA to recorder to UA.
> Right?
>
> If so, good point - I assume that's one possible deployment model to use.=
  I
> don't think it's how some (most?) people do it, because they can't or don=
't
> want to move the media path from where it began, especially for the UA si=
de
> that's outside their domain.  (though I dunno much about why they do what
> they do)  But it is one interesting solution.
>
> -hadriel
>
>
> > -----Original Message-----
> > From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> > Behalf Of Roni Even
> > Sent: Friday, June 12, 2009 5:07 PM
> > To: Hadriel Kaplan; 'James M. Polk'; 'Paul Kyzivat'
> > Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'
> > Subject: Re: [dispatch] Session recording in SIP
> >
> > Hadriel,
> > This is what mediactrl is about. Look in
> > http://tools.ietf.org/html/rfc5567
> >
> > Roni
> >
> > -----Original Message-----
> > From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> > Behalf
> > Of Hadriel Kaplan
> > Sent: Friday, June 12, 2009 11:08 PM
> > To: Roni Even; 'James M. Polk'; 'Paul Kyzivat'
> > Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'
> > Subject: Re: [dispatch] Session recording in SIP
> >
> >
> > Huh, interesting idea.  I don't see how that could easily work though.
> > The
> > recorder needs to be able to associate specific audio streams with
> > specific
> > calls, and from which called/calling party they came.  Unless there's a
> > control channel between each UA and the recorder, you'd have to somehow
> > get
> > the B2BUA to know what IP:port each UA can send the forked media from, =
and
> > provide each UA with the addr:port of the Recorder to send the forked
> > media
> > to.  Not to mention if we decide to allow a different codec to be used =
on
> > the forked-media connection vs. the in-call side (which I personally do=
n't
> > like the idea of but others do).  At the end of the day, I'd bet you'd
> > have
> > one complicated protocol for all that to occur, with a bunch of
> > re-INVITEs/UPDATEs and all kinds of whacked corner-cases with
> > pre-conditions, and IPv4/v6, and god-knows-what else.  :)
> >
> > -hadriel
> >
> > > -----Original Message-----
> > > From: Roni Even [mailto:Even.roni@huawei.com]
> > > Sent: Friday, June 12, 2009 3:27 AM
> > >
> > > James,
> > > The B2BUA does not need to see the RTP/RTCP stream they can go direct=
ly
> > > from
> > > the source to the recorder
> > >
> > >
> > > +-----+              +-----+              +-----+
> > > |     |     Call     |     |   Call       |     |
> > > |UA-A +<------------>+B2BUA+<------------>+UA-B |
> > > |     |              |     |              |     |
> > > +-----+              +-++--+              +-----+
> > >     .                   \\                   .
> > >       .                  \\ control          .
> > >          .  RTP       TBD \\                 .RTP
> > >              .             \\+----------+    .
> > >                 .           \+          |.....
> > >                       .      + Recorder |
> > >                              |          |
> > >                              +----------+
> > >
> > > Roni
> >
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> >
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
>
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From mizuno.shintaro@lab.ntt.co.jp  Mon Jun 15 03:52:04 2009
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Date: Mon, 15 Jun 2009 19:50:19 +0900
From: Shintaro Mizuno <mizuno.shintaro@lab.ntt.co.jp>
To: "'Henry Sinnreich'" <hsinnrei@adobe.com>, "'Eric Burger'" <eburger@standardstrack.com>, "Dan Wing" <dwing@cisco.com>
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Subject: Re: [dispatch] Charter Proposal
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Mr. Sinnreich,

 Besides the fact that SIP is a well-deployed infrastructure, we also think that SIP has an advantage for its capabilities to negotiate parameters using SDP offer/answer. In case of IKE, these parameters are, for example, authentication parameters (fingerprint of a certificate) and role of IKE sessions (e.g. server/client in remote access), which usually need to be exchanged previously by the peers.
 This enables IKE to be established in a peer-to-peer situation.

Best Regards,
Shintaro

On Thu, 11 Jun 2009 09:05:16 -0700
"Dan Wing" <dwing@cisco.com> wrote:

>  
> 
> > -----Original Message-----
> > From: dispatch-bounces@ietf.org 
> > [mailto:dispatch-bounces@ietf.org] On Behalf Of Henry Sinnreich
> > Sent: Tuesday, June 09, 2009 6:32 AM
> > To: Shintaro Mizuno; Eric Burger
> > Cc: dispatch@ietf.org
> > Subject: Re: [dispatch] Charter Proposal
> > 
> > You may also want to consider HIP, since it solves NAT 
> > traversal for all application layer protocols, including SIP.
> > 
> > http://www.ietf.org/internet-drafts/draft-ietf-hip-nat-travers
> > al-07.txt
> 
> Yes, I am aware of HIP's NAT traversal capabilities, as I 
> encouraged its adoption of ICE methodologies.
> 
> > It also takes care of the IPv4 to IPv6 transition for 
> > endpoints, mobility and multi-homing.
> > Open source code is plentifully available for reference 
> > implementations.
> > 
> > http://www.ietf.org/html.charters/hip-charter.html
> 
> There are always new protocols.  But it is often pragmatic to leverage the
> investment of deployed protocols -- much like TCP which continues to evolve
> even though SCTP and DCCP provide superior functionality, much like how IPsec
> was originally intended for IPv6 but was then supported on IPv4, etc.
> 
> We are interested in using SIP for establishing VPN sessions in much the same
> way that SIP has embraced TLS (RFC4572) and embraced file transfer (RFC5547).
> 
> -d
> 
> 
> > Henry
> > 
> > 
> > 
> > On 6/9/09 8:02 AM, "Shintaro Mizuno" 
> > <mizuno.shintaro@lab.ntt.co.jp> wrote:
> > 
> > 
> > 
> > 	Mr. Burger,
> > 	 Thank you for the comment.
> > 	
> > 	 We don't necessary see a need for a WG to be formed to 
> > document the 
> > 	informational document we have listed, the document is 
> > there to see if
> > 	the proposed way forward (informational document possibly
> > 	as an independent submission) for the document listed 
> > is on track or 
> > 	whether something should be done at the WG level.
> > 	
> > 	 On the other hand, we believe that use of SIP similar 
> > to what we 
> > 	have done for VPN will likely to happen more. And on 
> > that point, we are
> > 	wanting to see whether IETF is interested in 
> > documenting a guideline
> > 	for dealing with application that's not voice or video.
> > 	As written in the discussion point of the charter, the 
> > question is how
> > 	much should SIP be involved.
> > 	
> > 	 1. Should SIP just be used as a rendezvous protocol 
> > and leave the 
> > 	rest of the negotiation to whatever application?
> > 	 2. Should SIP involve in management of the established 
> > session as it
> > 	does for audio/video?
> > 	 3. Should use of SIP not be standardized for peer to 
> > peer application
> > 	beside audio/video?
> > 	
> > 	
> > 	Regards,
> > 	Shintaro
> > 	
> > 	
> > 	On Mon, 8 Jun 2009 06:38:20 -0400
> > 	Eric Burger <eburger@standardstrack.com> wrote:
> > 	
> > 	> Given the proposal is not really for an Internet 
> > standard, and the 
> > 	> product is not a protocol or standards track 
> > document, why are we 
> > 	> working on it here?
> > 	>
> > 	> On Jun 8, 2009, at 5:20 AM, Shintaro Mizuno wrote:
> > 	>
> > 	> > Dear DISPATCH Chair, ML members,
> > 	> >
> > 	> > I've finalized the charter proposal for on-demand 
> > media/application
> > 	> > sharing.
> > 	> > Added milestone/deliverables section and some 
> > descriptions on the 
> > 	> > usage
> > 	> > of SIP, but most of the text remains the same.
> > 	> >
> > 	> > --------------------------------------------------
> > 	> > "Charter Proposal for on-demand Media/Application 
> > Sharing between 
> > 	> > peers
> > 	> > over VPN"
> > 	> >
> > 	> > Intent:
> > 	> > The intent of this charter is to see interests in 
> > documenting what's
> > 	> > being deployed as an informational for on-demand 
> > Media/Application
> > 	> > sharing between peers over VPN using SIP and 
> > discuss what in the
> > 	> > overall effort may be standardized at IETF in the future.
> > 	> >
> > 	> >
> > 	> > Problem Statement:
> > 	> > Home servers or network-capable consumer electronic 
> > devices are
> > 	> > becoming widely deployed and people using such 
> > devices are wanting to
> > 	> > share contents or applications and are seeking a 
> > way to establish
> > 	> > multitudes of communication with each other.
> > 	> >
> > 	> > In order to allow two people to share contents or 
> > application securely
> > 	> > using popular LAN based communication tool such as 
> > DLNA on devices 
> > 	> > that
> > 	> > are sitting at home, secure LAN (VPN) 
> > establishments between these 
> > 	> > user
> > 	> > devices are necessary. Applications that use proprietary or
> > 	> > multi-session protocols such as in some of 
> > LAN-based gaming will also
> > 	> > benifit from VPN connections when users attempt to 
> > communicate 
> > 	> > remotely.
> > 	> >
> > 	> > For a deployment that's underway and considered by 
> > several other
> > 	> > standard bodies, problem of establishing VPN 
> > between these user 
> > 	> > devices
> > 	> > were that current standards for establishing VPN 
> > didn't provide all 
> > 	> > the
> > 	> > toolset necessary to overcome some of the obstacles.
> > 	> >
> > 	> > Namely following issues were found;
> > 	> >  (1) Name resolution mechanism under floating IP 
> > addresses and port
> > 	> > numbers
> > 	> >  (2) NAT traversal and hole-punching of firewall
> > 	> >  (3) Authentication/authorization of users.
> > 	> >  (4) Pre-sharing of key for establishing VPN isn't 
> > easy when VPN is to
> > 	> > be established on-demand between two user devices 
> > without any previous
> > 	> > affiliation.
> > 	> >
> > 	> > As a result current deployment based on 
> > "draft-saito-mmusic-sdp-
> > 	> > ike-04"
> > 	> > and other standard bodies are looking at the use of 
> > SIP for the
> > 	> > following reasons.
> > 	> >
> > 	> > (1)Name resolution can be achieved by using 
> > REGISTER method and SDP
> > 	> > negotiation
> > 	> > (2)NAT traversal and hole-punching of firewall can 
> > be achieved by
> > 	> > applying SIP+ICE.
> > 	> > (3)Authentication/authorization can be achieved by 
> > having trusted SIP
> > 	> > proxy and user friendly SIP-URIs.
> > 	> >
> > 	> > SIP/SDP will be used to negotiate parameters 
> > necessary to establish
> > 	> > IKE/IPsec such as IP address, port numbers and 
> > authentication
> > 	> > information of peers. Current deployment uses SIP 
> > also to manage IPsec
> > 	> > connection, e.g. use BYE to shut down media 
> > sessions for IKE/IPsec,
> > 	> > however, further discussions can be made to find 
> > out if there are
> > 	> > intrests or use cases in using SIP only as a 
> > rendezvous protocol.
> > 	> >
> > 	> > Use-cases that may see use for such standard effort;
> > 	> >  - Media sharing using DLNA or similar protocol 
> > over VPN between 2
> > 	> > users' devices.
> > 	> >  - Remote desktop sharing for Customer service over 
> > VPN initiated via
> > 	> > SIP call.
> > 	> >   * Similar to click to call, customer service 
> > agent can access user's
> > 	> > pc remotely to troubleshoot the problem customer is 
> > facing from the 
> > 	> > call.
> > 	> >  - Accessing medical equipments(medical robotics) 
> > remotely and
> > 	> > possibly controlling medical equipments to monitor 
> > elders in a rural
> > 	> > area that (remote care services).
> > 	> >  - LAN based gaming protocol to be established peer 
> > to peer rather
> > 	> > than via gaming server.
> > 	> >
> > 	> >
> > 	> > Discussion Point:
> > 	> > - To see interests in documenting what's being 
> > deployed based on
> > 	> > "draft-saito-mmusic-sdp-ike-04" as an informational 
> > document.
> > 	> > http://tools.ietf.org/html/draft-saito-mmusic-sdp-ike-04
> > 	> > (to be minor-updated to -05 to conform with this 
> > charter proposal)
> > 	> > - To see interests in standardizing toolset for 
> > realizing use-cases
> > 	> > similar to what's mentioned above, even 
> > contemplating the possibility
> > 	> > of defining a guideline for SIP usage where SIP is 
> > used only as a
> > 	> > rendezvous protocol.
> > 	> >
> > 	> >
> > 	> > Milestone/Deliverables:
> > 	> > - Aug 09 Issue Internet-Draft
> > 	> >    (by updating "draft-saito-mmusic-sdp-ike-05")
> > 	> > - Nov 09 Submit I-D to IESG for publication as an 
> > RFC (Informational)
> > 	> >
> > 	> > --------------------------------------------------
> > 	> > Best Regards,
> > 	> > Shintaro
> > 	> >
> > 	> > ----------------------------------------
> > 	> > Shintaro MIZUNO
> > 	> > mizuno.shintaro@lab.ntt.co.jp
> > 	> >
> > 	> > NTT Information Sharing Platform Laboratories
> > 	> > NTT Corporation
> > 	> > ----------------------------------------
> > 	> >
> > 	> > _______________________________________________
> > 	> > dispatch mailing list
> > 	> > dispatch@ietf.org
> > 	> > https://www.ietf.org/mailman/listinfo/dispatch
> > 	>
> > 	>
> > 	
> > 	
> > 	----------------------------------------
> > 	Shintaro MIZUNO
> > 	mizuno.shintaro@lab.ntt.co.jp
> > 	
> > 	NTT Information Sharing Platform Laboratories
> > 	NTT Corporation
> > 	----------------------------------------
> > 	
> > 	_______________________________________________
> > 	dispatch mailing list
> > 	dispatch@ietf.org
> > 	https://www.ietf.org/mailman/listinfo/dispatch
> > 	
> > 	
> > 
> > 
> 
> 


