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From: Ted Hardie <ted.ietf@gmail.com>
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Subject: [rtcweb] Agenda for IETF 92
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Howdy,

We're coming up on IETF 92, and RTCWEB currently has two sessions planned.
In the preliminary schedule we are Wednesday 13:00-15:00 and Thursday
17:40-18:40.  This is a much smaller amount of time that we've previously
requested, reflecting the progress we've made on a number of fronts.  Our
current thought on divvying up the time is:

Wednesday:

Administrivia:  5 minutes
Chair Doc Review: 15 minutes
JSEP:  60 minutes
FEC: 30 minutes

Thursday:

Transports: 20 minutes
WGLC resolutions: 40 minutes

Let the chairs know if you have a request for time and we can adjust.

Ted, Sean, Cullen

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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:tahoma,s=
ans-serif;font-size:small"><div class=3D"gmail_default" style=3D"font-famil=
y:tahoma,sans-serif;font-size:small">Howdy,<br><br></div><div class=3D"gmai=
l_default" style=3D"font-family:tahoma,sans-serif;font-size:small">We&#39;r=
e
 coming up on IETF 92, and RTCWEB currently has two sessions planned.=C2=A0=
=20
In the preliminary schedule we are Wednesday 13:00-15:00 and Thursday=20
17:40-18:40.=C2=A0 This is a much smaller amount of time that we&#39;ve=20
previously requested, reflecting the progress we&#39;ve made on a number of=
=20
fronts.=C2=A0 Our current thought on divvying up the time is:<br><br></div>=
<div class=3D"gmail_default" style=3D"font-family:tahoma,sans-serif;font-si=
ze:small">Wednesday: <br><br></div><div class=3D"gmail_default" style=3D"fo=
nt-family:tahoma,sans-serif;font-size:small">Administrivia:=C2=A0 5 minutes=
<br>Chair Doc Review: 15 minutes<br></div><div class=3D"gmail_default" styl=
e=3D"font-family:tahoma,sans-serif;font-size:small">JSEP:=C2=A0 60 minutes<=
br></div><div class=3D"gmail_default" style=3D"font-family:tahoma,sans-seri=
f;font-size:small">FEC: 30 minutes<br><br></div><div class=3D"gmail_default=
" style=3D"font-family:tahoma,sans-serif;font-size:small">Thursday:<br><br>=
</div><div class=3D"gmail_default" style=3D"font-family:tahoma,sans-serif;f=
ont-size:small">Transports: 20 minutes<br></div><div class=3D"gmail_default=
" style=3D"font-family:tahoma,sans-serif;font-size:small">WGLC resolutions:=
 40 minutes<br><br></div><div class=3D"gmail_default" style=3D"font-family:=
tahoma,sans-serif;font-size:small">Let the chairs know if you have a reques=
t for time and we can adjust.<br><br></div>Ted, Sean, Cullen</div></div>

--001a1142999047e0b2051052c0ef--


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Date: Wed, 04 Mar 2015 13:12:18 -0500
From: Simon Perreault <sperreault@jive.com>
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Subject: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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This is a follow-up to this thread on discuss-webrtc:
https://groups.google.com/forum/#!topic/discuss-webrtc/i67xq-FWtoM

I see two distinct problems:

1. It must be possible for a DTLS connection to jump from one ICE 
candidate to another. That is, a DTLS connection must not be bound to a 
single 5-tuple. This can be deduced from a good understanding of ICE, 
but it could certainly be stated more clearly.

2. SRTP keying material extracted from a DTLS connection must also not 
be bound to a single 5-tuple, in case media jumps from one ICE candidate 
to another. That's more problematic because RFC 5764 states very clearly 
that keying material is to be bound to the DTLS 5-tuple, modulo forking.

If I'm not mistaken, the right thing to do would be to bind keying 
material to all local ICE candidates (multiple 3-tuples), and treat any 
incoming media flow as indicated in RFC 5764 page 15, with one mapping 
table per 3-tuple. For outgoing flows, I guess it would make most sense 
to bind keying material to the 5-tuples for which the sender has 
obtained consent.

Is this correct? If so, would it be useful to add text explaining this 
in detail?

Simon


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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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--001a1134bd8067847605107afa09
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On Wed, Mar 4, 2015 at 1:12 PM, Simon Perreault <sperreault@jive.com> wrote:

> This is a follow-up to this thread on discuss-webrtc:
> https://groups.google.com/forum/#!topic/discuss-webrtc/i67xq-FWtoM
>
> If I'm not mistaken, the right thing to do would be to bind keying
> material to all local ICE candidates (multiple 3-tuples), and treat any
> incoming media flow as indicated in RFC 5764 page 15, with one mapping
> table per 3-tuple. For outgoing flows, I guess it would make most sense to
> bind keying material to the 5-tuples for which the sender has obtained
> consent.
>
> Is this correct? If so, would it be useful to add text explaining this in
> detail?
>
>
This is not correct. End point can have multiple flows with different
keying material or different DTLS sessions on the same local ICE 3-tuple
due to forking. More correct implementation would be to associate multiple
5-tuples with the same logical transport stream based on ICE connectivity
checks using ICE ufrag to identify which logical stream to associate with
each 5-tuple.

There is also another interesting consequence of this -- end point should
not re-use the same ICE candidate IP/port with a different ufrag during
session update offer/answer exchange. Otherwise you might end up with
ambiguous association between the logical streams and keying material or
DTLS session.

All of this probably needs to be defined somewhere and I am not aware which
RFC or draft defines this at this time.
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On W=
ed, Mar 4, 2015 at 1:12 PM, Simon Perreault <span dir=3D"ltr">&lt;<a href=
=3D"mailto:sperreault@jive.com" target=3D"_blank">sperreault@jive.com</a>&g=
t;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0=
px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);borde=
r-left-style:solid;padding-left:1ex">This is a follow-up to this thread on =
discuss-webrtc:<br>
<a href=3D"https://groups.google.com/forum/#!topic/discuss-webrtc/i67xq-FWt=
oM" target=3D"_blank">https://groups.google.com/<u></u>forum/#!topic/discus=
s-webrtc/<u></u>i67xq-FWtoM</a><br>
<br>If I&#39;m not mistaken, the right thing to do would be to bind keying =
material to all local ICE candidates (multiple 3-tuples), and treat any inc=
oming media flow as indicated in RFC 5764 page 15, with one mapping table p=
er 3-tuple. For outgoing flows, I guess it would make most sense to bind ke=
ying material to the 5-tuples for which the sender has obtained consent.<br=
>
<br>
Is this correct? If so, would it be useful to add text explaining this in d=
etail?<br>
<br></blockquote><div><br></div><div>This is not correct. End point can hav=
e multiple flows with different keying material or different DTLS sessions =
on the same local ICE 3-tuple due to forking. More correct implementation w=
ould be to associate multiple 5-tuples with the same logical transport stre=
am based on ICE connectivity checks using ICE ufrag to identify which logic=
al stream to associate with each 5-tuple.</div><div><br></div><div>There is=
 also another interesting consequence of this -- end point should not re-us=
e the same ICE candidate IP/port with a different ufrag during session upda=
te offer/answer exchange. Otherwise you might end up with ambiguous associa=
tion between the logical streams and keying material or DTLS session.=C2=A0=
</div><div><br></div><div>All of this probably needs to be defined somewher=
e and I am not aware which RFC or draft defines this at this time.</div><di=
v><div class=3D"gmail_signature">_____________<br>Roman Shpount</div></div>=
<div>=C2=A0</div></div></div></div>

--001a1134bd8067847605107afa09--


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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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Le 2015-03-04 13:54, Roman Shpount a Ã©crit :
> This is not correct. End point can have multiple flows with different
> keying material or different DTLS sessions on the same local ICE 3-tuple
> due to forking. More correct implementation would be to associate
> multiple 5-tuples with the same logical transport stream based on ICE
> connectivity checks using ICE ufrag to identify which logical stream to
> associate with each 5-tuple.

Agreed.

> There is also another interesting consequence of this -- end point
> should not re-use the same ICE candidate IP/port with a different ufrag
> during session update offer/answer exchange. Otherwise you might end up
> with ambiguous association between the logical streams and keying
> material or DTLS session.

Right. And wait MSL seconds before reusing a candidate to let the pipes 
drain.

> All of this probably needs to be defined somewhere and I am not aware
> which RFC or draft defines this at this time.

I guess for DTLS/DTLS-SRTP the obvious target would be -security...

Simon


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Simon Perreault <sperreault@jive.com>, Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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Hi,

Didn't we last week agree that, if the underlying transport changes, the DT=
LS connection MUST be re-established?

Jumping from one candidate to another is a transport change, isn't it?

Regards,

Christer

Sent from my Windows Phone
________________________________
From: Simon Perreault<mailto:sperreault@jive.com>
Sent: =FD04/=FD03/=FD2015 21:02
To: Roman Shpount<mailto:roman@telurix.com>
Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples

Le 2015-03-04 13:54, Roman Shpount a =E9crit :
> This is not correct. End point can have multiple flows with different
> keying material or different DTLS sessions on the same local ICE 3-tuple
> due to forking. More correct implementation would be to associate
> multiple 5-tuples with the same logical transport stream based on ICE
> connectivity checks using ICE ufrag to identify which logical stream to
> associate with each 5-tuple.

Agreed.

> There is also another interesting consequence of this -- end point
> should not re-use the same ICE candidate IP/port with a different ufrag
> during session update offer/answer exchange. Otherwise you might end up
> with ambiguous association between the logical streams and keying
> material or DTLS session.

Right. And wait MSL seconds before reusing a candidate to let the pipes
drain.

> All of this probably needs to be defined somewhere and I am not aware
> which RFC or draft defines this at this time.

I guess for DTLS/DTLS-SRTP the obvious target would be -security...

Simon

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

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<div>
<div>
<div style=3D"font-family:Calibri,sans-serif; font-size:11pt">Hi,<br>
<br>
Didn't we last week agree that, if the underlying transport changes, the DT=
LS connection MUST be re-established?<br>
<br>
Jumping from one candidate to another is a transport change, isn't it?<br>
<br>
Regards,<br>
<br>
Christer<br>
<br>
Sent from my Windows Phone</div>
</div>
<div dir=3D"ltr">
<hr>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">From:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:sperreault@jive.com">Simon Perreault</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Sent:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">=FD04=
/=FD03/=FD2015 21:02</span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">To:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:roman@telurix.com">Roman Shpount</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Cc:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Subject:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">Re: [=
rtcweb] DTLS, DTLS-SRTP, and 5-tuples</span><br>
<br>
</div>
</div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText">Le 2015-03-04 13:54, Roman Shpount a =E9crit :<br>
&gt; This is not correct. End point can have multiple flows with different<=
br>
&gt; keying material or different DTLS sessions on the same local ICE 3-tup=
le<br>
&gt; due to forking. More correct implementation would be to associate<br>
&gt; multiple 5-tuples with the same logical transport stream based on ICE<=
br>
&gt; connectivity checks using ICE ufrag to identify which logical stream t=
o<br>
&gt; associate with each 5-tuple.<br>
<br>
Agreed.<br>
<br>
&gt; There is also another interesting consequence of this -- end point<br>
&gt; should not re-use the same ICE candidate IP/port with a different ufra=
g<br>
&gt; during session update offer/answer exchange. Otherwise you might end u=
p<br>
&gt; with ambiguous association between the logical streams and keying<br>
&gt; material or DTLS session.<br>
<br>
Right. And wait MSL seconds before reusing a candidate to let the pipes <br=
>
drain.<br>
<br>
&gt; All of this probably needs to be defined somewhere and I am not aware<=
br>
&gt; which RFC or draft defines this at this time.<br>
<br>
I guess for DTLS/DTLS-SRTP the obvious target would be -security...<br>
<br>
Simon<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
rtcweb@ietf.org<br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.o=
rg/mailman/listinfo/rtcweb</a><br>
</div>
</span></font>
</body>
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On Wed, Mar 4, 2015 at 2:54 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>   Didn't we last week agree that, if the underlying transport changes,
> the DTLS connection MUST be re-established?
>
> Jumping from one candidate to another is a transport change, isn't it?
>
>
In cases where ICE is used jump from on candidate to another should not
constitute a transport change. A change in ICE ufrag should constitute the
transport change. All ICE candidates are a single virtual transport
channel. Without this a lot of ICE setup scenarios, such as rapid
nomination, break down. Even changes in the c= line address, m=line port
port, or list of candidates is not a transport change if ufrag stays the
same.

In cases where ICE is not used, changes of address in c= line or port in m=
line are a transport change.
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On W=
ed, Mar 4, 2015 at 2:54 PM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=
=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmb=
erg@ericsson.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote"=
 style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:=
rgb(204,204,204);border-left-style:solid;padding-left:1ex">





<div>
<div>
<div>
<div style=3D"font-family:Calibri,sans-serif;font-size:11pt">Didn&#39;t we =
last week agree that, if the underlying transport changes, the DTLS connect=
ion MUST be re-established?<br>
<br>
Jumping from one candidate to another is a transport change, isn&#39;t it?<=
br>
<br></div></div></div></div></blockquote><div><br></div><div>In cases where=
 ICE is used jump from on candidate to another should not constitute a tran=
sport change. A change in ICE ufrag should constitute the transport change.=
 All ICE candidates are a single virtual transport channel. Without this a =
lot of ICE setup scenarios, such as rapid nomination, break down. Even chan=
ges in the c=3D line address, m=3Dline port port, or list of candidates is =
not a transport change if ufrag stays the same.</div><div><br></div><div>In=
 cases where ICE is not used, changes of address in c=3D line or port in m=
=3D line are a transport change.</div><div><div class=3D"gmail_signature">_=
____________<br>Roman Shpount</div></div><div>=C2=A0</div></div></div></div=
>

--089e01229aaa6ba18c05107be472--


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From: Simon Perreault <sperreault@jive.com>
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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Le 2015-03-04 14:59, Roman Shpount a Ã©crit :
> On Wed, Mar 4, 2015 at 2:54 PM, Christer Holmberg
> <christer.holmberg@ericsson.com <mailto:christer.holmberg@ericsson.com>>
> wrote:
>
>     Didn't we last week agree that, if the underlying transport changes,
>     the DTLS connection MUST be re-established?
>
>     Jumping from one candidate to another is a transport change, isn't it?
>
>
> In cases where ICE is used jump from on candidate to another should not
> constitute a transport change. A change in ICE ufrag should constitute
> the transport change. All ICE candidates are a single virtual transport
> channel. Without this a lot of ICE setup scenarios, such as rapid
> nomination, break down. Even changes in the c= line address, m=line port
> port, or list of candidates is not a transport change if ufrag stays the
> same.
>
> In cases where ICE is not used, changes of address in c= line or port in
> m= line are a transport change.

I'm in full agreement.

Simon


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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--_000_7594FB04B1934943A5C02806D1A2204B1D726B71ESESSMB209erics_
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Hi,

What if you jump from an UDP candidate to a TCP candidate? At the same time=
 you would be jumping from DTLS to TLS (even for SRTP, where DTLS is only u=
sed for key management).

Regards,

Christer

Sent from my Windows Phone
________________________________
From: Roman Shpount<mailto:roman@telurix.com>
Sent: =FD04/=FD03/=FD2015 21:59
To: Christer Holmberg<mailto:christer.holmberg@ericsson.com>
Cc: Simon Perreault<mailto:sperreault@jive.com>; rtcweb@ietf.org<mailto:rtc=
web@ietf.org>
Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples

On Wed, Mar 4, 2015 at 2:54 PM, Christer Holmberg <christer.holmberg@ericss=
on.com<mailto:christer.holmberg@ericsson.com>> wrote:
Didn't we last week agree that, if the underlying transport changes, the DT=
LS connection MUST be re-established?

