
From rmohanr@cisco.com  Wed Jan  4 22:15:33 2012
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From: "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>
To: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>, <dispatch@ietf.org>
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Subject: Re: [dispatch] SIP media load balancer requirement
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Hi Partha,

I would be interested in this work. See more inline

> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Ravindran, Parthasarathi
> Sent: Thursday, December 22, 2011 8:02 AM
> To: dispatch@ietf.org
> Subject: [dispatch] SIP media load balancer requirement
>=20
> Hi all,
>=20
> As a continuation of IETF-81 Dispatch minutes
> (http://www.ietf.org/mail-archive/web/dispatch/current/msg03742.html)
> about SIP load balancing, I thought of discussing about SIP media
> servers load balancing as mentioned in the SIP load balancing proposed
> charter (http://www.ietf.org/mail-
> archive/web/dispatch/current/msg03615.html).
>=20
> "As SIP request resource consumption in SIP signaling only server
> varies drastically from SIP media servers [3], should the solution be
> split such that load balancing of a pure signaling server is different
> than that of a SIP server that handles signaling as well as media?"
>=20
> SIP media server load balancing solution will be different from
> signaling only because the downstream resource consumption is mainly
> because of media  (RTP) than signaling (SIP) and the units to indicate
> the load balancing information will be different in both solution.
> There is a need to indicate those media resource consumption to the
> load balancer in the standard manner. Please let me know your comment
> on this.

I think what we need is some text/document saying how media overload is
different from signaling overload and what are the different resources
that need to considered for media overload. In that way we will have
good starting point to discuss on this topic.

Regards,
Ram
>=20
> Thanks
> Partha
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

From pravindran@sonusnet.com  Wed Jan  4 22:27:44 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Thread-Topic: [dispatch] SIP media load balancer requirement
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Subject: Re: [dispatch] SIP media load balancer requirement
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Hi Ram,

Nice to see your interest. Please read inline for more comments.

Thanks
Partha

>-----Original Message-----
>From: Ram Mohan R (rmohanr) [mailto:rmohanr@cisco.com]
>Sent: Thursday, January 05, 2012 11:45 AM
>To: Ravindran, Parthasarathi; dispatch@ietf.org
>Subject: RE: [dispatch] SIP media load balancer requirement
>
>Hi Partha,
>
>I would be interested in this work. See more inline
>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>> Behalf Of Ravindran, Parthasarathi
>> Sent: Thursday, December 22, 2011 8:02 AM
>> To: dispatch@ietf.org
>> Subject: [dispatch] SIP media load balancer requirement
>>
>> Hi all,
>>
>> As a continuation of IETF-81 Dispatch minutes
>> (http://www.ietf.org/mail-archive/web/dispatch/current/msg03742.html)
>> about SIP load balancing, I thought of discussing about SIP media
>> servers load balancing as mentioned in the SIP load balancing proposed
>> charter (http://www.ietf.org/mail-
>> archive/web/dispatch/current/msg03615.html).
>>
>> "As SIP request resource consumption in SIP signaling only server
>> varies drastically from SIP media servers [3], should the solution be
>> split such that load balancing of a pure signaling server is different
>> than that of a SIP server that handles signaling as well as media?"
>>
>> SIP media server load balancing solution will be different from
>> signaling only because the downstream resource consumption is mainly
>> because of media  (RTP) than signaling (SIP) and the units to indicate
>> the load balancing information will be different in both solution.
>> There is a need to indicate those media resource consumption to the
>> load balancer in the standard manner. Please let me know your comment
>> on this.
>
>I think what we need is some text/document saying how media overload is
>different from signaling overload and what are the different resources
>that need to considered for media overload. In that way we will have
>good starting point to discuss on this topic.

<partha> The focus of this charter is about SIP media load balancing and no=
t media overload handling. I agree with you that SIP media load balancing a=
lgorithm works in conjunction with media overload mechanism and selection o=
f the load balancing algorithm majorly depends upon media overload mechanis=
m.=20

SIP media load balancer requirement is to consider media resource consumpti=
on of the peer apart from the signaling resource consumption.  </partha>

>
>Regards,
>Ram
>>
>> Thanks
>> Partha
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch

From mary.ietf.barnes@gmail.com  Mon Jan  9 09:29:40 2012
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From: Mary Barnes <mary.ietf.barnes@gmail.com>
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Subject: [dispatch] Reminder: DISPATCH WG Deadlines for IETF-83
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Hi folks,

A reminder that the first deadline for the upcoming meeting is in one
week's time (Monday, January 16th).  This is an easy one - you just
need to send the chairs an email that you intend to put forth a topic
and please identify the topic.   Then you have a week to put together
details for the topic proposal.  If your topic garners WG interest on
the mailing list over the next two weeks, then it will be considered
in developing the agenda for the f2f meeting.  You have up until the
normal meeting deadlines to submit any related drafts.

Thanks,
Mary.
---------- Forwarded message ----------
From: Mary Barnes <mary.ietf.barnes@gmail.com>
Date: Thu, Dec 8, 2011 at 2:28 PM
Subject: Deadlines for IETF-83
To: DISPATCH <dispatch@ietf.org>
Cc: Cullen Jennings <fluffy@cisco.com>, rai-ads@tools.ietf.org


Hi folks,

As presented in Taipei and per the wiki, the deadlines for IETF-83 are
as follows:
http://trac.tools.ietf.org/wg/dispatch/trac/wiki

* Jan 16th, 2012. Cutoff date to notify the chairs/DISPATCH WG of
plans to submit a proposal. [Two weeks prior to deadline for
scheduling WG sessions]
* Jan 23, 2012. Cutoff for charter proposals for topics. [Three weeks
prior to BoF proposal deadline, two weeks before announcement of
topics]
* Feb 6, 2012. Topics that are to be the focus of IETF-83 are
announced. [One week before AD BoF approval, 4 weeks before -00
deadline, *one week* after deadline for requesting WG session]
* March 5, 2012. -00 draft deadline.
* March 12, 2012, 2011. Draft deadline.

So, please send proposals ASAP and comment on those topics that were
posted after the IETF-82 deadlines (e.g., STRAW). =A0The sooner
discussion is started, the better prepared we will be to dispatch the
item before or at the meeting.

Thanks,
Mary
DISPATCH WG co-chair

From petithug@acm.org  Mon Jan 16 10:25:43 2012
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Subject: [dispatch] Fwd: I-D Action: draft-petithuguenin-dispatch-unique-overlay-01.txt
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA256

I would like to request a timeslot in the DISPATCH session in Paris to talk
about infrastructure overlays, which is a special class of RELOAD overlays
that by their own definition require that only one instance of this overlay
exists on the Internet (excluding instances deployed for test and
experimentation).

This kind of overlay could be useful for stuff like bitcoin or boinc, but the
main motivation is to support VIPR, which works only if there is a unique
overlay instance.

The discussion will be around the problems that infrastructure overlays solve
and the list of requirements for the mechanism used to implement it but,
although this is not part of the discussion, the new version of the draft
contains an appendix explaining one possible solution.


- -------- Original Message --------
Subject: I-D Action: draft-petithuguenin-dispatch-unique-overlay-01.txt
Date: Mon, 16 Jan 2012 09:54:44 -0800
From: internet-drafts@ietf.org
Reply-To: internet-drafts@ietf.org
To: i-d-announce@ietf.org


A New Internet-Draft is available from the on-line Internet-Drafts directories.

	Title           : Infrastructure Overlay
	Author(s)       : Marc Petit-Huguenin
	Filename        : draft-petithuguenin-dispatch-unique-overlay-01.txt
	Pages           : 6
	Date            : 2012-01-16

   This document provides requirements for infrastructure overlays, a
   special kind of peer-to-peer overlay whose main purpose would be
   defeated if more than one instance would exist on the Internet.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-petithuguenin-dispatch-unique-overlay-01.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-petithuguenin-dispatch-unique-overlay-01.txt

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From sapnas@s-mail.com  Tue Jan 17 05:15:23 2012
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Date: Tue, 17 Jan 2012 13:15:20 +0000
From: sapna s <sapnas@s-mail.com>
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Subject: [dispatch] Avoidance draft review
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Hi,

  Please review the *silent* draft in dispatch group for security issue
avoidance.

  Author was earlier talking abt top prize to top person. Top Person was
having Prior Knowledge about whereabouts of Most Wanted Person (who was
dead in APR last or May first week , who was in the region whose head was
sometime Musharraf). Web site at www.<tr ial><last name of top person> .com


  Looks like due to which top person is blocking the review. Please review
draft.

Thx
Sapna S
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From rifatyu@avaya.com  Fri Jan 20 10:55:47 2012
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From: "Shekh-Yusef, Rifaat (Rifaat)" <rifatyu@avaya.com>
To: "dispatch@ietf.org" <dispatch@ietf.org>
Date: Fri, 20 Jan 2012 13:55:35 -0500
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Cullen, Mary,

This is to request a DISPATCH timeslot to discuss the draft "Conference Foc=
us Indicating CCMP Support".
http://tools.ietf.org/id/draft-yusef-xcon-ccmp-indication-00.txt

The CCMP protocol allows a UA to discover if a conference server supports C=
CMP.
The draft it trying to address the need for a UA to discover if a conferenc=
e focus supports CCMP.
The draft presents multiple options and pros and cons of each one.

We would appreciate any comments.

Regards,
Rifaat


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nk=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal>Cullen, Mary,<o:=
p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>=
This is to request a DISPATCH timeslot to discuss the draft &#8220;Conferen=
ce Focus Indicating CCMP Support&#8221;.<o:p></o:p></p><p class=3DMsoNormal=
><a href=3D"http://tools.ietf.org/id/draft-yusef-xcon-ccmp-indication-00.tx=
t">http://tools.ietf.org/id/draft-yusef-xcon-ccmp-indication-00.txt</a><o:p=
></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>T=
he CCMP protocol allows a UA to discover if a conference server supports CC=
MP.<o:p></o:p></p><p class=3DMsoNormal>The draft it trying to address the n=
eed for a UA to discover if a conference focus supports CCMP.<o:p></o:p></p=
><p class=3DMsoNormal>The draft presents multiple options and pros and cons=
 of each one.<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p cl=
ass=3DMsoNormal>We would appreciate any comments.<o:p></o:p></p><p class=3D=
MsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Regards,<o:p></o:p></p>=
<p class=3DMsoNormal> Rifaat<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;=
</o:p></p></div></body></html>=

--_000_6369CB70BFD88942B9705AC1E639A3382310F063F6DCUS1MBEX4glo_--

From bo.burman@ericsson.com  Mon Jan 23 07:50:28 2012
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From: Bo Burman <bo.burman@ericsson.com>
To: "dispatch@ietf.org" <dispatch@ietf.org>
Date: Mon, 23 Jan 2012 16:50:24 +0100
Thread-Topic: Media Stream related signaling
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Dispatch,

We have requested agenda time for the upcoming meeting to discuss the subje=
ct of Media Stream related signaling. From the feedback we got in AVTEXT WG=
 on the discussion for a certain problem, the one related to pausing and re=
suming sender to middlebox media traffic, it is clear that a more high leve=
l discussion needs to happen.

We have two Internet drafts related to the issues we would like to discuss.
Media Stream Selection (MESS)
 - draft-westerlund-dispatch-stream-selection-00
RTP Media Stream Pause and Resume
 - draft-westerlund-avtext-rtp-stream-pause-00

The first one (MESS) proposes that for the general selection of media strea=
ms, an end-point within a centralized conferencing scenario desires to rece=
ive a media plane oriented rather than RTP/RTCP based signaling, which appe=
ars to have benefits as discussed in the draft.

As mentioned, the pause resume draft is targeted for a different use case w=
here a media plane solution has potential for other benefits. This was disc=
ussed and the main outcome of that discussion was three quite different opi=
nions;

a) We already do this signaling using TMMBR (RFC 5104) by setting the bit-r=
ate value to 0, which the current specs doesn't allow.

b) This should be done in signalling plane, RTP/RTCP shouldn't be bloated. =
(But I did interpret this as not necessarily needing to be SIP).

c) When it comes to pause/resume, unless it is done using SIP/SDP it will n=
ot work through any Session Border Gateway as the session will be torn down=
 after n seconds.

Thus I think it important that we take a discussion around the use cases, e=
xisting solutions, limitations in those or the proposal to try to determine=
 what is the most reasonable way of solving the problems and how to enable;

1) Receiver based selection of media streams from a larger set of streams, =
including different quality levels like video resolutions, in a centralized=
 conferencing application.

2) Optimize the transport so that media streams currently not used can be t=
urned off temporarily and then quickly resumed when someone determines a ne=
ed for the media stream. For example using mechanism (1) but also for voice=
 activity based, centrally controlled or other mechanisms to select which m=
edia streams are delivered to other session participants.

We are working to revise our Internet drafts. We seek as much input as poss=
ible into this prior to the meeting so that we can ensure that any discussi=
on are as on topic and focused as possible.

Regards

Bo Burman & Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7141311
F=E4r=F6gatan 6                | Mobile +46 73 0949021
SE-164 80 Stockholm, Sweden| mailto: bo.burman at ericsson.com
----------------------------------------------------------------------


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Date: Mon, 23 Jan 2012 11:44:00 -0500
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From: Michael Hammer <mphmmr@gmail.com>
To: Bo Burman <bo.burman@ericsson.com>
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Cc: "dispatch@ietf.org" <dispatch@ietf.org>
Subject: Re: [dispatch] Media Stream related signaling
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--f46d040892c3c76ae304b734bc49
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Bo,

Rather than "turning [the media] stream off temporarily", you should
consider putting the media stream into a low energy state, which would not
be bit rate =3D 0 but would have some limited keep-alive stream that the SB=
C
recognizes to keep the pinhole open.

There are probably multiple ways to make that happen.  Hope that is
semantically what you meant.

Mike


On Mon, Jan 23, 2012 at 10:50 AM, Bo Burman <bo.burman@ericsson.com> wrote:

> Dispatch,
>
> We have requested agenda time for the upcoming meeting to discuss the
> subject of Media Stream related signaling. From the feedback we got in
> AVTEXT WG on the discussion for a certain problem, the one related to
> pausing and resuming sender to middlebox media traffic, it is clear that =
a
> more high level discussion needs to happen.
>
> We have two Internet drafts related to the issues we would like to discus=
s.
> Media Stream Selection (MESS)
>  - draft-westerlund-dispatch-stream-selection-00
> RTP Media Stream Pause and Resume
>  - draft-westerlund-avtext-rtp-stream-pause-00
>
> The first one (MESS) proposes that for the general selection of media
> streams, an end-point within a centralized conferencing scenario desires =
to
> receive a media plane oriented rather than RTP/RTCP based signaling, whic=
h
> appears to have benefits as discussed in the draft.
>
> As mentioned, the pause resume draft is targeted for a different use case
> where a media plane solution has potential for other benefits. This was
> discussed and the main outcome of that discussion was three quite differe=
nt
> opinions;
>
> a) We already do this signaling using TMMBR (RFC 5104) by setting the
> bit-rate value to 0, which the current specs doesn't allow.
>
> b) This should be done in signalling plane, RTP/RTCP shouldn't be bloated=
.
> (But I did interpret this as not necessarily needing to be SIP).
>
> c) When it comes to pause/resume, unless it is done using SIP/SDP it will
> not work through any Session Border Gateway as the session will be torn
> down after n seconds.
>
> Thus I think it important that we take a discussion around the use cases,
> existing solutions, limitations in those or the proposal to try to
> determine what is the most reasonable way of solving the problems and how
> to enable;
>
> 1) Receiver based selection of media streams from a larger set of streams=
,
> including different quality levels like video resolutions, in a centraliz=
ed
> conferencing application.
>
> 2) Optimize the transport so that media streams currently not used can be
> turned off temporarily and then quickly resumed when someone determines a
> need for the media stream. For example using mechanism (1) but also for
> voice activity based, centrally controlled or other mechanisms to select
> which media streams are delivered to other session participants.
>
> We are working to revise our Internet drafts. We seek as much input as
> possible into this prior to the meeting so that we can ensure that any
> discussion are as on topic and focused as possible.
>
> Regards
>
> Bo Burman & Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7141311
> F=E4r=F6gatan 6                | Mobile +46 73 0949021
> SE-164 80 Stockholm, Sweden| mailto: bo.burman at ericsson.com
> ----------------------------------------------------------------------
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

--f46d040892c3c76ae304b734bc49
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Bo,<div><br></div><div>Rather than &quot;turning [the media] stream off tem=
porarily&quot;, you should consider putting the media stream into a low ene=
rgy state, which would not be bit rate =3D 0 but would have some limited ke=
ep-alive stream that the SBC recognizes to keep the pinhole open.</div>
<div><br></div><div>There are probably multiple ways to make that happen. =
=A0Hope that is semantically what you meant.</div><div><br></div><div>Mike<=
/div><div><br><br><div class=3D"gmail_quote">On Mon, Jan 23, 2012 at 10:50 =
AM, Bo Burman <span dir=3D"ltr">&lt;<a href=3D"mailto:bo.burman@ericsson.co=
m">bo.burman@ericsson.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Dispatch,<br>
<br>
We have requested agenda time for the upcoming meeting to discuss the subje=
ct of Media Stream related signaling. From the feedback we got in AVTEXT WG=
 on the discussion for a certain problem, the one related to pausing and re=
suming sender to middlebox media traffic, it is clear that a more high leve=
l discussion needs to happen.<br>

<br>
We have two Internet drafts related to the issues we would like to discuss.=
<br>
Media Stream Selection (MESS)<br>
=A0- draft-westerlund-dispatch-stream-selection-00<br>
RTP Media Stream Pause and Resume<br>
=A0- draft-westerlund-avtext-rtp-stream-pause-00<br>
<br>
The first one (MESS) proposes that for the general selection of media strea=
ms, an end-point within a centralized conferencing scenario desires to rece=
ive a media plane oriented rather than RTP/RTCP based signaling, which appe=
ars to have benefits as discussed in the draft.<br>

<br>
As mentioned, the pause resume draft is targeted for a different use case w=
here a media plane solution has potential for other benefits. This was disc=
ussed and the main outcome of that discussion was three quite different opi=
nions;<br>

<br>
a) We already do this signaling using TMMBR (RFC 5104) by setting the bit-r=
ate value to 0, which the current specs doesn&#39;t allow.<br>
<br>
b) This should be done in signalling plane, RTP/RTCP shouldn&#39;t be bloat=
ed. (But I did interpret this as not necessarily needing to be SIP).<br>
<br>
c) When it comes to pause/resume, unless it is done using SIP/SDP it will n=
ot work through any Session Border Gateway as the session will be torn down=
 after n seconds.<br>
<br>
Thus I think it important that we take a discussion around the use cases, e=
xisting solutions, limitations in those or the proposal to try to determine=
 what is the most reasonable way of solving the problems and how to enable;=
<br>

<br>
1) Receiver based selection of media streams from a larger set of streams, =
including different quality levels like video resolutions, in a centralized=
 conferencing application.<br>
<br>
2) Optimize the transport so that media streams currently not used can be t=
urned off temporarily and then quickly resumed when someone determines a ne=
ed for the media stream. For example using mechanism (1) but also for voice=
 activity based, centrally controlled or other mechanisms to select which m=
edia streams are delivered to other session participants.<br>

<br>
We are working to revise our Internet drafts. We seek as much input as poss=
ible into this prior to the meeting so that we can ensure that any discussi=
on are as on topic and focused as possible.<br>
<br>
Regards<br>
<br>
Bo Burman &amp; Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Multimedia Technologies, Ericsson Research EAB/TVM<br>
----------------------------------------------------------------------<br>
Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0<a href=3D"tel:%2B46%=
2010%207141311" value=3D"+46107141311">+46 10 7141311</a><br>
F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile <a href=3D"tel:%2B4=
6%2073%200949021" value=3D"+46730949021">+46 73 0949021</a><br>
SE-164 80 Stockholm, Sweden| mailto: bo.burman at <a href=3D"http://ericsso=
n.com" target=3D"_blank">ericsson.com</a><br>
----------------------------------------------------------------------<br>
<br>
_______________________________________________<br>
dispatch mailing list<br>
<a href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" target=3D"_blank=
">https://www.ietf.org/mailman/listinfo/dispatch</a><br>
</blockquote></div><br></div>

--f46d040892c3c76ae304b734bc49--

From richard@shockey.us  Mon Jan 23 10:37:17 2012
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From: "Richard Shockey" <richard@shockey.us>
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Cc: dispatch@ietf.org
Subject: Re: [dispatch] Interested in giving a talk at the FCC?
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I'd like to whole heartedly endorse this suggestion and encourage a variety
of IETF Subject matter experts to give talks relevant to the FCC.