----------------------------------------
Shintaro MIZUNO
mizuno.shintaro@lab.ntt.co.jp

NTT Information Sharing Platform Laboratories
NTT Corporation
----------------------------------------


From hannes.tschofenig@nsn.com  Mon Jun 15 09:56:59 2009
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Without going too far into the solution space but we received some
review comments from Ekr for
http://tools.ietf.org/html/draft-wing-sipping-srtp-key-04. I believe it
is valuable technical input. Here is his review:
http://www.ietf.org/mail-archive/web/sipping/current/msg16652.html

Ciao
Hannes

From HKaplan@acmepacket.com  Wed Jun 17 11:29:48 2009
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To: "sip-clf@ietf.org" <sip-clf@ietf.org>
Date: Wed, 17 Jun 2009 14:29:47 -0400
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Howdy,
I have submitted an I-D for Yet Another SIP-CLF Format.  This one is pure b=
inary, and follows the libpcap/PCAP file format, encoding the fields into a=
 format which just happens to be decode-able as RADIUS messages by PCAP dec=
oders.  One of the advantages to this format is off-line decoders such as W=
ireshark can decode it right now, with only a dictionary for the VSA used f=
or SIP-CLF.  For wireshark, for example, this can be done without re-compil=
ing/upgrading wireshark.  Just a 30-second procedure and restarting Wiresha=
rk enables one to view, filter, and search through the SIP-CLF fields as is=
.

The I-D's link is below, and there's a sample CLF file and dictionary file =
and Wireshark instructions (and even screenshot!) available on:
https://sourceforge.net/projects/sip-clf/

There's also an open-source C-code library implementation available there.

-hadriel

> -----Original Message-----
> From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-bounces@ietf.org=
]
> On Behalf Of Internet-Drafts@ietf.org
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>=20
> 	Title           : PCAP-compatible Binary Syntax for SIP Common Log
> File Format
> 	Author(s)       : H. Kaplan
> 	Filename        : draft-kaplan-sipping-clf-pcap-00.txt
> 	Pages           : 20
> 	Date            : 2009-06-16
>=20
> This document proposes a libpcap/PCAP-compatible binary syntax for
> the SIP common log format (CLF).  It does not cover semantic
> issues, and is meant to be evaluated in the context of the other
> efforts discussing SIP CLF.
>=20
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-kaplan-sipping-clf-pcap-00.txt



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From mary.barnes@nortel.com  Wed Jun 17 16:36:42 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
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Subject: [dispatch] IETF-75 plans for DISPATCH
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Hi folks,

A tentative agenda for the IETF-75 DISPATCH WG has been submitted:
http://www.ietf.org/proceedings/09jul/agenda/dispatch.html

Per the various emails posted last week, we dispatched (or not) all the
topics that had been put forth by June 8th, per the following summary:
1) 3rd party auth - on the agenda to discuss what part or parts of the
problem need to be solved
2) CBUS - Christer to update his proposal at a later date
3) cc-uui - on the agenda to discuss a proposed charter (and not he
solutions)  for a new "mini" WG to progress the work which we anticipate
is a single document. =20
4) CLF - a new WG has been proposed and is in the process of being
formed. We will only allocate agenda time if open issues that cannot be
resolved on the ML are identified.=20
5) Codec - a separate BoF has been requested. =20
6) Sound level indicator - work/discussion moved to the AVT WG
7) On demand media/app sharing - in order to dispatch this work, there
needs to be more interest and feedback on this topic

The additional topic on Session Recording was put forth after June 8th
and has not yet been considered in the IETF-75 plans.  However, if there
is enough interest in the topic and folks committed to the work, then we
may be able to allocate agenda time. =20

The following summarizes the status of the other two topics that are on
the current DISPATCH WG charter:
- Overload. A new mailing list is being setup to discuss solution
options and determine the scope of work that might be done. The
expectation would be that like the other work items above, a problem
statement and proposed chartered work item(s) will be posted to the
DISPATCH WG mailing list for discussion. I'll announce the new ML as
soon as it's up and running.
- Configuration/Datasets. Dale has been looking at the documents and
trying to get WG feedback. But, there seems to be little interest in
this work. An option being considered, since this work is a dependency
for several SIP/SIPPING WG items to complete, would be to complete the
work to satisfy the current dependencies and progress as AD sponsored.=20

As a reminder, the deadline for any -00 documents is July 6th, 17:00 PDT
(less than three weeks!) and the final submission deadline is July 13th,
17:00 PDT (less than 4 weeks!).=20

We do appreciate everyone's patience with the new model, as well as the
level of participation.