Jumping from one candidate to another is a transport change, isn't it?


In cases where ICE is used jump from on candidate to another should not con=
stitute a transport change. A change in ICE ufrag should constitute the tra=
nsport change. All ICE candidates are a single virtual transport channel. W=
ithout this a lot of ICE setup scenarios, such as rapid nomination, break d=
own. Even changes in the c=3D line address, m=3Dline port port, or list of =
candidates is not a transport change if ufrag stays the same.

In cases where ICE is not used, changes of address in c=3D line or port in =
m=3D line are a transport change.
_____________
Roman Shpount


--_000_7594FB04B1934943A5C02806D1A2204B1D726B71ESESSMB209erics_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dwindows-1=
256">
</head>
<body>
<div>
<div style=3D"font-family:Calibri,sans-serif; font-size:11pt">Hi,<br>
<br>
What if you jump from an UDP candidate to a TCP candidate? At the same time=
 you would be jumping from DTLS to TLS (even for SRTP, where DTLS is only u=
sed for key management).<br>
<br>
Regards,<br>
<br>
Christer<br>
<br>
Sent from my Windows Phone</div>
</div>
<div dir=3D"ltr">
<hr>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">From:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:roman@telurix.com">Roman Shpount</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Sent:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">=FD04=
/=FD03/=FD2015 21:59</span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">To:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:christer.holmberg@ericsson.com">Christer Holmberg</a></span><b=
r>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Cc:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:sperreault@jive.com">Simon Perreault</a>;
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Subject:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">Re: [=
rtcweb] DTLS, DTLS-SRTP, and 5-tuples</span><br>
<br>
</div>
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">On Wed, Mar 4, 2015 at 2:54 PM, Christer Holmber=
g <span dir=3D"ltr">
&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chr=
ister.holmberg@ericsson.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex; border=
-left-width:1px; border-left-color:rgb(204,204,204); border-left-style:soli=
d; padding-left:1ex">
<div>
<div>
<div>
<div style=3D"font-family:Calibri,sans-serif; font-size:11pt">Didn't we las=
t week agree that, if the underlying transport changes, the DTLS connection=
 MUST be re-established?<br>
<br>
Jumping from one candidate to another is a transport change, isn't it?<br>
<br>
</div>
</div>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>In cases where ICE is used jump from on candidate to another should no=
t constitute a transport change. A change in ICE ufrag should constitute th=
e transport change. All ICE candidates are a single virtual transport chann=
el. Without this a lot of ICE setup
 scenarios, such as rapid nomination, break down. Even changes in the c=3D =
line address, m=3Dline port port, or list of candidates is not a transport =
change if ufrag stays the same.</div>
<div><br>
</div>
<div>In cases where ICE is not used, changes of address in c=3D line or por=
t in m=3D line are a transport change.</div>
<div>
<div class=3D"gmail_signature">_____________<br>
Roman Shpount</div>
</div>
<div>&nbsp;</div>
</div>
</div>
</div>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1D726B71ESESSMB209erics_--


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To: Christer Holmberg <christer.holmberg@ericsson.com>, Roman Shpount <roman@telurix.com>
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Archived-At: <http://mailarchive.ietf.org/arch/msg/rtcweb/rvB4YWw888fukwpsLsuRz3eZ6lk>
Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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Correction. You are not jumping to TLS, if you use framing for DTLS when TC=
P is used.

Regards,

Christer

Sent from my Windows Phone
________________________________
From: Christer Holmberg<mailto:christer.holmberg@ericsson.com>
Sent: =FD04/=FD03/=FD2015 22:05
To: Roman Shpount<mailto:roman@telurix.com>
Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples

Hi,

What if you jump from an UDP candidate to a TCP candidate? At the same time=
 you would be jumping from DTLS to TLS (even for SRTP, where DTLS is only u=
sed for key management).

Regards,

Christer

Sent from my Windows Phone
________________________________
From: Roman Shpount<mailto:roman@telurix.com>
Sent: =FD04/=FD03/=FD2015 21:59
To: Christer Holmberg<mailto:christer.holmberg@ericsson.com>
Cc: Simon Perreault<mailto:sperreault@jive.com>; rtcweb@ietf.org<mailto:rtc=
web@ietf.org>
Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples

On Wed, Mar 4, 2015 at 2:54 PM, Christer Holmberg <christer.holmberg@ericss=
on.com<mailto:christer.holmberg@ericsson.com>> wrote:
Didn't we last week agree that, if the underlying transport changes, the DT=
LS connection MUST be re-established?

Jumping from one candidate to another is a transport change, isn't it?


In cases where ICE is used jump from on candidate to another should not con=
stitute a transport change. A change in ICE ufrag should constitute the tra=
nsport change. All ICE candidates are a single virtual transport channel. W=
ithout this a lot of ICE setup scenarios, such as rapid nomination, break d=
own. Even changes in the c=3D line address, m=3Dline port port, or list of =
candidates is not a transport change if ufrag stays the same.

In cases where ICE is not used, changes of address in c=3D line or port in =
m=3D line are a transport change.
_____________
Roman Shpount


--_000_7594FB04B1934943A5C02806D1A2204B1D726BFEESESSMB209erics_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dwindows-1=
256">
</head>
<body>
<div>
<div style=3D"font-family:Calibri,sans-serif; font-size:11pt">Correction. Y=
ou are not jumping to TLS, if you use framing for DTLS when TCP is used.<br=
>
<br>
Regards,<br>
<br>
Christer <br>
<br>
Sent from my Windows Phone</div>
</div>
<div dir=3D"ltr">
<hr>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">From:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:christer.holmberg@ericsson.com">Christer Holmberg</a></span><b=
r>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Sent:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">=FD04=
/=FD03/=FD2015 22:05</span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">To:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:roman@telurix.com">Roman Shpount</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Cc:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Subject:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">Re: [=
rtcweb] DTLS, DTLS-SRTP, and 5-tuples</span><br>
<br>
</div>
<div>
<div>
<div style=3D"font-family:Calibri,sans-serif; font-size:11pt">Hi,<br>
<br>
What if you jump from an UDP candidate to a TCP candidate? At the same time=
 you would be jumping from DTLS to TLS (even for SRTP, where DTLS is only u=
sed for key management).<br>
<br>
Regards,<br>
<br>
Christer<br>
<br>
Sent from my Windows Phone</div>
</div>
<div dir=3D"ltr">
<hr>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">From:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:roman@telurix.com">Roman Shpount</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Sent:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">=FD04=
/=FD03/=FD2015 21:59</span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">To:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:christer.holmberg@ericsson.com">Christer Holmberg</a></span><b=
r>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Cc:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:sperreault@jive.com">Simon Perreault</a>;
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Subject:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">Re: [=
rtcweb] DTLS, DTLS-SRTP, and 5-tuples</span><br>
<br>
</div>
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">On Wed, Mar 4, 2015 at 2:54 PM, Christer Holmber=
g <span dir=3D"ltr">
&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chr=
ister.holmberg@ericsson.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex; border=
-left-width:1px; border-left-color:rgb(204,204,204); border-left-style:soli=
d; padding-left:1ex">
<div>
<div>
<div>
<div style=3D"font-family:Calibri,sans-serif; font-size:11pt">Didn't we las=
t week agree that, if the underlying transport changes, the DTLS connection=
 MUST be re-established?<br>
<br>
Jumping from one candidate to another is a transport change, isn't it?<br>
<br>
</div>
</div>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>In cases where ICE is used jump from on candidate to another should no=
t constitute a transport change. A change in ICE ufrag should constitute th=
e transport change. All ICE candidates are a single virtual transport chann=
el. Without this a lot of ICE setup
 scenarios, such as rapid nomination, break down. Even changes in the c=3D =
line address, m=3Dline port port, or list of candidates is not a transport =
change if ufrag stays the same.</div>
<div><br>
</div>
<div>In cases where ICE is not used, changes of address in c=3D line or por=
t in m=3D line are a transport change.</div>
<div>
<div class=3D"gmail_signature">_____________<br>
Roman Shpount</div>
</div>
<div>&nbsp;</div>
</div>
</div>
</div>
</div>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1D726BFEESESSMB209erics_--


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From: Eric Rescorla <ekr@rtfm.com>
Date: Wed, 4 Mar 2015 12:07:22 -0800
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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On Wed, Mar 4, 2015 at 12:04 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Hi,
>
> What if you jump from an UDP candidate to a TCP candidate? At the same
> time you would be jumping from DTLS to TLS (even for SRTP, where DTLS is
> only used for key management).
>

No. You do DTLS even with TCP.

-Ekr


>
> Regards,
>
> Christer
>
> Sent from my Windows Phone
>  ------------------------------
> From: Roman Shpount <roman@telurix.com>
> Sent: =E2=80=8E04/=E2=80=8E03/=E2=80=8E2015 21:59
> To: Christer Holmberg <christer.holmberg@ericsson.com>
> Cc: Simon Perreault <sperreault@jive.com>; rtcweb@ietf.org
> Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
>
>   On Wed, Mar 4, 2015 at 2:54 PM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
>
>>   Didn't we last week agree that, if the underlying transport changes,
>> the DTLS connection MUST be re-established?
>>
>> Jumping from one candidate to another is a transport change, isn't it?
>>
>>
>  In cases where ICE is used jump from on candidate to another should not
> constitute a transport change. A change in ICE ufrag should constitute th=
e
> transport change. All ICE candidates are a single virtual transport
> channel. Without this a lot of ICE setup scenarios, such as rapid
> nomination, break down. Even changes in the c=3D line address, m=3Dline p=
ort
> port, or list of candidates is not a transport change if ufrag stays the
> same.
>
>  In cases where ICE is not used, changes of address in c=3D line or port =
in
> m=3D line are a transport change.
>  _____________
> Roman Shpount
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">=
On Wed, Mar 4, 2015 at 12:04 PM, Christer Holmberg <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.=
holmberg@ericsson.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_q=
uote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1e=
x">



<div>
<div>
<div style=3D"font-family:Calibri,sans-serif;font-size:11pt">Hi,<br>
<br>
What if you jump from an UDP candidate to a TCP candidate? At the same time=
 you would be jumping from DTLS to TLS (even for SRTP, where DTLS is only u=
sed for key management).</div></div></div></blockquote><div><br></div><div>=
No. You do DTLS even with TCP.</div><div><br></div><div>-Ekr</div><div>=C2=
=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;borde=
r-left:1px #ccc solid;padding-left:1ex"><div><div><div style=3D"font-family=
:Calibri,sans-serif;font-size:11pt"><span class=3D""><br>
Regards,<br>
<br>
Christer<br>
<br>
Sent from my Windows Phone</span></div>
</div>
<div dir=3D"ltr">
<hr>
<span style=3D"font-family:Calibri,sans-serif;font-size:11pt;font-weight:bo=
ld">From:
</span><span style=3D"font-family:Calibri,sans-serif;font-size:11pt"><a hre=
f=3D"mailto:roman@telurix.com" target=3D"_blank">Roman Shpount</a></span><b=
r>
<span style=3D"font-family:Calibri,sans-serif;font-size:11pt;font-weight:bo=
ld">Sent:
</span><span style=3D"font-family:Calibri,sans-serif;font-size:11pt">=E2=80=
=8E04/=E2=80=8E03/=E2=80=8E2015 21:59</span><br>
<span style=3D"font-family:Calibri,sans-serif;font-size:11pt;font-weight:bo=
ld">To:
</span><span style=3D"font-family:Calibri,sans-serif;font-size:11pt"><a hre=
f=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">Christer Holm=
berg</a></span><br>
<span style=3D"font-family:Calibri,sans-serif;font-size:11pt;font-weight:bo=
ld">Cc:
</span><span style=3D"font-family:Calibri,sans-serif;font-size:11pt"><a hre=
f=3D"mailto:sperreault@jive.com" target=3D"_blank">Simon Perreault</a>;
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a></s=
pan><span class=3D""><br>
<span style=3D"font-family:Calibri,sans-serif;font-size:11pt;font-weight:bo=
ld">Subject:
</span><span style=3D"font-family:Calibri,sans-serif;font-size:11pt">Re: [r=
tcweb] DTLS, DTLS-SRTP, and 5-tuples</span><br>
<br>
</span></div><div><div class=3D"h5">
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">On Wed, Mar 4, 2015 at 2:54 PM, Christer Holmber=
g <span dir=3D"ltr">
&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chr=
ister.holmberg@ericsson.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<div>
<div>
<div>
<div style=3D"font-family:Calibri,sans-serif;font-size:11pt">Didn&#39;t we =
last week agree that, if the underlying transport changes, the DTLS connect=
ion MUST be re-established?<br>
<br>
Jumping from one candidate to another is a transport change, isn&#39;t it?<=
br>
<br>
</div>
</div>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>In cases where ICE is used jump from on candidate to another should no=
t constitute a transport change. A change in ICE ufrag should constitute th=
e transport change. All ICE candidates are a single virtual transport chann=
el. Without this a lot of ICE setup
 scenarios, such as rapid nomination, break down. Even changes in the c=3D =
line address, m=3Dline port port, or list of candidates is not a transport =
change if ufrag stays the same.</div>
<div><br>
</div>
<div>In cases where ICE is not used, changes of address in c=3D line or por=
t in m=3D line are a transport change.</div>
<div>
<div>_____________<br>
Roman Shpount</div>
</div>
<div>=C2=A0</div>
</div>
</div>
</div>
</div>
</div></div></div>

<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

--f46d0418257c3709b305107c024b--


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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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--089e01229aaab49ffe05107c9d28
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On Wed, Mar 4, 2015 at 3:07 PM, Eric Rescorla <ekr@rtfm.com> wrote:

>
> On Wed, Mar 4, 2015 at 12:04 PM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
>
>>  What if you jump from an UDP candidate to a TCP candidate? At the same
>> time you would be jumping from DTLS to TLS (even for SRTP, where DTLS is
>> only used for key management).
>>
>
> No. You do DTLS even with TCP.
>

And this is still continues to be the same logical channel, i.e. this does
not constitute a transport change.
_____________
Roman Shpount

--089e01229aaab49ffe05107c9d28
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On W=
ed, Mar 4, 2015 at 3:07 PM, Eric Rescorla <span dir=3D"ltr">&lt;<a href=3D"=
mailto:ekr@rtfm.com" target=3D"_blank">ekr@rtfm.com</a>&gt;</span> wrote:<b=
r><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;borde=
r-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid=
;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><br><div cla=
ss=3D"gmail_quote"><span class=3D"">On Wed, Mar 4, 2015 at 12:04 PM, Christ=
er Holmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:christer.holmberg@erics=
son.com" target=3D"_blank">christer.holmberg@ericsson.com</a>&gt;</span> wr=
ote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex=
;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style=
:solid;padding-left:1ex">



<div>
<div>
<div style=3D"font-family:Calibri,sans-serif;font-size:11pt">What if you ju=
mp from an UDP candidate to a TCP candidate? At the same time you would be =
jumping from DTLS to TLS (even for SRTP, where DTLS is only used for key ma=
nagement).</div></div></div></blockquote><div><br></div></span><div>No. You=
 do DTLS even with TCP.</div></div></div></div></blockquote><div>=C2=A0</di=
v><div>And this is still continues to be the same logical channel, i.e. thi=
s does not constitute a transport change.</div><div><div class=3D"gmail_sig=
nature">_____________<br>Roman Shpount</div></div><div>=C2=A0</div></div></=
div></div>

--089e01229aaab49ffe05107c9d28--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Roman Shpount <roman@telurix.com>, Eric Rescorla <ekr@rtfm.com>
Thread-Topic: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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--_000_7594FB04B1934943A5C02806D1A2204B1D726DC1ESESSMB209erics_
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Hi,

I am not objecting to the idea, but as far as I know this "logical channel"=
 concept is not defined anywhere, so before we can start referring to it it=
 needs to be anchored somewhere on DTLS level.