I personally help arrange two seminars at the FCC at the invitation of Doug
Sicker, Henning's predecessor as CTO

The first on was a tutorial on SIP and the second on MPLS. 

As a rule these kinds of talks should be vendor/policy neutral. 

-----Original Message-----
From: ietf-bounces@ietf.org [mailto:ietf-bounces@ietf.org] On Behalf Of
Henning Schulzrinne
Sent: Sunday, January 22, 2012 11:10 AM
To: ietf@ietf.org
Subject: Interested in giving a talk at the FCC?

If you're interested in giving a seminar to technical staff and/or
economists at the US Federal Communications Commission (FCC), please let me
know. The FCC won't be able to pay for travel, so we'd likely schedule any
talks when you are in the DC area. Also, if you know of any colleagues the
Commission staff should hear from, feel free to suggest them.

I won't make any promises that I'll be able to accommodate all or even most
offers, but this is an experiment to see if we can broaden the perspectives
heard at the FCC. Most FCC staff are generalists, so topics should be
accessible and of interest to this community. (In general, the FCC deals
with many aspects of wired and wireless communications, and is interested in
applications, as well as lower layers. Examples of topics of particular
interest are spectrum-related issues, increasing broadband availability,
security/privacy, IPv6, PSTN-transition issues, public safety and
accessibility.

Thus, if you're interested, please get in touch with me with a rough
description of what you'd talk about. If there's local interest in your
topic, I will get back to you to work out details. There is no deadline as
such, so consider this a standing offer.

Henning
------
Henning Schulzrinne
CTO, FCC
7C252, (202) 418-1544
_______________________________________________
Ietf mailing list
Ietf@ietf.org
https://www.ietf.org/mailman/listinfo/ietf


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Date: Mon, 23 Jan 2012 12:50:52 -0800 (PST)
From: Sohel Khan <sohel_khan777@yahoo.com>
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Subject: [dispatch] SPID Sequence and Loop Detection
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--1229592896-1874021676-1327351852=:65491
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SIP calls=C2=A0routing in an infinite loop in a subset of networks is a kno=
wn phenomenon. =0AVarious SIP methods exist today may be used to detect and=
 mitigate loops. Operational engineers would like to see a method that indi=
cates the sequence of providers=E2=80=99 IDs in the call path. Based on the=
 input from the operational engineers, we are proposing a=C2=A0new SIP head=
er <SPID Sequence> to detect SIP routing loop. This method is preferred by =
the operational engineers of large providers=E2=80=99 networks.=0A=C2=A0=0A=
If a SIP INVITE with SPID sequence {33333333, 55555555, 77777777} arrives t=
o Alpha=E2=80=99s network, Alpha=C2=A0will detect 77777777 in the SPID sequ=
ence and conclude that a loop has occurred. =0A=C2=A0=0AIn the above use ca=
se, an example =E2=80=9CSPID-sequence=E2=80=9Dheader is as follows:=0ASPID-=
Sequence:<sip:33333333>;index=3D1;=0ASPID-Sequence:<sip:55555555>;index=3D1=
.1;=0ASPID-Sequence:<sip:77777777>;index=3D1.2;=0A=C2=A0=0ASPID-Sequence:<s=
ip:33333333>;index=3D1;=0ASPID-Sequence:<sip:55555555>;index=3D1.1;=0ASPID-=
Sequence:<sip:77777777>;index=3D1.2=0ASPID-Sequence:<sip:PRIVATE>;index=3D1=
.3;=0AA provider should not delete any value from the=C2=A0SPID sequence un=
less the SPID is its own.=0AWe had some discussion in the DISPATCH meetings=
 on ideas such as this. We would need a broader community support and help =
to initiate a work in IETF RTC WG on SPID sequence.=0APlease let us know yo=
ur opinion.=0A=C2=A0=0A=0ARegards, Sohel Khan=0AA provider wishing not to d=
ivulge its SPID, may insert a private or a blank value.=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 =0AWith =
the new method, the SIP INVITE contains the new header SPID sequence {33333=
333, 55555555, 77777777}. =0A=C2=A0=0A=C2=A0=0AAssume that a SIP call origi=
nating from=C2=A0provider Alpha (SPID: 77777777) and traversing through=C2=
=A0provider Bravo=C2=A0(SPID: 55555555),=C2=A0Charlie (SPID: 33333333), and=
 looping back to=C2=A0Alpha.=0A
--1229592896-1874021676-1327351852=:65491
Content-Type: text/html; charset=utf-8
Content-Transfer-Encoding: quoted-printable

<html><body><div style=3D"color:#000; background-color:#fff; font-family:ti=
mes new roman, new york, times, serif;font-size:12pt"><SPAN style=3D"RIGHT:=
 auto">
<div style=3D"RIGHT: auto"><SPAN style=3D"FONT-FAMILY: 'Cambria','serif'; C=
OLOR: black; mso-ascii-theme-font: major-latin; mso-hansi-theme-font: major=
-latin">SIP calls&nbsp;routing in an infinite loop in a subset of networks =
is a known phenomenon. </SPAN><?xml:namespace prefix =3D o ns =3D "urn:sche=
mas-microsoft-com:office:office" /><o:p style=3D"RIGHT: auto"></o:p></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto"><SPAN style=3D"FONT-=
FAMILY: 'Cambria','serif'; COLOR: black; mso-ascii-theme-font: major-latin;=
 mso-hansi-theme-font: major-latin">Various SIP methods exist today may be =
used to detect and mitigate loops. </SPAN>Operational engineers would like =
to see a method that indicates the sequence of providers=E2=80=99 IDs in th=
e call path. Based on the input from the operational engineers, <SPAN style=
=3D"FONT-FAMILY: 'Cambria','serif'; COLOR: black; mso-ascii-theme-font: maj=
or-latin; mso-hansi-theme-font: major-latin">we are proposing a&nbsp;new SI=
P header &lt;SPID Sequence&gt; to detect SIP routing loop. This method is p=
referred by the operational engineers of large providers=E2=80=99 networks.=
</SPAN><o:p></o:p></div>
<div><SPAN style=3D"FONT-FAMILY: 'Cambria','serif'; COLOR: black; mso-ascii=
-theme-font: major-latin; mso-hansi-theme-font: major-latin"><o:p>&nbsp;</o=
:p></SPAN></div>
<div><SPAN style=3D"FONT-FAMILY: 'Cambria','serif'; COLOR: black; mso-ascii=
-theme-font: major-latin; mso-hansi-theme-font: major-latin">Assume that <S=
PAN>a SIP call originating from&nbsp;provider Alpha (SPID: 77777777) and tr=
aversing through&nbsp;provider Bravo&nbsp;(SPID: 55555555),&nbsp;Charlie (S=
PID: 33333333), and looping back to&nbsp;Alpha.</SPAN><o:p></o:p></SPAN></d=
iv></SPAN>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto"><SPAN><SPAN style=3D=
"FONT-FAMILY: 'Cambria','serif'; COLOR: black; mso-ascii-theme-font: major-=
latin; mso-hansi-theme-font: major-latin"><o:p>&nbsp;</o:p></SPAN></div>
<div><SPAN style=3D"FONT-FAMILY: 'Cambria','serif'; COLOR: black; mso-ascii=
-theme-font: major-latin; mso-hansi-theme-font: major-latin">With the new m=
ethod, the SIP INVITE contains the new header SPID sequence {33333333, 5555=
5555, 77777777}. <o:p></o:p></SPAN></div>
<div><o:p>&nbsp;</o:p></div></SPAN>
<div><SPAN style=3D"FONT-FAMILY: 'Cambria','serif'; COLOR: black; mso-ascii=
-theme-font: major-latin; mso-hansi-theme-font: major-latin"></SPAN>If a SI=
P INVITE with SPID sequence {33333333, 55555555, 77777777} arrives to Alpha=
=E2=80=99s network, Alpha&nbsp;will detect 77777777 in the SPID sequence an=
d conclude that a loop has occurred. <o:p></o:p></SPAN></div>
<div style=3D"MARGIN: 0in 0in 0pt" class=3DMsoNormal><o:p><FONT face=3DCali=
bri>&nbsp;</FONT></o:p></div>
<div style=3D"MARGIN: 0in 0in 0pt; RIGHT: auto" class=3DMsoNormal><SPAN><SP=
AN style=3D"FONT-FAMILY: 'Cambria','serif'; COLOR: black; mso-ascii-theme-f=
ont: major-latin; mso-hansi-theme-font: major-latin"></SPAN><FONT face=3DCa=
libri>In the above use case, an example </FONT></SPAN><SPAN style=3D"FONT-F=
AMILY: 'Cambria','serif'; COLOR: black; mso-ascii-theme-font: major-latin; =
mso-hansi-theme-font: major-latin; mso-bidi-font-family: Calibri">=E2=80=9C=
SPID-sequence=E2=80=9D</SPAN><SPAN style=3D"FONT-FAMILY: 'Cambria','serif';=
 COLOR: black; mso-ascii-theme-font: major-latin; mso-hansi-theme-font: maj=
or-latin"> header is as follows:<o:p></o:p></SPAN></div>
<div style=3D"LINE-HEIGHT: 16.5pt; MARGIN: 0in 0in 0pt; RIGHT: auto" class=
=3DMsoNormal><SPAN><SPAN style=3D"FONT-FAMILY: 'Cambria','serif'; BACKGROUN=
D: white; COLOR: black; mso-ascii-theme-font: major-latin; mso-hansi-theme-=
font: major-latin">SPID-Sequence:&lt;<A href=3D"sip:33333333">sip:33333333<=
/A>&gt;;index=3D1;<o:p></o:p></SPAN></div></SPAN>
<div style=3D"LINE-HEIGHT: 16.5pt; MARGIN: 0in 0in 0pt; RIGHT: auto" class=
=3DMsoNormal><SPAN><SPAN style=3D"FONT-FAMILY: 'Cambria','serif'; BACKGROUN=
D: white; COLOR: black; mso-ascii-theme-font: major-latin; mso-hansi-theme-=
font: major-latin">SPID-Sequence:&lt;<A href=3D"sip:55555555">sip:55555555<=
/A>&gt;;index=3D1.1;<o:p></o:p></SPAN></div></SPAN>
<div style=3D"LINE-HEIGHT: 16.5pt; MARGIN: 0in 0in 0pt; RIGHT: auto" class=
=3DMsoNormal><SPAN><SPAN style=3D"FONT-FAMILY: 'Cambria','serif'; BACKGROUN=
D: white; COLOR: black; mso-ascii-theme-font: major-latin; mso-hansi-theme-=
font: major-latin">SPID-Sequence:&lt;<A href=3D"sip:77777777">sip:77777777<=
/A>&gt;;index=3D1.2;<o:p></o:p></SPAN></div></SPAN>
<div style=3D"MARGIN: 0in 0in 0pt; RIGHT: auto" class=3DMsoNormal><SPAN><SP=
AN style=3D"FONT-FAMILY: 'Cambria','serif'; COLOR: black; mso-ascii-theme-f=
ont: major-latin; mso-hansi-theme-font: major-latin"><o:p>&nbsp;</o:p></SPA=
N></div>
<div style=3D"MARGIN: 0in 0in 0pt" class=3DMsoNormal><SPAN style=3D"FONT-FA=
MILY: 'Cambria','serif'; COLOR: black; mso-ascii-theme-font: major-latin; m=
so-hansi-theme-font: major-latin">A provider wishing not to divulge its SPI=
D, may insert a private or a blank value.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <o:p></o:p></SPAN></div>=
</SPAN>
<div style=3D"LINE-HEIGHT: 16.5pt; MARGIN: 0in 0in 0pt; RIGHT: auto" class=
=3DMsoNormal><SPAN><SPAN style=3D"FONT-FAMILY: 'Cambria','serif'; BACKGROUN=
D: white; COLOR: black; mso-ascii-theme-font: major-latin; mso-hansi-theme-=
font: major-latin">SPID-Sequence:&lt;<A href=3D"sip:33333333">sip:33333333<=
/A>&gt;;index=3D1;<o:p></o:p></SPAN></div></SPAN>
<div style=3D"LINE-HEIGHT: 16.5pt; MARGIN: 0in 0in 0pt; RIGHT: auto" class=
=3DMsoNormal><SPAN><SPAN style=3D"FONT-FAMILY: 'Cambria','serif'; BACKGROUN=
D: white; COLOR: black; mso-ascii-theme-font: major-latin; mso-hansi-theme-=
font: major-latin">SPID-Sequence:&lt;<A href=3D"sip:55555555">sip:55555555<=
/A>&gt;;index=3D1.1;<o:p></o:p></SPAN></div></SPAN>
<div style=3D"LINE-HEIGHT: 16.5pt; MARGIN: 0in 0in 0pt; RIGHT: auto" class=
=3DMsoNormal><SPAN><SPAN style=3D"FONT-FAMILY: 'Cambria','serif'; BACKGROUN=
D: white; COLOR: black; mso-ascii-theme-font: major-latin; mso-hansi-theme-=
font: major-latin">SPID-Sequence:&lt;<A href=3D"sip:77777777">sip:77777777<=
/A>&gt;;index=3D1.2</SPAN></SPAN><SPAN style=3D"FONT-FAMILY: 'Cambria','ser=
if'; COLOR: black; mso-ascii-theme-font: major-latin; mso-hansi-theme-font:=
 major-latin"><o:p></o:p></SPAN></div>
<div style=3D"LINE-HEIGHT: 16.5pt; MARGIN: 0in 0in 0pt; RIGHT: auto" class=
=3DMsoNormal><SPAN><SPAN style=3D"FONT-FAMILY: 'Cambria','serif'; BACKGROUN=
D: white; COLOR: black; mso-ascii-theme-font: major-latin; mso-hansi-theme-=
font: major-latin">SPID<SPAN></SPAN>-Sequence:&lt;<A href=3D"sip:PRIVATE">s=
ip:PRIVATE</A>&gt;;index=3D1.3;<o:p></o:p></div></SPAN>
<div style=3D"LINE-HEIGHT: 16.5pt; MARGIN: 0in 0in 0pt; RIGHT: auto" class=
=3DMsoNormal><SPAN><SPAN style=3D"FONT-FAMILY: 'Cambria','serif'; BACKGROUN=
D: white; COLOR: black; mso-ascii-theme-font: major-latin; mso-hansi-theme-=
font: major-latin"></SPAN><FONT face=3DCalibri>A provider should not delete=
 any value from the&nbsp;SPID sequence unless the SPID is its own.</FONT></=
SPAN></SPAN><SPAN style=3D"FONT-FAMILY: 'Cambria','serif'; COLOR: black; ms=
o-ascii-theme-font: major-latin; mso-hansi-theme-font: major-latin"><o:p></=
o:p></SPAN></div>
<div style=3D"LINE-HEIGHT: 16.5pt; MARGIN: 0in 0in 0pt; RIGHT: auto" class=
=3DMsoNormal><SPAN><SPAN style=3D"FONT-FAMILY: 'Cambria','serif'; BACKGROUN=
D: white; COLOR: black; mso-ascii-theme-font: major-latin; mso-hansi-theme-=
font: major-latin">We had some discussion in the DISPATCH meetings on ideas=
 such as this. We would need a broader community support and help to initia=
te a work in IETF RTC WG on SPID sequence.</SPAN></SPAN><SPAN style=3D"FONT=
-FAMILY: 'Cambria','serif'; COLOR: black; mso-ascii-theme-font: major-latin=
; mso-hansi-theme-font: major-latin"><o:p></o:p></SPAN></div>
<div style=3D"LINE-HEIGHT: 16.5pt; MARGIN: 0in 0in 0pt; RIGHT: auto" class=
=3DMsoNormal><SPAN><SPAN style=3D"FONT-FAMILY: 'Cambria','serif'; BACKGROUN=
D: white; COLOR: black; mso-ascii-theme-font: major-latin; mso-hansi-theme-=
font: major-latin">Please let us know your opinion.</SPAN></SPAN><SPAN styl=
e=3D"FONT-FAMILY: 'Cambria','serif'; COLOR: black; mso-ascii-theme-font: ma=
jor-latin; mso-hansi-theme-font: major-latin"><o:p></o:p></SPAN></div>
<div style=3D"MARGIN: 0in 0in 0pt" class=3DMsoNormal><o:p><FONT face=3DCali=
bri>&nbsp;</FONT></o:p></div>
<div style=3D"RIGHT: auto"></SPAN></SPAN></SPAN><VAR id=3Dyui-ie-cursor></V=
AR><BR style=3D"RIGHT: auto" class=3Dyui-cursor></SPAN></div>
<DIV></DIV>
<DIV>&nbsp;</DIV>
<div style=3D"RIGHT: auto">Regards, Sohel Khan</div></div></body></html>
--1229592896-1874021676-1327351852=:65491--

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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: Sohel Khan <sohel_khan777@yahoo.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Mon, 23 Jan 2012 16:18:53 -0500
Thread-Topic: [dispatch] SPID Sequence and Loop Detection
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Subject: Re: [dispatch] SPID Sequence and Loop Detection
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> From: Sohel Khan [sohel_khan777@yahoo.com]
>=20
> If a SIP INVITE with SPID sequence {33333333, 55555555, 77777777}
> arrives to Alpha=92s network, Alpha will detect 77777777 in the SPID
> sequence and conclude that a loop has occurred.