Regards,
Mary.
DISPATCH WG co-chair



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At 06:39 PM 6/17/2009, Mary Barnes wrote:
>The additional topic on Session Recording was put forth after June 8th
>and has not yet been considered in the IETF-75 plans.  However, if there
>is enough interest in the topic and folks committed to the work, then we
>may be able to allocate agenda time.

I'm rather stunned that demonstrated interest isn't being considered 
sufficiently clear on this topic to the WG chairs or ADs.

Of the last 75 messages to this list, more than half have been on 
this topic alone.

I do agree this topic was late, but that ought to be stated as the 
only reason this topic doesn't progress at IETF75. Evidence of the 
vast debate we've had on this list proves folks are embracing this 
idea - even if they do not exactly have an idea of how it will look 
when finished - which is actually what we want in a WG, right?

James




From HKaplan@acmepacket.com  Wed Jun 17 17:23:23 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "James M. Polk" <jmpolk@cisco.com>, Mary Barnes <mary.barnes@nortel.com>,  "dispatch@ietf.org" <dispatch@ietf.org>
Date: Wed, 17 Jun 2009 20:23:21 -0400
Thread-Topic: [dispatch] IETF-75 plans for DISPATCH
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> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of James M. Polk
> Sent: Wednesday, June 17, 2009 8:06 PM
>=20
> At 06:39 PM 6/17/2009, Mary Barnes wrote:
> >The additional topic on Session Recording was put forth after June 8th
> >and has not yet been considered in the IETF-75 plans.  However, if there
> >is enough interest in the topic and folks committed to the work, then we
> >may be able to allocate agenda time.
>=20
> I'm rather stunned that demonstrated interest isn't being considered
> sufficiently clear on this topic to the WG chairs or ADs.

+1.  And considering the topic was put forth 10hrs and 1 minute after the d=
eadline, it's not like the fewer 10 hours (and 1 minute) left people withou=
t time to think about it.  :)

-hadriel

From mary.barnes@nortel.com  Wed Jun 17 18:19:15 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Hadriel Kaplan" <HKaplan@acmepacket.com>, "James M. Polk" <jmpolk@cisco.com>, <dispatch@ietf.org>
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Subject: Re: [dispatch] IETF-75 plans for DISPATCH
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Actually, the topic was 2 weeks + 10hrs and 1 minute after the deadline
- i.e., we asked for topics to be sent by May 25th, with any updates by
June 8th.

That said, certainly if folks have an interest, then we can consider
discussing in Stockholm. It would be good for someone to summarize where
the topic is - there seems to be a very broad range of views as to the
scope of the problem and what would be potential deliverables.

Thanks,
Mary.=20

-----Original Message-----
From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]=20
Sent: Wednesday, June 17, 2009 7:23 PM
To: James M. Polk; Barnes, Mary (RICH2:AR00); dispatch@ietf.org
Cc: rai-ads@tools.ietf.org
Subject: RE: [dispatch] IETF-75 plans for DISPATCH


> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On=20
> Behalf Of James M. Polk
> Sent: Wednesday, June 17, 2009 8:06 PM
>=20
> At 06:39 PM 6/17/2009, Mary Barnes wrote:
> >The additional topic on Session Recording was put forth after June=20
> >8th and has not yet been considered in the IETF-75 plans.  However,=20
> >if there is enough interest in the topic and folks committed to the=20
> >work, then we may be able to allocate agenda time.
>=20
> I'm rather stunned that demonstrated interest isn't being considered=20
> sufficiently clear on this topic to the WG chairs or ADs.

+1.  And considering the topic was put forth 10hrs and 1 minute after=20
+the deadline, it's not like the fewer 10 hours (and 1 minute) left=20
+people without time to think about it.  :)

-hadriel

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From: "Doken, Serhad" <sdoken@qualcomm.com>
To: "James M. Polk" <jmpolk@cisco.com>, Roni Even <Even.roni@huawei.com>, "'Alan Johnston'" <alan@sipstation.com>, "'Romascanu, Dan (Dan)'" <dromasca@avaya.com>
Date: Thu, 18 Jun 2009 01:04:15 -0700
Thread-Topic: [dispatch] Session recording in SIP
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Cc: 'Leon Portman' <Leon.Portman@nice.com>, "dispatch@ietf.org" <dispatch@ietf.org>, "'Jain, Rajnish'" <Rajnish.Jain@ipc.com>, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [dispatch] Session recording in SIP
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> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of James M. Polk
> Sent: Wednesday, June 10, 2009 3:00 PM
> To: Roni Even; 'Alan Johnston'; 'Romascanu, Dan (Dan)'
> Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'; 'Hadriel
> Kaplan'
> Subject: Re: [dispatch] Session recording in SIP
>=20
> Hey
>=20
> I agree with Roni about this probably being best suited for XCON,
> after all, any point to point recording is likely (merely) adding a
> 3rd party (the recorder) to the session.
>=20
> This brings up probably the most glaring whole (in my mind) that is
> not mentioned below - the ability to have a recorder without at least
> one side knowing it's there.  Not every time, and in fact probably
> most times, you don't want the other person knowing you are recording
> them. It's likely more of a case of not explicitly asking the called
> party "can I record this conversation", it just happens.

As someone who have implemented SIP based recording which has been shipping=
 for several years, this thread caught my eye though I haven't read the who=
le lively set of e-mails yet.=20

In most enterprise contact center environments(say 1-800-2CALLME), caller(l=
et's call him customer) will get an announcement of some sort like "This ca=
ll may/will be monitored/recorded for quality purposes" at the beginning of=
 the call. For limited cases, customer has a chance to accept/reject that o=
r if refuses he/she can hang up. Called party(Agent) as an employee is almo=
st always aware that call may be recorded since contact center businesses c=
ollect quality and business analytic data based on those recordings. Moreov=
er, in some deployments, periodic recording tones are played towards custom=
er and agent phones to alert/remind them that call is being recorded. It co=
uld be the case that sometimes all calls are recorded or sometimes calls to=
 a particular set of agent phones are recorded.

On the other hand, a supervisor monitoring a conversation between a custome=
r and agent may decide to record that conversation, at any random point, si=
mply triggering it via a CTI application that is running on his desktop for=
 the whole call duration or may wish to stop that recording at any random p=
oint.

Serhad

>=20
> James
>=20
> At 11:18 AM 6/10/2009, Roni Even wrote:
> >Hi,
> >I agree that this is an important work but I am not sure about a WG.
> This
> >may be done in XCON and AVT.
> >Note that the recording is not only for point to point call but at
> least for
> >three parties like in call center scenarios
> >
> >I also did not notice any mention of the playback part but it should
> also be
> >addressed
> >
> >Roni Even
> >
> >-----Original Message-----
> >From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf
> >Of Alan Johnston
> >Sent: Wednesday, June 10, 2009 4:42 PM
> >To: Romascanu, Dan (Dan)
> >Cc: Leon Portman; dispatch@ietf.org; Jain, Rajnish; Hadriel Kaplan
> >Subject: Re: [dispatch] Session recording in SIP
> >
> >This is important work to be done in RAI.  Key management for secure
> >recording is an important component that needs to be addressed.
> >
> >draft-wing-sipping-srtp-key-04 addresses many of these topics.
> >
> >Thanks,
> >Alan
> >
> >
> >Romascanu, Dan (Dan) wrote:
> > > I believe that this is an important application, and I support
> doing
> > > work in this direction.
> > >
> > > Three comments on the preliminary words:
> > >
> > > 1. The requirement is for both signaling and media recording,
> right?
> > > Probably good to say it.
> > > 2. Security and regulations on how the information is accessed and
> > > protected are of high importance. They would probably be reflected
> in
> > > the requirements, but explicit wording in the future charter can
> help.
> > > 3. I suggest that we look at the IPFIX protocol as a possible
> technology
> > > to re-use
> > >
> > > Dan
> > >
> > >
> > >
> > >> -----Original Message-----
> > >> From: dispatch-bounces@ietf.org
> > >> [mailto:dispatch-bounces@ietf.org] On Behalf Of Vijay K. Gurbani
> > >> Sent: Wednesday, June 10, 2009 12:02 AM
> > >> To: dispatch@ietf.org
> > >> Cc: Leon Portman; Jain,Rajnish; Hadriel Kaplan
> > >> Subject: [dispatch] Session recording in SIP
> > >>
> > >> Hi: I realize that the deadline for charter proposals was
> > >> yesterday, but I hope that it is not too late to submit one more.
> > >>
> > >> A few interested people (Hadriel Kaplan, Dan Wing, Rajnish
> > >> Jain, Leon Portman, Andrew Hutton and I) have been interested
> > >> in RTP session recording in SIP.  The requirement draft will
> > >> be released shortly.
> > >>
> > >> We would like to request agenda time in dispatch to propose
> > >> the formation of a new working group to define protocol
> > >> extensions and an architecture for RTP recording.
> > >>
> > >> Session recording in SIP
> > >> Mailing Lists: TBD
> > >> Chairs: TBD
> > >> Area Directorate: Real Time Applications (RAI)
> > >>
> > >> Purpose:
> > >>
> > >> Session recording is a critical operational requirement in
> > >> many businesses, especially where voice is used as a medium
> > >> for commerce and customer support. A prime example where
> > >> voice is used for trade is the financial industry. The call
> > >> recording requirements in this industry are quite stringent.
> > >> The recorded calls are used for dispute resolution and
> > >> regulatory compliance. Other businesses such as customer
> > >> support call centers typically employ call recording for
> > >> quality control or business analytics.
> > >>
> > >> Depending on the country and its regulatory requirements,
> > >> financial trading floors typically must record all calls. The
> > >> recorded media content must be an exact copy of the actual
> > >> conversation (i.e.
> > >> clipping and loss of media are unacceptable).  Some
> > >> deployments and regulations require that calls be aborted or
> > >> rejected if the recording device is unavailable.
> > >>
> > >> This group will specify requirements for a SIP based protocol
> > >> interface between a communications system and a recorder. The
> > >> Communications System is responsible for establishing media
> > >> sessions where the actual business is conducted. The Recorder
> > >> is the sink of the recorded media.
> > >>
> > >> The recorded sessions can be of any kind such as voice, video
> > >> and instant messaging. A recorded session is typically
> > >> comprised of actual media content and the call metadata. The
> > >> call metadata allows recording archives to be searched and
> > >> filtered at a later time.
> > >> The conveyance of call metadata from the communications
> > >> system to the recorder is outside the scope of this document.
> > >>
> > >> This group will only looks into active recording, where the
> > >> recorded system purposefully streams media to a recording
> > >> device. Passive recording, where a recording device detects
> > >> media directly from the network, is outside the scope of this
> > >> document. In addition, lawful intercept is outside the scope
> > >> of the group.
> > >>
> > >> Proposed deliverables:
> > >>
> > >> 1) Requirements document;
> > >> 2) Solutions document, including reference architecture.
> > >>
> > >> Thanks,
> > >>
> > >> - vijay
> > >> --
> > >> Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent 1960
> > >> Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
> > >> Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
> > >> Web:   http://ect.bell-labs.com/who/vkg/
> > >> _______________________________________________
> > >> dispatch mailing list
> > >> dispatch@ietf.org
> > >> https://www.ietf.org/mailman/listinfo/dispatch
> > >>
> > >>
> > > _______________________________________________
> > > dispatch mailing list
> > > dispatch@ietf.org
> > > https://www.ietf.org/mailman/listinfo/dispatch
> > >
> > >
> >
> >_______________________________________________
> >dispatch mailing list
> >dispatch@ietf.org
> >https://www.ietf.org/mailman/listinfo/dispatch
> >
> >_______________________________________________
> >dispatch mailing list
> >dispatch@ietf.org
> >https://www.ietf.org/mailman/listinfo/dispatch
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