Regards,

Christer

Sent from my Windows Phone
________________________________
From: Roman Shpount<mailto:roman@telurix.com>
Sent: =FD04/=FD03/=FD2015 22:51
To: Eric Rescorla<mailto:ekr@rtfm.com>
Cc: Christer Holmberg<mailto:christer.holmberg@ericsson.com>; rtcweb@ietf.o=
rg<mailto:rtcweb@ietf.org>
Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples

On Wed, Mar 4, 2015 at 3:07 PM, Eric Rescorla <ekr@rtfm.com<mailto:ekr@rtfm=
.com>> wrote:

On Wed, Mar 4, 2015 at 12:04 PM, Christer Holmberg <christer.holmberg@erics=
son.com<mailto:christer.holmberg@ericsson.com>> wrote:
What if you jump from an UDP candidate to a TCP candidate? At the same time=
 you would be jumping from DTLS to TLS (even for SRTP, where DTLS is only u=
sed for key management).

No. You do DTLS even with TCP.

And this is still continues to be the same logical channel, i.e. this does =
not constitute a transport change.
_____________
Roman Shpount


--_000_7594FB04B1934943A5C02806D1A2204B1D726DC1ESESSMB209erics_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dwindows-1=
256">
</head>
<body>
<div>
<div style=3D"font-family:Calibri,sans-serif; font-size:11pt">Hi,<br>
<br>
I am not objecting to the idea, but as far as I know this &quot;logical cha=
nnel&quot; concept is not defined anywhere, so before we can start referrin=
g to it it needs to be anchored somewhere on DTLS level.<br>
<br>
Regards,<br>
<br>
Christer<br>
<br>
Sent from my Windows Phone</div>
</div>
<div dir=3D"ltr">
<hr>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">From:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:roman@telurix.com">Roman Shpount</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Sent:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">=FD04=
/=FD03/=FD2015 22:51</span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">To:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:ekr@rtfm.com">Eric Rescorla</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Cc:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:christer.holmberg@ericsson.com">Christer Holmberg</a>;
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Subject:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">Re: [=
rtcweb] DTLS, DTLS-SRTP, and 5-tuples</span><br>
<br>
</div>
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">On Wed, Mar 4, 2015 at 3:07 PM, Eric Rescorla <s=
pan dir=3D"ltr">
&lt;<a href=3D"mailto:ekr@rtfm.com" target=3D"_blank">ekr@rtfm.com</a>&gt;<=
/span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex; border=
-left-width:1px; border-left-color:rgb(204,204,204); border-left-style:soli=
d; padding-left:1ex">
<div dir=3D"ltr">
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote"><span class=3D"">On Wed, Mar 4, 2015 at 12:04 PM=
, Christer Holmberg
<span dir=3D"ltr">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" tar=
get=3D"_blank">christer.holmberg@ericsson.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex; border=
-left-width:1px; border-left-color:rgb(204,204,204); border-left-style:soli=
d; padding-left:1ex">
<div>
<div>
<div style=3D"font-family:Calibri,sans-serif; font-size:11pt">What if you j=
ump from an UDP candidate to a TCP candidate? At the same time you would be=
 jumping from DTLS to TLS (even for SRTP, where DTLS is only used for key m=
anagement).</div>
</div>
</div>
</blockquote>
<div><br>
</div>
</span>
<div>No. You do DTLS even with TCP.</div>
</div>
</div>
</div>
</blockquote>
<div>&nbsp;</div>
<div>And this is still continues to be the same logical channel, i.e. this =
does not constitute a transport change.</div>
<div>
<div class=3D"gmail_signature">_____________<br>
Roman Shpount</div>
</div>
<div>&nbsp;</div>
</div>
</div>
</div>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1D726DC1ESESSMB209erics_--


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 5 Mar 2015 01:44:01 +0100
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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2015-03-04 23:05 GMT+01:00 Christer Holmberg <christer.holmberg@ericsson.co=
m>:
> I am not objecting to the idea, but as far as I know this "logical channe=
l"
> concept is not defined anywhere, so before we can start referring to it i=
t
> needs to be anchored somewhere on DTLS level.

Take into account that when aggressive ICE nomination is being done, a
peer sends multiple STUN requests with USE-CANDIDATE at the same time
and DTLS ClientHello after each of them. At the end this means that
the receiver must be ready to receive DTLS packets via different
5-tuples at the same time, all of them belonging to the same DTLS
association/connection.

So stating that "if the underlying transport changes the DTLS
connection MUST be re-established" seems a unfortunate simplification
to me.

Said that, I fully agree with you. This kind of "virtual transport"
created by ICE must be defined somewhere because existing specs (for
example DTLS-SRTP RFC) do not mention ICE at all.

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


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From: Christer Holmberg <christer.holmberg@ericsson.com>
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Thread-Topic: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 5 Mar 2015 10:52:44 +0100
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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2015-03-05 7:58 GMT+01:00 Christer Holmberg <christer.holmberg@ericsson.com=
>:
> As far as I understand, sending of multiple ClientHellos, on different
> 5-tuples, would require a change to core DTLS (not only DTLS-SRTP), and w=
ho
> knows how long such work would take. Do we really want to add such
> dependency to our deliveries at this point?

Sorry, I think I didn't explain well.

It is not about multiple DTLS connections, but about the fact that
DTLS packets belonging to the *same* DTLS connection/association can
be carried over different 5-tuples. This may happen during agressive
ICE nomination in which media (so the DTLS ClientHello) follows the
USE-CANDIDATE Binding request without even waiting for a response.


> In addition, do we expect existing DTLS implementations to support this?

There is nothing new to implement in DTLS libraries. It is just the
the DTLS user must not assume a single and fixed 5-tuple


> If one doesn't want to re-establish DTLS when jumping between candidates,
> doesn't it work by creating separate DTLS connections for each candidate?

This subject has been heavily discussed time ago:

http://www.ietf.org/mail-archive/web/mmusic/current/msg13637.html

No, there MUST NOT be separate DTLS connection for each candidate-pair.


Regards.



--=20
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<ibc@aliax.net>


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--90e6ba1ef2ea50071a051087c63d
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On Wed, Mar 4, 2015 at 7:44 PM, I=C3=B1aki Baz Castillo <ibc@aliax.net> wro=
te:

> Take into account that when aggressive ICE nomination is being done, a
> peer sends multiple STUN requests with USE-CANDIDATE at the same time
> and DTLS ClientHello after each of them. At the end this means that
> the receiver must be ready to receive DTLS packets via different
> 5-tuples at the same time, all of them belonging to the same DTLS
> association/connection.
>

I am not sure this is entirely correct.  My assumption was that during
aggressive ICE nomination, an end point will send DTLS ClientHello over the
first candidate pair where ICE connectivity check succeeds. The end point
should be ready to receive DTLS messages over any ICE candidate pair. If
end point does not receive response to DTLS ClientHello it should
re-transmit ClientHello over the highest priority ICE candidate pair at the
time of the re-transmission. I do not think there is any need to send
multiple DTLS ClientHello messages at the same time over multiple candidate
pairs. In other words, an end point should treat ICE as a single virtual
channel, where data is transmitted over the highest priority ICE candidate
pair for which the connectivity check succeeded and process data received
from any non-pruned ICE candidate pair. There is no need to trombone the
data over all the candidate pairs when it is being sent out. This should
fit nicely with existing DTLS implementations.
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On W=
ed, Mar 4, 2015 at 7:44 PM, I=C3=B1aki Baz Castillo <span dir=3D"ltr">&lt;<=
a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt;</sp=
an> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px=
 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left=
-style:solid;padding-left:1ex">Take into account that when aggressive ICE n=
omination is being done, a<br>
peer sends multiple STUN requests with USE-CANDIDATE at the same time<br>
and DTLS ClientHello after each of them. At the end this means that<br>
the receiver must be ready to receive DTLS packets via different<br>
5-tuples at the same time, all of them belonging to the same DTLS<br>
association/connection.<br></blockquote><div><br></div><div>I am not sure t=
his is entirely correct.=C2=A0 My assumption was that during aggressive ICE=
 nomination, an end point will send DTLS ClientHello over the first candida=
te pair where ICE connectivity check succeeds. The end point should be read=
y to receive DTLS messages over any ICE candidate pair. If end point does n=
ot receive response to DTLS ClientHello it should re-transmit ClientHello o=
ver the highest priority ICE candidate pair at the time of the re-transmiss=
ion. I do not think there is any need to send multiple DTLS ClientHello mes=
sages at the same time over multiple candidate pairs. In other words, an en=
d point should treat ICE as a single virtual channel, where data is transmi=
tted over the highest priority ICE candidate pair for which the connectivit=
y check succeeded and process data received from any non-pruned ICE candida=
te pair. There is no need to trombone the data over all the candidate pairs=
 when it is being sent out. This should fit nicely with existing DTLS imple=
mentations.</div><div><div class=3D"gmail_signature">_____________<br>Roman=
 Shpount</div></div><div>=C2=A0</div></div></div></div>

--90e6ba1ef2ea50071a051087c63d--


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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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On Thu, Mar 5, 2015 at 4:52 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> wro=
te:

> 2015-03-05 7:58 GMT+01:00 Christer Holmberg <
> christer.holmberg@ericsson.com>:
> > As far as I understand, sending of multiple ClientHellos, on different
> > 5-tuples, would require a change to core DTLS (not only DTLS-SRTP), and
> who
> > knows how long such work would take. Do we really want to add such
> > dependency to our deliveries at this point?
>
> Sorry, I think I didn't explain well.
>
> It is not about multiple DTLS connections, but about the fact that
> DTLS packets belonging to the *same* DTLS connection/association can
> be carried over different 5-tuples. This may happen during agressive
> ICE nomination in which media (so the DTLS ClientHello) follows the
> USE-CANDIDATE Binding request without even waiting for a response.
>

When was it agreed that DTLS ClientHello can be sent before the
connectivity check succeeded? I understand that this will increase the
connection setup time, but I though that no data should be sent before the
connectivity check response (consent) from the remote party.

> In addition, do we expect existing DTLS implementations to support this?
>
> There is nothing new to implement in DTLS libraries. It is just the
> the DTLS user must not assume a single and fixed 5-tuple
>
>
> > If one doesn't want to re-establish DTLS when jumping between candidate=
s,
> > doesn't it work by creating separate DTLS connections for each candidat=
e?
>
> This subject has been heavily discussed time ago:
>
> http://www.ietf.org/mail-archive/web/mmusic/current/msg13637.html
>
> No, there MUST NOT be separate DTLS connection for each candidate-pair.
>

Agreed
_____________
Roman Shpount

--001a1141ca9ef6a4ff051087d5fe
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On T=
hu, Mar 5, 2015 at 4:52 AM, I=C3=B1aki Baz Castillo <span dir=3D"ltr">&lt;<=
a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt;</sp=
an> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px=
 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left=
-style:solid;padding-left:1ex"><span class=3D"">2015-03-05 7:58 GMT+01:00 C=
hrister Holmberg &lt;<a href=3D"mailto:christer.holmberg@ericsson.com">chri=
ster.holmberg@ericsson.com</a>&gt;:<br>
&gt; As far as I understand, sending of multiple ClientHellos, on different=
<br>
&gt; 5-tuples, would require a change to core DTLS (not only DTLS-SRTP), an=
d who<br>
&gt; knows how long such work would take. Do we really want to add such<br>
&gt; dependency to our deliveries at this point?<br>
<br>
</span>Sorry, I think I didn&#39;t explain well.<br>
<br>
It is not about multiple DTLS connections, but about the fact that<br>
DTLS packets belonging to the *same* DTLS connection/association can<br>
be carried over different 5-tuples. This may happen during agressive<br>
ICE nomination in which media (so the DTLS ClientHello) follows the<br>
USE-CANDIDATE Binding request without even waiting for a response.<br></blo=
ckquote><div><br></div><div>When was it agreed that DTLS ClientHello can be=
 sent before the connectivity check succeeded? I understand that this will =
increase the connection setup time, but I though that no data should be sen=
t before the connectivity check response (consent) from the remote party.</=
div><div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0p=
x 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border=
-left-style:solid;padding-left:1ex"><span class=3D"">
&gt; In addition, do we expect existing DTLS implementations to support thi=
s?<br>
<br>
</span>There is nothing new to implement in DTLS libraries. It is just the<=
br>
the DTLS user must not assume a single and fixed 5-tuple<br>
<span class=3D""><br>
<br>
&gt; If one doesn&#39;t want to re-establish DTLS when jumping between cand=
idates,<br>
&gt; doesn&#39;t it work by creating separate DTLS connections for each can=
didate?<br>
<br>
</span>This subject has been heavily discussed time ago:<br>
<br>
<a href=3D"http://www.ietf.org/mail-archive/web/mmusic/current/msg13637.htm=
l" target=3D"_blank">http://www.ietf.org/mail-archive/web/mmusic/current/ms=
g13637.html</a><br>
<br>
No, there MUST NOT be separate DTLS connection for each candidate-pair.<br>=
</blockquote><div><br></div><div>Agreed</div><div><div class=3D"gmail_signa=
ture">_____________<br>Roman Shpount</div></div><div>=C2=A0</div></div></di=
v></div>

--001a1141ca9ef6a4ff051087d5fe--


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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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From nobody Thu Mar  5 02:44:53 2015
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Date: Thu, 5 Mar 2015 11:44:25 +0100
From: Lorenzo Miniero <lorenzo@meetecho.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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On Thu, 5 Mar 2015 10:30:31 +0000
Christer Holmberg <christer.holmberg@ericsson.com> wrote:

> Hi,
>=20
> >When was it agreed that DTLS ClientHello can be sent before the
> >connectivity check succeeded? I understand >that this will increase
> >the connection setup time, but I though that no data should be sent
> >before the >connectivity check response (consent) from the remote
> >party.
>=20
> And, if the ClientHello reaches the server BEFORE the connectivity
> check STUN, how does the server know that it is associated with a
> given =E2=80=9Cvirtual connection=E2=80=9D?
>=20
> Regards,
>=20
> Christer
>=20


I guess sending a ClientHello before the ICE process has been completed
would be pointless, as you wouldn't be sure where to send the message,
and even if it reached the right destination it might be discarded. The
"virtual connection" is only available when you have at least a working
pair on both sides.

Lorenzo


From nobody Thu Mar  5 02:46:54 2015
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 5 Mar 2015 11:46:30 +0100
Message-ID: <CALiegfncTzNKgia_zDVkYZn3WHmWKZ7B7Qz9Yc7QmSmLVT9Drg@mail.gmail.com>
To: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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2015-03-05 11:14 GMT+01:00 Roman Shpount <roman@telurix.com>:
>> It is not about multiple DTLS connections, but about the fact that
>> DTLS packets belonging to the *same* DTLS connection/association can
>> be carried over different 5-tuples. This may happen during agressive
>> ICE nomination in which media (so the DTLS ClientHello) follows the
>> USE-CANDIDATE Binding request without even waiting for a response.
>
>
> When was it agreed that DTLS ClientHello can be sent before the connectiv=
ity
> check succeeded? I understand that this will increase the connection setu=
p
> time, but I though that no data should be sent before the connectivity ch=
eck
> response (consent) from the remote party.