And that conclusion is incorrect, because at the second entry to
Alpha, Alpha has no way of knowing that the request-URI is at all the
same as the request-URI at the time of first entry to Alpha.

In practice, if user A1 on network Alpha call-forwards his phone to
user B on network Bravo, and user B call-forwards his phone to user A2
on network Alpha, all calls to A1 will be routed to B, and then to A2,
and will be rejected upon the second entry to Alpha.

Something closer to the loop detection algorithm of RFC 5393 will be
needed.

Dale

From vkg@bell-labs.com  Mon Jan 23 13:35:54 2012
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Date: Mon, 23 Jan 2012 15:39:52 -0600
From: "Vijay K. Gurbani" <vkg@bell-labs.com>
Organization: Bell Laboratories, Alcatel-Lucent
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Subject: [dispatch] SIP Load Balancing
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Folks: During the summer IETF (Quebec City, summer 2011) there was a
discussion on SIP load balancing [1].

The general hum indicated that folks consider this work important
enough to be at least looked at by the IETF, especially given gear
from different vendors and the need to perform feedback-based
load balancing.

The details of what "this work" entails need to be fleshed out.
For instance, there were good discussions on whether signaling-
only and media-based servers needed different load balancing
techniques, and what exactly needs standardization here.

Robert rightly suggested that having a document describing
existing SIP load balancing techniques would be a good first step to
seeing what work needs to be done here.

So, responding to that starting point, Partha, Paul and I have put
together the beginnings of a survey document that looks at
various techniques whereby load balancing is done today [2].  The work
is not complete, but a good start towards getting a document that
describes different techniques in use today.

We will like to start a larger discussion on the survey document
to see if we have adequately captured the different ways to do
load balancing today.  Our hope is that this work will serve
as input to the questions of exactly what should be standardized.
To that extent, we kindly solicit your comments and thoughts on
the survey draft.

Thanks,

[1] http://www.ietf.org/proceedings/81/minutes/dispatch.txt
[2] 
http://tools.ietf.org/html/draft-partha-dispatch-siploadbalancing-survey-00

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{bell-labs.com,acm.org} / vijay.gurbani@alcatel-lucent.com
Web:   http://ect.bell-labs.com/who/vkg/

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "dispatch@ietf.org list" <dispatch@ietf.org>
Thread-Topic: STRAW Charter Proposal
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Date: Tue, 24 Jan 2012 07:15:38 +0000
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Subject: [dispatch] STRAW Charter Proposal
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Howdy,
As instructed by DISPATCH chairs, I am re-posting the STRAW Charter Proposa=
l, as a topic for the Paris meeting.
The proposed charter is as follows:

Description of STRAW Working Group
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Name: Sip Traversal Required for Applications to Work (STRAW)

Problem Statement:=20
Within the context of the SIP protocol and architecture, a Back-to-Back Use=
r Agent (B2BUA) is any SIP device in the logical path between two User Agen=
ts performing a role beyond that of a Proxy as defined in RFC 3261.  The B2=
BUA may be as simple as a session-stateful Proxy becoming a B2BUA in order =
to terminate dead sessions by generating BYEs; or it may be a 3PCC-style ag=
ent only modifying SDP; or it may be a Session Border Controller performing=
 such functions as in RFC 5853; or it may be an Enterprise PBX terminating =
REFERs and such; or it may be a complete UAS and UAC implementation with a =
PRI loopback in-between.=20

In its most extreme form, the scope of the SIP protocol ends at the UAS of =
the B2BUA, and a new SIP protocol scope begins on its UAC side.  In practic=
e, however, users expect some SIP protocol aspects to go beyond the scope o=
f the B2BUA's UAS side, and be traversed onto its UAC side, as if the B2BUA=
 was not an end unto itself; this is similar to the expectation that emails=
 work when they cross from POP and IMAP to/from SMTP.

It is impossible to normatively define all the behaviors of B2BUAs in gener=
al, or even subsets of them such as SBCs. Unlike consumer NATs, B2BUAs perf=
orm widely varying functions for purposes which may be unique to their envi=
ronment, unique to their architecture, or unique to the wishes of their adm=
inistrator.  Instead of defining all things a given type of B2BUA must do, =
a more practical objective would be to define what very few things any B2BU=
A must do to make a specific SIP mechanism work, and let the market decide =
whether to do those things.

The name of this working group reflects that practical objective: if there =
were a thin straw between the SIP UAS and UAC of a B2BUA, what must be pass=
ed through that straw and used on each side.  Or viewed another way, if a B=
2BUA were in fact a UAS and UAC connected with a PRI loopback circuit, and =
if we could extend ISDN, what information would we carry in ISDN across the=
 PRI for a specific SIP mechanism to work end-to-end.

For example, the WG could produce a document which specifies that the Max-F=
orwards header field value should be copied and decremented across the B2BU=
A, if the B2BUA wishes to prevent loops.  Administrators could then tell th=
eir B2BUA vendors to comply with the document, if the administrator so wish=
es.

Objectives:
The objectives of the STRAW Working Group are to publish normative document=
s which define which SIP header fields, parameters, MIME bodies, body conte=
nt fields/information, or media-plane characteristics are required to trave=
rse between the User Agent "sides" of a B2BUA for specific functions to wor=
k. =20

Deliverables would indicate which types of B2BUAs would apply or not.  For =
example, a document defining the requirements for end-to-end DTLS-SRTP woul=
d not apply to B2BUAs which terminate media, such as transcoders or recorde=
rs.

The intention of this WG is to document such requirements in separate docum=
ents, per SIP or media function, instead of as one document for all functio=
ns.  That will both reduce the time to publication, as well a provide B2BUA=
 administrators and manufacturers with simple comply/no-comply criteria.


Initial Deliverables:
1) A taxonomy document defining role-types of B2BUAs, as a reference for ot=
her deliverables.
2) A document defining the requirements for B2BUAs with respect to loop det=
ection/prevention.
3) A document defining the requirements for B2BUAs to support end-to-end an=
d hop-by-hop media-loopback test calls.
4) A document defining the requirements for B2BUAs to support DTLS-SRTP (RF=
C 5764) end-to-end.
5) A document defining the requirements for B2BUAs to support STUN connecti=
vity checks end-to-end.



From bo.burman@ericsson.com  Tue Jan 24 00:22:28 2012
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From: Bo Burman <bo.burman@ericsson.com>
To: Michael Hammer <mphmmr@gmail.com>
Date: Tue, 24 Jan 2012 09:22:22 +0100
Thread-Topic: [dispatch] Media Stream related signaling
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Subject: Re: [dispatch] Media Stream related signaling
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Hi Mike,

yes, that is in principle what was intended with "pause" - something that i=
s as low energy as possible but that is readily available and can be quickl=
y and efficiently resumed when needed again.

We did think about what to send instead of the paused media. It is however =
not clear to us if it will be sufficient to use e.g. ICE or RTP-RTCP Mux as=
 keepalive to make the SBC keep the pinhole, or if it will be necessary to =
use actual RTP media packets to create a keepalive stream. That is also the=
 case when neither ICE nor RTP-RTCP Mux are used. Any ideas?

BR
/Bo

________________________________
From: Michael Hammer [mailto:mphmmr@gmail.com]
Sent: Monday, January 23, 2012 5:44 PM
To: Bo Burman
Cc: dispatch@ietf.org
Subject: Re: [dispatch] Media Stream related signaling

Bo,

Rather than "turning [the media] stream off temporarily", you should consid=
er putting the media stream into a low energy state, which would not be bit=
 rate =3D 0 but would have some limited keep-alive stream that the SBC reco=
gnizes to keep the pinhole open.

There are probably multiple ways to make that happen.  Hope that is semanti=
cally what you meant.

Mike


On Mon, Jan 23, 2012 at 10:50 AM, Bo Burman <bo.burman@ericsson.com<mailto:=
bo.burman@ericsson.com>> wrote:
Dispatch,

We have requested agenda time for the upcoming meeting to discuss the subje=
ct of Media Stream related signaling. From the feedback we got in AVTEXT WG=
 on the discussion for a certain problem, the one related to pausing and re=
suming sender to middlebox media traffic, it is clear that a more high leve=
l discussion needs to happen.

We have two Internet drafts related to the issues we would like to discuss.
Media Stream Selection (MESS)
 - draft-westerlund-dispatch-stream-selection-00
RTP Media Stream Pause and Resume
 - draft-westerlund-avtext-rtp-stream-pause-00

The first one (MESS) proposes that for the general selection of media strea=
ms, an end-point within a centralized conferencing scenario desires to rece=
ive a media plane oriented rather than RTP/RTCP based signaling, which appe=
ars to have benefits as discussed in the draft.

As mentioned, the pause resume draft is targeted for a different use case w=
here a media plane solution has potential for other benefits. This was disc=
ussed and the main outcome of that discussion was three quite different opi=
nions;

a) We already do this signaling using TMMBR (RFC 5104) by setting the bit-r=
ate value to 0, which the current specs doesn't allow.

b) This should be done in signalling plane, RTP/RTCP shouldn't be bloated. =
(But I did interpret this as not necessarily needing to be SIP).

c) When it comes to pause/resume, unless it is done using SIP/SDP it will n=
ot work through any Session Border Gateway as the session will be torn down=
 after n seconds.

Thus I think it important that we take a discussion around the use cases, e=
xisting solutions, limitations in those or the proposal to try to determine=
 what is the most reasonable way of solving the problems and how to enable;

1) Receiver based selection of media streams from a larger set of streams, =
including different quality levels like video resolutions, in a centralized=
 conferencing application.

2) Optimize the transport so that media streams currently not used can be t=
urned off temporarily and then quickly resumed when someone determines a ne=
ed for the media stream. For example using mechanism (1) but also for voice=
 activity based, centrally controlled or other mechanisms to select which m=
edia streams are delivered to other session participants.

We are working to revise our Internet drafts. We seek as much input as poss=
ible into this prior to the meeting so that we can ensure that any discussi=
on are as on topic and focused as possible.

Regards

Bo Burman & Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7141311<tel:%2B46%2010%207141311=
>
F=E4r=F6gatan 6                | Mobile +46 73 0949021<tel:%2B46%2073%20094=
9021>
SE-164 80 Stockholm, Sweden| mailto: bo.burman at ericsson.com<http://erics=
son.com>
----------------------------------------------------------------------

_______________________________________________
dispatch mailing list
dispatch@ietf.org<mailto: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>
<META http-equiv=3DContent-Type content=3D"text/html; charset=3Diso-8859-1"=
>
<META content=3D"MSHTML 6.00.6002.18538" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D831264907-24012012><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Hi Mike,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D831264907-24012012><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D831264907-24012012><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>yes, that is in principle what was intended with "=
pause" -=20
something that is as low energy as possible but that is readily available a=
nd=20
can be quickly and efficiently resumed when needed again.</FONT></SPAN></DI=
V>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D831264907-24012012><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D831264907-24012012><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>We did think about what to send instead of the pau=
sed=20
media. It is however not clear to us if it will be sufficient to use e.g. I=
CE or=20
RTP-RTCP Mux as keepalive to make the SBC keep the pinhole, or if it will b=
e=20
necessary to use actual RTP media packets to create a keepalive stream. Tha=
t is=20
also the case when neither ICE nor RTP-RTCP Mux&nbsp;are used. Any=20
ideas?</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D831264907-24012012><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D831264907-24012012><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>BR</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D831264907-24012012><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>/Bo</FONT></SPAN></DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> Michael Hammer [mailto:mphmmr@gma=
il.com]=20
<BR><B>Sent:</B> Monday, January 23, 2012 5:44 PM<BR><B>To:</B> Bo=20
Burman<BR><B>Cc:</B> dispatch@ietf.org<BR><B>Subject:</B> Re: [dispatch] Me=
dia=20
Stream related signaling<BR></FONT><BR></DIV>
<DIV></DIV>Bo,
<DIV><BR></DIV>
<DIV>Rather than "turning [the media] stream off temporarily", you should=20
consider putting the media stream into a low energy state, which would not =
be=20
bit rate =3D 0 but would have some limited keep-alive stream that the SBC=20
recognizes to keep the pinhole open.</DIV>
<DIV><BR></DIV>
<DIV>There are probably multiple ways to make that happen. &nbsp;Hope that =
is=20
semantically what you meant.</DIV>
<DIV><BR></DIV>
<DIV>Mike</DIV>
<DIV><BR><BR>
<DIV class=3Dgmail_quote>On Mon, Jan 23, 2012 at 10:50 AM, Bo Burman <SPAN=
=20
dir=3Dltr>&lt;<A=20
href=3D"mailto:bo.burman@ericsson.com">bo.burman@ericsson.com</A>&gt;</SPAN=
>=20
wrote:<BR>
<BLOCKQUOTE class=3Dgmail_quote=20
style=3D"PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1p=
x solid">Dispatch,<BR><BR>We=20
  have requested agenda time for the upcoming meeting to discuss the subjec=
t of=20
  Media Stream related signaling. From the feedback we got in AVTEXT WG on =
the=20
  discussion for a certain problem, the one related to pausing and resuming=
=20
  sender to middlebox media traffic, it is clear that a more high level=20
  discussion needs to happen.<BR><BR>We have two Internet drafts related to=
 the=20
  issues we would like to discuss.<BR>Media Stream Selection (MESS)<BR>&nbs=
p;-=20
  draft-westerlund-dispatch-stream-selection-00<BR>RTP Media Stream Pause a=
nd=20
  Resume<BR>&nbsp;- draft-westerlund-avtext-rtp-stream-pause-00<BR><BR>The =
first=20
  one (MESS) proposes that for the general selection of media streams, an=20
  end-point within a centralized conferencing scenario desires to receive a=
=20
  media plane oriented rather than RTP/RTCP based signaling, which appears =
to=20
  have benefits as discussed in the draft.<BR><BR>As mentioned, the pause r=
esume=20
  draft is targeted for a different use case where a media plane solution h=
as=20
  potential for other benefits. This was discussed and the main outcome of =
that=20
  discussion was three quite different opinions;<BR><BR>a) We already do th=
is=20
  signaling using TMMBR (RFC 5104) by setting the bit-rate value to 0, whic=
h the=20
  current specs doesn't allow.<BR><BR>b) This should be done in signalling=
=20
  plane, RTP/RTCP shouldn't be bloated. (But I did interpret this as not=20
  necessarily needing to be SIP).<BR><BR>c) When it comes to pause/resume,=
=20
  unless it is done using SIP/SDP it will not work through any Session Bord=
er=20
  Gateway as the session will be torn down after n seconds.<BR><BR>Thus I t=
hink=20
  it important that we take a discussion around the use cases, existing=20
  solutions, limitations in those or the proposal to try to determine what =
is=20
  the most reasonable way of solving the problems and how to enable;<BR><BR=
>1)=20
  Receiver based selection of media streams from a larger set of streams,=20
  including different quality levels like video resolutions, in a centraliz=
ed=20
  conferencing application.<BR><BR>2) Optimize the transport so that media=
=20
  streams currently not used can be turned off temporarily and then quickly=
=20
  resumed when someone determines a need for the media stream. For example =
using=20
  mechanism (1) but also for voice activity based, centrally controlled or =
other=20
  mechanisms to select which media streams are delivered to other session=20
  participants.<BR><BR>We are working to revise our Internet drafts. We see=
k as=20
  much input as possible into this prior to the meeting so that we can ensu=
re=20
  that any discussion are as on topic and focused as=20
  possible.<BR><BR>Regards<BR><BR>Bo Burman &amp; Magnus=20
  Westerlund<BR><BR>-------------------------------------------------------=
---------------<BR>Multimedia=20
  Technologies, Ericsson Research=20
  EAB/TVM<BR>--------------------------------------------------------------=
--------<BR>Ericsson=20
  AB &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;| Phone &nbsp;<=
A=20
  href=3D"tel:%2B46%2010%207141311" value=3D"+46107141311">+46 10=20
  7141311</A><BR>F=E4r=F6gatan 6 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
  &nbsp;| Mobile <A href=3D"tel:%2B46%2073%200949021" value=3D"+46730949021=
">+46 73=20
  0949021</A><BR>SE-164 80 Stockholm, Sweden| mailto: bo.burman at <A=20
  href=3D"http://ericsson.com"=20
  target=3D_blank>ericsson.com</A><BR>-------------------------------------=
---------------------------------<BR><BR>__________________________________=
_____________<BR>dispatch=20
  mailing list<BR><A href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</A=
><BR><A=20
  href=3D"https://www.ietf.org/mailman/listinfo/dispatch"=20
  target=3D_blank>https://www.ietf.org/mailman/listinfo/dispatch</A><BR></B=
LOCKQUOTE></DIV><BR></DIV></BODY></HTML>

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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Bo Burman'" <bo.burman@ericsson.com>, <dispatch@ietf.org>
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Date: Tue, 24 Jan 2012 13:02:04 +0200
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Subject: Re: [dispatch] Media Stream related signaling
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Hi,

About
a) We already do this signaling using TMMBR (RFC 5104) by setting the
bit-rate value to 0, which the current specs doesn't allow.