From vkg@alcatel-lucent.com  Thu Jun 18 06:16:49 2009
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From: "Vijay K. Gurbani" <vkg@alcatel-lucent.com>
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Mary Barnes wrote:
> That said, certainly if folks have an interest, then we can consider 
> discussing in Stockholm.

Mary: The interest is there, and I agree with James and Hadriel
that we should consider discussions in Stockholm.

Rajnish Jain, the primary editor of the session recording requirement
document, has the -00 version distributed among the co-authors
and will be updating the document with list discussions
before submission shortly.  So there will be a concrete
document to discuss by Stockholm.

> It would be good for someone to summarize where the topic is - there
> seems to be a very broad range of views as to the scope of the
> problem and what would be potential deliverables.

The potential deliverables in the form of 2-3 documents
are listed in the call for proposals.  Regarding the summary
of list discussions, sure, I will attempt one over the weekend.

Thanks,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/

From mary.barnes@nortel.com  Thu Jun 18 06:21:42 2009
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Thread-Topic: [dispatch] IETF-75 plans for DISPATCH
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Vijay K. Gurbani" <vkg@alcatel-lucent.com>
Cc: dispatch@ietf.org, rai-ads@tools.ietf.org
Subject: Re: [dispatch] IETF-75 plans for DISPATCH
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Just keep in mind that the focus of discussions in DISPATCH is on the
level of the scope of work, where it should be done and deliverables and
not necessarily the technical details. Understanding of course that some
consideration of technical details is required to figure out where the
work should be done.=20

Thanks,
Mary.=20

-----Original Message-----
From: Vijay K. Gurbani [mailto:vkg@alcatel-lucent.com]=20
Sent: Thursday, June 18, 2009 8:17 AM
To: Barnes, Mary (RICH2:AR00)
Cc: dispatch@ietf.org; rai-ads@tools.ietf.org
Subject: Re: [dispatch] IETF-75 plans for DISPATCH

Mary Barnes wrote:
> That said, certainly if folks have an interest, then we can consider=20
> discussing in Stockholm.

Mary: The interest is there, and I agree with James and Hadriel that we
should consider discussions in Stockholm.

Rajnish Jain, the primary editor of the session recording requirement
document, has the -00 version distributed among the co-authors and will
be updating the document with list discussions before submission
shortly.  So there will be a concrete document to discuss by Stockholm.

> It would be good for someone to summarize where the topic is - there=20
> seems to be a very broad range of views as to the scope of the problem

> and what would be potential deliverables.

The potential deliverables in the form of 2-3 documents are listed in
the call for proposals.  Regarding the summary of list discussions,
sure, I will attempt one over the weekend.

Thanks,

- vijay
--
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent 1960 Lucent Lane,
Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/

From vkg@alcatel-lucent.com  Thu Jun 18 06:21:42 2009
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Subject: Re: [dispatch] IETF-75 plans for DISPATCH
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Mary Barnes wrote:
> 4) CLF - a new WG has been proposed and is in the process of being
> formed. We will only allocate agenda time if open issues that cannot be
> resolved on the ML are identified. 

Mary, Gonzalo: This is a process-related question -- when we
form a WG as a result of deliberations in dispatch, is this
a formal WG that needs to be scheduled and managed as such?
Or does the dispatch-sponsored WG meet during an ad-hoc time,
or even chew up some time in the dispatch slot?

In other words, once dispatch agrees to create a WG, what
are the next steps that should be done?

Thanks,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/

From mary.barnes@nortel.com  Thu Jun 18 06:26:50 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Vijay K. Gurbani" <vkg@alcatel-lucent.com>, "Gonzalo Camarillo" <Gonzalo.Camarillo@ericsson.com>
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The ADs go through the normal process to get a WG setup. These will be
regular WGs - i.e., own slots at meetings, chairs, etc. I refer to these
as "mini" WGs because they may only have one WG deliverable.  The CLF
work is in the plans as a regular WG.  Until the WG is setup, if you
need time at the meeting to discuss issues, then we can use some of the
DISPATCH WG allocated time. =20

Mary.=20

-----Original Message-----
From: Vijay K. Gurbani [mailto:vkg@alcatel-lucent.com]=20
Sent: Thursday, June 18, 2009 8:22 AM
To: Barnes, Mary (RICH2:AR00); Gonzalo Camarillo
Cc: dispatch@ietf.org; rai-ads@tools.ietf.org
Subject: Re: [dispatch] IETF-75 plans for DISPATCH

Mary Barnes wrote:
> 4) CLF - a new WG has been proposed and is in the process of being=20
> formed. We will only allocate agenda time if open issues that cannot=20
> be resolved on the ML are identified.

Mary, Gonzalo: This is a process-related question -- when we form a WG
as a result of deliberations in dispatch, is this a formal WG that needs
to be scheduled and managed as such?
Or does the dispatch-sponsored WG meet during an ad-hoc time, or even
chew up some time in the dispatch slot?

In other words, once dispatch agrees to create a WG, what are the next
steps that should be done?

Thanks,

- vijay
--
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent 1960 Lucent Lane,
Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/

From mary.barnes@nortel.com  Thu Jun 18 07:11:52 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
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I just want to pull one thing out of here - not necessarily specific to
this topic or picking on James, but some general comments on the model
we're trying to achieve.  Yes, there has been lots of discussion on this
topic, but that's actually a concern with docs that come in late. The
attention span in RAI tends to be very short - a good example being
Identity that was such a hot topic over the past year - we burned a lot
of f2f meeting time on that topic over the past year (at least).   The
idea of the early deadlines was so that we could limit the topics that
folks would need to pay attention to.  I am actually concerned that
there hasn't been more discussion for some of the other topics - e.g.,
3rd party auth - that doc was one of the earliest to be posted for
discussion, yet we haven't had any discussion since May 27th.       =20

The whole idea of the model we're trying to achieve is that we want to
avoid the f2f meetings being entirely on the initial discussions of
shiny new topics (in particular -00 documents submitted at the
deadlines), but rather be having the f2f discussions at the point where
the problems have been adequately discussed such that the problem
statement and scope is quite clear and achievable deliverables have been
defined.=20

Thanks,
Mary.=20

-----Original Message-----
From: James M. Polk [mailto:jmpolk@cisco.com]=20
Sent: Wednesday, June 17, 2009 7:06 PM
To: Barnes, Mary (RICH2:AR00); dispatch@ietf.org
Cc: rai-ads@tools.ietf.org
Subject: Re: [dispatch] IETF-75 plans for DISPATCH

At 06:39 PM 6/17/2009, Mary Barnes wrote:
>The additional topic on Session Recording was put forth after June 8th=20
>and has not yet been considered in the IETF-75 plans.  However, if=20
>there is enough interest in the topic and folks committed to the work,=20
>then we may be able to allocate agenda time.

I'm rather stunned that demonstrated interest isn't being considered
sufficiently clear on this topic to the WG chairs or ADs.

Of the last 75 messages to this list, more than half have been on this
topic alone.