I don't remember at 100%, but AFAIR that happens (may be I'm wrong
sorry, I checked it in deep long time ago but cannot remember right
now).



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


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 5 Mar 2015 11:53:02 +0100
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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2015-03-05 11:30 GMT+01:00 Christer Holmberg <christer.holmberg@ericsson.co=
m>:
>>When was it agreed that DTLS ClientHello can be sent before the
>> connectivity check succeeded? I understand >that this will increase the
>> connection setup time, but I though that no data should be sent before t=
he
>> >connectivity check response (consent) from the remote party.
>
>
>
> And, if the ClientHello reaches the server BEFORE the connectivity check
> STUN, how does the server know that it is associated with a given =E2=80=
=9Cvirtual
> connection=E2=80=9D?

It just ignores it. Client retransmissions will fix that.

Anyhow, I'm sorry but I no longer remember whether DTLS (which is just
media from the ICE point of view) can be sent before receiving the
STUN response or not. From the point of view of the peer performing
aggressive ICE nomination, it is sending USE-CANDIDATE Binding
requests so I can understand that it assumes a "valid-pair" (even if
it's not replied/confirmed yet by the remote) so media can begin (and
DTLS is just media).

I have to re-check.


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


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 5 Mar 2015 14:06:29 +0100
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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2015-03-05 14:01 GMT+01:00 Makaraju, Maridi Raju (Raju)
<Raju.Makaraju@alcatel-lucent.com>:
>> Anyhow, I'm sorry but I no longer remember whether DTLS (which is just
>> media from the ICE point of view) can be sent before receiving the
>> STUN response or not. From the point of view of the peer performing
>> aggressive ICE nomination, it is sending USE-CANDIDATE Binding
>> requests so I can understand that it assumes a "valid-pair" (even if
>> it's not replied/confirmed yet by the remote) so media can begin (and
>> DTLS is just media).
>
> <Raju>
> Aggressive nomination is to avoid 2nd round of checks with USE-CANDIDATE,=
 but not to send media before the nomination is complete. Per [1], nominati=
on is complete only when checks on all valid pairs are complete. A pair bec=
omes "valid" only when you get response. That means, you can't send media (=
DTLS ClientHello) before you get STUN response.

Thanks. I remember now what it may happens:

- A sends USE-CANDIDATE requests in parallel.
- A receives the ok response from one of them.
- A sends DTLS ClientHello.
- A receives the ok response with higher priority from another pair.
- A then continues sending media (maybe remaining DTLS stuff or RTP)
for that pair.

At the end DTLS packets can be sent over any valid and confirmed
candidate-pair. That's why we need to think about it as a "virtual
transport" and forget the 5-tuple when it comes to ICE.


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


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 5 Mar 2015 14:44:18 +0100
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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2015-03-05 14:26 GMT+01:00 Makaraju, Maridi Raju (Raju)
<Raju.Makaraju@alcatel-lucent.com>:
>>
>> - A sends USE-CANDIDATE requests in parallel.
>> - A receives the ok response from one of them.
>> - A sends DTLS ClientHello.
>
> <Raju>
> This is not allowed by http://tools.ietf.org/html/rfc5245#section-8.1.1.2=
 , unless webrtc overrides it.
> A can only send DTLS once final nomination, which happens after all valid=
 pairs are found, is done.
> </Raju>


   However, because the agent included the USE-CANDIDATE
   attribute in all of its checks, another check may yet
   complete, causing another valid pair to have its nominated flag set.
   ICE always selects the highest-priority nominated candidate pair from
   the valid list as the one used for media.  Consequently, the selected
   pair may actually change briefly as ICE checks complete, resulting in
   a set of transient selections until it stabilizes.


Also consider the case in which the ICE-CONTROLLING agent is not the
DTLS client. This is:

- A sends USE-CANDIDATE requests in parallel.
- B replies OK to the first one.
- B decides that it is the selected-pair (common in ICE-Lite servers).
- B sends DTLS ClientHello.
- A replies DTLS ServerHello etc.
- A continues sending USE-CANDIDATE requests.
- After a while another pair may be selected but "media" already started.


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


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To: =?utf-8?B?ScOxYWtpIEJheiBDYXN0aWxsbw==?= <ibc@aliax.net>, "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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On Thu, Mar 5, 2015 at 8:26 AM, Makaraju, Maridi Raju (Raju) <
Raju.Makaraju@alcatel-lucent.com> wrote:

> This is not allowed by http://tools.ietf.org/html/rfc5245#section-8.1.1.2
> , unless webrtc overrides it.
> A can only send DTLS once final nomination, which happens after all valid
> pairs are found, is done.
>

I think you are misreading
http://tools.ietf.org/html/rfc5245#section-8.1.1.2: An end point can start
sending media once a valid pair is discovered for each component (which is
actually one component in case of bundle, one component per m= line in case
of no bundle and rtcp-mux, and two components per m= line in case of
neither). New valid pairs can still be found when media is being sent and
media should be switched to the valid pair with the highest priority. When
all the pairs are checked the final nomination request is sent, but the
media will start flowing way before that. Alternative would be almost
unusable since pair of two private IPs for two end-point each behind NAT
will never get nominated and it will take a long time to time out,
preventing media from flowing long after call is connected.
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On T=
hu, Mar 5, 2015 at 8:26 AM, Makaraju, Maridi Raju (Raju) <span dir=3D"ltr">=
&lt;<a href=3D"mailto:Raju.Makaraju@alcatel-lucent.com" target=3D"_blank">R=
aju.Makaraju@alcatel-lucent.com</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;bo=
rder-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">=
This is not allowed by <a href=3D"http://tools.ietf.org/html/rfc5245#sectio=
n-8.1.1.2" target=3D"_blank">http://tools.ietf.org/html/rfc5245#section-8.1=
.1.2</a> , unless webrtc overrides it.<br>
A can only send DTLS once final nomination, which happens after all valid p=
airs are found, is done.<br></blockquote><div><br></div><div>I think you ar=
e misreading=C2=A0<a href=3D"http://tools.ietf.org/html/rfc5245#section-8.1=
.1.2" target=3D"_blank">http://tools.ietf.org/html/rfc5245#section-8.1.1.2<=
/a>: An end point can start sending media once a valid pair is discovered f=
or each component (which is actually one component in case of bundle, one c=
omponent per m=3D line in case of no bundle and rtcp-mux, and two component=
s per m=3D line in case of neither). New valid pairs can still be found whe=
n media is being sent and media should be switched to the valid pair with t=
he highest priority. When all the pairs are checked the final nomination re=
quest is sent, but the media will start flowing way before that. Alternativ=
e would be almost unusable since pair of two private IPs for two end-point =
each behind NAT will never get nominated and it will take a long time to ti=
me out, preventing media from flowing long after call is connected.</div><d=
iv><div class=3D"gmail_signature">_____________<br>Roman Shpount</div></div=
><div>=C2=A0</div></div></div></div>

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Date: Thu, 5 Mar 2015 09:03:41 -0500
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From: Roman Shpount <roman@telurix.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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On Thu, Mar 5, 2015 at 8:44 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> >- A sends USE-CANDIDATE requests in parallel.
> >- A receives the ok response from one of them.
> >- A sends DTLS ClientHello.
> >- A receives the ok response with higher priority from another pair.
> >- A then continues sending media (maybe remaining DTLS stuff or RTP) for
> that pair.
>
> Just to clarify: when you say "continues sending", you are NOT saying that
> A "starts over" by sending a new initial ClientHello on the new pair -
> instead A only switches to the other pair and continues with normal DTLS
> setup procedures?
>
> ...which means that both the client and server may receive DTLS messages,
> associated with the same DTLS connection setup, on different 5-tuples?
>

End point should not "start over". End point should continue the normal
DTLS setup over a new pair. This does mean that the same DTLS association
can go over different 5-tuples over time. With some of the new proposals,
such as continuous nomination, DTLS association can be switching the
underlying 5-tuple at any time while the connection is active, not just
during the initial setup. As far as DTLS is concerned, all 5-tuples setup
with the same ICE ufrag pair are the same transport and all the packets
received over any of those 5-tuples should be processed by the same DTLS
association.
_____________
Roman Shpount

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br clear=3D"all"><div><div=
 class=3D"gmail_signature">On Thu, Mar 5, 2015 at 8:44 AM, Christer Holmber=
g <span dir=3D"ltr">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" t=
arget=3D"_blank">christer.holmberg@ericsson.com</a>&gt;</span> wrote:<br></=
div></div><div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" styl=
e=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(2=
04,204,204);border-left-style:solid;padding-left:1ex"><span class=3D"">&gt;=
- A sends USE-CANDIDATE requests in parallel.<br>
&gt;- A receives the ok response from one of them.<br>
&gt;- A sends DTLS ClientHello.<br>
&gt;- A receives the ok response with higher priority from another pair.<br=
>
&gt;- A then continues sending media (maybe remaining DTLS stuff or RTP) fo=
r that pair.<br>
<br>
</span>Just to clarify: when you say &quot;continues sending&quot;, you are=
 NOT saying that A &quot;starts over&quot; by sending a new initial ClientH=
ello on the new pair - instead A only switches to the other pair and contin=
ues with normal DTLS setup procedures?<br>
<br>
...which means that both the client and server may receive DTLS messages, a=
ssociated with the same DTLS connection setup, on different 5-tuples?<br></=
blockquote><div><br></div><div>End point should not &quot;start over&quot;.=
 End point should continue the normal DTLS setup over a new pair. This does=
 mean that the same DTLS association can go over different 5-tuples over ti=
me. With some of the new proposals, such as continuous nomination, DTLS ass=
ociation can be switching the underlying 5-tuple at any time while the conn=
ection is active, not just during the initial setup. As far as DTLS is conc=
erned, all 5-tuples setup with the same ICE ufrag pair are the same transpo=
rt and all the packets received over any of those 5-tuples should be proces=
sed by the same DTLS association.</div><div><div class=3D"gmail_signature">=
_____________<br>Roman Shpount</div></div><br><div>=C2=A0</div></div></div>=
</div>

--001a11343b920ad13c05108b091f--


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 5 Mar 2015 15:32:54 +0100
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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2015-03-05 14:44 GMT+01:00 Christer Holmberg <christer.holmberg@ericsson.co=
m>:
>>- A sends USE-CANDIDATE requests in parallel.
>>- A receives the ok response from one of them.
>>- A sends DTLS ClientHello.
>>- A receives the ok response with higher priority from another pair.
>>- A then continues sending media (maybe remaining DTLS stuff or RTP) for =
that pair.
>
> Just to clarify: when you say "continues sending", you are NOT saying tha=
t A "starts over" by sending a new initial ClientHello on the new pair - in=
stead A only switches to the other pair and continues with normal DTLS setu=
p procedures?
>
> ...which means that both the client and server may receive DTLS messages,=
 associated with the same DTLS connection setup, on different 5-tuples?

yep, and that happens.


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


From nobody Thu Mar  5 06:37:53 2015
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 5 Mar 2015 15:33:54 +0100
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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2015-03-05 14:58 GMT+01:00 Roman Shpount <roman@telurix.com>:
> I think you are misreading
> http://tools.ietf.org/html/rfc5245#section-8.1.1.2: An end point can star=
t
> sending media once a valid pair is discovered for each component (which i=
s
> actually one component in case of bundle, one component per m=3D line in =
case
> of no bundle and rtcp-mux, and two components per m=3D line in case of
> neither). New valid pairs can still be found when media is being sent and
> media should be switched to the valid pair with the highest priority. Whe=
n
> all the pairs are checked the final nomination request is sent, but the
> media will start flowing way before that. Alternative would be almost
> unusable since pair of two private IPs for two end-point each behind NAT
> will never get nominated and it will take a long time to time out,
> preventing media from flowing long after call is connected.


Exactly. And that is also stated in the RFC somewhere.


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


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From: Jonathan Lennox <jonathan@vidyo.com>
To: Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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--_000_CA5E97EE99F844D8B05BC9EFDED1A9BBvidyocom_
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On Mar 5, 2015, at 5:14 AM, Roman Shpount <roman@telurix.com<mailto:roman@t=
elurix.com>> wrote:

On Thu, Mar 5, 2015 at 4:52 AM, I=F1aki Baz Castillo <ibc@aliax.net<mailto:=
ibc@aliax.net>> wrote:
2015-03-05 7:58 GMT+01:00 Christer Holmberg <christer.holmberg@ericsson.com=
<mailto:christer.holmberg@ericsson.com>>:
> As far as I understand, sending of multiple ClientHellos, on different
> 5-tuples, would require a change to core DTLS (not only DTLS-SRTP), and w=
ho
> knows how long such work would take. Do we really want to add such
> dependency to our deliveries at this point?

Sorry, I think I didn't explain well.

It is not about multiple DTLS connections, but about the fact that
DTLS packets belonging to the *same* DTLS connection/association can
be carried over different 5-tuples. This may happen during agressive
ICE nomination in which media (so the DTLS ClientHello) follows the
USE-CANDIDATE Binding request without even waiting for a response.

When was it agreed that DTLS ClientHello can be sent before the connectivit=
y check succeeded? I understand that this will increase the connection setu=
p time, but I though that no data should be sent before the connectivity ch=
eck response (consent) from the remote party.

As I understand the current rough consensus, the endpoint has to have recei=
ved a successful response to a connectivity check on the 5-tuple, proving c=
onsent, but it doesn=92t have to have received (or sent) a check with a USE=
-CANDIDATE attribute in it yet.

This was the outcome of our discussion =93merging" regular and aggressive I=
CE nomination.


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252">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space;">
<br>
<div>
<div>On Mar 5, 2015, at 5:14 AM, Roman Shpount &lt;<a href=3D"mailto:roman@=
telurix.com">roman@telurix.com</a>&gt; wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">On Thu, Mar 5, 2015 at 4:52 AM, I=F1aki Baz Cast=
illo <span dir=3D"ltr">
&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt=
;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<span class=3D"">2015-03-05 7:58 GMT&#43;01:00 Christer Holmberg &lt;<a hre=
f=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson.com<=
/a>&gt;:<br>
&gt; As far as I understand, sending of multiple ClientHellos, on different=
<br>
&gt; 5-tuples, would require a change to core DTLS (not only DTLS-SRTP), an=
d who<br>
&gt; knows how long such work would take. Do we really want to add such<br>
&gt; dependency to our deliveries at this point?<br>
<br>
</span>Sorry, I think I didn't explain well.<br>
<br>
It is not about multiple DTLS connections, but about the fact that<br>
DTLS packets belonging to the *same* DTLS connection/association can<br>
be carried over different 5-tuples. This may happen during agressive<br>
ICE nomination in which media (so the DTLS ClientHello) follows the<br>
USE-CANDIDATE Binding request without even waiting for a response.<br>
</blockquote>
<div><br>
</div>
<div>When was it agreed that DTLS ClientHello can be sent before the connec=
tivity check succeeded? I understand that this will increase the connection=
 setup time, but I though that no data should be sent before the connectivi=
ty check response (consent) from
 the remote party.</div>
</div>
</div>
</div>
</blockquote>
<br>
</div>
<div>As I understand the current rough consensus, the endpoint has to have =
received a successful response to a connectivity check on the 5-tuple, prov=
ing consent, but it doesn=92t have to have received (or sent) a check with =
a USE-CANDIDATE attribute in it yet.</div>
<div><br>
</div>
<div>This was the outcome of our discussion =93merging&quot; regular and ag=
gressive ICE nomination.</div>
<br>
</body>
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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--_000_E1FE4C082A89A246A11D7F32A95A17828E7273B4US70UWXCHMBA02z_--


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--001a1135c8b86184a90510a03772
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New draft of RETURN is up:
http://tools.ietf.org/html/draft-schwartz-rtcweb-return-05

This draft contains a variety of clarifications and changes of emphasis,
and should hopefully now be easier to read and understand.