I am not sure why? The text in 3.5.4 "The limit may change to any value
between zero and the session maximum, as negotiated during session
establishment
   signaling." And there is also the message format in section 4.2.1.1
allows the value of 0


Thanks
Roni Even

> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Bo Burman
> Sent: Monday, January 23, 2012 5:50 PM
> To: dispatch@ietf.org
> Subject: [dispatch] Media Stream related signaling
>=20
> Dispatch,
>=20
> We have requested agenda time for the upcoming meeting to discuss the
> subject of Media Stream related signaling. From the feedback we got in
> AVTEXT WG on the discussion for a certain problem, the one related to
> pausing and resuming sender to middlebox media traffic, it is clear
> that a more high level discussion needs to happen.
>=20
> We have two Internet drafts related to the issues we would like to
> discuss.
> Media Stream Selection (MESS)
>  - draft-westerlund-dispatch-stream-selection-00
> RTP Media Stream Pause and Resume
>  - draft-westerlund-avtext-rtp-stream-pause-00
>=20
> The first one (MESS) proposes that for the general selection of media
> streams, an end-point within a centralized conferencing scenario
> desires to receive a media plane oriented rather than RTP/RTCP based
> signaling, which appears to have benefits as discussed in the draft.
>=20
> As mentioned, the pause resume draft is targeted for a different use
> case where a media plane solution has potential for other benefits.
> This was discussed and the main outcome of that discussion was three
> quite different opinions;
>=20
> a) We already do this signaling using TMMBR (RFC 5104) by setting the
> bit-rate value to 0, which the current specs doesn't allow.
>=20
> b) This should be done in signalling plane, RTP/RTCP shouldn't be
> bloated. (But I did interpret this as not necessarily needing to be
> SIP).
>=20
> c) When it comes to pause/resume, unless it is done using SIP/SDP it
> will not work through any Session Border Gateway as the session will =
be
> torn down after n seconds.
>=20
> Thus I think it important that we take a discussion around the use
> cases, existing solutions, limitations in those or the proposal to try
> to determine what is the most reasonable way of solving the problems
> and how to enable;
>=20
> 1) Receiver based selection of media streams from a larger set of
> streams, including different quality levels like video resolutions, in
> a centralized conferencing application.
>=20
> 2) Optimize the transport so that media streams currently not used can
> be turned off temporarily and then quickly resumed when someone
> determines a need for the media stream. For example using mechanism =
(1)
> but also for voice activity based, centrally controlled or other
> mechanisms to select which media streams are delivered to other =
session
> participants.
>=20
> We are working to revise our Internet drafts. We seek as much input as
> possible into this prior to the meeting so that we can ensure that any
> discussion are as on topic and focused as possible.
>=20
> Regards
>=20
> Bo Burman & Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7141311
> F=E4r=F6gatan 6                | Mobile +46 73 0949021
> SE-164 80 Stockholm, Sweden| mailto: bo.burman at ericsson.com
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


From magnus.westerlund@ericsson.com  Tue Jan 24 04:42:47 2012
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Cc: "dispatch@ietf.org" <dispatch@ietf.org>
Subject: Re: [dispatch] Media Stream related signaling : Why not TMMBR
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On 2012-01-24 12:02, Roni Even wrote:
> Hi,
> 
> About
> a) We already do this signaling using TMMBR (RFC 5104) by setting the
> bit-rate value to 0, which the current specs doesn't allow.
> 
> I am not sure why? The text in 3.5.4 "The limit may change to any value
> between zero and the session maximum, as negotiated during session
> establishment
>    signaling." And there is also the message format in section 4.2.1.1
> allows the value of 0

Roni,

Our main argument is that the semantics of TMMBR is not generally
matching the one needed for pause an resume. As indicated in the AVTEXT
slides unless you are absolutely certain that there are no other RTP
entity on the path between the TMMBR issuing and the TMMBR responding
entity using it to set to 0 can cause significant issues for that other
party in that the stream is pause for that entity also.

Another problem with TMMBR is that is has the following NORMATIVE
requirement on how it goes about increasing the bit-rate:

   If the media sender concludes that it can increase the maximum total
   media bit rate value, it SHALL wait before actually doing so, for a
   period long enough to allow a media receiver to respond to the TMMBN
   if it determines that its tuple belongs in the bounding set.   This
   delay period is estimated by the formula:

      2 * RTT + T_Dither_Max,

Thus a receiver should not get media until 2*RTT + T_Dither_Max after it
request "RESUME". But following this clearly makes TMMBR unusable for
PAUSE RESUME. And this is required also in point to point:

   Even in
   point-to-point sessions, a media sender MUST obey the aforementioned
   rule, as it is not guaranteed that a participant is able to determine
   correctly whether all the sources are co-located in a single node,
   and are coordinated.

Thus at a minimum if one like to use TMMBR we need to redefine TMMBR
such that it allows for the functionality. That likely means separating
the two use cases from each other to avoid the risk of miss
classification of topologies. Which from my perspective means different
signalling code-points to make it clear to the end-points what to do.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From lennox@cs.columbia.edu  Tue Jan 24 08:00:24 2012
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Date: Tue, 24 Jan 2012 11:00:22 -0500
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From: lennox@cs.columbia.edu
Cc: "dispatch@ietf.org list" <dispatch@ietf.org>
Subject: Re: [dispatch] STRAW Charter Proposal
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On Tuesday, January 24 2012, "Hadriel Kaplan" wrote to "dispatch@ietf.org list" saying:

> Howdy,
> As instructed by DISPATCH chairs, I am re-posting the STRAW Charter Proposal, as a topic for the Paris meeting.
> The proposed charter is as follows:

One item that I think is important to raise here -- though it should perhaps
be part of the taxonomy document, rather than in the charter itself -- is
that there needs to be a fairly rigorous notion of what kinds of devices are
*not* B2BUAs in STRAW's scope.

In particular, while an MCU can be regarded as a collection of two or more
UAs back-to-back-to-back, I think it's not anyone's intention that MCU
behavior be in scope for STRAW.

It seems to me that the cases where STRAW should apply -- i.e., where the
intended semantic is that two dialogs are the "same call" -- are very
similar to (perhaps the same as) the cases where two dialogs should have the
same Session-ID.  Thus reducing to a previous (intractable) problem.

-- 
Jonathan Lennox
lennox@cs.columbia.edu / jonathan@vidyo.com

From pkyzivat@alum.mit.edu  Tue Jan 24 08:20:11 2012
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Subject: Re: [dispatch] STRAW Charter Proposal
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On 1/24/12 11:00 AM, lennox@cs.columbia.edu wrote:

> It seems to me that the cases where STRAW should apply -- i.e., where the
> intended semantic is that two dialogs are the "same call" -- are very
> similar to (perhaps the same as) the cases where two dialogs should have the
> same Session-ID.  Thus reducing to a previous (intractable) problem.

:-)

From richard@shockey.us  Tue Jan 24 12:28:37 2012
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Subject: [dispatch] FYI : Canada's CRTC mandates all IP Interconnection for voice.
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In this decision, the Commission decides that it is in the public interest
to establish a set of principles to facilitate IP voice network
interconnections between network operators while allowing market forces to
shape the details of the arrangements. Specifically, in areas where a
carrier provides IP voice interconnection to an affiliate, a division of its
operations, or an unrelated service provider, the carrier must negotiate a
similar arrangement with any other carrier that requests such an
arrangement. Within six months of a formal request, an arrangement is to be
concluded.

 

http://www.crtc.gc.ca/eng/com100/2012/r120119.htm

 

Richard Shockey
Shockey Consulting
Chairman of the Board of Directors SIP Forum
PSTN Mobile: +1 703.593.2683
< <mailto:richard(at)shockey.us> mailto:richard(at)shockey.us>
skype-linkedin-facebook: rshockey101
http//www.sipforum.org

 


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style=3D'font-family:"Calibri","sans-serif"'>In this decision, the =
Commission decides that it is in the public interest to establish a set =
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an unrelated service provider, the carrier must negotiate a similar =
arrangement with any other carrier that requests such an arrangement. =
Within six months of a formal request, an arrangement is to be =
concluded.</span></em><o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal><a =
href=3D"http://www.crtc.gc.ca/eng/com100/2012/r120119.htm">http://www.crt=
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Date: Tue, 24 Jan 2012 13:59:45 -0800 (PST)
From: Sohel Khan <sohel_khan777@yahoo.com>
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Subject: Re: [dispatch] SPID Sequence and Loop Detection
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---1328350513-1008145155-1327442385=:69551
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We are not discussing routing loop due to call forwarding.=0AVarious loop m=
itigation mechanism=C2=A0may=C2=A0be applicable=C2=A0to=C2=A0mitigate diffe=
rent types of routing loops.=0ASPID sequence mechanism may apply to routing=
 loops due to least cost routing=C2=A0=0AProvider Alpha routes the call to =
Provider Bravo because Provider Alpha considers that Provider Bravo is a le=
ast cost route.=0AProvider Bravo routes the call to Provider Charlie becaus=
e Provider Bravo considers that Provider Charlie is a least cost route.=0AP=
rovider Charlie routes the call back to Provider Alpha because Provider Cha=
rlie considers that Provider Alpha is a least cost route. Thus, an infinite=
 routing loop=C2=A0occurs.=C2=A0 SPID Sequence can be hypothetically viewed=
 as the BGP AS_PATH route vector=0ARegards, Sohel Khan=0A=0A=0A____________=
____________________=0AFrom: "Worley, Dale R (Dale)" <dworley@avaya.com>=0A=
To: Sohel Khan <sohel_khan777@yahoo.com>; "dispatch@ietf.org" <dispatch@iet=
f.org> =0ASent: Monday, January 23, 2012 4:18 PM=0ASubject: RE: [dispatch] =
SPID Sequence and Loop Detection=0A=0A> From: Sohel Khan [sohel_khan777@yah=
oo.com]=0A> =0A> If a SIP INVITE with SPID sequence {33333333, 55555555, 77=
777777}=0A> arrives to Alpha=E2=80=99s network, Alpha will detect 77777777 =
in the SPID=0A> sequence and conclude that a loop has occurred.=0A=0AAnd th=
at conclusion is incorrect, because at the second entry to=0AAlpha, Alpha h=
as no way of knowing that the request-URI is at all the=0Asame as the reque=
st-URI at the time of first entry to Alpha.=0A=0AIn practice, if user A1 on=
 network Alpha call-forwards his phone to=0Auser B on network Bravo, and us=
er B call-forwards his phone to user A2=0Aon network Alpha, all calls to A1=
 will be routed to B, and then to A2,=0Aand will be rejected upon the secon=
d entry to Alpha.=0A=0ASomething closer to the loop detection algorithm of =
RFC 5393 will be=0Aneeded.=0A=0ADale
---1328350513-1008145155-1327442385=:69551
Content-Type: text/html; charset=utf-8
Content-Transfer-Encoding: quoted-printable

<html><body><div style=3D"color:#000; background-color:#fff; font-family:ti=
mes new roman, new york, times, serif;font-size:12pt"><SPAN style=3D"RIGHT:=
 auto">
<div><SPAN style=3D"COLOR: black">We are not discussing routing loop due to=
 call forwarding.</SPAN><?xml:namespace prefix =3D o ns =3D "urn:schemas-mi=
crosoft-com:office:office" /><o:p></o:p></SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto"><SPAN style=3D"COLOR=
: black">Various loop mitigation mechanism&nbsp;may&nbsp;be applicable&nbsp=
;to&nbsp;mitigate different types of routing loops.</SPAN><o:p></o:p></SPAN=
></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto"><SPAN style=3D"COLOR=
: black">SPID sequence mechanism may apply to routing loops due to least co=
st routing<o:p></o:p></SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto"><SPAN style=3D"COLOR=
: black">Provider Alpha routes the call to Provider Bravo because Provider =
Alpha considers that Provider Bravo is a least cost route.</SPAN><o:p></o:p=
></SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto"><SPAN style=3D"COLOR=
: black">Provider Bravo routes the call to Provider Charlie because Provide=
r Bravo considers that Provider Charlie is a least cost route.</SPAN><o:p><=
/o:p></SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto"><SPAN style=3D"COLOR=
: black">Provider Charlie routes the call back to Provider Alpha because Pr=
ovider Charlie considers that Provider Alpha is a least cost route. Thus, a=
n infinite routing loop&nbsp;occurs.&nbsp; SPID Sequence can be hypothetica=
lly viewed as the BGP AS_PATH route vector</SPAN><o:p></o:p></SPAN></div>
<div style=3D"RIGHT: auto"></SPAN></SPAN><VAR id=3Dyui-ie-cursor></VAR>&nbs=
p;</div>
<div>Regards, Sohel Khan<BR></div>
<DIV style=3D"FONT-FAMILY: times new roman, new york, times, serif; FONT-SI=
ZE: 12pt">
<DIV style=3D"FONT-FAMILY: times new roman, new york, times, serif; FONT-SI=
ZE: 12pt">
<DIV dir=3Dltr><FONT size=3D2 face=3DArial>
<DIV style=3D"BORDER-BOTTOM: #ccc 1px solid; BORDER-LEFT: #ccc 1px solid; P=
ADDING-BOTTOM: 0px; LINE-HEIGHT: 0; MARGIN: 5px 0px; PADDING-LEFT: 0px; PAD=
DING-RIGHT: 0px; HEIGHT: 0px; FONT-SIZE: 0px; BORDER-TOP: #ccc 1px solid; B=
ORDER-RIGHT: #ccc 1px solid; PADDING-TOP: 0px" class=3Dhr contentEditable=
=3Dfalse readonly=3D"true"></DIV><B><SPAN style=3D"FONT-WEIGHT: bold">From:=
</SPAN></B> "Worley, Dale R (Dale)" &lt;dworley@avaya.com&gt;<BR><B><SPAN s=
tyle=3D"FONT-WEIGHT: bold">To:</SPAN></B> Sohel Khan &lt;sohel_khan777@yaho=
o.com&gt;; "dispatch@ietf.org" &lt;dispatch@ietf.org&gt; <BR><B><SPAN style=
=3D"FONT-WEIGHT: bold">Sent:</SPAN></B> Monday, January 23, 2012 4:18 PM<BR=
><B><SPAN style=3D"FONT-WEIGHT: bold">Subject:</SPAN></B> RE: [dispatch] SP=
ID Sequence and Loop Detection<BR></FONT></DIV><BR>&gt; From: Sohel Khan [<=
A href=3D"mailto:sohel_khan777@yahoo.com" ymailto=3D"mailto:sohel_khan777@y=
ahoo.com">sohel_khan777@yahoo.com</A>]<BR>&gt; <BR>&gt; If a SIP INVITE wit=
h SPID sequence
 {33333333, 55555555, 77777777}<BR>&gt; arrives to Alpha=E2=80=99s network,=
 Alpha will detect 77777777 in the SPID<BR>&gt; sequence and conclude that =
a loop has occurred.<BR><BR>And that conclusion is incorrect, because at th=
e second entry to<BR>Alpha, Alpha has no way of knowing that the request-UR=
I is at all the<BR>same as the request-URI at the time of first entry to Al=
pha.<BR><BR>In practice, if user A1 on network Alpha call-forwards his phon=
e to<BR>user B on network Bravo, and user B call-forwards his phone to user=
 A2<BR>on network Alpha, all calls to A1 will be routed to B, and then to A=
2,<BR>and will be rejected upon the second entry to Alpha.<BR><BR>Something=
 closer to the loop detection algorithm of RFC 5393 will be<BR>needed.<BR><=
BR>Dale<BR><BR><BR></DIV></DIV></div></body></html>
---1328350513-1008145155-1327442385=:69551--

From pkyzivat@alum.mit.edu  Tue Jan 24 19:21:03 2012
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Subject: Re: [dispatch] SPID Sequence and Loop Detection
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On 1/24/12 4:59 PM, Sohel Khan wrote:
> We are not discussing routing loop due to call forwarding.
> Various loop mitigation mechanism may be applicable to mitigate
> different types of routing loops.

You may not be discussing/considering those things, but you should be, 
because (as Dale explained) the mechanism you are proposing would 
*break* non-loop calls that revisit the same provider.

	Thanks,
	Paul

> SPID sequence mechanism may apply to routing loops due to least cost routing
> Provider Alpha routes the call to Provider Bravo because Provider Alpha
> considers that Provider Bravo is a least cost route.
> Provider Bravo routes the call to Provider Charlie because Provider
> Bravo considers that Provider Charlie is a least cost route.
> Provider Charlie routes the call back to Provider Alpha because Provider
> Charlie considers that Provider Alpha is a least cost route. Thus, an
> infinite routing loop occurs. SPID Sequence can be hypothetically viewed
> as the BGP AS_PATH route vector
> Regards, Sohel Khan
> *From:* "Worley, Dale R (Dale)" <dworley@avaya.com>
> *To:* Sohel Khan <sohel_khan777@yahoo.com>; "dispatch@ietf.org"
> <dispatch@ietf.org>
> *Sent:* Monday, January 23, 2012 4:18 PM
> *Subject:* RE: [dispatch] SPID Sequence and Loop Detection
>
>>  From: Sohel Khan [sohel_khan777@yahoo.com
> <mailto:sohel_khan777@yahoo.com>]
>>
>>  If a SIP INVITE with SPID sequence {33333333, 55555555, 77777777}
>>  arrives to Alpha’s network, Alpha will detect 77777777 in the SPID
>>  sequence and conclude that a loop has occurred.
>
> And that conclusion is incorrect, because at the second entry to
> Alpha, Alpha has no way of knowing that the request-URI is at all the
> same as the request-URI at the time of first entry to Alpha.
>
> In practice, if user A1 on network Alpha call-forwards his phone to
> user B on network Bravo, and user B call-forwards his phone to user A2
> on network Alpha, all calls to A1 will be routed to B, and then to A2,
> and will be rejected upon the second entry to Alpha.
>
> Something closer to the loop detection algorithm of RFC 5393 will be
> needed.
>
> Dale
>
>
>
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 25 Jan 2012 13:57:12 +0100
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Subject: Re: [dispatch] SPID Sequence and Loop Detection
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2012/1/23 Sohel Khan <sohel_khan777@yahoo.com>:
> SIP calls=C2=A0routing in an infinite loop in a subset of networks is a k=
nown
> phenomenon.
> Various SIP methods exist today may be used to detect and mitigate loops.
> Operational engineers would like to see a method that indicates the seque=
nce
> of providers=E2=80=99 IDs in the call path. Based on the input from the o=
perational
> engineers, we are proposing a=C2=A0new SIP header <SPID Sequence> to dete=
ct SIP
> routing loop. This method is preferred by the operational engineers of la=
rge
> providers=E2=80=99 networks.