I do agree this topic was late, but that ought to be stated as the only
reason this topic doesn't progress at IETF75. Evidence of the vast
debate we've had on this list proves folks are embracing this idea -
even if they do not exactly have an idea of how it will look when
finished - which is actually what we want in a WG, right?

James




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To: Francois Audet <audet@nortel.com>, "James M. Polk" <jmpolk@cisco.com>, Paul Kyzivat <pkyzivat@cisco.com>, Hadriel Kaplan <HKaplan@acmepacket.com>
Date: Thu, 18 Jun 2009 10:24:21 -0700
Thread-Topic: [dispatch] Session recording in SIP
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Cc: 'Leon Portman' <Leon.Portman@nice.com>, "dispatch@ietf.org" <dispatch@ietf.org>, "'Jain, Rajnish'" <Rajnish.Jain@ipc.com>, Roni Even <Even.roni@huawei.com>
Subject: Re: [dispatch] Session recording in SIP
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Comment on (6) :=20

Once people pay for an expensive SIP Phone with fair Dsp power, they expect=
 that endpoint trigger the recording/fork the stream towards the recorder(s=
) as well. For those cases, it's hard to convince folks/justify the need to=
 increase the resources on the server/B2BUA/gateway to do recording for man=
y endpoints when an endpoint itself is capable of doing so.

Moreover, forking a stream is not all that expensive(cpu intensive). A phon=
e, even a basic one, that can do 3-way conferencing can easily make a copy =
of a stream and send somewhere else.=20

Serhad

> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Francois Audet
> Sent: Thursday, June 11, 2009 8:43 PM
> To: James M. Polk; Paul Kyzivat; Hadriel Kaplan
> Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'; Roni Even
> Subject: Re: [dispatch] Session recording in SIP
>=20
> I agree with James.
>=20
> There are *some* scenarios where the UA-initiated recording will work,
> but
> they are very limited. They have the following disadvantages:
> 1- It makes the client "aware" that it is being recorded
> 2- It makes it very easy to avoid being recorded by interrupting the
> media
> stream
> 3 - It makes it much easier to "dispute" (i.e., lawyers) the
> authenticity of
> a recorded conversation if there is potential for significant
> differences in
> the streams as seen by different ends
> 4 - It doubles the bandwidth requirements on the UA, which can be a
> problem
> in some scenarios
> 5 - It doubles the processing power required in the client (think
> SRTP/key
> negotiation & al). This can be a problem with many phones.
> 6 - They require phones that actually support this. Which means, you
> can't
> use "normal" SIP phones. Which means,  if you get the $30 phone, you
> don't
> get tapped into, but if you get the $300 phone, you don't.
>=20
>=20
> But UA-initiated recording has very few advantages:
> 1 - Less work on servers. This is actually a big advantage because lots
> of
> servers are so complex, it is difficult to do anything new on them.
> 2 - Looks much more abstract and cool in an IETF RFC :-)
>=20
>=20
> On Jun11 2009 19:13 , "James M. Polk" <jmpolk@cisco.com> wrote:
>=20
> > At 06:11 PM 6/11/2009, Paul Kyzivat wrote:
> >
> >
> >> Hadriel Kaplan wrote:
> >>> Hey Roni,
> >>> Being a middlebox man myself, a B2BUA doing the media forking makes
> >>> the most sense to me too, but some other folks don't do it/want-it
> >>> that way.  So the proposed work is to cover both potential models.
> >>> The statement you cite reflects the belief that the solution to the
> >>> requirements will be to use SIP between the UA doing the media
> >>> forking, and the recorder doing the collection; and that the work
> >>> of specification will be to define that.
> >>> So one diagram could be this, where "TBD" is the To-Be-Defined
> work:
> >>>    +-----+              +-----+
> >>>    |     |   Call       |     |
> >>>    |UA-A +<------------>+UA-B |
> >>>    |     |              |     |
> >>>    +-++--+              +-----+
> >>>       \\
> >>>        \\
> >>>     TBD \\
> >>>          \\+----------+
> >>>           \+          |
> >>>            + Recorder |
> >>>            |          |
> >>>            +----------+
> >>
> >> This model actually can make a lot of sense in *some* circumstances.
> >> For instance, in a call center, where you may have control over what
> >> components are used throughout, it could make sense to do the media
> >> forking for recording and monitoring right at the agent phone.
> >
> > Paul - how is your call center example *not* better accomplished with
> > the figure below instead of the figure above?
> >
> > It seems to me that any enterprise based call center would have to
> > have a SIP server of some form (i.e., the caller is not going to have
> > a pt-to-pt SIP established session directly with a call center call
> > taker). It seems much easier if that call center wants to record
> > calls (even some of the calls, that to the caller, the B2BUA is the
> > UAS, meaning the sRTP keys only have to be between the caller and the
> > B2BUA, and not all the way to the call taker.  In this scenario, the
> > B2BUA merely replicates the RTP stream out towards the recording
> > device with RTCP in tact, an can allow the B2BUA provide any meta
> > data about the specifics of the call signaling it wants to also
> store.
> >
> > This scenario also means the caller is never aware the call is being
> > recorded, creating anonymity wrt that facet of the solution.
> >
> > I'm not advocating B2BUAs here, I just see this as a better solution
> > for call center type recording than pt-to-pt between caller and call
> taker.
> >
> >
> >> But in those sorts of monitoring/recording we don't talk about in
> >> IETF, its essential that the recorded parties not be able to
> >> distinguish calls that are monitored from those that are not.
> >> Anything that requires cooperation by one of the endpoints is a
> >> no-go. Even something as simple as a middlebox replacing one media
> >> port with another in order to initiate monitoring might be enough to
> >> clue in the monitored parties. So  in those cases you had better
> >> have a way that looks exactly the same whether recording is
> >> happening or not, which generally requires something like your
> picture below.
> >>
> >>         Thanks,
> >>         Paul
> >>
> >>> Or this:
> >>>
> >>> +-----+              +-----+              +-----+
> >>> |     |     Call     |     |   Call       |     |
> >>> |UA-A +<------------>+B2BUA+<------------>+UA-B |
> >>> |     |              |     |              |     |
> >>> +-----+              +-++--+              +-----+
> >>>                         \\
> >>>                          \\
> >>>                       TBD \\
> >>>                            \\+----------+
> >>>                             \+          |
> >>>                              + Recorder |
> >>>                              |          |
> >>>                              +----------+
> >>> -hadriel
> >>>
> >>>> -----Original Message-----
> >>>> From: Roni Even [mailto:Even.roni@huawei.com]
> >>>> Sent: Thursday, June 11, 2009 3:54 PM
> >>>> To: Hadriel Kaplan; 'Hutton, Andrew'; 'Alan Johnston'; 'Romascanu,
> Dan
> >>>> (Dan)'
> >>>> Cc: 'Leon Portman'; dispatch@ietf.org; 'Jain, Rajnish'
> >>>> Subject: RE: [dispatch] Session recording in SIP
> >>>>
> >>>> Hadriel,
> >>>> I am still confused by the following text which was the reason for
> the
> >>>> previous question
> >>>>
> >>>> "This group will specify requirements for a SIP based protocol
> interface
> >>>> between a communications system and a recorder. The Communications
> System
> >>>> is
> >>>> responsible for establishing media sessions where the actual
> business is
> >>>> conducted. The Recorder is the sink of the recorded media."
> >>>>
> >>>>
> >>>> Can someone draw a figure showing the talking parties the
> Communication
> >>>> system and the recorder and explain where is this interface.
> According to
> >>>> your response I can think that there are at least two separate
> >>>> architectures
> >>>> your are proposing one where the parties (or one of them) in the
> call know
> >>>> the recorder and somehow forwards the media to it. The other
> architecture
> >>>> has some B2BUA in the call path who is either forking the media to
> the
> >>>> recorder and to the parties or the recorder is in the middle of
> the media
> >>>> path.
> >>> _______________________________________________
> >>> dispatch mailing list
> >>> dispatch@ietf.org
> >>> https://www.ietf.org/mailman/listinfo/dispatch
> >> _______________________________________________
> >> dispatch mailing list
> >> dispatch@ietf.org
> >> https://www.ietf.org/mailman/listinfo/dispatch
> >
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

From fluffy@cisco.com  Thu Jun 18 11:58:05 2009
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Subject: Re: [dispatch] Session recording in SIP
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Scott and Mary nicely expressed what I was thinking about where to run  
this discussion.

On Jun 12, 2009, at 14:04 , Scott Lawrence wrote:

>  I also agree with previous posters that while this overlaps
> with XCON and some other things, it's really a distinct problem and
> would probably be solvable more quickly in a more focused setting.   
> One
> of the major ideas behind our recent reorg was that smaller, more
> time-limited WGs will work better than the perpetual ones we've been
> using.



On Jun 10, 2009, at 19:08 , Mary Barnes wrote:

> At this stage, DISPATCH is the appropriate place to be having these
> discussions. Since this topic was just introduced to the WG yesterday,
> it's a bit too early, and not obvious based on the various opinions
> expressed thus far, to decide that it might belong elsewhere, need a  
> new
> WG/BoF, etc. So, let's continue the discussion on DISPATCH and refine
> the problem space and solution scope.

Lets focus the conversation on what is the problem to solve, what's  
the big picture, what needs to be standardized then we can send the  
right parts off to an existing group or form a new WG and do the  
technical work of developing the standard.