Thanks to Alan Johnston and John Yoakum for contributing new figures and
detailed text review for this revision.

--Ben

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<div dir=3D"ltr"><span style=3D"font-size:12.8000001907349px">New draft of =
RETURN is up:</span><div style=3D"font-size:12.8000001907349px"><a href=3D"=
http://tools.ietf.org/html/draft-schwartz-rtcweb-return-05" target=3D"_blan=
k">http://tools.ietf.org/html/draft-schwartz-rtcweb-return-05</a><br></div>=
<div style=3D"font-size:12.8000001907349px"><br></div><div style=3D"font-si=
ze:12.8000001907349px">This draft contains a variety of clarifications and =
changes of emphasis, and should hopefully now be easier to read and underst=
and.</div><div style=3D"font-size:12.8000001907349px"><br></div><div style=
=3D"font-size:12.8000001907349px">Thanks to Alan Johnston and John Yoakum f=
or contributing new figures and detailed text review for this revision.</di=
v><div style=3D"font-size:12.8000001907349px"><br></div><div style=3D"font-=
size:12.8000001907349px">--Ben</div></div>

--001a1135c8b86184a90510a03772--


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Subject: [rtcweb] Conditions for long-term permissions grants
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https://tools.ietf.org/html/draft-ietf-rtcweb-security-arch-10#section-5.2
requires
that JS be able to ask for short or long-term permissions grants:



   API Requirement:  The API MUST provide a mechanism for the requesting
      JS to indicate which of these forms of permissions it is
      requesting.  This allows the browser client to know what sort of
      user interface experience to provide to the user, including what
      permissions to request from the user and hence what to enforce
      later.  For instance, browsers might display a non-invasive door
      hanger ("some features of this site may not work..." when asking
      for long-term permissions) but a more invasive UI ("here is your
      own video") for single-call permissions.  The API MAY grant weaker
      permissions than the JS asked for if the user chooses to authorize
      only those permissions, but if it intends to grant stronger ones
      it SHOULD display the appropriate UI for those permissions and
      MUST clearly indicate what permissions are being requested.


However, there's no such affordance in the API and neither Chrome nor Firefox

comply with this. Currently:


- Chrome grants short-term permissions for HTTP and long-term permissions for

  HTTPS.

- Firefox by default grants short-term permissions but allows the user to select

  long-term permissions if the site is HTTPS.


It seems like some consistency would be nice here.


My personal view is that it would still be nice to require sites to
ask for persistent

permissions if they want them and that there should be a getUserMedia()

flag to indicate that. If people agree with me, I'll file an issue on the media

capture specification to add this affordance. However, if people think this

is wrong, we should remove this requirement in the security architecture

document.


-Ekr

--089e013d14c2b860740510b8df64
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<div dir=3D"ltr"><a href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-s=
ecurity-arch-10#section-5.2">https://tools.ietf.org/html/draft-ietf-rtcweb-=
security-arch-10#section-5.2</a> requires<div>that JS be able to ask for sh=
ort or long-term permissions grants:</div><div><pre class=3D"" style=3D"fon=
t-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br class=3D"=
"><font face=3D"arial, helvetica, sans-serif">
   API Requirement:  The API MUST provide a mechanism for the requesting
      JS to indicate which of these forms of permissions it is
      requesting.  This allows the browser client to know what sort of
      user interface experience to provide to the user, including what
      permissions to request from the user and hence what to enforce
      later.  For instance, browsers might display a non-invasive door
      hanger (&quot;some features of this site may not work...&quot; when a=
sking
      for long-term permissions) but a more invasive UI (&quot;here is your
      own video&quot;) for single-call permissions.  The API MAY grant weak=
er
      permissions than the JS asked for if the user chooses to authorize
      only those permissions, but if it intends to grant stronger ones
      it SHOULD display the appropriate UI for those permissions and
      MUST clearly indicate what permissions are being requested.</font></p=
re><pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;=
color:rgb(0,0,0)"><font face=3D"arial, helvetica, sans-serif"><br></font></=
pre><pre class=3D"" style=3D"margin-top:0px;margin-bottom:0px;color:rgb(0,0=
,0)"><font face=3D"arial, helvetica, sans-serif"><span style=3D"font-size:1=
3px">However, there&#39;s no such affordance in the API and neither Chrome =
nor Firefox</span></font></pre><pre class=3D"" style=3D"margin-top:0px;marg=
in-bottom:0px;color:rgb(0,0,0)"><font face=3D"arial, helvetica, sans-serif"=
><span style=3D"font-size:13px">comply with this. Currently:</span></font><=
/pre><pre class=3D"" style=3D"margin-top:0px;margin-bottom:0px;color:rgb(0,=
0,0)"><font face=3D"arial, helvetica, sans-serif"><span style=3D"font-size:=
13px"><br></span></font></pre><pre class=3D"" style=3D"margin-top:0px;margi=
n-bottom:0px;color:rgb(0,0,0)"><font face=3D"arial, helvetica, sans-serif">=
<span style=3D"font-size:13px">- Chrome grants short-term permissions for H=
TTP and long-term permissions for</span></font></pre><pre class=3D"" style=
=3D"margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><font face=3D"arial,=
 helvetica, sans-serif"><span style=3D"font-size:13px">  HTTPS.</span></fon=
t></pre><pre class=3D"" style=3D"margin-top:0px;margin-bottom:0px;color:rgb=
(0,0,0)"><font face=3D"arial, helvetica, sans-serif"><span style=3D"font-si=
ze:13px">- Firefox by default grants short-term permissions but allows the =
user to select</span></font></pre><pre class=3D"" style=3D"margin-top:0px;m=
argin-bottom:0px;color:rgb(0,0,0)"><font face=3D"arial, helvetica, sans-ser=
if"><span style=3D"font-size:13px">  long-term permissions if the site is H=
TTPS.</span></font></pre><pre class=3D"" style=3D"margin-top:0px;margin-bot=
tom:0px;color:rgb(0,0,0)"><font face=3D"arial, helvetica, sans-serif"><span=
 style=3D"font-size:13px"><br></span></font></pre><pre class=3D"" style=3D"=
margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><font face=3D"arial, hel=
vetica, sans-serif"><span style=3D"font-size:13px">It seems like some consi=
stency would be nice here.</span></font></pre><pre class=3D"" style=3D"marg=
in-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><font face=3D"arial, helveti=
ca, sans-serif"><span style=3D"font-size:13px"><br></span></font></pre><pre=
 class=3D"" style=3D"margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><fo=
nt face=3D"arial, helvetica, sans-serif">My personal view is that it would =
still be nice to require sites to ask for persistent</font></pre><pre class=
=3D"" style=3D"margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><font fac=
e=3D"arial, helvetica, sans-serif">permissions if they want them and that t=
here should be a getUserMedia()</font></pre><pre class=3D"" style=3D"margin=
-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><font face=3D"arial, helvetica=
, sans-serif">flag to indicate that. If people agree with me, I&#39;ll file=
 an issue on the media</font></pre><pre class=3D"" style=3D"margin-top:0px;=
margin-bottom:0px;color:rgb(0,0,0)"><font face=3D"arial, helvetica, sans-se=
rif">capture specification to add this affordance. However, if people think=
 this</font></pre><pre class=3D"" style=3D"margin-top:0px;margin-bottom:0px=
;color:rgb(0,0,0)"><font face=3D"arial, helvetica, sans-serif">is wrong, we=
 should remove this requirement in the security architecture</font></pre><p=
re class=3D"" style=3D"margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><=
font face=3D"arial, helvetica, sans-serif">document.</font></pre><pre class=
=3D"" style=3D"margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><font fac=
e=3D"arial, helvetica, sans-serif"><br></font></pre><pre class=3D"" style=
=3D"margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><font face=3D"arial,=
 helvetica, sans-serif">-Ekr</font></pre><pre class=3D"" style=3D"margin-to=
p:0px;margin-bottom:0px;color:rgb(0,0,0)"><font face=3D"arial, helvetica, s=
ans-serif"><br></font></pre><pre class=3D"" style=3D"margin-top:0px;margin-=
bottom:0px;color:rgb(0,0,0)"><font face=3D"arial, helvetica, sans-serif"><s=
pan style=3D"font-size:13px"><br></span></font></pre><pre class=3D"" style=
=3D"margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><font face=3D"arial,=
 helvetica, sans-serif"><span style=3D"font-size:13px"><br></span></font></=
pre><pre class=3D"" style=3D"margin-top:0px;margin-bottom:0px;color:rgb(0,0=
,0)"><font face=3D"arial, helvetica, sans-serif"><span style=3D"font-size:1=
3px"><br></span></font></pre><pre class=3D"" style=3D"font-size:1em;margin-=
top:0px;margin-bottom:0px;color:rgb(0,0,0)"><font face=3D"arial, helvetica,=
 sans-serif"><br></font></pre></div></div>

--089e013d14c2b860740510b8df64--


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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : WebRTC Security Architecture
        Author          : Eric Rescorla
	Filename        : draft-ietf-rtcweb-security-arch-11.txt
	Pages           : 43
	Date            : 2015-03-07

Abstract:
   The Real-Time Communications on the Web (RTCWEB) working group is
   tasked with standardizing protocols for enabling real-time
   communications within user-agents using web technologies (commonly
   called "WebRTC").  This document defines the security architecture
   for WebRTC.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-security-arch/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-security-arch-11

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-security-arch-11


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From nobody Sun Mar  8 15:56:55 2015
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On 3/5/15 8:44 AM, Christer Holmberg wrote:
> Hi,
>
>> Thanks. I remember now what it may happens:
>>
>> - A sends USE-CANDIDATE requests in parallel.
>> - A receives the ok response from one of them.
>> - A sends DTLS ClientHello.
>> - A receives the ok response with higher priority from another pair.
>> - A then continues sending media (maybe remaining DTLS stuff or RTP) for that pair.
>
> Just to clarify: when you say "continues sending", you are NOT saying that A "starts over" by sending a new initial ClientHello on the new pair - instead A only switches to the other pair and continues with normal DTLS setup procedures?
>
> ...which means that both the client and server may receive DTLS messages, associated with the same DTLS connection setup, on different 5-tuples?

To get back to a point that Christer made some time ago on this thread:

This stuff is *not* "rtcweb" stuff - it has much broader import. It 
needs to be specified more generally.

	Thanks,
	Paul


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On Mar 8, 2015, at 6:57 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>=20
> To get back to a point that Christer made some time ago on this thread:
>=20
> This stuff is *not* "rtcweb" stuff - it has much broader import. It needs t=
o be specified more generally.

[BA] It is appropriate for discussion in RTCWEB since WebRTC implementations=
 are implementing this and need to know how things work. However I agree tha=
t there should be a document (probably in MMUSIC) that describes how it work=
s. This was discussed in MMUSIC at the last IETF.=


From nobody Sun Mar  8 21:47:44 2015
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : WebRTC Forward Error Correction Requirements
        Author          : Justin Uberti
	Filename        : draft-ietf-rtcweb-fec-01.txt
	Pages           : 7
	Date            : 2015-03-08

Abstract:
   This document provides information and requirements for how Forward
   Error Correction (FEC) should be used by WebRTC applications.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/

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http://tools.ietf.org/html/draft-ietf-rtcweb-fec-01

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http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-fec-01


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From nobody Sun Mar  8 22:18:30 2015
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Bernard Aboba <bernard.aboba@gmail.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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Hi,

In MMUSIC we define how it's negotiated using SDP.

But, shouldn't the "virtual connection" concept, i.e. usage of multiple 5-t=
uples for a DTLS, be defined in a security WG?

Regards,

Christer

Sent from my Windows Phone
________________________________
From: Bernard Aboba<mailto:bernard.aboba@gmail.com>
Sent: =FD09/=FD03/=FD2015 01:28
To: Paul Kyzivat<mailto:pkyzivat@alum.mit.edu>
Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples

On Mar 8, 2015, at 6:57 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>
> To get back to a point that Christer made some time ago on this thread:
>
> This stuff is *not* "rtcweb" stuff - it has much broader import. It needs=
 to be specified more generally.

[BA] It is appropriate for discussion in RTCWEB since WebRTC implementation=
s are implementing this and need to know how things work. However I agree t=
hat there should be a document (probably in MMUSIC) that describes how it w=
orks. This was discussed in MMUSIC at the last IETF.
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

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256">
<meta name=3D"Generator" content=3D"Microsoft Exchange Server">
<!-- converted from text --><style><!-- .EmailQuote { margin-left: 1pt; pad=
ding-left: 4pt; border-left: #800000 2px solid; } --></style>
</head>
<body>
<div>
<div>
<div style=3D"font-family:Calibri,sans-serif; font-size:11pt">Hi,<br>
<br>
In MMUSIC we define how it's negotiated using SDP.<br>
<br>
But, shouldn't the &quot;virtual connection&quot; concept, i.e. usage of mu=
ltiple 5-tuples for a DTLS, be defined in a security WG?<br>
<br>
Regards,<br>
<br>
Christer<br>
<br>
Sent from my Windows Phone</div>
</div>
<div dir=3D"ltr">
<hr>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">From:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:bernard.aboba@gmail.com">Bernard Aboba</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Sent:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">=FD09=
/=FD03/=FD2015 01:28</span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">To:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:pkyzivat@alum.mit.edu">Paul Kyzivat</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Cc:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Subject:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">Re: [=
rtcweb] DTLS, DTLS-SRTP, and 5-tuples</span><br>
<br>
</div>
</div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText">On Mar 8, 2015, at 6:57 PM, Paul Kyzivat &lt;pkyzi=
vat@alum.mit.edu&gt; wrote:<br>
&gt; <br>
&gt; To get back to a point that Christer made some time ago on this thread=
:<br>
&gt; <br>
&gt; This stuff is *not* &quot;rtcweb&quot; stuff - it has much broader imp=
ort. It needs to be specified more generally.<br>
<br>
[BA] It is appropriate for discussion in RTCWEB since WebRTC implementation=
s are implementing this and need to know how things work. However I agree t=
hat there should be a document (probably in MMUSIC) that describes how it w=
orks. This was discussed in MMUSIC
 at the last IETF.<br>
_______________________________________________<br>
rtcweb mailing list<br>
rtcweb@ietf.org<br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.o=
rg/mailman/listinfo/rtcweb</a><br>
</div>
</span></font>
</body>
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--_000_7594FB04B1934943A5C02806D1A2204B1D72EE30ESESSMB209erics_--


From nobody Mon Mar  9 02:08:01 2015
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Subject: Re: [rtcweb] Call for adoption of draft-alvestrand-rtcweb-gateways
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Hello everyone,

2014-12-20 17:55 GMT+01:00 Cullen Jennings (fluffy) <fluffy@cisco.com>:

>
> Thanks you to everyone that provided input.
>
> The chairs are calling that there is consensus to adopt this draft as a W=
G
> draft.
>
> We will start discussions with the AD to see about adding a milestone. Th=
e
> chairs are mostly on vacation till the new year so this may not move much
> till then.
>
> Thanks,
> The Chairs (Cullen, Sean, Ted)
>

Is there any new about the adoption of the draft?