IMHO RFC 5393 is the correct way to go. Your proposal breaks valid
scenarios. Check RC 5393.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: Sohel Khan <sohel_khan777@yahoo.com>
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--4756291-252513321-1327530310=:86812
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Thanks Paul and Dale.=0AI think the provider interal rule may vary.=0ADepen=
ding upon a provider internal rule, a provider may or may not take action b=
ased upon the information in the SPID sequence. Even inside a provider, the=
re can be two different types of SIP agents. One may choose to take particu=
lar action based on the information on the SPID and the other session agent=
 may not take any action based on the information on the SPID. =0APlease le=
t me know if I still missing something.=0AFor the call forwarding loops, th=
e operatioonal engineers prefer to use Diversion counter to break loops. =
=0AI will read the RFC that Dale suggested and will get back to you on this=
=A0further.=0A=A0=0ASohel=0A=A0=0AMessage: 3=0ADate: Tue, 24 Jan 2012 22:21=
:00 -0500=0AFrom: Paul Kyzivat <pkyzivat@alum.mit.edu>=0ATo: dispatch@ietf.=
org=0ASubject: Re: [dispatch] SPID Sequence and Loop Detection=0AMessage-ID=
: <4F1F751C.2000309@alum.mit.edu>=0AContent-Type: text/plain; charset=3Dwin=
dows-1252; format=3Dflowed=0A=0AOn 1/24/12 4:59 PM, Sohel Khan wrote:=0A> W=
e are not discussing routing loop due to call forwarding.=0A> Various loop =
mitigation mechanism may be applicable to mitigate=0A> different types of r=
outing loops.=0A=0AYou may not be discussing/considering those things, but =
you should be, =0Abecause (as Dale explained) the mechanism you are proposi=
ng would =0A*break* non-loop calls that revisit the same provider.=0A=0A=A0=
=A0=A0 Thanks,=0A=A0=A0=A0 Paul=0A=0A=0ARegards, Sohel Khan
--4756291-252513321-1327530310=:86812
Content-Type: text/html; charset=iso-8859-1
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<html><body><div style=3D"color:#000; background-color:#fff; font-family:ti=
mes new roman, new york, times, serif;font-size:12pt"><div style=3D"RIGHT: =
auto"><SPAN style=3D"RIGHT: auto">Thanks Paul and Dale.</SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto">I think the provider=
 interal rule may vary.</SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto">Depending upon a pro=
vider internal rule, a provider may or may not take action based upon the i=
nformation in the SPID sequence. </SPAN><SPAN style=3D"RIGHT: auto">Even in=
side a provider, there can be two different types of SIP agents. One may ch=
oose to take particular action based on the information on the SPID and <VA=
R id=3Dyui-ie-cursor></VAR>the other session agent may not take any action =
based on the information on the SPID. </SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto">Please let me know i=
f I still missing something.</SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto">For the call forward=
ing loops, the operatioonal engineers prefer to use Diversion counter to br=
eak loops. </SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto">I will read the RFC =
that Dale suggested and will get back to you on this&nbsp;further.</SPAN></=
div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto"></SPAN>&nbsp;</div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto">Sohel</SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto"></SPAN>&nbsp;</div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto">Message: 3<BR>Date: =
Tue, 24 Jan 2012 22:21:00 -0500<BR>From: Paul Kyzivat &lt;<A style=3D"RIGHT=
: auto" href=3D"mailto:pkyzivat@alum.mit.edu" ymailto=3D"mailto:pkyzivat@al=
um.mit.edu"><FONT style=3D"RIGHT: auto" color=3D#234786>pkyzivat@alum.mit.e=
du</FONT></A>&gt;<BR>To: <A href=3D"mailto:dispatch@ietf.org" ymailto=3D"ma=
ilto:dispatch@ietf.org"><FONT color=3D#234786>dispatch@ietf.org</FONT></A><=
BR>Subject: Re: [dispatch] SPID Sequence and Loop Detection<BR>Message-ID: =
&lt;<A style=3D"RIGHT: auto" href=3D"mailto:4F1F751C.2000309@alum.mit.edu" =
ymailto=3D"mailto:4F1F751C.2000309@alum.mit.edu"><FONT style=3D"RIGHT: auto=
" color=3D#234786>4F1F751C.2000309@alum.mit.edu</FONT></A>&gt;<BR>Content-T=
ype: text/plain; charset=3Dwindows-1252; format=3Dflowed<BR><BR>On 1/24/12 =
4:59 PM, Sohel Khan wrote:<BR>&gt; We are not discussing routing loop due t=
o call forwarding.<BR>&gt; Various loop mitigation mechanism may be applica=
ble to mitigate<BR>&gt;
 different types of routing loops.<BR><BR>You may not be discussing/conside=
ring those things, but you should be, <BR>because (as Dale explained) the m=
echanism you are proposing would <BR>*break* non-loop calls that revisit th=
e same provider.<BR><BR>&nbsp;&nbsp;&nbsp; Thanks,<BR>&nbsp;&nbsp;&nbsp; Pa=
ul<BR></div></SPAN>
<DIV></DIV>
<DIV>&nbsp;</DIV>
<div>Regards, Sohel Khan</div></div></body></html>
--4756291-252513321-1327530310=:86812--

From pkyzivat@alum.mit.edu  Wed Jan 25 19:17:34 2012
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Subject: Re: [dispatch] SPID Sequence and Loop Detection
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On 1/25/12 5:25 PM, Sohel Khan wrote:
> Thanks Paul and Dale.
> I think the provider interal rule may vary.
> Depending upon a provider internal rule, a provider may or may not take
> action based upon the information in the SPID sequence. Even inside a
> provider, there can be two different types of SIP agents. One may choose
> to take particular action based on the information on the SPID and the
> other session agent may not take any action based on the information on
> the SPID.

But in some cases it may decide there is an error based on the message 
revisiting a particular provider??? (After all, that is what your 
original message asserted.)

That policy could break some *valid* non-looping situations that revisit 
a provider based on call forwarding or other actions.

> Please let me know if I still missing something.
> For the call forwarding loops, the operatioonal engineers prefer to use
> Diversion counter to break loops.

Where do you get the diversion counter? From the Diversion header? Or 
from H-I?

This *might* work in a closed garden if all messages are policed to 
adhere to some proprietary constraints. But it won't work in general.

And I don't think you can fall back on SBCs to "make this right" for 
you. They can't necessarily identify and tunnel sufficient information.

	Thanks,
	Paul

> I will read the RFC that Dale suggested and will get back to you on this
> further.
> Sohel
> Message: 3
> Date: Tue, 24 Jan 2012 22:21:00 -0500
> From: Paul Kyzivat <pkyzivat@alum.mit.edu <mailto:pkyzivat@alum.mit.edu>>
> To: dispatch@ietf.org <mailto:dispatch@ietf.org>
> Subject: Re: [dispatch] SPID Sequence and Loop Detection
> Message-ID: <4F1F751C.2000309@alum.mit.edu
> <mailto:4F1F751C.2000309@alum.mit.edu>>
> Content-Type: text/plain; charset=windows-1252; format=flowed
>
> On 1/24/12 4:59 PM, Sohel Khan wrote:
>>  We are not discussing routing loop due to call forwarding.
>>  Various loop mitigation mechanism may be applicable to mitigate
>>  different types of routing loops.
>
> You may not be discussing/considering those things, but you should be,
> because (as Dale explained) the mechanism you are proposing would
> *break* non-loop calls that revisit the same provider.
>
> Thanks,
> Paul
> Regards, Sohel Khan
>
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


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Note - a related draft is this one (expired):

draft-lennox-mmusic-sdp-source-selection-02.txt

It proposes an SDP-based mechanism for turning streams on and off.
This subject has also been discussed in RTCWEB, and may also be in scope 
for CLUE.

           Harald


On 01/23/2012 04:50 PM, Bo Burman wrote:
> Dispatch,
>
> We have requested agenda time for the upcoming meeting to discuss the subject of Media Stream related signaling. From the feedback we got in AVTEXT WG on the discussion for a certain problem, the one related to pausing and resuming sender to middlebox media traffic, it is clear that a more high level discussion needs to happen.
>
> We have two Internet drafts related to the issues we would like to discuss.
> Media Stream Selection (MESS)
>   - draft-westerlund-dispatch-stream-selection-00
> RTP Media Stream Pause and Resume
>   - draft-westerlund-avtext-rtp-stream-pause-00
>
> The first one (MESS) proposes that for the general selection of media streams, an end-point within a centralized conferencing scenario desires to receive a media plane oriented rather than RTP/RTCP based signaling, which appears to have benefits as discussed in the draft.
>
> As mentioned, the pause resume draft is targeted for a different use case where a media plane solution has potential for other benefits. This was discussed and the main outcome of that discussion was three quite different opinions;
>
> a) We already do this signaling using TMMBR (RFC 5104) by setting the bit-rate value to 0, which the current specs doesn't allow.
>
> b) This should be done in signalling plane, RTP/RTCP shouldn't be bloated. (But I did interpret this as not necessarily needing to be SIP).
>
> c) When it comes to pause/resume, unless it is done using SIP/SDP it will not work through any Session Border Gateway as the session will be torn down after n seconds.
>
> Thus I think it important that we take a discussion around the use cases, existing solutions, limitations in those or the proposal to try to determine what is the most reasonable way of solving the problems and how to enable;
>
> 1) Receiver based selection of media streams from a larger set of streams, including different quality levels like video resolutions, in a centralized conferencing application.
>
> 2) Optimize the transport so that media streams currently not used can be turned off temporarily and then quickly resumed when someone determines a need for the media stream. For example using mechanism (1) but also for voice activity based, centrally controlled or other mechanisms to select which media streams are delivered to other session participants.
>
> We are working to revise our Internet drafts. We seek as much input as possible into this prior to the meeting so that we can ensure that any discussion are as on topic and focused as possible.
>
> Regards
>
> Bo Burman&  Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7141311
> Färögatan 6                | Mobile +46 73 0949021
> SE-164 80 Stockholm, Sweden| mailto: bo.burman at ericsson.com
> ----------------------------------------------------------------------
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>


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To: Sohel Khan <sohel_khan777@yahoo.com>
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Subject: Re: [dispatch] SPID Sequence and Loop Detection
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When doing least cost routing, my experience has been that if A delivers =
the call to B that B will then send to C, B wants to hide this fact from =
C and certainly wants to hide from A that they the call went to C. =
Otherwise B gets cut out and A and C just set up a peering deal =
directly.=20

So I worry a scheme like this would not work as B's SBC would be =
configured to remove this.=20


On Jan 23, 2012, at 1:50 PM, Sohel Khan wrote:

> SIP calls routing in an infinite loop in a subset of networks is a =
known phenomenon.
> Various SIP methods exist today may be used to detect and mitigate =
loops. Operational engineers would like to see a method that indicates =
the sequence of providers=92 IDs in the call path. Based on the input =
from the operational engineers, we are proposing a new SIP header <SPID =
Sequence> to detect SIP routing loop. This method is preferred by the =
operational engineers of large providers=92 networks.
> =20
> Assume that a SIP call originating from provider Alpha (SPID: =
77777777) and traversing through provider Bravo (SPID: 55555555), =
Charlie (SPID: 33333333), and looping back to Alpha.
> =20
> With the new method, the SIP INVITE contains the new header SPID =
sequence {33333333, 55555555, 77777777}.
> =20
> If a SIP INVITE with SPID sequence {33333333, 55555555, 77777777} =
arrives to Alpha=92s network, Alpha will detect 77777777 in the SPID =
sequence and conclude that a loop has occurred.
> =20
> In the above use case, an example =93SPID-sequence=94 header is as =
follows:
> SPID-Sequence:<sip:33333333>;index=3D1;
> SPID-Sequence:<sip:55555555>;index=3D1.1;
> SPID-Sequence:<sip:77777777>;index=3D1.2;
> =20
> A provider wishing not to divulge its SPID, may insert a private or a =
blank value.             =20
> SPID-Sequence:<sip:33333333>;index=3D1;
> SPID-Sequence:<sip:55555555>;index=3D1.1;
> SPID-Sequence:<sip:77777777>;index=3D1.2
> SPID-Sequence:<sip:PRIVATE>;index=3D1.3;
> A provider should not delete any value from the SPID sequence unless =
the SPID is its own.
> We had some discussion in the DISPATCH meetings on ideas such as this. =
We would need a broader community support and help to initiate a work in =
IETF RTC WG on SPID sequence.
> Please let us know your opinion.
> =20
>=20
> =20
> Regards, Sohel Khan
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


From sohel_khan777@yahoo.com  Mon Jan 30 12:14:38 2012
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From: Sohel Khan <sohel_khan777@yahoo.com>
To: Cullen Jennings <fluffy@iii.ca>
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Subject: Re: [dispatch] SPID Sequence and Loop Detection
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--4756291-657517060-1327954476=:49537
Content-Type: text/plain; charset=utf-8
Content-Transfer-Encoding: quoted-printable

Thanks Cullen.=0AActually, in my opinion, providers neither=C2=A0will add a=
=C2=A0=C2=A0SPID nor will add a=C2=A0known deterministic=C2=A0number.=0AIns=
tead, providers will implement a stochastic model, and will add a private e=
ncrypted number so that only a provider can decrypt its own identity in the=
 chain.=0A=C2=A0=0A=C2=A0Regards, Sohel Khan=0A=0A=0A______________________=
__________=0AFrom: Cullen Jennings <fluffy@iii.ca>=0ATo: Sohel Khan <sohel_=
khan777@yahoo.com> =0ACc: "dispatch@ietf.org" <dispatch@ietf.org> =0ASent: =
Friday, January 27, 2012 1:56 PM=0ASubject: Re: [dispatch] SPID Sequence an=
d Loop Detection=0A=0A=0AWhen doing least cost routing, my experience has b=
een that if A delivers the call to B that B will then send to C, B wants to=
 hide this fact from C and certainly wants to hide from A that they the cal=
l went to C. Otherwise B gets cut out and A and C just set up a peering dea=
l directly. =0A=0ASo I worry a scheme like this would not work as B's SBC w=
ould be configured to remove this. =0A=0A=0AOn Jan 23, 2012, at 1:50 PM, So=
hel Khan wrote:=0A=0A> SIP calls routing in an infinite loop in a subset of=
 networks is a known phenomenon.=0A> Various SIP methods exist today may be=
 used to detect and mitigate loops. Operational engineers would like to see=
 a method that indicates the sequence of providers=E2=80=99 IDs in the call=
 path. Based on the input from the operational engineers, we are proposing =
a new SIP header <SPID Sequence> to detect SIP routing loop. This method is=
 preferred by the operational engineers of large providers=E2=80=99 network=
s.=0A>=C2=A0 =0A> Assume that a SIP call originating from provider Alpha (S=
PID: 77777777) and traversing through provider Bravo (SPID: 55555555), Char=
lie (SPID: 33333333), and looping back to Alpha.=0A>=C2=A0 =0A> With the ne=
w method, the SIP INVITE contains the new header SPID sequence {33333333, 5=
5555555, 77777777}.=0A>=C2=A0 =0A> If a SIP INVITE with SPID sequence {3333=
3333, 55555555, 77777777} arrives to Alpha=E2=80=99s network, Alpha will de=
tect 77777777 in the SPID sequence and conclude that a loop has occurred.=
=0A>=C2=A0 =0A> In the above use case, an example =E2=80=9CSPID-sequence=E2=
=80=9D header is as follows:=0A> SPID-Sequence:<sip:33333333>;index=3D1;=0A=
> SPID-Sequence:<sip:55555555>;index=3D1.1;=0A> SPID-Sequence:<sip:77777777=
>;index=3D1.2;=0A>=C2=A0 =0A> A provider wishing not to divulge its SPID, m=
ay insert a private or a blank value.=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =0A> SPID-Sequence:<sip:33333333>;index=3D1;=0A> SPID-Sequence:<=
sip:55555555>;index=3D1.1;=0A> SPID-Sequence:<sip:77777777>;index=3D1.2=0A>=
 SPID-Sequence:<sip:PRIVATE>;index=3D1.3;=0A> A provider should not delete =
any value from the SPID sequence unless the SPID is its own.=0A> We had som=
e discussion in the DISPATCH meetings on ideas such as this. We would need =
a broader community support and help to initiate a work in IETF RTC WG on S=
PID sequence.=0A> Please let us know your opinion.=0A>=C2=A0 =0A> =0A>=C2=
=A0 =0A> Regards, Sohel Khan=0A> __________________________________________=
_____=0A> dispatch mailing list=0A> dispatch@ietf.org=0A> https://www.ietf.=
org/mailman/listinfo/dispatch
--4756291-657517060-1327954476=:49537
Content-Type: text/html; charset=utf-8
Content-Transfer-Encoding: quoted-printable