Cullen <RAI AD>



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Subject: Re: [dispatch] Session recording in SIP
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I look forward to reading the requirements draft when that comes out  
as it may answer this but ...

>>> A prime example where
>>> voice is used for trade is the financial industry. The call
>>> recording requirements in this industry are quite stringent.
>>> The recorded calls are used for dispute resolution and
>>> regulatory compliance. Other businesses such as customer
>>> support call centers typically employ call recording for
>>> quality control or business analytics.

So in the case where a customer is talking to an agent, my  
understanding is that some of the deployments (often legal compliance  
type situations) require an accurate recording of what the agent heard  
while other sorts of deployments may (perhaps a post call statistical  
analysis) only care about  an best effort approximation of the call.  
So in the first case, if the call could not be recorded, the call  
would not be set up but in the second case, the call would go ahead  
regardless.

I'm assuming the scope of the work would cover both case. Does that  
sound about right to others?

Cullen <with my individual contributor hat on>


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Subject: Re: [dispatch] IETF-75 plans for DISPATCH
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On Jun 18, 2009, at 6:21 , Vijay K. Gurbani wrote:

>
> In other words, once dispatch agrees to create a WG, what
> are the next steps that should be done?

It's the standard IETF process. Typical flow for things coming from  
DISPATCH would be submit a charter to ADs. ADs will put it on IESG  
agenda for approval to go out to IETF. When this is approved it will  
be sent out for IETF Last Call and comments collect. Then it goes back  
on IESG agenda for approval as a working group. From that point on it  
is run as a normal WG and get time allocated at IETF meetings.

Cullen


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Does this mean that "dispatch" is effectively a continuous BoF?

Or does it mean that AFTER you get the dispatch "seal of approval", THEN 
you have a BoF?

Janet

dispatch-bounces@ietf.org wrote on 06/18/2009 03:16:19 PM:

> 
> On Jun 18, 2009, at 6:21 , Vijay K. Gurbani wrote:
> 
> >
> > In other words, once dispatch agrees to create a WG, what
> > are the next steps that should be done?
> 
> It's the standard IETF process. Typical flow for things coming from 
> DISPATCH would be submit a charter to ADs. ADs will put it on IESG 
> agenda for approval to go out to IETF. When this is approved it will 
> be sent out for IETF Last Call and comments collect. Then it goes back 
> on IESG agenda for approval as a working group. From that point on it 
> is run as a normal WG and get time allocated at IETF meetings.
> 
> Cullen
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

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<br><font size=2 face="sans-serif"><br>
Does this mean that &quot;dispatch&quot; is effectively a continuous BoF?</font>
<br>
<br><font size=2 face="sans-serif">Or does it mean that AFTER you get the
dispatch &quot;seal of approval&quot;, THEN you have a BoF?</font>
<br>
<br><font size=2 face="sans-serif">Janet</font>
<br>
<br><font size=2><tt>dispatch-bounces@ietf.org wrote on 06/18/2009 03:16:19
PM:<br>
<br>
&gt; <br>
&gt; On Jun 18, 2009, at 6:21 , Vijay K. Gurbani wrote:<br>
&gt; <br>
&gt; &gt;<br>
&gt; &gt; In other words, once dispatch agrees to create a WG, what<br>
&gt; &gt; are the next steps that should be done?<br>
&gt; <br>
&gt; It's the standard IETF process. Typical flow for things coming from
&nbsp;<br>
&gt; DISPATCH would be submit a charter to ADs. ADs will put it on IESG
&nbsp;<br>
&gt; agenda for approval to go out to IETF. When this is approved it will
&nbsp;<br>
&gt; be sent out for IETF Last Call and comments collect. Then it goes
back &nbsp;<br>
&gt; on IESG agenda for approval as a working group. From that point on
it &nbsp;<br>
&gt; is run as a normal WG and get time allocated at IETF meetings.<br>
&gt; <br>
&gt; Cullen<br>
&gt; <br>
&gt; _______________________________________________<br>
&gt; dispatch mailing list<br>
&gt; dispatch@ietf.org<br>
&gt; https://www.ietf.org/mailman/listinfo/dispatch<br>
</tt></font>
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From mary.barnes@nortel.com  Thu Jun 18 12:51:03 2009
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I wouldn't look at it that way.  For example, for CLF, there was not a
BoF and a new WG is planned -  i.e., you don't have to have a BoF to
start a new WG and I would think that would be more the rule than the
exception. Out of the 7 items that were put on the table by June 8th,
only one BoF is planned.  So, dispatch is more of a continous discussion
of new ideas and farming out the work to other WGs, a BoF, a new WG or
as individual/AD sponsored work.  That all said, some of the topic
discussions  in the DISPATCH WG f2f meetings might look like "mini" BoFs
- for example the cc-uui topic.=20
=20
Regards,
Mary.=20

________________________________

From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Janet P Gunn
Sent: Thursday, June 18, 2009 2:27 PM
To: Cullen Jennings
Cc: dispatch-bounces@ietf.org; dispatch@ietf.org; rai-ads@tools.ietf.org
Subject: Re: [dispatch] IETF-75 plans for DISPATCH




Does this mean that "dispatch" is effectively a continuous BoF?=20

Or does it mean that AFTER you get the dispatch "seal of approval", THEN
you have a BoF?=20

Janet=20

dispatch-bounces@ietf.org wrote on 06/18/2009 03:16:19 PM:

>=20
> On Jun 18, 2009, at 6:21 , Vijay K. Gurbani wrote:
>=20
> >
> > In other words, once dispatch agrees to create a WG, what
> > are the next steps that should be done?
>=20
> It's the standard IETF process. Typical flow for things coming from =20
> DISPATCH would be submit a charter to ADs. ADs will put it on IESG =20
> agenda for approval to go out to IETF. When this is approved it will =20
> be sent out for IETF Last Call and comments collect. Then it goes back

> on IESG agenda for approval as a working group. From that point on it

> is run as a normal WG and get time allocated at IETF meetings.
>=20
> Cullen
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


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<BODY>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D593564819-18062009>I=20
wouldn't look at it that way. &nbsp;For example, for CLF, there was not =
a BoF=20
and a new WG is planned -&nbsp; i.e., you don't have to have a BoF to =
start a=20
new WG and I would think that would be more the rule than the exception. =
Out of=20
the 7 items that were put on the table by June 8th, only one BoF is=20
planned.&nbsp; So, dispatch is more of a continous discussion of new =
ideas and=20
farming out the work to other WGs, a BoF, a new WG or as individual/AD =
sponsored=20
work.&nbsp;&nbsp;That all said,&nbsp;some of the topic discussions =
&nbsp;in the=20
DISPATCH WG f2f meetings might look like "mini" BoFs - for example the =
cc-uui=20
topic. </SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D593564819-18062009></SPAN></FONT><FONT face=3DArial =
color=3D#0000ff=20
size=3D2><SPAN class=3D593564819-18062009></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D593564819-18062009>Regards,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D593564819-18062009>Mary.=20
</SPAN></FONT></DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> dispatch-bounces@ietf.org=20
[mailto:dispatch-bounces@ietf.org] <B>On Behalf Of </B>Janet P=20
Gunn<BR><B>Sent:</B> Thursday, June 18, 2009 2:27 PM<BR><B>To:</B> =
Cullen=20
Jennings<BR><B>Cc:</B> dispatch-bounces@ietf.org; dispatch@ietf.org;=20
rai-ads@tools.ietf.org<BR><B>Subject:</B> Re: [dispatch] IETF-75 plans =
for=20
DISPATCH<BR></FONT><BR></DIV>
<DIV></DIV><BR><FONT face=3Dsans-serif size=3D2><BR>Does this mean that =
"dispatch"=20
is effectively a continuous BoF?</FONT> <BR><BR><FONT face=3Dsans-serif =
size=3D2>Or=20
does it mean that AFTER you get the dispatch "seal of approval", THEN =
you have a=20
BoF?</FONT> <BR><BR><FONT face=3Dsans-serif size=3D2>Janet</FONT> =
<BR><BR><FONT=20
size=3D2><TT>dispatch-bounces@ietf.org wrote on 06/18/2009 03:16:19=20
PM:<BR><BR>&gt; <BR>&gt; On Jun 18, 2009, at 6:21 , Vijay K. Gurbani=20
wrote:<BR>&gt; <BR>&gt; &gt;<BR>&gt; &gt; In other words, once dispatch =
agrees=20
to create a WG, what<BR>&gt; &gt; are the next steps that should be=20
done?<BR>&gt; <BR>&gt; It's the standard IETF process. Typical flow for =
things=20
coming from &nbsp;<BR>&gt; DISPATCH would be submit a charter to ADs. =
ADs will=20
put it on IESG &nbsp;<BR>&gt; agenda for approval to go out to IETF. =
When this=20
is approved it will &nbsp;<BR>&gt; be sent out for IETF Last Call and =
comments=20
collect. Then it goes back &nbsp;<BR>&gt; on IESG agenda for approval as =
a=20
working group. From that point on it &nbsp;<BR>&gt; is run as a normal =
WG and=20
get time allocated at IETF meetings.<BR>&gt; <BR>&gt; Cullen<BR>&gt; =
<BR>&gt;=20
_______________________________________________<BR>&gt; dispatch mailing =

list<BR>&gt; dispatch@ietf.org<BR>&gt;=20
https://www.ietf.org/mailman/listinfo/dispatch<BR></TT></FONT></BODY></HT=
ML>