Thanks


--=20
Jos=C3=A9 Luis Mill=C3=A1n

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<div dir=3D"ltr">Hello everyone,<div class=3D"gmail_extra"><br><div class=
=3D"gmail_quote">2014-12-20 17:55 GMT+01:00 Cullen Jennings (fluffy) <span =
dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@cisco.com" target=3D"_blank" oncli=
ck=3D"window.open(&#39;https://mail.google.com/mail/?view=3Dcm&amp;tf=3D1&a=
mp;to=3Dfluffy@cisco.com&amp;cc=3D&amp;bcc=3D&amp;su=3D&amp;body=3D&#39;,&#=
39;_blank&#39;);return false;">fluffy@cisco.com</a>&gt;</span>:<br><blockqu=
ote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc s=
olid;padding-left:1ex"><br>
Thanks you to everyone that provided input.<br>
<br>
The chairs are calling that there is consensus to adopt this draft as a WG =
draft.<br>
<br>
We will start discussions with the AD to see about adding a milestone. The =
chairs are mostly on vacation till the new year so this may not move much t=
ill then.<br>
<span class=3D"im HOEnZb"><br>
Thanks,<br>
The Chairs (Cullen, Sean, Ted)<br></span></blockquote><div><br></div><div>I=
s there any new about the adoption of the draft?</div><div><br></div><div>T=
hanks</div></div><br clear=3D"all"><div><br></div>-- <br><div class=3D"gmai=
l_signature">Jos=C3=A9 Luis Mill=C3=A1n</div>
</div></div>

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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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Le 2015-03-09 01:18, Christer Holmberg a écrit :
> But, shouldn't the "virtual connection" concept, i.e. usage of multiple
> 5-tuples for a DTLS, be defined in a security WG?

Not at all. The "virtual connection" concept applies to any protocol 
being transported over ICE. DTLS is just one such protocol. Another 
obvious one is... RTP! :)

Maybe ICE-bis should explain all of this?

Simon


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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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2015-03-09 13:47 GMT+01:00 Simon Perreault <sperreault@jive.com>:
> Not at all. The "virtual connection" concept applies to any protocol bein=
g
> transported over ICE. DTLS is just one such protocol. Another obvious one
> is... RTP! :)
>
> Maybe ICE-bis should explain all of this?

Yes. ICE-bis should define what a "virtual transport" is, and should
explain that from the point of view of the application is must be used
as a single 5-tuple transport.


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


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Simon Perreault <sperreault@jive.com>, Bernard Aboba <bernard.aboba@gmail.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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Hi,

DTLS itself knows nothing about ICE.

So, IF DTLS assumes (implicitly or explicitly) that a single 5-tuple is use=
d, the appropriate WG at least need to be consulted about whether usage of =
multiple 5-tuples will cause any issues - technical or security.=20

Regards,

Christer

-----Original Message-----
From: Simon Perreault [mailto:sperreault@jive.com]=20
Sent: 9. maaliskuuta 2015 14:47
To: Christer Holmberg; Bernard Aboba; Paul Kyzivat
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples

Le 2015-03-09 01:18, Christer Holmberg a =E9crit :
> But, shouldn't the "virtual connection" concept, i.e. usage of=20
> multiple 5-tuples for a DTLS, be defined in a security WG?

Not at all. The "virtual connection" concept applies to any protocol being =
transported over ICE. DTLS is just one such protocol. Another obvious one i=
s... RTP! :)

Maybe ICE-bis should explain all of this?

Simon


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 9 Mar 2015 13:56:48 +0100
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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2015-03-09 13:54 GMT+01:00 Christer Holmberg <christer.holmberg@ericsson.co=
m>:
> DTLS itself knows nothing about ICE.
>
> So, IF DTLS assumes (implicitly or explicitly) that a single 5-tuple is u=
sed, the appropriate WG at least need to be consulted about whether usage o=
f multiple 5-tuples will cause any issues - technical or security.

DTLS wrongly assumes a single 5-tuple. It should assume a single
transport, and such a transport may be a classic 5-tuple or a ICE
transport. It is a task of ICE to define what such a transport is.


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


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: =?utf-8?B?ScOxYWtpIEJheiBDYXN0aWxsbw==?= <ibc@aliax.net>
Thread-Topic: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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2015-03-09 14:00 GMT+01:00 Christer Holmberg <christer.holmberg@ericsson.co=
m>:
>> DTLS wrongly assumes a single 5-tuple. It should assume a single transpo=
rt, and such a transport may be a classic 5-tuple or a ICE transport. It is=
 a task of ICE to define what such a transport is.
>
> Yes, I agree that ICE can define such virtual transport.
>
> But, before we do that, the DTLS folks need to agree that assuming a sing=
le 5-tuple IS wrong  :)

It is as wrong as assuming that using a single **UDP** 5-tuple is safe :)


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


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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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Le 2015-03-09 08:54, Christer Holmberg a écrit :
> DTLS itself knows nothing about ICE.
> 
> So, IF DTLS assumes (implicitly or explicitly) that a single 5-tuple is used, the appropriate WG at least need to be consulted about whether usage of multiple 5-tuples will cause any issues - technical or security. 

Definitely.

Simon


From nobody Mon Mar  9 06:43:08 2015
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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-constraints-registry-02.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : IANA Registry for RTCWeb Constrainable Properties
        Author          : Daniel C. Burnett
	Filename        : draft-ietf-rtcweb-constraints-registry-02.txt
	Pages           : 5
	Date            : 2015-03-09

Abstract:
   Specifications in W3C's Media Capture Task Force and WebRTC Working
   Group have need of a registry in which to maintain a list of
   constrainable properties for HTML media and other constrainable
   objects.  This document defines this registry.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-constraints-registry/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-constraints-registry-02

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-constraints-registry-02


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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


From nobody Mon Mar  9 07:17:08 2015
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To: Simon Perreault <sperreault@jive.com>,  Christer Holmberg <christer.holmberg@ericsson.com>, Bernard Aboba <bernard.aboba@gmail.com>
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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On 3/9/15 9:17 AM, Simon Perreault wrote:
> Le 2015-03-09 08:54, Christer Holmberg a écrit :
>> DTLS itself knows nothing about ICE.
>>
>> So, IF DTLS assumes (implicitly or explicitly) that a single 5-tuple is used, the appropriate WG at least need to be consulted about whether usage of multiple 5-tuples will cause any issues - technical or security.

ISTM that this subject is closely related to multipath transports - TCP, 
RTP, UDP, SCTP. ICE provides a way to negotiate the multiple paths.

Looking around, work on multipath transports is spread around. For TCP 
and SCTP it was done in transport. There is a draft for multipath RTP in 
avtcore. I see a little email discussion of multipath UDP in transport, 
but (based on my one minute spent googling) not any real action.

To date ICE hasn't considered this to be a general multipath transport 
problem - just a way to negotiate a single path transport. But the 
discussion here is showing flaws in that approach. But 
draft-wing-mmusic-ice-mobility was closer to really dealing with multipath.

This may need a tactical solution to keep rtcweb going in a timely way 
and a more strategic approach to get this all properly cleaned up.

	Thanks,
	Paul


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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2015-03-09 15:16 GMT+01:00 Paul Kyzivat <pkyzivat@alum.mit.edu>:
> To date ICE hasn't considered this to be a general multipath transport
> problem - just a way to negotiate a single path transport. But the
> discussion here is showing flaws in that approach.

I don't think there are flaws. The fact is that in certain ICE usages
(aggressive nomination) it may perfectly happen that media (say RTP or
DTLS) can be sent over different 5-tuples at the ~same time. IMHO
that's not a flaw to fix, but something that we must assume as it is.


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


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References: <54F74B02.1070902@jive.com> <7594FB04B1934943A5C02806D1A2204B1D727570@ESESSMB209.ericsson.se> <CALiegfmfvz3NWSjcovGBytiOTbR6kFfyh0vx5cXoMJtytfGzRA@mail.gmail.com> <CAD5OKxsu3D0xHY-zYbDu1hyH_+4=3mWDvW2i98WCVZ+29BpKCw@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D728297@ESESSMB209.ericsson.se> <CALiegf=uPN+g546Ucv9s89z14cUTEme55y7B1siXZe97yj7Lig@mail.gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E726EEC@US70UWXCHMBA02.zam.alcatel-lucent.com> <CALiegf=oVWk-8UcbQE2Edh=QSXSRUnSC=X-WMyGpvHYQ9SD1yg@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D728BE2@ESESSMB209.ericsson.se> <54FCD3BC.4070900@alum.mit.edu> <F37736EA-2AEE-4022-A813-E21469420038@gmail.com> <7594FB04B1934943A5C02806D1A2204B1D72EE30@ESESSMB209.ericsson.se> <54FD964F.2070105@jive.com> <7594FB04B1934943A5C02806D1A2204B1D73015C@ESESSMB209.ericsson.se> <54FD9D50.4070202@jive.com> <54FDAB23.7070202@alum.mit.edu> <CALiegf=EPWs1LPpU4_dH5-ZFSJY6BDhP-f0yPBXVYxEMKNcDSQ@mail.gmail.com>
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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On 3/9/15 11:51 AM, IÃ±aki Baz Castillo wrote:
> 2015-03-09 15:16 GMT+01:00 Paul Kyzivat <pkyzivat@alum.mit.edu>:
>> To date ICE hasn't considered this to be a general multipath transport
>> problem - just a way to negotiate a single path transport. But the
>> discussion here is showing flaws in that approach.
>
> I don't think there are flaws. The fact is that in certain ICE usages
> (aggressive nomination) it may perfectly happen that media (say RTP or
> DTLS) can be sent over different 5-tuples at the ~same time. IMHO
> that's not a flaw to fix, but something that we must assume as it is.

I didn't mean that it is a flaw to be discussing this in the context of 
RTCWEB.

What I meant is that it is a flaw that the more general nature of this 
as an aspect of multipath transports hasn't been noticed. (Or maybe it 
has, and I'm just not well informed. I don't follow the transport area.)

	Thanks,
	Paul


From nobody Mon Mar  9 10:39:24 2015
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Date: Mon, 9 Mar 2015 12:39:09 -0500
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From: Alan Johnston <alan.b.johnston@gmail.com>
To: Benjamin Schwartz <bemasc@webrtc.org>
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Subject: Re: [rtcweb] RETURN-05
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--20cf3033449dfcafdc0510de82b4
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I think RETURN is essential for WebRTC to work well across enterprise
borders.  We will increasingly see border TURN servers used by enterprises
to manage and control WebRTC usage, most likely as additions to existing
firewalls and SBCs.  Without RETURN, the usage of border TURN servers will
result in loss of functionality in WebRTC applications that use a TURN
server to provide features in addition to simple NAT traversal.  RETURN
should work perfectly to allow both TURN servers to work, while still
allowing end-to-end encrypted and authenticated media flows.

I=E2=80=99m not aware of any other solutions to the enterprise requirements
identified in Section 3.3.5 of
draft-ietf-rtcweb-use-cases-and-requirements.  I think we need to move
forward with this draft in RTCWEB.

- Alan -

On Fri, Mar 6, 2015 at 9:19 AM, Benjamin Schwartz <bemasc@webrtc.org> wrote=
:

> New draft of RETURN is up:
> http://tools.ietf.org/html/draft-schwartz-rtcweb-return-05
>
> This draft contains a variety of clarifications and changes of emphasis,
> and should hopefully now be easier to read and understand.
>
> Thanks to Alan Johnston and John Yoakum for contributing new figures and
> detailed text review for this revision.
>
> --Ben
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--20cf3033449dfcafdc0510de82b4
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<div dir=3D"ltr"><div>I think RETURN is essential for WebRTC to work well a=
cross enterprise borders.=C2=A0 We will increasingly see border TURN server=
s used by enterprises to manage and control WebRTC usage, most likely as ad=
ditions to existing firewalls and SBCs.=C2=A0 Without RETURN, the usage of =
border TURN servers will result in loss of functionality in WebRTC applicat=
ions that use a TURN server to provide features in addition to simple NAT t=
raversal.=C2=A0 RETURN should work perfectly to allow both TURN servers to =
work, while still allowing end-to-end encrypted and authenticated media flo=
ws.<br></div><div><br></div><div>I=E2=80=99m not aware of any other solutio=
ns to the enterprise requirements identified in Section 3.3.5 of draft-ietf=
-rtcweb-use-cases-and-requirements.=C2=A0 I think we need to move forward w=
ith this draft in RTCWEB.</div><div><br></div><div>- Alan -</div></div><div=
 class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri, Mar 6, 2015 a=
t 9:19 AM, Benjamin Schwartz <span dir=3D"ltr">&lt;<a href=3D"mailto:bemasc=
@webrtc.org" target=3D"_blank">bemasc@webrtc.org</a>&gt;</span> wrote:<br><=
blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px=
 #ccc solid;padding-left:1ex"><div dir=3D"ltr"><span style=3D"font-size:12.=
8000001907349px">New draft of RETURN is up:</span><div style=3D"font-size:1=
2.8000001907349px"><a href=3D"http://tools.ietf.org/html/draft-schwartz-rtc=
web-return-05" target=3D"_blank">http://tools.ietf.org/html/draft-schwartz-=
rtcweb-return-05</a><br></div><div style=3D"font-size:12.8000001907349px"><=
br></div><div style=3D"font-size:12.8000001907349px">This draft contains a =
variety of clarifications and changes of emphasis, and should hopefully now=
 be easier to read and understand.</div><div style=3D"font-size:12.80000019=
07349px"><br></div><div style=3D"font-size:12.8000001907349px">Thanks to Al=
an Johnston and John Yoakum for contributing new figures and detailed text =
review for this revision.</div><div style=3D"font-size:12.8000001907349px">=
<br></div><div style=3D"font-size:12.8000001907349px">--Ben</div></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--20cf3033449dfcafdc0510de82b4--


From nobody Mon Mar  9 10:41:18 2015
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From: Ted Hardie <ted.ietf@gmail.com>
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Subject: Re: [rtcweb] Call for adoption of draft-alvestrand-rtcweb-gateways
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--20cf301d3a286738200510de8ab7
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On Mon, Mar 9, 2015 at 2:07 AM, Jos=C3=A9 Luis Mill=C3=A1n <jmillan@aliax.n=
et> wrote:

> Is there any new about the adoption of the draft?
>
>
=E2=80=8BHowdy,

It was adopted, but we have not determined when it should be put in the
timeline.
During the "Doc review" portion of the upcoming meeting, we plan to discuss
our
overall timeline, and can cover this then.

regards,

Ted Hardie



> Thanks
>
>
> --
> Jos=C3=A9 Luis Mill=C3=A1n
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--20cf301d3a286738200510de8ab7
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<div dir=3D"ltr"><div class=3D"gmail_extra">On Mon, Mar 9, 2015 at 2:07 AM,=
 Jos=C3=A9 Luis Mill=C3=A1n <span dir=3D"ltr">&lt;<a href=3D"mailto:jmillan=
@aliax.net" target=3D"_blank">jmillan@aliax.net</a>&gt;</span> wrote:<br><d=
iv class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margin:=
0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><d=
iv class=3D"gmail_quote"><span class=3D""></span><div>Is there any new abou=
t the adoption of the draft?</div><div><br></div></div></div></blockquote><=
div><br><div class=3D"gmail_default" style=3D"font-family:tahoma,sans-serif=
;font-size:small">=E2=80=8BHowdy,<br><br></div><div class=3D"gmail_default"=
 style=3D"font-family:tahoma,sans-serif;font-size:small">It was adopted, bu=
t we have not determined when it should be put in the timeline.<br></div><d=
iv class=3D"gmail_default" style=3D"font-family:tahoma,sans-serif;font-size=
:small">During the &quot;Doc review&quot; portion of the upcoming meeting, =
we plan to discuss our<br></div><div class=3D"gmail_default" style=3D"font-=
family:tahoma,sans-serif;font-size:small">overall timeline, and can cover t=
his then.<br><br></div><div class=3D"gmail_default" style=3D"font-family:ta=
homa,sans-serif;font-size:small">regards,<br><br></div><div class=3D"gmail_=
default" style=3D"font-family:tahoma,sans-serif;font-size:small">Ted Hardie=
<br></div><br>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin=
:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><=
div class=3D"gmail_extra"><div class=3D"gmail_quote"><div></div><div>Thanks=
</div></div><span class=3D"HOEnZb"><font color=3D"#888888"><br clear=3D"all=
"><div><br></div>-- <br><div>Jos=C3=A9 Luis Mill=C3=A1n</div>
</font></span></div></div>
<br>_______________________________________________<br>
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<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