<html><body><div style=3D"color:#000; background-color:#fff; font-family:ti=
mes new roman, new york, times, serif;font-size:12pt"><div style=3D"RIGHT: =
auto"><SPAN style=3D"RIGHT: auto">Thanks Cullen.</SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto">Actually, in my opin=
ion, providers neither&nbsp;will add a&nbsp;&nbsp;SPID nor will add a&nbsp;=
known deterministic&nbsp;number.</SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto">Instead, providers w=
ill implement a stochastic model, and will add a private encrypted number s=
o that only a provider can decrypt its own identity in the chain.</SPAN></d=
iv>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto"></SPAN>&nbsp;</div>
<div style=3D"RIGHT: auto"><VAR id=3Dyui-ie-cursor></VAR>&nbsp;Regards, Soh=
el Khan<BR></div>
<DIV style=3D"FONT-FAMILY: times new roman, new york, times, serif; FONT-SI=
ZE: 12pt">
<DIV style=3D"FONT-FAMILY: times new roman, new york, times, serif; FONT-SI=
ZE: 12pt">
<DIV dir=3Dltr><FONT size=3D2 face=3DArial>
<DIV style=3D"BORDER-BOTTOM: #ccc 1px solid; BORDER-LEFT: #ccc 1px solid; P=
ADDING-BOTTOM: 0px; LINE-HEIGHT: 0; MARGIN: 5px 0px; PADDING-LEFT: 0px; PAD=
DING-RIGHT: 0px; HEIGHT: 0px; FONT-SIZE: 0px; BORDER-TOP: #ccc 1px solid; B=
ORDER-RIGHT: #ccc 1px solid; PADDING-TOP: 0px" class=3Dhr contentEditable=
=3Dfalse readonly=3D"true"></DIV><B><SPAN style=3D"FONT-WEIGHT: bold">From:=
</SPAN></B> Cullen Jennings &lt;fluffy@iii.ca&gt;<BR><B><SPAN style=3D"FONT=
-WEIGHT: bold">To:</SPAN></B> Sohel Khan &lt;sohel_khan777@yahoo.com&gt; <B=
R><B><SPAN style=3D"FONT-WEIGHT: bold">Cc:</SPAN></B> "dispatch@ietf.org" &=
lt;dispatch@ietf.org&gt; <BR><B><SPAN style=3D"FONT-WEIGHT: bold">Sent:</SP=
AN></B> Friday, January 27, 2012 1:56 PM<BR><B><SPAN style=3D"FONT-WEIGHT: =
bold">Subject:</SPAN></B> Re: [dispatch] SPID Sequence and Loop Detection<B=
R></FONT></DIV><BR><BR>When doing least cost routing, my experience has bee=
n that if A delivers the call to B that B will then send to C, B wants to h=
ide this fact
 from C and certainly wants to hide from A that they the call went to C. Ot=
herwise B gets cut out and A and C just set up a peering deal directly. <BR=
><BR>So I worry a scheme like this would not work as B's SBC would be confi=
gured to remove this. <BR><BR><BR>On Jan 23, 2012, at 1:50 PM, Sohel Khan w=
rote:<BR><BR>&gt; SIP calls routing in an infinite loop in a subset of netw=
orks is a known phenomenon.<BR>&gt; Various SIP methods exist today may be =
used to detect and mitigate loops. Operational engineers would like to see =
a method that indicates the sequence of providers=E2=80=99 IDs in the call =
path. Based on the input from the operational engineers, we are proposing a=
 new SIP header &lt;SPID Sequence&gt; to detect SIP routing loop. This meth=
od is preferred by the operational engineers of large providers=E2=80=99 ne=
tworks.<BR>&gt;&nbsp; <BR>&gt; Assume that a SIP call originating from prov=
ider Alpha (SPID: 77777777) and traversing through provider Bravo (SPID:
 55555555), Charlie (SPID: 33333333), and looping back to Alpha.<BR>&gt;&nb=
sp; <BR>&gt; With the new method, the SIP INVITE contains the new header SP=
ID sequence {33333333, 55555555, 77777777}.<BR>&gt;&nbsp; <BR>&gt; If a SIP=
 INVITE with SPID sequence {33333333, 55555555, 77777777} arrives to Alpha=
=E2=80=99s network, Alpha will detect 77777777 in the SPID sequence and con=
clude that a loop has occurred.<BR>&gt;&nbsp; <BR>&gt; In the above use cas=
e, an example =E2=80=9CSPID-sequence=E2=80=9D header is as follows:<BR>&gt;=
 SPID-Sequence:&lt;sip:33333333&gt;;index=3D1;<BR>&gt; SPID-Sequence:&lt;si=
p:55555555&gt;;index=3D1.1;<BR>&gt; SPID-Sequence:&lt;sip:77777777&gt;;inde=
x=3D1.2;<BR>&gt;&nbsp; <BR>&gt; A provider wishing not to divulge its SPID,=
 may insert a private or a blank value.&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &=
nbsp; &nbsp; <BR>&gt; SPID-Sequence:&lt;sip:33333333&gt;;index=3D1;<BR>&gt;=
 SPID-Sequence:&lt;sip:55555555&gt;;index=3D1.1;<BR>&gt;
 SPID-Sequence:&lt;sip:77777777&gt;;index=3D1.2<BR>&gt; SPID-Sequence:&lt;s=
ip:PRIVATE&gt;;index=3D1.3;<BR>&gt; A provider should not delete any value =
from the SPID sequence unless the SPID is its own.<BR>&gt; We had some disc=
ussion in the DISPATCH meetings on ideas such as this. We would need a broa=
der community support and help to initiate a work in IETF RTC WG on SPID se=
quence.<BR>&gt; Please let us know your opinion.<BR>&gt;&nbsp; <BR>&gt; <BR=
>&gt;&nbsp; <BR>&gt; Regards, Sohel Khan<BR>&gt; __________________________=
_____________________<BR>&gt; dispatch mailing list<BR>&gt; <A href=3D"mail=
to:dispatch@ietf.org" ymailto=3D"mailto:dispatch@ietf.org">dispatch@ietf.or=
g</A><BR>&gt; <A href=3D"https://www.ietf.org/mailman/listinfo/dispatch" ta=
rget=3D_blank>https://www.ietf.org/mailman/listinfo/dispatch</A><BR><BR><BR=
><BR></DIV></DIV></div></body></html>
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From pkyzivat@alum.mit.edu  Mon Jan 30 12:30:21 2012
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Subject: Re: [dispatch] SPID Sequence and Loop Detection
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On 1/30/12 3:14 PM, Sohel Khan wrote:
> Thanks Cullen.
> Actually, in my opinion, providers neither will add a SPID nor will add
> a known deterministic number.
> Instead, providers will implement a stochastic model, and will add a
> private encrypted number so that only a provider can decrypt its own
> identity in the chain.

So, are you withdrawing your proposal?
Or are you suggesting modifying it to carry these encrypted ids?

If the latter, then I guess what you are proposing is a new mechanism by 
which a provider can tell that a message has returned to it after having 
been routed to some other provider? To use that it would have to attempt 
to decrypt every value in the list. And most of the time none of those 
would turn out to be its own. And the revisit to the provider may be 
legitimate, rather than a "loop".

Rather than that, you might want to review 
draft-kaplan-dispatch-b2bua-loop-detection-00 and 
draft-jones-ipmc-session-id-reqts-01, which might serve your purpose.

	Thanks,
	Paul

> Regards, Sohel Khan
> *From:* Cullen Jennings <fluffy@iii.ca>
> *To:* Sohel Khan <sohel_khan777@yahoo.com>
> *Cc:* "dispatch@ietf.org" <dispatch@ietf.org>
> *Sent:* Friday, January 27, 2012 1:56 PM
> *Subject:* Re: [dispatch] SPID Sequence and Loop Detection
>
>
> When doing least cost routing, my experience has been that if A delivers
> the call to B that B will then send to C, B wants to hide this fact from
> C and certainly wants to hide from A that they the call went to C.
> Otherwise B gets cut out and A and C just set up a peering deal directly.
>
> So I worry a scheme like this would not work as B's SBC would be
> configured to remove this.
>
>
> On Jan 23, 2012, at 1:50 PM, Sohel Khan wrote:
>
>  > SIP calls routing in an infinite loop in a subset of networks is a
> known phenomenon.
>  > Various SIP methods exist today may be used to detect and mitigate
> loops. Operational engineers would like to see a method that indicates
> the sequence of providers’ IDs in the call path. Based on the input from
> the operational engineers, we are proposing a new SIP header <SPID
> Sequence> to detect SIP routing loop. This method is preferred by the
> operational engineers of large providers’ networks.
>  >
>  > Assume that a SIP call originating from provider Alpha (SPID:
> 77777777) and traversing through provider Bravo (SPID: 55555555),
> Charlie (SPID: 33333333), and looping back to Alpha.
>  >
>  > With the new method, the SIP INVITE contains the new header SPID
> sequence {33333333, 55555555, 77777777}.
>  >
>  > If a SIP INVITE with SPID sequence {33333333, 55555555, 77777777}
> arrives to Alpha’s network, Alpha will detect 77777777 in the SPID
> sequence and conclude that a loop has occurred.
>  >
>  > In the above use case, an example “SPID-sequence” header is as follows:
>  > SPID-Sequence:<sip:33333333>;index=1;
>  > SPID-Sequence:<sip:55555555>;index=1.1;
>  > SPID-Sequence:<sip:77777777>;index=1.2;
>  >
>  > A provider wishing not to divulge its SPID, may insert a private or a
> blank value.
>  > SPID-Sequence:<sip:33333333>;index=1;
>  > SPID-Sequence:<sip:55555555>;index=1.1;
>  > SPID-Sequence:<sip:77777777>;index=1.2
>  > SPID-Sequence:<sip:PRIVATE>;index=1.3;
>  > A provider should not delete any value from the SPID sequence unless
> the SPID is its own.
>  > We had some discussion in the DISPATCH meetings on ideas such as
> this. We would need a broader community support and help to initiate a
> work in IETF RTC WG on SPID sequence.
>  > Please let us know your opinion.
>  >
>  >
>  >
>  > Regards, Sohel Khan
>  > _______________________________________________
>  > 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


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Subject: Re: [dispatch] SPID Sequence and Loop Detection (Paul Kyzivat)
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--1969517296-869412809-1327960027=:29111
Content-Type: text/plain; charset=iso-8859-1
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Thank you Paul for the good input.=0ALet us discuss the two scenarios in de=
tails.=0AThe scenarios:=0A1) Loop due to call forwarding=0A2) Route analysi=
s=0A=A0=0ALoop due to call forwarding:=0AMy=A0analysis of=A0the empirical d=
ata shows that the majority of the loops due to call forwarding occur due i=
n a Fixed-Mobile convergence scenario. A user Jon Doe has two phones: Wirel=
ines and Wireless with two telephone numbers (TNine and TNess). Mr. Doe set=
s call forward of his Wirline phone to the Wireless phone prior to heading =
for the office.=A0 All calls arriving at his home are forwarded by the Wire=
line phone to his Wireless phone. By 2:00 PM, Mr. Doe is tierd of receiving=
 calls and sets call forward of his Wireless phone to the Wireline phone. N=
ow the game=A0 starts: Wireline phone forwards calls to the Wireless phone =
and the Wireless Phone forwards calls to the Wireline phone. It just takes =
one or two of this scenario to exhaust a significant amount of processors (=
e.g., SIP proxies) =A0in the network. Generally, it occurs between a Wireli=
ne provider and a Wireless provider peering with each other. We call solve =
this either by=A0 the proposed "SPID
 Sequence" header or by diversion-counter of the Diversion Indication heade=
r (RFC 5806).=0A=A0=0ALoop due to least cost route analysis:=0ALet us assum=
e that in a Utopian Internet, three SIP Multimeida networks exist: Alpha, B=
ravo, and Charlie.=0AAll these three networks, no SBC or B2BUA exist !=0A=
=A0=0AEach network has only one SIP Proxy and one SIP Redirect Server. The =
SIP Redirect servers in these three networks=A0are intelligent. Before=A0a =
SIP Redirect Server=A0decides a redirection, it performs a relational algeb=
ra to find the best redirection of a particular INVITE (not all invites). A=
lpha SIP Redirect Server =A0finds a redirection in Bravo's network; thus, A=
lpha's Proxy redirects.the INVITE Bravo's network. The Bravo's redirect ser=
ver also determines a redirection based on a intelligent logic; thus. Bravo=
's proxy redirects the INVITE to Charlie Network. The Charlie's network red=
irects the INVITE to Alpha's Network. Note that while Alpha, Bravo, and Cha=
irle network's redirection logic is either intelligent or consider it "inte=
lligent", the aggregate effect in the three networks is infinite loop!=0A=
=A0=0AAs=A0 I wrote in the other post, providers may not use the legible SP=
ID or a=A0 legible know deterministic value; providers may use encrypted st=
ochastic values so that no providers unders the identity of other providers=
 in the path.=0A=A0=0A=A0=0A=A0Date: Wed, 25 Jan 2012 22:17:31 -0500=0AFrom=
: Paul Kyzivat <pkyzivat@alum.mit.edu>=0ATo: dispatch@ietf.org=0ASubject: R=
e: [dispatch] SPID Sequence and Loop Detection=0AMessage-ID: <4F20C5CB.7030=
203@alum.mit.edu>=0AContent-Type: text/plain; charset=3DISO-8859-1; format=
=3Dflowed=0A=0AOn 1/25/12 5:25 PM, Sohel Khan wrote:=0A> Thanks Paul and Da=
le.=0A> I think the provider interal rule may vary.=0A> Depending upon a pr=
ovider internal rule, a provider may or may not take=0A> action based upon =
the information in the SPID sequence. Even inside a=0A> provider, there can=
 be two different types of SIP agents. One may choose=0A> to take particula=
r action based on the information on the SPID and the=0A> other session age=
nt may not take any action based on the information on=0A> the SPID.=0A=0AB=
ut in some cases it may decide there is an error based on the message =0Are=
visiting a particular provider??? (After all, that is what your =0Aoriginal=
 message asserted.)=0A=0AThat policy could break some *valid* non-looping s=
ituations that revisit =0Aa provider based on call forwarding or other acti=
ons.=0A=0A> Please let me know if I still missing something.=0A> For the ca=
ll forwarding loops, the operatioonal engineers prefer to use=0A> Diversion=
 counter to break loops.=0A=0AWhere do you get the diversion counter? From =
the Diversion header? Or =0Afrom H-I?=0A=0AThis *might* work in a closed ga=
rden if all messages are policed to =0Aadhere to some proprietary constrain=
ts. But it won't work in general.=0A=0AAnd I don't think you can fall back =
on SBCs to "make this right" for =0Ayou. They can't necessarily identify an=
d tunnel sufficient information.=0A=0A=A0=A0=A0 Thanks,=0A=A0=A0=A0 Paul
--1969517296-869412809-1327960027=:29111
Content-Type: text/html; charset=iso-8859-1
Content-Transfer-Encoding: quoted-printable

<html><body><div style=3D"color:#000; background-color:#fff; font-family:ti=
mes new roman, new york, times, serif;font-size:12pt"><div style=3D"RIGHT: =
auto"><SPAN style=3D"RIGHT: auto">Thank you Paul for the good input.</SPAN>=
</div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto">Let us discuss the t=
wo scenarios in details.</SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto">The scenarios:</SPAN=
></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto"><SPAN style=3D"RIGHT=
: auto">1) Loop due to call forwarding</SPAN></SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto">2) Route analysis</S=
PAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto"></SPAN>&nbsp;</div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto">Loop due to call for=
warding:</SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto">My&nbsp;analysis of&=
nbsp;t</SPAN><SPAN style=3D"RIGHT: auto">he empirical data shows that the m=
ajority of the loops due to call forwarding occur due in a Fixed-Mobile con=
vergence scenario. A user Jon Doe has two phones: Wirelines and Wireless wi=
th two telephone numbers (TNine and TNess). Mr. Doe sets call forward of hi=
s Wirline phone to the Wireless phone prior to heading for the office.&nbsp=
; All calls arriving at his home are forwarded by the Wireline phone to his=
 Wireless phone. By 2:00 PM, Mr. Doe is tierd of receiving calls and sets c=
all forward of his Wireless phone to the Wireline phone. </SPAN><SPAN style=
=3D"RIGHT: auto">Now the game&nbsp; starts: Wireline phone forwards calls t=
o the Wireless phone and the Wireless Phone forwards calls to the Wireline =
phone. It just takes one or two of this scenario to exhaust a significant a=
mount of processors (e.g., SIP proxies) &nbsp;in the network. Generally, it
 occurs between a Wireline provider and a Wireless provider peering with ea=
ch other. We call solve this either by&nbsp; the proposed "SPID Sequence" h=
eader or by diversion-counter of the Diversion Indication header (RFC 5806)=
.</SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto"></SPAN>&nbsp;</div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto">Loop due to least co=
st route analysis:</SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto">Let us assume that i=
n a Utopian Internet, three SIP Multimeida networks exist: Alpha, Bravo, an=
d Charlie.</SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto">All these three netw=
orks, no SBC or B2BUA exist !</SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto"></SPAN>&nbsp;</div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto">Each network has onl=
y one SIP Proxy and one SIP Redirect Server. </SPAN><SPAN style=3D"RIGHT: a=
uto">The SIP Redirect servers in these three networks&nbsp;are intelligent.=
 Before&nbsp;a SIP Redirect Server&nbsp;decides a redirection, it performs =
a relational algebra to find the best redirection of a particular INVITE (n=
ot all invites). Alpha SIP Redirect Server &nbsp;finds a redirection in Bra=
vo's network; thus, Alpha's Proxy redirects.the INVITE Bravo's network. The=
 Bravo's redirect server also determines a redirection based on a intellige=
nt logic; thus. Bravo's proxy redirects the INVITE to Charlie Network. The =
Charlie's network redirects the INVITE to Alpha's Network. Note that while =
Alpha, Bravo, and Chairle network's redirection logic is either intelligent=
 or consider it "intelligent", the aggregate effect in the three networks i=
s infinite loop!</SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto"></SPAN>&nbsp;</div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto">As&nbsp; I wrote in =
the other post, providers may not use the legible SPID or a&nbsp; legible k=
now deterministic value; providers may use encrypted stochastic values so t=
hat no providers unders the identity of other providers in the path.<VAR id=
=3Dyui-ie-cursor></VAR></SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto"></SPAN>&nbsp;</div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto"></SPAN>&nbsp;</div>
<div style=3D"RIGHT: auto">&nbsp;Date: Wed, 25 Jan 2012 22:17:31 -0500<BR>F=
rom: Paul Kyzivat &lt;<A href=3D"mailto:pkyzivat@alum.mit.edu" ymailto=3D"m=
ailto:pkyzivat@alum.mit.edu">pkyzivat@alum.mit.edu</A>&gt;<BR>To: <A href=
=3D"mailto:dispatch@ietf.org" ymailto=3D"mailto:dispatch@ietf.org">dispatch=
@ietf.org</A><BR>Subject: Re: [dispatch] SPID Sequence and Loop Detection<B=
R>Message-ID: &lt;<A href=3D"mailto:4F20C5CB.7030203@alum.mit.edu" ymailto=
=3D"mailto:4F20C5CB.7030203@alum.mit.edu">4F20C5CB.7030203@alum.mit.edu</A>=
&gt;<BR>Content-Type: text/plain; charset=3DISO-8859-1; format=3Dflowed<BR>=
<BR>On 1/25/12 5:25 PM, Sohel Khan wrote:<BR>&gt; Thanks Paul and Dale.<BR>=
&gt; I think the provider interal rule may vary.<BR>&gt; Depending upon a p=
rovider internal rule, a provider may or may not take<BR>&gt; action based =
upon the information in the SPID sequence. Even inside a<BR>&gt; provider, =
there can be two different types of SIP agents. One may choose<BR>&gt; to t=
ake particular
 action based on the information on the SPID and the<BR>&gt; other session =
agent may not take any action based on the information on<BR>&gt; the SPID.=
<BR><BR>But in some cases it may decide there is an error based on the mess=
age <BR>revisiting a particular provider??? (After all, that is what your <=
BR>original message asserted.)<BR><BR>That policy could break some *valid* =
non-looping situations that revisit <BR>a provider based on call forwarding=
 or other actions.<BR><BR>&gt; Please let me know if I still missing someth=
ing.<BR>&gt; For the call forwarding loops, the operatioonal engineers pref=
er to use<BR>&gt; Diversion counter to break loops.<BR><BR>Where do you get=
 the diversion counter? From the Diversion header? Or <BR>from H-I?<BR><BR>=
This *might* work in a closed garden if all messages are policed to <BR>adh=
ere to some proprietary constraints. But it won't work in general.<BR><BR>A=
nd I don't think you can fall back on SBCs to "make this right" for
 <BR>you. They can't necessarily identify and tunnel sufficient information=
.<BR><BR>&nbsp;&nbsp;&nbsp; Thanks,<BR>&nbsp;&nbsp;&nbsp; Paul<BR><BR></div=
></div></body></html>
--1969517296-869412809-1327960027=:29111--