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From alan@sipstation.com  Thu Jun 18 14:26:25 2009
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Subject: Re: [dispatch] Session recording in SIP
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Cullen Jennings wrote:
>
> I look forward to reading the requirements draft when that comes out 
> as it may answer this but ...
>
>>>> A prime example where
>>>> voice is used for trade is the financial industry. The call
>>>> recording requirements in this industry are quite stringent.
>>>> The recorded calls are used for dispute resolution and
>>>> regulatory compliance. Other businesses such as customer
>>>> support call centers typically employ call recording for
>>>> quality control or business analytics.
>
> So in the case where a customer is talking to an agent, my 
> understanding is that some of the deployments (often legal compliance 
> type situations) require an accurate recording of what the agent heard 
> while other sorts of deployments may (perhaps a post call statistical 
> analysis) only care about  an best effort approximation of the call. 
> So in the first case, if the call could not be recorded, the call 
> would not be set up but in the second case, the call would go ahead 
> regardless.
>
> I'm assuming the scope of the work would cover both case. Does that 
> sound about right to others?
>

Yes - we need to cover the various modes.  We described four in our 
draft (http://tools.ietf.org/html/draft-wing-sipping-srtp-key):

1. Always On Recording
2. Recording On Demand
3. Required Recording
4. Pause and Resume Recording

There may be others as well, but these cover all the ones we support in 
our systems today.  Some are different by policy, others are different 
in actual call flows.

Thanks,
Alan

> Cullen <with my individual contributor hat on>
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>


From volkerh@bell-labs.com  Fri Jun 19 06:52:42 2009
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Date: Fri, 19 Jun 2009 15:55:46 +0200
From: Volker Hilt <volkerh@bell-labs.com>
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Subject: [dispatch] SIP overload control mailing list
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All,

we have set up a new mailing list for discussions around SIP overload 
control. We will move the discussions of the SIP overload control design 
team to the new list. Everyone interested in actively contributing to 
the topic is welcome to join.

The URL of the list is:
https://www.ietf.org/mailman/listinfo/sip-overload

Thanks,

Volker

From vkg@alcatel-lucent.com  Sun Jun 21 21:08:42 2009
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Date: Sun, 21 Jun 2009 23:09:26 -0500
From: Vijay Gurbani <vkg@alcatel-lucent.com>
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Subject: [dispatch] Summary on Session Recording in SIP
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All: Here is the summary of the Session Recording thread from June 9
to June 19, 2009.  If I misrepresented anything, please let me know.

There was a fair amount of discussion on where to conduct this
work.  In the end, it was decided to continue discussions on
dispatch, further decisions what to do -- new working group or
put this work in an existing working group -- will be decided
later.  Regardless of where the work is done, opinions were
expressed that the work is important that it should commence
in the IETF.

Need to be crisp about the purpose of this work: is it that users
record sessions to hear later, or sessions need to be recorded
because of some business needs?  Or both?  Is SRTP covered or
only RTP?

Solutions should cover:
  Always on recording
  Recording on demand
  Required recording
  Pause and resume recording
  Playback facilities and how they interact with recording
  Recording formats and protocols for controlling the stored recording.

A lot of discussion ensued on the specific architecture or
architectures that would be possible.  No decision was reached on
which specific architecture is good or bad, although various opinions
were expressed in support for each.

There are essentially four architectures to realize recording
services: two that discuss where the media stream is forked
out to the recorder -- at the UA or in a middlebox of some
sort are outlined here:
   http://www.ietf.org/mail-archive/web/dispatch/current/msg00226.html

The third architecture has both UAs sending media to a
recorder.  See:
     http://www.ietf.org/mail-archive/web/dispatch/current/msg00236.html

A fourth architecture that is a superset of the third one is at:
     http://www.ietf.org/mail-archive/web/dispatch/current/msg00256.html

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
WWW:   http://ect.bell-labs.com/who/vkg

From fluffy@cisco.com  Mon Jun 22 10:58:01 2009
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From: Cullen Jennings <fluffy@cisco.com>
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Cc: dispatch@ietf.org, rai-ads@tools.ietf.org
Subject: Re: [dispatch] IETF-75 plans for DISPATCH
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On Jun 18, 2009, at 12:26 , Janet P Gunn wrote:

>
>
> Does this mean that "dispatch" is effectively a continuous BoF?
>
> Or does it mean that AFTER you get the dispatch "seal of approval",  
> THEN you have a BoF?
>

Closer to the first than the second. We certainly don't want to add an  
extra step to make it even take longer to form new WGs. The general  
plan is for things that get consensus or form a WG in DISPATCH, to go  
directly to IESG approval step without a BOF.

> Janet
>
> dispatch-bounces@ietf.org wrote on 06/18/2009 03:16:19 PM:
>
> >
> > On Jun 18, 2009, at 6:21 , Vijay K. Gurbani wrote:
> >
> > >
> > > In other words, once dispatch agrees to create a WG, what
> > > are the next steps that should be done?
> >
> > It's the standard IETF process. Typical flow for things coming from
> > DISPATCH would be submit a charter to ADs. ADs will put it on IESG
> > agenda for approval to go out to IETF. When this is approved it will
> > be sent out for IETF Last Call and comments collect. Then it goes  
> back
> > on IESG agenda for approval as a working group. From that point on  
> it
> > is run as a normal WG and get time allocated at IETF meetings.
> >
> > Cullen
> >
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch


From pkyzivat@cisco.com  Mon Jun 22 11:35:55 2009
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From: Paul Kyzivat <pkyzivat@cisco.com>
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Cc: dispatch@ietf.org
Subject: Re: [dispatch] Summary on Session Recording in SIP
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A real simple question about intended scope:

Does this include a direct session between a UAC and a UAS whose purpose 
is recording, such as for a VM system? (A VM system does seem to be a 
special case of "recording on demand", and also often includes playback 
facilities.)

I had the impression that this work was more scoped toward cases where 
there is an interactive session between two or more parties, and that 
recording is intended to be supplemental to that.

	Thanks,
	Paul

Vijay Gurbani wrote:
> All: Here is the summary of the Session Recording thread from June 9
> to June 19, 2009.  If I misrepresented anything, please let me know.
> 
> There was a fair amount of discussion on where to conduct this
> work.  In the end, it was decided to continue discussions on
> dispatch, further decisions what to do -- new working group or
> put this work in an existing working group -- will be decided
> later.  Regardless of where the work is done, opinions were
> expressed that the work is important that it should commence
> in the IETF.
> 
> Need to be crisp about the purpose of this work: is it that users
> record sessions to hear later, or sessions need to be recorded
> because of some business needs?  Or both?  Is SRTP covered or
> only RTP?
> 
> Solutions should cover:
>  Always on recording
>  Recording on demand
>  Required recording
>  Pause and resume recording
>  Playback facilities and how they interact with recording
>  Recording formats and protocols for controlling the stored recording.
> 
> A lot of discussion ensued on the specific architecture or
> architectures that would be possible.  No decision was reached on
> which specific architecture is good or bad, although various opinions
> were expressed in support for each.
> 
> There are essentially four architectures to realize recording
> services: two that discuss where the media stream is forked
> out to the recorder -- at the UA or in a middlebox of some
> sort are outlined here:
>   http://www.ietf.org/mail-archive/web/dispatch/current/msg00226.html
> 
> The third architecture has both UAs sending media to a
> recorder.  See:
>     http://www.ietf.org/mail-archive/web/dispatch/current/msg00236.html
> 
> A fourth architecture that is a superset of the third one is at:
>     http://www.ietf.org/mail-archive/web/dispatch/current/msg00256.html
> 
> - vijay

From jmh@joelhalpern.com  Mon Jun 22 11:55:44 2009
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I am having trouble understanding why playback is part of this at all.
There appear to be two cases:
1) Playback after the call has ended.  That is clearly a distinct 
application.  Whether it is delivered by SIP, HTTP, or some other 
applicaiton protocol does not matter.

2) Playback during a call.  That would seem to be a combination of (1) 
plus the general problem of inserting media from another application 
into an ongoing call.  Again not part of the recording problem.

There do seem to be ligitmate items that need work in the SIP area on 
enabling several different kinds of call recording.  So I think there is 
work to be done.  I am just confused by the inclusion of this dimension 
(other than "because someone asked for it.")