--20cf301d3a286738200510de8ab7--


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From: Cullen Jennings <fluffy@cisco.com>
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Date: Mon, 9 Mar 2015 09:28:23 -0600
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Message-Id: <2F467A7E-7A6C-4B1B-985A-0D9C089BE973@cisco.com>
References: <54F74B02.1070902@jive.com> <CAD5OKxs8JYG3-Vvndi59ZrdPE7UTj22ozD4tcWTHgzWrHv=q7Q@mail.gmail.com> <54F756B2.60408@jive.com> <7594FB04B1934943A5C02806D1A2204B1D726AD8@ESESSMB209.ericsson.se> <CAD5OKxu7py3HbrFjxTDZS5ECFzx7vd=wpjve-gT6gWwksjEu+g@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D726B71@ESESSMB209.ericsson.se> <CABcZeBO1O6sA8MqvWkCDu3RPLz5-P2G65Us28i0baOavDnRT7Q@mail.gmail.com> <CAD5OKxuWCdgMR5Kxjv9BSwZ3Jm9kGXx9Pi-9FrfsnuQZ_91jAA@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D726DC1@ESESSMB209.ericsson.se> <CALiegfkipJhsy7-40+=d9xMUf4RJGdn3_fABL3NN2KuFNvS2BA@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D727570@ESESSMB209.ericsson.se> <CALiegfmfvz3NWSjcovGBytiOTbR6kFfyh0vx5cXoMJtytfGzRA@mail.gmail.com> <CAD5OKxsu3D0xHY-zYbDu1hyH_+4=3mWDvW2i98WCVZ+29BpKCw@mail.gmail.com> <CA5E97EE-99F8-44D8-B05B-C9EFDED1A9BB@vidyo.com>
To: Jonathan Lennox <jonathan@vidyo.com>
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Archived-At: <http://mailarchive.ietf.org/arch/msg/rtcweb/i0KhlTALNyCbcvL6Ury9kOwaiws>
Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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> On Mar 5, 2015, at 8:22 AM, Jonathan Lennox <jonathan@vidyo.com> =
wrote:
>=20
>=20
> On Mar 5, 2015, at 5:14 AM, Roman Shpount <roman@telurix.com> wrote:
>=20
>> On Thu, Mar 5, 2015 at 4:52 AM, I=F1aki Baz Castillo <ibc@aliax.net> =
wrote:
>> 2015-03-05 7:58 GMT+01:00 Christer Holmberg =
<christer.holmberg@ericsson.com>:
>> > As far as I understand, sending of multiple ClientHellos, on =
different
>> > 5-tuples, would require a change to core DTLS (not only DTLS-SRTP), =
and who
>> > knows how long such work would take. Do we really want to add such
>> > dependency to our deliveries at this point?
>>=20
>> Sorry, I think I didn't explain well.
>>=20
>> It is not about multiple DTLS connections, but about the fact that
>> DTLS packets belonging to the *same* DTLS connection/association can
>> be carried over different 5-tuples. This may happen during agressive
>> ICE nomination in which media (so the DTLS ClientHello) follows the
>> USE-CANDIDATE Binding request without even waiting for a response.
>>=20
>> When was it agreed that DTLS ClientHello can be sent before the =
connectivity check succeeded? I understand that this will increase the =
connection setup time, but I though that no data should be sent before =
the connectivity check response (consent) from the remote party.
>=20
> As I understand the current rough consensus, the endpoint has to have =
received a successful response to a connectivity check on the 5-tuple, =
proving consent, but it doesn=92t have to have received (or sent) a =
check with a USE-CANDIDATE attribute in it yet.
>=20
> This was the outcome of our discussion =93merging" regular and =
aggressive ICE nomination.

That matches my understanding of how things work.=20=


From nobody Mon Mar  9 16:24:07 2015
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : Javascript Session Establishment Protocol
        Authors         : Justin Uberti
                          Cullen Jennings
                          Eric Rescorla
	Filename        : draft-ietf-rtcweb-jsep-09.txt
	Pages           : 73
	Date            : 2015-03-09

Abstract:
   This document describes the mechanisms for allowing a Javascript
   application to control the signaling plane of a multimedia session
   via the interface specified in the W3C RTCPeerConnection API, and
   discusses how this relates to existing signaling protocols.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-jsep-09

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-jsep-09


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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


From nobody Mon Mar  9 16:36:26 2015
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From: Justin Uberti <juberti@google.com>
Date: Mon, 9 Mar 2015 16:36:01 -0700
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--90e6ba61465a6336970510e38028
Content-Type: text/plain; charset=UTF-8

This version incorporates the feedback from Honolulu, as well as several
comments from the list.

The Honolulu feedback suggested removing support for redundant audio with
codecs like PCMU, but I did get a request for this on the list, so I left
it in.

On Sun, Mar 8, 2015 at 9:47 PM, <internet-drafts@ietf.org> wrote:

>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>  This draft is a work item of the Real-Time Communication in WEB-browsers
> Working Group of the IETF.
>
>         Title           : WebRTC Forward Error Correction Requirements
>         Author          : Justin Uberti
>         Filename        : draft-ietf-rtcweb-fec-01.txt
>         Pages           : 7
>         Date            : 2015-03-08
>
> Abstract:
>    This document provides information and requirements for how Forward
>    Error Correction (FEC) should be used by WebRTC applications.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rtcweb-fec-01
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-fec-01
>
>
> Please note that it may take a couple of minutes from the time of
> submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--90e6ba61465a6336970510e38028
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<div dir=3D"ltr">This version incorporates the feedback from Honolulu, as w=
ell as several comments from the list.<div><br></div><div>The Honolulu feed=
back suggested removing support for redundant audio with codecs like PCMU, =
but I did get a request for this on the list, so I left it in.</div></div><=
div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Sun, Mar 8, 201=
5 at 9:47 PM,  <span dir=3D"ltr">&lt;<a href=3D"mailto:internet-drafts@ietf=
.org" target=3D"_blank">internet-drafts@ietf.org</a>&gt;</span> wrote:<br><=
blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px=
 #ccc solid;padding-left:1ex"><br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
=C2=A0This draft is a work item of the Real-Time Communication in WEB-brows=
ers Working Group of the IETF.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Title=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 WebRTC Forward Error Correction Requirements<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Author=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : Just=
in Uberti<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Filename=C2=A0 =C2=A0 =C2=A0 =C2=A0 : draft-iet=
f-rtcweb-fec-01.txt<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Pages=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 7<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Date=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :=
 2015-03-08<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0This document provides information and requirements for how Fo=
rward<br>
=C2=A0 =C2=A0Error Correction (FEC) should be used by WebRTC applications.<=
br>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/" target=
=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/</a><br>
<br>
There&#39;s also a htmlized version available at:<br>
<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-fec-01" target=3D"_=
blank">http://tools.ietf.org/html/draft-ietf-rtcweb-fec-01</a><br>
<br>
A diff from the previous version is available at:<br>
<a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-fec-01" tar=
get=3D"_blank">http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-fec-01<=
/a><br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp://ftp=
.ietf.org/internet-drafts/</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div>

--90e6ba61465a6336970510e38028--


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From: Justin Uberti <juberti@google.com>
Date: Mon, 9 Mar 2015 16:41:34 -0700
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To: Cullen Jennings <fluffy@cisco.com>
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Cc: Jonathan Lennox <jonathan@vidyo.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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On Mon, Mar 9, 2015 at 8:28 AM, Cullen Jennings <fluffy@cisco.com> wrote:

>
> > On Mar 5, 2015, at 8:22 AM, Jonathan Lennox <jonathan@vidyo.com> wrote:
> >
> >
> > On Mar 5, 2015, at 5:14 AM, Roman Shpount <roman@telurix.com> wrote:
> >
> >> On Thu, Mar 5, 2015 at 4:52 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net=
>
> wrote:
> >> 2015-03-05 7:58 GMT+01:00 Christer Holmberg <
> christer.holmberg@ericsson.com>:
> >> > As far as I understand, sending of multiple ClientHellos, on differe=
nt
> >> > 5-tuples, would require a change to core DTLS (not only DTLS-SRTP),
> and who
> >> > knows how long such work would take. Do we really want to add such
> >> > dependency to our deliveries at this point?
> >>
> >> Sorry, I think I didn't explain well.
> >>
> >> It is not about multiple DTLS connections, but about the fact that
> >> DTLS packets belonging to the *same* DTLS connection/association can
> >> be carried over different 5-tuples. This may happen during agressive
> >> ICE nomination in which media (so the DTLS ClientHello) follows the
> >> USE-CANDIDATE Binding request without even waiting for a response.
> >>
> >> When was it agreed that DTLS ClientHello can be sent before the
> connectivity check succeeded? I understand that this will increase the
> connection setup time, but I though that no data should be sent before th=
e
> connectivity check response (consent) from the remote party.
> >
> > As I understand the current rough consensus, the endpoint has to have
> received a successful response to a connectivity check on the 5-tuple,
> proving consent, but it doesn=E2=80=99t have to have received (or sent) a=
 check
> with a USE-CANDIDATE attribute in it yet.
> >
> > This was the outcome of our discussion =E2=80=9Cmerging" regular and ag=
gressive
> ICE nomination.
>
> That matches my understanding of how things work.
>
>
That discussion is now codified in a new draft,
https://tools.ietf.org/html/draft-uberti-mmusic-nombis-00.

To handle the issue raised by the OP, I think we should either
1) Fix the DTLS draft (and any others) to not be bound to a 5-tuple when
ICE is used
2) Update ICE-bis to say that any language about 5-tuples is invalidated by
ICE, and should be treated as a virtual transport.

#2 seems like the simplest path at this point, given that ICE-bis work is
underway.

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Mon, Mar 9, 2015 at 8:28 AM, Cullen Jennings <span dir=3D"ltr">&lt;<=
a href=3D"mailto:fluffy@cisco.com" target=3D"_blank">fluffy@cisco.com</a>&g=
t;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0=
px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);borde=
r-left-style:solid;padding-left:1ex"><div><div><br>
&gt; On Mar 5, 2015, at 8:22 AM, Jonathan Lennox &lt;<a href=3D"mailto:jona=
than@vidyo.com" target=3D"_blank">jonathan@vidyo.com</a>&gt; wrote:<br>
&gt;<br>
&gt;<br>
&gt; On Mar 5, 2015, at 5:14 AM, Roman Shpount &lt;<a href=3D"mailto:roman@=
telurix.com" target=3D"_blank">roman@telurix.com</a>&gt; wrote:<br>
&gt;<br>
&gt;&gt; On Thu, Mar 5, 2015 at 4:52 AM, I=C3=B1aki Baz Castillo &lt;<a hre=
f=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt; wrote:<b=
r>
&gt;&gt; 2015-03-05 7:58 GMT+01:00 Christer Holmberg &lt;<a href=3D"mailto:=
christer.holmberg@ericsson.com" target=3D"_blank">christer.holmberg@ericsso=
n.com</a>&gt;:<br>
&gt;&gt; &gt; As far as I understand, sending of multiple ClientHellos, on =
different<br>
&gt;&gt; &gt; 5-tuples, would require a change to core DTLS (not only DTLS-=
SRTP), and who<br>
&gt;&gt; &gt; knows how long such work would take. Do we really want to add=
 such<br>
&gt;&gt; &gt; dependency to our deliveries at this point?<br>
&gt;&gt;<br>
&gt;&gt; Sorry, I think I didn&#39;t explain well.<br>
&gt;&gt;<br>
&gt;&gt; It is not about multiple DTLS connections, but about the fact that=
<br>
&gt;&gt; DTLS packets belonging to the *same* DTLS connection/association c=
an<br>
&gt;&gt; be carried over different 5-tuples. This may happen during agressi=
ve<br>
&gt;&gt; ICE nomination in which media (so the DTLS ClientHello) follows th=
e<br>
&gt;&gt; USE-CANDIDATE Binding request without even waiting for a response.=
<br>
&gt;&gt;<br>
&gt;&gt; When was it agreed that DTLS ClientHello can be sent before the co=
nnectivity check succeeded? I understand that this will increase the connec=
tion setup time, but I though that no data should be sent before the connec=
tivity check response (consent) from the remote party.<br>
&gt;<br>
&gt; As I understand the current rough consensus, the endpoint has to have =
received a successful response to a connectivity check on the 5-tuple, prov=
ing consent, but it doesn=E2=80=99t have to have received (or sent) a check=
 with a USE-CANDIDATE attribute in it yet.<br>
&gt;<br>
&gt; This was the outcome of our discussion =E2=80=9Cmerging&quot; regular =
and aggressive ICE nomination.<br>
<br>
</div></div>That matches my understanding of how things work.<br>
<div><div><br></div></div></blockquote><div>=C2=A0</div><div>That discussio=
n is now codified in a new draft, <a href=3D"https://tools.ietf.org/html/dr=
aft-uberti-mmusic-nombis-00" target=3D"_blank">https://tools.ietf.org/html/=
draft-uberti-mmusic-nombis-00</a>.</div><div><br></div><div>To handle the i=
ssue raised by the OP, I think we should either</div><div>1) Fix the DTLS d=
raft (and any others) to not be bound to a 5-tuple when ICE is used</div><d=
iv>2) Update ICE-bis to say that any language about 5-tuples is invalidated=
 by ICE, and should be treated as a virtual transport.</div><div><br></div>=
<div>#2 seems like the simplest path at this point, given that ICE-bis work=
 is underway.</div></div></div></div>

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From: Justin Uberti <juberti@google.com>
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[fixing cc list]

On Mon, Mar 9, 2015 at 4:36 PM, Justin Uberti <juberti@google.com> wrote:

> This version incorporates the feedback from Honolulu, as well as several
> comments from the list.
>
> The Honolulu feedback suggested removing support for redundant audio with
> codecs like PCMU, but I did get a request for this on the list, so I left
> it in.
>
> On Sun, Mar 8, 2015 at 9:47 PM, <internet-drafts@ietf.org> wrote:
>
>>
>> A New Internet-Draft is available from the on-line Internet-Drafts
>> directories.
>>  This draft is a work item of the Real-Time Communication in WEB-browsers
>> Working Group of the IETF.
>>
>>         Title           : WebRTC Forward Error Correction Requirements
>>         Author          : Justin Uberti
>>         Filename        : draft-ietf-rtcweb-fec-01.txt
>>         Pages           : 7
>>         Date            : 2015-03-08
>>
>> Abstract:
>>    This document provides information and requirements for how Forward
>>    Error Correction (FEC) should be used by WebRTC applications.
>>
>>
>> The IETF datatracker status page for this draft is:
>> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/
>>
>> There's also a htmlized version available at:
>> http://tools.ietf.org/html/draft-ietf-rtcweb-fec-01
>>
>> A diff from the previous version is available at:
>> http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-fec-01
>>
>>
>> Please note that it may take a couple of minutes from the time of
>> submission
>> until the htmlized version and diff are available at tools.ietf.org.
>>
>> Internet-Drafts are also available by anonymous FTP at:
>> ftp://ftp.ietf.org/internet-drafts/
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
>