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From: Sohel Khan <sohel_khan777@yahoo.com>
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--4756291-1280636108-1327961727=:69408
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Thanks Paul. No, I am not withdrawing the proposal.=0AI am modifying=A0the =
proposal=A0to=A0mitigate concerns raised in this discussion list.=0AYou are=
 correct that=A0 I proposing=A0 a new mechanism by =0Awhich a provider can =
tell that a message has returned to it after having =0Abeen routed to some =
other provider. =0AAssume, there are three encrypted values in a sequence. =
Only one proxy of a provider has to decrypt these three fixed sized encrypt=
ed values. =0AOther than a looped case, the calls arriving back is most pro=
bably for=A0call forwarding=A0case.=0AOperation Engineers like to use Diver=
sion Indication header for call forwarding although the RFC is a historical=
 RFC.=0AIf an INVITE contains a Diversion header, then the provider should =
not analyze the proposed provider sequence header.=0AMax-Forwards and Max-B=
readths techniques also need to be implemented along with above methods sin=
ce Max-Forwards and Max-Breadths have also limitations.=0A=A0=0A=A0=0ASohel=
 Khan
--4756291-1280636108-1327961727=:69408
Content-Type: text/html; charset=iso-8859-1
Content-Transfer-Encoding: quoted-printable

<html><body><div style=3D"color:#000; background-color:#fff; font-family:ti=
mes new roman, new york, times, serif;font-size:12pt"><div style=3D"RIGHT: =
auto"><SPAN style=3D"RIGHT: auto">Thanks Paul. No, I am not withdrawing the=
 proposal.</SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto">I am modifying&nbsp;=
the proposal&nbsp;to&nbsp;mitigate concerns raised in this discussion list.=
</SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto">You are correct that=
&nbsp; I proposing&nbsp; a new mechanism by <BR>which a provider can tell t=
hat a message has returned to it after having <BR>been routed to some other=
 provider. </SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto">Assume, there are th=
ree encrypted values in a sequence. Only one proxy of a provider has to dec=
rypt these three fixed sized encrypted values. </SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto">Other than a looped =
case, the calls arriving back is most probably for&nbsp;call forwarding&nbs=
p;case.</SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto">Operation Engineers =
like to use Diversion Indication header for call forwarding although the RF=
C is a historical RFC.</SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto">If an INVITE contain=
s a Diversion header, then the provider should not analyze the proposed pro=
vider sequence header.</SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto">Max-Forwards and Max=
-Breadths techniques also need to be implemented along with above methods s=
ince Max-Forwards and Max-Breadths have also limitations.</SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto"></SPAN>&nbsp;</div>
<div><SPAN style=3D"RIGHT: auto"></SPAN><SPAN style=3D"RIGHT: auto"></SPAN>=
&nbsp;</div>
<div style=3D"RIGHT: auto"><SPAN style=3D"RIGHT: auto">Sohel Khan<VAR id=3D=
yui-ie-cursor></VAR></div>
<div style=3D"RIGHT: auto"></SPAN><BR style=3D"RIGHT: auto">&nbsp;</div></d=
iv></body></html>
--4756291-1280636108-1327961727=:69408--

From pkyzivat@alum.mit.edu  Mon Jan 30 15:12:41 2012
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Subject: Re: [dispatch] SPID Sequence and Loop Detection
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On 1/30/12 5:15 PM, Sohel Khan wrote:
> Thanks Paul. No, I am not withdrawing the proposal.
> I am modifying the proposal to mitigate concerns raised in this
> discussion list.
> You are correct that I proposing a new mechanism by
> which a provider can tell that a message has returned to it after having
> been routed to some other provider.
> Assume, there are three encrypted values in a sequence. Only one proxy
> of a provider has to decrypt these three fixed sized encrypted values.

IIUC, you proposed that each provider would insert an encrypted 
identifier for itself. To achieve your goal of not identifying the 
provider to other providers, it needs to:
- vary from call to call
- not be superficially recognizable as being from you

Thus, *somewhere* after the message enters your control you will need to 
decode all the ids based on your algorithm in order to determine which 
are yours. You will need to do that for every id in the message. (But I 
agree you can probably find a way to only do it once, regardless of how 
many of your nodes the message traverses before leaving.

I also suspect it will be difficult to find a way to construct this 
encrypted id that will achieve your objectives.

> Other than a looped case, the calls arriving back is most probably for
> call forwarding case.
> Operation Engineers like to use Diversion Indication header for call
> forwarding although the RFC is a historical RFC.
> If an INVITE contains a Diversion header, then the provider should not
> analyze the proposed provider sequence header.

You cannot assume that other providers use Diversion.
With your approach, if a call passes through your provider, then to 
another where it is forwarded (without Diversion) back to you, then you 
will erroneously consider it a loop.

> Max-Forwards and Max-Breadths techniques also need to be implemented
> along with above methods since Max-Forwards and Max-Breadths have also
> limitations.
> Sohel Khan

I fail to see how what you propose adds anything other than problems.

	Thanks,
	Paul


From pkyzivat@alum.mit.edu  Mon Jan 30 15:15:54 2012
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Subject: Re: [dispatch] SPID Sequence and Loop Detection (Paul Kyzivat)
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On 1/30/12 4:47 PM, Sohel Khan wrote:
> Thank you Paul for the good input.
> Let us discuss the two scenarios in details.
> The scenarios:
> 1) Loop due to call forwarding
> 2) Route analysis
> Loop due to call forwarding:
> My analysis of the empirical data shows that the majority of the loops
> due to call forwarding occur due in a Fixed-Mobile convergence scenario.
> A user Jon Doe has two phones: Wirelines and Wireless with two telephone
> numbers (TNine and TNess). Mr. Doe sets call forward of his Wirline
> phone to the Wireless phone prior to heading for the office. All calls
> arriving at his home are forwarded by the Wireline phone to his Wireless
> phone. By 2:00 PM, Mr. Doe is tierd of receiving calls and sets call
> forward of his Wireless phone to the Wireline phone. Now the game
> starts: Wireline phone forwards calls to the Wireless phone and the
> Wireless Phone forwards calls to the Wireline phone. It just takes one
> or two of this scenario to exhaust a significant amount of processors
> (e.g., SIP proxies) in the network. Generally, it occurs between a
> Wireline provider and a Wireless provider peering with each other. We
> call solve this either by the proposed "SPID Sequence" header or by
> diversion-counter of the Diversion Indication header (RFC 5806).

You can solve this with max-forwards.
You can't count on Diversion being used.
And, as I noted in other reply, the spid sequence doesn't reliably work 
when you realize you can't count on Diversion.

> Loop due to least cost route analysis:
> Let us assume that in a Utopian Internet, three SIP Multimeida networks
> exist: Alpha, Bravo, and Charlie.
> All these three networks, no SBC or B2BUA exist !
> Each network has only one SIP Proxy and one SIP Redirect Server. The SIP
> Redirect servers in these three networks are intelligent. Before a SIP
> Redirect Server decides a redirection, it performs a relational algebra
> to find the best redirection of a particular INVITE (not all invites).
> Alpha SIP Redirect Server finds a redirection in Bravo's network; thus,
> Alpha's Proxy redirects.the INVITE Bravo's network. The Bravo's redirect
> server also determines a redirection based on a intelligent logic; thus.
> Bravo's proxy redirects the INVITE to Charlie Network. The Charlie's
> network redirects the INVITE to Alpha's Network. Note that while Alpha,
> Bravo, and Chairle network's redirection logic is either intelligent or
> consider it "intelligent", the aggregate effect in the three networks is
> infinite loop!
> As I wrote in the other post, providers may not use the legible SPID or
> a legible know deterministic value; providers may use encrypted
> stochastic values so that no providers unders the identity of other
> providers in the path.

Again, see my other reply.

	Thanks,
	Paul

> Date: Wed, 25 Jan 2012 22:17:31 -0500
> From: Paul Kyzivat <pkyzivat@alum.mit.edu <mailto:pkyzivat@alum.mit.edu>>
> To: dispatch@ietf.org <mailto:dispatch@ietf.org>
> Subject: Re: [dispatch] SPID Sequence and Loop Detection
> Message-ID: <4F20C5CB.7030203@alum.mit.edu
> <mailto:4F20C5CB.7030203@alum.mit.edu>>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> On 1/25/12 5:25 PM, Sohel Khan wrote:
>  > Thanks Paul and Dale.
>  > I think the provider interal rule may vary.
>  > Depending upon a provider internal rule, a provider may or may not take
>  > action based upon the information in the SPID sequence. Even inside a
>  > provider, there can be two different types of SIP agents. One may choose
>  > to take particular action based on the information on the SPID and the
>  > other session agent may not take any action based on the information on
>  > the SPID.
>
> But in some cases it may decide there is an error based on the message
> revisiting a particular provider??? (After all, that is what your
> original message asserted.)
>
> That policy could break some *valid* non-looping situations that revisit
> a provider based on call forwarding or other actions.
>
>  > Please let me know if I still missing something.
>  > For the call forwarding loops, the operatioonal engineers prefer to use
>  > Diversion counter to break loops.
>
> Where do you get the diversion counter? From the Diversion header? Or
> from H-I?
>
> This *might* work in a closed garden if all messages are policed to
> adhere to some proprietary constraints. But it won't work in general.
>
> And I don't think you can fall back on SBCs to "make this right" for
> you. They can't necessarily identify and tunnel sufficient information.
>
> Thanks,
> Paul
>


From keith.drage@alcatel-lucent.com  Tue Jan 31 03:15:13 2012
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Cc: "dispatch@ietf.org" <dispatch@ietf.org>
Subject: Re: [dispatch] SPID Sequence and Loop Detection
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If you are into anything more complicated that identifying the number of ho=
ps so far routed over, then you need to count more than the number of diver=
sions. Some multiple diversion scenarios can be perfectly legitimate, and i=
ndeed transit multiple operators, particularly when some of those users may=
 be roaming or on enterprise networks beyond a public network.

To do that analysis correctly you need more information akin to the History=
-Info header field (or at least the tree structure within it).

It seems we provide mechanisms, and then people refuse to use them because =
they reveal information, and we have to invent another mechanism, .... etc.

I would also note that I would oppose any chartered work investigating the =
applicability of the Diversion header.

Regards

Keith

> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Paul Kyzivat
> Sent: 30 January 2012 23:13
> To: Sohel Khan
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] SPID Sequence and Loop Detection
>=20
> On 1/30/12 5:15 PM, Sohel Khan wrote:
> > Thanks Paul. No, I am not withdrawing the proposal.
> > I am modifying the proposal to mitigate concerns raised in this
> > discussion list.
> > You are correct that I proposing a new mechanism by
> > which a provider can tell that a message has returned to it after havin=
g
> > been routed to some other provider.
> > Assume, there are three encrypted values in a sequence. Only one proxy
> > of a provider has to decrypt these three fixed sized encrypted values.
>=20
> IIUC, you proposed that each provider would insert an encrypted
> identifier for itself. To achieve your goal of not identifying the
> provider to other providers, it needs to:
> - vary from call to call
> - not be superficially recognizable as being from you
>=20
> Thus, *somewhere* after the message enters your control you will need to
> decode all the ids based on your algorithm in order to determine which
> are yours. You will need to do that for every id in the message. (But I
> agree you can probably find a way to only do it once, regardless of how
> many of your nodes the message traverses before leaving.
>=20
> I also suspect it will be difficult to find a way to construct this
> encrypted id that will achieve your objectives.
>=20
> > Other than a looped case, the calls arriving back is most probably for
> > call forwarding case.
> > Operation Engineers like to use Diversion Indication header for call
> > forwarding although the RFC is a historical RFC.
> > If an INVITE contains a Diversion header, then the provider should not
> > analyze the proposed provider sequence header.
>=20
> You cannot assume that other providers use Diversion.
> With your approach, if a call passes through your provider, then to
> another where it is forwarded (without Diversion) back to you, then you
> will erroneously consider it a loop.
>=20
> > Max-Forwards and Max-Breadths techniques also need to be implemented
> > along with above methods since Max-Forwards and Max-Breadths have also
> > limitations.
> > Sohel Khan
>=20
> I fail to see how what you propose adds anything other than problems.
>=20
> 	Thanks,
> 	Paul
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

From pkyzivat@alum.mit.edu  Tue Jan 31 06:20:10 2012
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Cc: "dispatch@ietf.org" <dispatch@ietf.org>, Sohel Khan <sohel_khan777@yahoo.com>
Subject: Re: [dispatch] SPID Sequence and Loop Detection
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On 1/31/12 6:15 AM, DRAGE, Keith (Keith) wrote:
> If you are into anything more complicated that identifying the number of hops so far routed over, then you need to count more than the number of diversions. Some multiple diversion scenarios can be perfectly legitimate, and indeed transit multiple operators, particularly when some of those users may be roaming or on enterprise networks beyond a public network.
>
> To do that analysis correctly you need more information akin to the History-Info header field (or at least the tree structure within it).
>
> It seems we provide mechanisms, and then people refuse to use them because they reveal information, and we have to invent another mechanism, .... etc.

+1

> I would also note that I would oppose any chartered work investigating the applicability of the Diversion header.

+1

	Thanks,
	Paul

> Regards
>
> Keith
>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>> Behalf Of Paul Kyzivat
>> Sent: 30 January 2012 23:13
>> To: Sohel Khan
>> Cc: dispatch@ietf.org
>> Subject: Re: [dispatch] SPID Sequence and Loop Detection
>>
>> On 1/30/12 5:15 PM, Sohel Khan wrote:
>>> Thanks Paul. No, I am not withdrawing the proposal.
>>> I am modifying the proposal to mitigate concerns raised in this
>>> discussion list.
>>> You are correct that I proposing a new mechanism by
>>> which a provider can tell that a message has returned to it after having
>>> been routed to some other provider.
>>> Assume, there are three encrypted values in a sequence. Only one proxy
>>> of a provider has to decrypt these three fixed sized encrypted values.
>>
>> IIUC, you proposed that each provider would insert an encrypted
>> identifier for itself. To achieve your goal of not identifying the
>> provider to other providers, it needs to:
>> - vary from call to call
>> - not be superficially recognizable as being from you
>>
>> Thus, *somewhere* after the message enters your control you will need to
>> decode all the ids based on your algorithm in order to determine which
>> are yours. You will need to do that for every id in the message. (But I
>> agree you can probably find a way to only do it once, regardless of how
>> many of your nodes the message traverses before leaving.
>>
>> I also suspect it will be difficult to find a way to construct this
>> encrypted id that will achieve your objectives.
>>
>>> Other than a looped case, the calls arriving back is most probably for
>>> call forwarding case.
>>> Operation Engineers like to use Diversion Indication header for call
>>> forwarding although the RFC is a historical RFC.
>>> If an INVITE contains a Diversion header, then the provider should not
>>> analyze the proposed provider sequence header.
>>
>> You cannot assume that other providers use Diversion.
>> With your approach, if a call passes through your provider, then to
>> another where it is forwarded (without Diversion) back to you, then you
>> will erroneously consider it a loop.
>>
>>> Max-Forwards and Max-Breadths techniques also need to be implemented
>>> along with above methods since Max-Forwards and Max-Breadths have also
>>> limitations.
>>> Sohel Khan
>>
>> I fail to see how what you propose adds anything other than problems.
>>
>> 	Thanks,
>> 	Paul
>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>