Yours,
Joel

Paul Kyzivat wrote:
> A real simple question about intended scope:
> 
> Does this include a direct session between a UAC and a UAS whose purpose 
> is recording, such as for a VM system? (A VM system does seem to be a 
> special case of "recording on demand", and also often includes playback 
> facilities.)
> 
> I had the impression that this work was more scoped toward cases where 
> there is an interactive session between two or more parties, and that 
> recording is intended to be supplemental to that.
> 
>     Thanks,
>     Paul
> 
> Vijay Gurbani wrote:
>> All: Here is the summary of the Session Recording thread from June 9
>> to June 19, 2009.  If I misrepresented anything, please let me know.
>>
>> There was a fair amount of discussion on where to conduct this
>> work.  In the end, it was decided to continue discussions on
>> dispatch, further decisions what to do -- new working group or
>> put this work in an existing working group -- will be decided
>> later.  Regardless of where the work is done, opinions were
>> expressed that the work is important that it should commence
>> in the IETF.
>>
>> Need to be crisp about the purpose of this work: is it that users
>> record sessions to hear later, or sessions need to be recorded
>> because of some business needs?  Or both?  Is SRTP covered or
>> only RTP?
>>
>> Solutions should cover:
>>  Always on recording
>>  Recording on demand
>>  Required recording
>>  Pause and resume recording
>>  Playback facilities and how they interact with recording
>>  Recording formats and protocols for controlling the stored recording.
>>
>> A lot of discussion ensued on the specific architecture or
>> architectures that would be possible.  No decision was reached on
>> which specific architecture is good or bad, although various opinions
>> were expressed in support for each.
>>
>> There are essentially four architectures to realize recording
>> services: two that discuss where the media stream is forked
>> out to the recorder -- at the UA or in a middlebox of some
>> sort are outlined here:
>>   http://www.ietf.org/mail-archive/web/dispatch/current/msg00226.html
>>
>> The third architecture has both UAs sending media to a
>> recorder.  See:
>>     http://www.ietf.org/mail-archive/web/dispatch/current/msg00236.html
>>
>> A fourth architecture that is a superset of the third one is at:
>>     http://www.ietf.org/mail-archive/web/dispatch/current/msg00256.html
>>
>> - vijay
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

From vkg@alcatel-lucent.com  Mon Jun 22 12:14:38 2009
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Subject: Re: [dispatch] Summary on Session Recording in SIP
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Paul Kyzivat wrote:
> A real simple question about intended scope:
> 
> Does this include a direct session between a UAC and a UAS whose purpose 
> is recording, such as for a VM system? (A VM system does seem to be a 
> special case of "recording on demand", and also often includes playback 
> facilities.)
> 
> I had the impression that this work was more scoped toward cases where 
> there is an interactive session between two or more parties, and that 
> recording is intended to be supplemental to that.

Paul: When we (i.e., the author team) had initially started
this work, the latter was more of what we had in mind.

Thanks,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/

From spencer@wonderhamster.org  Mon Jun 22 15:14:07 2009
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From: "Spencer Dawkins" <spencer@wonderhamster.org>
To: "Joel M. Halpern" <jmh@joelhalpern.com>, <dispatch@ietf.org>
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Subject: Re: [dispatch] Summary on Session Recording in SIP
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----- Original Message ----- 
From: "Joel M. Halpern" <jmh@joelhalpern.com>
To: <dispatch@ietf.org>
Sent: Monday, June 22, 2009 1:55 PM
Subject: Re: [dispatch] Summary on Session Recording in SIP


>I am having trouble understanding why playback is part of this at all.

As am I, FWIW...

> There appear to be two cases:
> 1) Playback after the call has ended.  That is clearly a distinct 
> application.  Whether it is delivered by SIP, HTTP, or some other 
> applicaiton protocol does not matter.
> 
> 2) Playback during a call.  That would seem to be a combination of (1) 
> plus the general problem of inserting media from another application 
> into an ongoing call.  Again not part of the recording problem.
> 
> There do seem to be ligitmate items that need work in the SIP area on 
> enabling several different kinds of call recording.  So I think there is 
> work to be done.  I am just confused by the inclusion of this dimension 
> (other than "because someone asked for it.")

What Joel said.

Thanks,

Spencer


From andrew.hutton@siemens-enterprise.com  Tue Jun 23 02:35:36 2009
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Date: Tue, 23 Jun 2009 10:35:52 +0100
From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
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Hi,

I would support dropping playback from the requirements,

Regards
Andy=20

>-----Original Message-----
>From: dispatch-bounces@ietf.org=20
>[mailto:dispatch-bounces@ietf.org] On Behalf Of Spencer Dawkins
>Sent: 22 June 2009 23:14
>To: Joel M. Halpern; dispatch@ietf.org
>Subject: Re: [dispatch] Summary on Session Recording in SIP
>
>
>----- Original Message -----=20
>From: "Joel M. Halpern" <jmh@joelhalpern.com>
>To: <dispatch@ietf.org>
>Sent: Monday, June 22, 2009 1:55 PM
>Subject: Re: [dispatch] Summary on Session Recording in SIP
>
>
>>I am having trouble understanding why playback is part of this at all.
>
>As am I, FWIW...
>
>> There appear to be two cases:
>> 1) Playback after the call has ended.  That is clearly a distinct=20
>> application.  Whether it is delivered by SIP, HTTP, or some other=20
>> applicaiton protocol does not matter.
>>=20
>> 2) Playback during a call.  That would seem to be a=20
>combination of (1)=20
>> plus the general problem of inserting media from another application=20
>> into an ongoing call.  Again not part of the recording problem.
>>=20
>> There do seem to be ligitmate items that need work in the=20
>SIP area on=20
>> enabling several different kinds of call recording.  So I=20
>think there is=20
>> work to be done.  I am just confused by the inclusion of=20
>this dimension=20
>> (other than "because someone asked for it.")
>
>What Joel said.
>
>Thanks,
>
>Spencer
>
>_______________________________________________
>dispatch mailing list
>dispatch@ietf.org
>https://www.ietf.org/mailman/listinfo/dispatch
>

From john.elwell@siemens-enterprise.com  Thu Jun 25 00:57:20 2009
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
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=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Vijay Gurbani
> Sent: 22 June 2009 05:09
> To: dispatch@ietf.org
> Subject: [dispatch] Summary on Session Recording in SIP
>=20
> All: Here is the summary of the Session Recording thread from June 9
> to June 19, 2009.  If I misrepresented anything, please let me know.
>=20
> There was a fair amount of discussion on where to conduct this
> work.  In the end, it was decided to continue discussions on
> dispatch, further decisions what to do -- new working group or
> put this work in an existing working group -- will be decided
> later.  Regardless of where the work is done, opinions were
> expressed that the work is important that it should commence
> in the IETF.
>=20
> Need to be crisp about the purpose of this work: is it that users
> record sessions to hear later, or sessions need to be recorded
> because of some business needs?  Or both?  Is SRTP covered or
> only RTP?
[JRE] Whatever architecture(s) we adopt, must allow SRTP to be used on
the original call. We can then discuss various options on the recorded
leg, e.g., SRTP using the same keys as the original call (i.e., without
decryption and re-encryption), SRTP with separate keys, or RTP.

John


>=20
> Solutions should cover:
>   Always on recording
>   Recording on demand
>   Required recording
>   Pause and resume recording
>   Playback facilities and how they interact with recording
>   Recording formats and protocols for controlling the stored=20
> recording.
>=20
> A lot of discussion ensued on the specific architecture or
> architectures that would be possible.  No decision was reached on
> which specific architecture is good or bad, although various opinions
> were expressed in support for each.
>=20
> There are essentially four architectures to realize recording
> services: two that discuss where the media stream is forked
> out to the recorder -- at the UA or in a middlebox of some
> sort are outlined here:
>    http://www.ietf.org/mail-archive/web/dispatch/current/msg00226.html
>=20
> The third architecture has both UAs sending media to a
> recorder.  See:
>     =20
> http://www.ietf.org/mail-archive/web/dispatch/current/msg00236.html
>=20
> A fourth architecture that is a superset of the third one is at:
>     =20
> http://www.ietf.org/mail-archive/web/dispatch/current/msg00256.html
>=20
> - vijay
> --=20
> Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
> 1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
> Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
> WWW:   http://ect.bell-labs.com/who/vkg
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

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Thread-Topic: Updated agenda for IETF-75
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: <dispatch@ietf.org>
Subject: [dispatch] Updated agenda for IETF-75
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Hi folks,

We have uploaded a revised (tentative) agenda for IETF-75 for the
DISPATCH WG session:
http://www.ietf.org/proceedings/09jul/agenda/dispatch.html

Due to popular demand, we have added a slot for Session Recording. Note,
that the current agenda has us meeting on Thursday afternoon, but Cullen
has asked for that to be changed to Monday. As always, the agenda is
subject to change up until the meeting.

As a reminder any new documents related to the agenda topics must be
submitted by July 6th and any updates to current docs by July 13th. And,
most importantly we really should be seeing traffic on the list on these
topics before the meeting so we can have focused discussions with
conclusions and a well defined way forward.=20

Please send any comments on the proposed agenda to the list and/or
Gonzalo and myself. =20

Regards,
Mary
DISPATCH WG co-chair

From john.elwell@siemens-enterprise.com  Sun Jun 28 23:46:54 2009
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Date: Mon, 29 Jun 2009 07:47:11 +0100
From: "Elwell, John" <john.elwell@siemens-enterprise.com>
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Thread-Topic: draft-elwell-dispatch-identity-reqs-00 posted
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There was an extensive discussion at IETF#74 on SIP Identity, where a =
good many views were put forward. The only consensus seemed to be to =
continue to discuss the topic.

One of the themes during that discussion was to focus more on =
requirements, which the authors have attempted to do in this new draft:
http://www.ietf.org/internet-drafts/draft-elwell-dispatch-identity-reqs-0=
0.txt

We would appreciate feedback as to whether this is helpful and going in =
the right direction, as well as detailed comments.

I had hoped to do this a lot earlier, to trigger a discussion in time to =
get something set up for DISPATCH at IETF#75, but I failed, and also I =
failed to talk to the many of you who have interest in the topic and =
expressed opinions in the past. But since the deadline is approaching, I =
thought it best to post the draft and let discussion continue on that =
basis.

John