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FYI

> -----Original Message-----
> From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-
> bounces@ietf.org] On Behalf Of internet-drafts@ietf.org
> Sent: Monday, January 30, 2012 12:43 PM
> To: i-d-announce@ietf.org
> Subject: I-D Action: draft-jones-ipmc-session-id-reqts-01.txt
> 
> 
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> 
> 	Title           : Requirements for an End-to-End Session
> Identification in IP-Based Multimedia Communication Networks
> 	Author(s)       : Paul E. Jones
>                           Gonzalo Salgueiro
>                           James Polk
>                           Parthasarathi Ravindran
>                           Laura Liess
>                           Roland Jesske
>                           Salvatore Loreto
>                           Hadriel Kaplan
> 	Filename        : draft-jones-ipmc-session-id-reqts-01.txt
> 	Pages           : 8
> 	Date            : 2012-01-30
> 
>    This document specifies the requirements for an end-to-end session
>    identifier in IP-based multimedia communication networks.  This
>    identifier would enable endpoints, intermediate devices, and
>    management and monitoring systems to identify a session end-to-end,
>    associate multiple endpoints with a given multipoint conference,
>    track communication sessions when they are redirected, and associate
>    one or more media flows with a given communication session.
> 
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-jones-ipmc-session-id-reqts-
> 01.txt
> 
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
> 
> This Internet-Draft can be retrieved at:
> ftp://ftp.ietf.org/internet-drafts/draft-jones-ipmc-session-id-reqts-
> 01.txt
> 
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org
> https://www.ietf.org/mailman/listinfo/i-d-announce
> Internet-Draft directories: http://www.ietf.org/shadow.html or
> ftp://ftp.ietf.org/ietf/1shadow-sites.txt


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FYI

> -----Original Message-----
> From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-
> bounces@ietf.org] On Behalf Of internet-drafts@ietf.org
> Sent: Monday, January 30, 2012 12:47 PM
> To: i-d-announce@ietf.org
> Subject: I-D Action: draft-jones-ipmc-session-id-03.txt
> 
> 
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> 
> 	Title           : End-to-End Session Identification in IP-Based
> Multimedia Communication Networks
> 	Author(s)       : Paul E. Jones
>                           Chris Pearce
>                           James Polk
>                           Gonzalo Salgueiro
> 	Filename        : draft-jones-ipmc-session-id-03.txt
> 	Pages           : 8
> 	Date            : 2012-01-30
> 
>    This document describes an end-to-end Session Identifier for use in
>    IP-based Multimedia Communication systems that enables endpoints,
>    intermediate devices, and management systems to identify a session
>    end-to-end, associate multiple endpoints with a given multipoint
>    conference, track communication sessions when they are redirected,
>    and associate one or more media flows with a given communication
>    session.
> 
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-jones-ipmc-session-id-03.txt
> 
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
> 
> This Internet-Draft can be retrieved at:
> ftp://ftp.ietf.org/internet-drafts/draft-jones-ipmc-session-id-03.txt
> 
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org
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> Internet-Draft directories: http://www.ietf.org/shadow.html or
> ftp://ftp.ietf.org/ietf/1shadow-sites.txt


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Date: Tue, 31 Jan 2012 09:17:10 -0800 (PST)
From: Sohel Khan <sohel_khan777@yahoo.com>
To: "DRAGE, Keith \(Keith\)" <keith.drage@alcatel-lucent.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>
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Cc: "dispatch@ietf.org" <dispatch@ietf.org>
Subject: Re: [dispatch] SPID Sequence and Loop Detection
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Thanks Keith and Paul.=0AFirst of all, I am not proposing to start a charte=
r at this time. =0AMy intention is now to have a discussion to find a best =
solution to detect that loop has occurred. In my opinion, the proposed SPID=
 Sequence or Domain Sequence is analogous to the AS-PATH of BGP. =C2=A0Some=
 of you raised concern on disclosing the identity of=C2=A0providers; thus, =
I suggested=C2=A0that we include=C2=A0encrypted provider ID in the sequence=
. I agree with Keith that we should try to use existing mechanism (e.g., Hi=
story-info) instead of re-inventing something else. The History-Info can be=
 used for redirection related functions. The History-info also helps operat=
ion to determine the sequence of all the proxies in the call path. =C2=A0We=
 observed that fqdn of proxies are generally included in the call paths. In=
 the peering context, it helps to analyze the entry/exit proxies of a provi=
der=E2=80=99s network. Note that a large network may contain 100s of entry/=
exit proxies. The proxy fqdn information in the History-Info not always is =
fixed sized. =C2=A0The history information
 may not contain the name of the provider. Sometimes, name of the provider =
added in the history-info by different proxies use different combinations; =
thus, it is difficult for an automated system in an entry/exit proxy to ana=
lyze and detect loop in real time.=C2=A0 On the other hand, the proposed SP=
ID sequence/domain sequence information is planned to be fixed length; thus=
, easier for the entry/exit points to discover loop. Let us continue to dis=
cuss, we will=C2=A0appreciate your help to come up with a comprehensive sol=
ution.=0A=C2=A0=0ARegards, Sohel Khan=0A=0A=0A_____________________________=
___=0AFrom: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>=0ATo: P=
aul Kyzivat <pkyzivat@alum.mit.edu>; Sohel Khan <sohel_khan777@yahoo.com> =
=0ACc: "dispatch@ietf.org" <dispatch@ietf.org> =0ASent: Tuesday, January 31=
, 2012 6:15 AM=0ASubject: RE: [dispatch] SPID Sequence and Loop Detection=
=0A=0AIf you are into anything more complicated that identifying the number=
 of hops so far routed over, then you need to count more than the number of=
 diversions. Some multiple diversion scenarios can be perfectly legitimate,=
 and indeed transit multiple operators, particularly when some of those use=
rs may be roaming or on enterprise networks beyond a public network.=0A=0AT=
o do that analysis correctly you need more information akin to the History-=
Info header field (or at least the tree structure within it).=0A=0AIt seems=
 we provide mechanisms, and then people refuse to use them because they rev=
eal information, and we have to invent another mechanism, .... etc.=0A=0AI =
would also note that I would oppose any chartered work investigating the ap=
plicability of the Diversion header.=0A=0ARegards=0A=0AKeith=0A=0A> -----Or=
iginal Message-----=0A> From: dispatch-bounces@ietf.org [mailto:dispatch-bo=
unces@ietf.org] On=0A> Behalf Of Paul Kyzivat=0A> Sent: 30 January 2012 23:=
13=0A> To: Sohel Khan=0A> Cc: dispatch@ietf.org=0A> Subject: Re: [dispatch]=
 SPID Sequence and Loop Detection=0A> =0A> On 1/30/12 5:15 PM, Sohel Khan w=
rote:=0A> > Thanks Paul. No, I am not withdrawing the proposal.=0A> > I am =
modifying the proposal to mitigate concerns raised in this=0A> > discussion=
 list.=0A> > You are correct that I proposing a new mechanism by=0A> > whic=
h a provider can tell that a message has returned to it after having=0A> > =
been routed to some other provider.=0A> > Assume, there are three encrypted=
 values in a sequence. Only one proxy=0A> > of a provider has to decrypt th=
ese three fixed sized encrypted values.=0A> =0A> IIUC, you proposed that ea=
ch provider would insert an encrypted=0A> identifier for itself. To achieve=
 your goal of not identifying the=0A> provider to other providers, it needs=
 to:=0A> - vary from call to call=0A> - not be superficially recognizable a=
s being from you=0A> =0A> Thus, *somewhere* after the message enters your c=
ontrol you will need to=0A> decode all the ids based on your algorithm in o=
rder to determine which=0A> are yours. You will need to do that for every i=
d in the message. (But I=0A> agree you can probably find a way to only do i=
t once, regardless of how=0A> many of your nodes the message traverses befo=
re leaving.=0A> =0A> I also suspect it will be difficult to find a way to c=
onstruct this=0A> encrypted id that will achieve your objectives.=0A> =0A> =
> Other than a looped case, the calls arriving back is most probably for=0A=
> > call forwarding case.=0A> > Operation Engineers like to use Diversion I=
ndication header for call=0A> > forwarding although the RFC is a historical=
 RFC.=0A> > If an INVITE contains a Diversion header, then the provider sho=
uld not=0A> > analyze the proposed provider sequence header.=0A> =0A> You c=
annot assume that other providers use Diversion.=0A> With your approach, if=
 a call passes through your provider, then to=0A> another where it is forwa=
rded (without Diversion) back to you, then you=0A> will erroneously conside=
r it a loop.=0A> =0A> > Max-Forwards and Max-Breadths techniques also need =
to be implemented=0A> > along with above methods since Max-Forwards and Max=
-Breadths have also=0A> > limitations.=0A> > Sohel Khan=0A> =0A> I fail to =
see how what you propose adds anything other than problems.=0A> =0A> =C2=A0=
=C2=A0=C2=A0 Thanks,=0A> =C2=A0=C2=A0=C2=A0 Paul=0A> =0A> _________________=
______________________________=0A> dispatch mailing list=0A> dispatch@ietf.=
org=0A> https://www.ietf.org/mailman/listinfo/dispatch
--1969517296-246355704-1328030230=:95017
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<html><body><div style=3D"color:#000; background-color:#fff; font-family:ti=
mes new roman, new york, times, serif;font-size:12pt"><SPAN style=3D"RIGHT:=
 auto">
<div style=3D"RIGHT: auto"><SPAN style=3D"COLOR: black">Thanks Keith and Pa=
ul.</SPAN><?xml:namespace prefix =3D o ns =3D "urn:schemas-microsoft-com:of=
fice:office" /><o:p></o:p></SPAN style=3D"RIGHT: auto"></div>
<div><SPAN style=3D"COLOR: black">First of all, I am not proposing to start=
 a charter at this time. </SPAN><o:p></o:p></SPAN></div>
<div style=3D"RIGHT: auto"><SPAN style=3D"COLOR: black">My intention is now=
 to have a discussion to find a best solution to detect that loop has occur=
red.</SPAN> In my opinion, the proposed SPID Sequence or Domain Sequence is=
 analogous to the AS-PATH of BGP. </SPAN><SPAN style=3D"mso-spacerun: yes">=
&nbsp;</SPAN>Some of you raised concern on disclosing the identity of&nbsp;=
providers; thus, I suggested&nbsp;that we include&nbsp;encrypted provider I=
D in the sequence. I agree with Keith that we should try to use existing me=
chanism (e.g., History-info) instead of re-inventing something else. The Hi=
story-Info can be used for redirection related functions. The History-info =
also helps operation to determine the sequence of all the proxies in the ca=
ll path. <SPAN style=3D"mso-spacerun: yes">&nbsp;</SPAN>We observed that fq=
dn of proxies are generally included in the call paths. In the peering cont=
ext, it helps to analyze the entry/exit proxies of a provider=E2=80=99s net=
work.
 Note that a large network may contain 100s of entry/exit proxies. The prox=
y fqdn information in the History-Info not always is fixed sized. <SPAN sty=
le=3D"mso-spacerun: yes">&nbsp;</SPAN>The history information may not conta=
in the name of the provider. Sometimes, name of the provider added in the h=
istory-info by different proxies use different combinations; thus, it is di=
fficult for an automated system in an entry/exit proxy to analyze and detec=
t loop in real time.&nbsp; On the other hand, the proposed SPID sequence/do=
main sequence information is planned to be fixed length; thus, easier for t=
he entry/exit points to discover loop. </SPAN><SPAN style=3D"RIGHT: auto">L=
et us continue to discuss, we will&nbsp;appreciate your help to come up wit=
h a comprehensive solution.</SPAN><o:p></o:p></SPAN></div>
<div style=3D"RIGHT: auto"></SPAN><VAR id=3Dyui-ie-cursor></VAR>&nbsp;</div=
>
<div style=3D"RIGHT: auto">Regards, Sohel Khan<BR></div>
<DIV style=3D"FONT-FAMILY: times new roman, new york, times, serif; FONT-SI=
ZE: 12pt">
<DIV style=3D"FONT-FAMILY: times new roman, new york, times, serif; FONT-SI=
ZE: 12pt">
<DIV dir=3Dltr><FONT size=3D2 face=3DArial>
<DIV style=3D"BORDER-BOTTOM: #ccc 1px solid; BORDER-LEFT: #ccc 1px solid; P=
ADDING-BOTTOM: 0px; LINE-HEIGHT: 0; MARGIN: 5px 0px; PADDING-LEFT: 0px; PAD=
DING-RIGHT: 0px; HEIGHT: 0px; FONT-SIZE: 0px; BORDER-TOP: #ccc 1px solid; B=
ORDER-RIGHT: #ccc 1px solid; PADDING-TOP: 0px" class=3Dhr contentEditable=
=3Dfalse readonly=3D"true"></DIV><B><SPAN style=3D"FONT-WEIGHT: bold">From:=
</SPAN></B> "DRAGE, Keith (Keith)" &lt;keith.drage@alcatel-lucent.com&gt;<B=
R><B><SPAN style=3D"FONT-WEIGHT: bold">To:</SPAN></B> Paul Kyzivat &lt;pkyz=
ivat@alum.mit.edu&gt;; Sohel Khan &lt;sohel_khan777@yahoo.com&gt; <BR><B><S=
PAN style=3D"FONT-WEIGHT: bold">Cc:</SPAN></B> "dispatch@ietf.org" &lt;disp=
atch@ietf.org&gt; <BR><B><SPAN style=3D"FONT-WEIGHT: bold">Sent:</SPAN></B>=
 Tuesday, January 31, 2012 6:15 AM<BR><B><SPAN style=3D"FONT-WEIGHT: bold">=
Subject:</SPAN></B> RE: [dispatch] SPID Sequence and Loop Detection<BR></FO=
NT></DIV><BR>If you are into anything more complicated that identifying the=
 number of
 hops so far routed over, then you need to count more than the number of di=
versions. Some multiple diversion scenarios can be perfectly legitimate, an=
d indeed transit multiple operators, particularly when some of those users =
may be roaming or on enterprise networks beyond a public network.<BR><BR>To=
 do that analysis correctly you need more information akin to the History-I=
nfo header field (or at least the tree structure within it).<BR><BR>It seem=
s we provide mechanisms, and then people refuse to use them because they re=
veal information, and we have to invent another mechanism, .... etc.<BR><BR=
>I would also note that I would oppose any chartered work investigating the=
 applicability of the Diversion header.<BR><BR>Regards<BR><BR>Keith<BR><BR>=
&gt; -----Original Message-----<BR>&gt; From: <A href=3D"mailto:dispatch-bo=
unces@ietf.org" ymailto=3D"mailto:dispatch-bounces@ietf.org">dispatch-bounc=
es@ietf.org</A> [mailto:<A href=3D"mailto:dispatch-bounces@ietf.org"
 ymailto=3D"mailto:dispatch-bounces@ietf.org">dispatch-bounces@ietf.org</A>=
] On<BR>&gt; Behalf Of Paul Kyzivat<BR>&gt; Sent: 30 January 2012 23:13<BR>=
&gt; To: Sohel Khan<BR>&gt; Cc: <A href=3D"mailto:dispatch@ietf.org" ymailt=
o=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</A><BR>&gt; Subject: Re: [=
dispatch] SPID Sequence and Loop Detection<BR>&gt; <BR>&gt; On 1/30/12 5:15=
 PM, Sohel Khan wrote:<BR>&gt; &gt; Thanks Paul. No, I am not withdrawing t=
he proposal.<BR>&gt; &gt; I am modifying the proposal to mitigate concerns =
raised in this<BR>&gt; &gt; discussion list.<BR>&gt; &gt; You are correct t=
hat I proposing a new mechanism by<BR>&gt; &gt; which a provider can tell t=
hat a message has returned to it after having<BR>&gt; &gt; been routed to s=
ome other provider.<BR>&gt; &gt; Assume, there are three encrypted values i=
n a sequence. Only one proxy<BR>&gt; &gt; of a provider has to decrypt thes=
e three fixed sized encrypted values.<BR>&gt; <BR>&gt; IIUC, you proposed
 that each provider would insert an encrypted<BR>&gt; identifier for itself=
. To achieve your goal of not identifying the<BR>&gt; provider to other pro=
viders, it needs to:<BR>&gt; - vary from call to call<BR>&gt; - not be supe=
rficially recognizable as being from you<BR>&gt; <BR>&gt; Thus, *somewhere*=
 after the message enters your control you will need to<BR>&gt; decode all =
the ids based on your algorithm in order to determine which<BR>&gt; are you=
rs. You will need to do that for every id in the message. (But I<BR>&gt; ag=
ree you can probably find a way to only do it once, regardless of how<BR>&g=
t; many of your nodes the message traverses before leaving.<BR>&gt; <BR>&gt=
; I also suspect it will be difficult to find a way to construct this<BR>&g=
t; encrypted id that will achieve your objectives.<BR>&gt; <BR>&gt; &gt; Ot=
her than a looped case, the calls arriving back is most probably for<BR>&gt=
; &gt; call forwarding case.<BR>&gt; &gt; Operation Engineers like
 to use Diversion Indication header for call<BR>&gt; &gt; forwarding althou=
gh the RFC is a historical RFC.<BR>&gt; &gt; If an INVITE contains a Divers=
ion header, then the provider should not<BR>&gt; &gt; analyze the proposed =
provider sequence header.<BR>&gt; <BR>&gt; You cannot assume that other pro=
viders use Diversion.<BR>&gt; With your approach, if a call passes through =
your provider, then to<BR>&gt; another where it is forwarded (without Diver=
sion) back to you, then you<BR>&gt; will erroneously consider it a loop.<BR=
>&gt; <BR>&gt; &gt; Max-Forwards and Max-Breadths techniques also need to b=
e implemented<BR>&gt; &gt; along with above methods since Max-Forwards and =
Max-Breadths have also<BR>&gt; &gt; limitations.<BR>&gt; &gt; Sohel Khan<BR=
>&gt; <BR>&gt; I fail to see how what you propose adds anything other than =
problems.<BR>&gt; <BR>&gt; &nbsp;&nbsp;&nbsp; Thanks,<BR>&gt; &nbsp;&nbsp;&=
nbsp; Paul<BR>&gt; <BR>&gt;
 _______________________________________________<BR>&gt; dispatch mailing l=
ist<BR>&gt; <A href=3D"mailto:dispatch@ietf.org" ymailto=3D"mailto:dispatch=
@ietf.org">dispatch@ietf.org</A><BR>&gt; <A href=3D"https://www.ietf.org/ma=
ilman/listinfo/dispatch" target=3D_blank>https://www.ietf.org/mailman/listi=
nfo/dispatch</A><BR><BR><BR></DIV></DIV></div></body></html>
--1969517296-246355704-1328030230=:95017--