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<div dir=3D"ltr">[fixing cc list]</div><div class=3D"gmail_extra"><br><div =
class=3D"gmail_quote">On Mon, Mar 9, 2015 at 4:36 PM, Justin Uberti <span d=
ir=3D"ltr">&lt;<a href=3D"mailto:juberti@google.com" target=3D"_blank">jube=
rti@google.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" s=
tyle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div=
 dir=3D"ltr">This version incorporates the feedback from Honolulu, as well =
as several comments from the list.<div><br></div><div>The Honolulu feedback=
 suggested removing support for redundant audio with codecs like PCMU, but =
I did get a request for this on the list, so I left it in.</div></div><div =
class=3D"HOEnZb"><div class=3D"h5"><div class=3D"gmail_extra"><br><div clas=
s=3D"gmail_quote">On Sun, Mar 8, 2015 at 9:47 PM,  <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:internet-drafts@ietf.org" target=3D"_blank">internet-drafts=
@ietf.org</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
=C2=A0This draft is a work item of the Real-Time Communication in WEB-brows=
ers Working Group of the IETF.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Title=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 WebRTC Forward Error Correction Requirements<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Author=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : Just=
in Uberti<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Filename=C2=A0 =C2=A0 =C2=A0 =C2=A0 : draft-iet=
f-rtcweb-fec-01.txt<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Pages=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 7<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Date=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :=
 2015-03-08<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0This document provides information and requirements for how Fo=
rward<br>
=C2=A0 =C2=A0Error Correction (FEC) should be used by WebRTC applications.<=
br>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/" target=
=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/</a><br>
<br>
There&#39;s also a htmlized version available at:<br>
<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-fec-01" target=3D"_=
blank">http://tools.ietf.org/html/draft-ietf-rtcweb-fec-01</a><br>
<br>
A diff from the previous version is available at:<br>
<a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-fec-01" tar=
get=3D"_blank">http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-fec-01<=
/a><br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp://ftp=
.ietf.org/internet-drafts/</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div>
</div></div></blockquote></div><br></div>

--001a113eeca825f6e30510e45904--


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From: Eric Rescorla <ekr@rtfm.com>
Date: Mon, 9 Mar 2015 17:49:08 -0700
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To: Justin Uberti <juberti@google.com>
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Cc: Cullen Jennings <fluffy@cisco.com>, Jonathan Lennox <jonathan@vidyo.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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On Mon, Mar 9, 2015 at 4:41 PM, Justin Uberti <juberti@google.com> wrote:

>
>
> On Mon, Mar 9, 2015 at 8:28 AM, Cullen Jennings <fluffy@cisco.com> wrote:
>
>>
>> > On Mar 5, 2015, at 8:22 AM, Jonathan Lennox <jonathan@vidyo.com> wrote=
:
>> >
>> >
>> > On Mar 5, 2015, at 5:14 AM, Roman Shpount <roman@telurix.com> wrote:
>> >
>> >> On Thu, Mar 5, 2015 at 4:52 AM, I=C3=B1aki Baz Castillo <ibc@aliax.ne=
t>
>> wrote:
>> >> 2015-03-05 7:58 GMT+01:00 Christer Holmberg <
>> christer.holmberg@ericsson.com>:
>> >> > As far as I understand, sending of multiple ClientHellos, on
>> different
>> >> > 5-tuples, would require a change to core DTLS (not only DTLS-SRTP),
>> and who
>> >> > knows how long such work would take. Do we really want to add such
>> >> > dependency to our deliveries at this point?
>> >>
>> >> Sorry, I think I didn't explain well.
>> >>
>> >> It is not about multiple DTLS connections, but about the fact that
>> >> DTLS packets belonging to the *same* DTLS connection/association can
>> >> be carried over different 5-tuples. This may happen during agressive
>> >> ICE nomination in which media (so the DTLS ClientHello) follows the
>> >> USE-CANDIDATE Binding request without even waiting for a response.
>> >>
>> >> When was it agreed that DTLS ClientHello can be sent before the
>> connectivity check succeeded? I understand that this will increase the
>> connection setup time, but I though that no data should be sent before t=
he
>> connectivity check response (consent) from the remote party.
>> >
>> > As I understand the current rough consensus, the endpoint has to have
>> received a successful response to a connectivity check on the 5-tuple,
>> proving consent, but it doesn=E2=80=99t have to have received (or sent) =
a check
>> with a USE-CANDIDATE attribute in it yet.
>> >
>> > This was the outcome of our discussion =E2=80=9Cmerging" regular and a=
ggressive
>> ICE nomination.
>>
>> That matches my understanding of how things work.
>>
>>
> That discussion is now codified in a new draft,
> https://tools.ietf.org/html/draft-uberti-mmusic-nombis-00.
>
> To handle the issue raised by the OP, I think we should either
> 1) Fix the DTLS draft (and any others) to not be bound to a 5-tuple when
> ICE is used
> 2) Update ICE-bis to say that any language about 5-tuples is invalidated
> by ICE, and should be treated as a virtual transport.
>
> #2 seems like the simplest path at this point, given that ICE-bis work is
> underway.
>

#2 seems like the right answer to me.

-Ekr

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

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Mon, Mar 9, 2015 at 4:41 PM, Justin Uberti <span dir=3D"ltr">&lt;<a =
href=3D"mailto:juberti@google.com" target=3D"_blank">juberti@google.com</a>=
&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0=
 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><br><=
div class=3D"gmail_extra"><br><div class=3D"gmail_quote"><div><div class=3D=
"h5">On Mon, Mar 9, 2015 at 8:28 AM, Cullen Jennings <span dir=3D"ltr">&lt;=
<a href=3D"mailto:fluffy@cisco.com" target=3D"_blank">fluffy@cisco.com</a>&=
gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px =
0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);bord=
er-left-style:solid;padding-left:1ex"><div><div><br>
&gt; On Mar 5, 2015, at 8:22 AM, Jonathan Lennox &lt;<a href=3D"mailto:jona=
than@vidyo.com" target=3D"_blank">jonathan@vidyo.com</a>&gt; wrote:<br>
&gt;<br>
&gt;<br>
&gt; On Mar 5, 2015, at 5:14 AM, Roman Shpount &lt;<a href=3D"mailto:roman@=
telurix.com" target=3D"_blank">roman@telurix.com</a>&gt; wrote:<br>
&gt;<br>
&gt;&gt; On Thu, Mar 5, 2015 at 4:52 AM, I=C3=B1aki Baz Castillo &lt;<a hre=
f=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt; wrote:<b=
r>
&gt;&gt; 2015-03-05 7:58 GMT+01:00 Christer Holmberg &lt;<a href=3D"mailto:=
christer.holmberg@ericsson.com" target=3D"_blank">christer.holmberg@ericsso=
n.com</a>&gt;:<br>
&gt;&gt; &gt; As far as I understand, sending of multiple ClientHellos, on =
different<br>
&gt;&gt; &gt; 5-tuples, would require a change to core DTLS (not only DTLS-=
SRTP), and who<br>
&gt;&gt; &gt; knows how long such work would take. Do we really want to add=
 such<br>
&gt;&gt; &gt; dependency to our deliveries at this point?<br>
&gt;&gt;<br>
&gt;&gt; Sorry, I think I didn&#39;t explain well.<br>
&gt;&gt;<br>
&gt;&gt; It is not about multiple DTLS connections, but about the fact that=
<br>
&gt;&gt; DTLS packets belonging to the *same* DTLS connection/association c=
an<br>
&gt;&gt; be carried over different 5-tuples. This may happen during agressi=
ve<br>
&gt;&gt; ICE nomination in which media (so the DTLS ClientHello) follows th=
e<br>
&gt;&gt; USE-CANDIDATE Binding request without even waiting for a response.=
<br>
&gt;&gt;<br>
&gt;&gt; When was it agreed that DTLS ClientHello can be sent before the co=
nnectivity check succeeded? I understand that this will increase the connec=
tion setup time, but I though that no data should be sent before the connec=
tivity check response (consent) from the remote party.<br>
&gt;<br>
&gt; As I understand the current rough consensus, the endpoint has to have =
received a successful response to a connectivity check on the 5-tuple, prov=
ing consent, but it doesn=E2=80=99t have to have received (or sent) a check=
 with a USE-CANDIDATE attribute in it yet.<br>
&gt;<br>
&gt; This was the outcome of our discussion =E2=80=9Cmerging&quot; regular =
and aggressive ICE nomination.<br>
<br>
</div></div>That matches my understanding of how things work.<br>
<div><div><br></div></div></blockquote><div>=C2=A0</div></div></div><div>Th=
at discussion is now codified in a new draft, <a href=3D"https://tools.ietf=
.org/html/draft-uberti-mmusic-nombis-00" target=3D"_blank">https://tools.ie=
tf.org/html/draft-uberti-mmusic-nombis-00</a>.</div><div><br></div><div>To =
handle the issue raised by the OP, I think we should either</div><div>1) Fi=
x the DTLS draft (and any others) to not be bound to a 5-tuple when ICE is =
used</div><div>2) Update ICE-bis to say that any language about 5-tuples is=
 invalidated by ICE, and should be treated as a virtual transport.</div><di=
v><br></div><div>#2 seems like the simplest path at this point, given that =
ICE-bis work is underway.</div></div></div></div></blockquote><div><br></di=
v><div>#2 seems like the right answer to me.</div><div><br></div><div>-Ekr<=
/div><div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0=
 .8ex;border-left:1px #ccc solid;padding-left:1ex">________________________=
_______________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

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From: Barry Dingle <btdingle@gmail.com>
Date: Tue, 10 Mar 2015 14:20:22 +1100
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To: Justin Uberti <juberti@google.com>
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-fec-01.txt
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Was the suggestion to change the Title to "WebRTC *Media* Forward Error
Correction Requirements" - because there is no mention of Data Channel FEC
in this draft - accepted?

If not, why not?

Barry Dingle
= = =

On Tue, Mar 10, 2015 at 11:36 AM, Justin Uberti <juberti@google.com> wrote:

> [fixing cc list]
>
> On Mon, Mar 9, 2015 at 4:36 PM, Justin Uberti <juberti@google.com> wrote:
>
>> This version incorporates the feedback from Honolulu, as well as several
>> comments from the list.
>>
>> The Honolulu feedback suggested removing support for redundant audio with
>> codecs like PCMU, but I did get a request for this on the list, so I left
>> it in.
>>
>> On Sun, Mar 8, 2015 at 9:47 PM, <internet-drafts@ietf.org> wrote:
>>
>>>
>>> A New Internet-Draft is available from the on-line Internet-Drafts
>>> directories.
>>>  This draft is a work item of the Real-Time Communication in
>>> WEB-browsers Working Group of the IETF.
>>>
>>>         Title           : WebRTC Forward Error Correction Requirements
>>>         Author          : Justin Uberti
>>>         Filename        : draft-ietf-rtcweb-fec-01.txt
>>>         Pages           : 7
>>>         Date            : 2015-03-08
>>>
>>> Abstract:
>>>    This document provides information and requirements for how Forward
>>>    Error Correction (FEC) should be used by WebRTC applications.
>>>
>>>
>>> The IETF datatracker status page for this draft is:
>>> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/
>>>
>>> There's also a htmlized version available at:
>>> http://tools.ietf.org/html/draft-ietf-rtcweb-fec-01
>>>
>>> A diff from the previous version is available at:
>>> http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-fec-01
>>>
>>>
>>> Please note that it may take a couple of minutes from the time of
>>> submission
>>> until the htmlized version and diff are available at tools.ietf.org.
>>>
>>> Internet-Drafts are also available by anonymous FTP at:
>>> ftp://ftp.ietf.org/internet-drafts/
>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>
>>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--001a11349192c435940510e6a27c
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div>Was the suggestion to change the Title to &quot;WebRT=
C <b>Media</b> Forward Error Correction Requirements&quot; - because there =
is no mention of Data Channel FEC in this draft - accepted? <br><br></div>I=
f not, why not?<br><div class=3D"gmail_extra"><br clear=3D"all"><div><div c=
lass=3D"gmail_signature"><div dir=3D"ltr">Barry Dingle<br></div></div></div=
>=3D =3D =3D <br><br><div class=3D"gmail_quote">On Tue, Mar 10, 2015 at 11:=
36 AM, Justin Uberti <span dir=3D"ltr">&lt;<a href=3D"mailto:juberti@google=
.com" target=3D"_blank">juberti@google.com</a>&gt;</span> wrote:<br><blockq=
uote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc =
solid;padding-left:1ex"><div dir=3D"ltr">[fixing cc list]</div><div class=
=3D"HOEnZb"><div class=3D"h5"><div class=3D"gmail_extra"><br><div class=3D"=
gmail_quote">On Mon, Mar 9, 2015 at 4:36 PM, Justin Uberti <span dir=3D"ltr=
">&lt;<a href=3D"mailto:juberti@google.com" target=3D"_blank">juberti@googl=
e.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"l=
tr">This version incorporates the feedback from Honolulu, as well as severa=
l comments from the list.<div><br></div><div>The Honolulu feedback suggeste=
d removing support for redundant audio with codecs like PCMU, but I did get=
 a request for this on the list, so I left it in.</div></div><div><div><div=
 class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Sun, Mar 8, 2015 a=
t 9:47 PM,  <span dir=3D"ltr">&lt;<a href=3D"mailto:internet-drafts@ietf.or=
g" target=3D"_blank">internet-drafts@ietf.org</a>&gt;</span> wrote:<br><blo=
ckquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #c=
cc solid;padding-left:1ex"><br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
=C2=A0This draft is a work item of the Real-Time Communication in WEB-brows=
ers Working Group of the IETF.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Title=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 WebRTC Forward Error Correction Requirements<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Author=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : Just=
in Uberti<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Filename=C2=A0 =C2=A0 =C2=A0 =C2=A0 : draft-iet=
f-rtcweb-fec-01.txt<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Pages=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 7<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Date=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :=
 2015-03-08<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0This document provides information and requirements for how Fo=
rward<br>
=C2=A0 =C2=A0Error Correction (FEC) should be used by WebRTC applications.<=
br>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/" target=
=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/</a><br>
<br>
There&#39;s also a htmlized version available at:<br>
<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-fec-01" target=3D"_=
blank">http://tools.ietf.org/html/draft-ietf-rtcweb-fec-01</a><br>
<br>
A diff from the previous version is available at:<br>
<a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-fec-01" tar=
get=3D"_blank">http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-fec-01<=
/a><br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp://ftp=
.ietf.org/internet-drafts/</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div>
</div></div></blockquote></div><br></div>
</div></div><br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

--001a11349192c435940510e6a27c--


From nobody Mon Mar  9 22:46:52 2015
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From: Justin Uberti <juberti@google.com>
Date: Mon, 9 Mar 2015 22:46:27 -0700
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-fec-01.txt
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--001a113f29bc2f31b50510e8ad9e
Content-Type: text/plain; charset=UTF-8

Based on the discussion on the list, I elected to include a section on
Application FEC:
https://tools.ietf.org/html/draft-ietf-rtcweb-fec-01#section-6

On Mon, Mar 9, 2015 at 8:20 PM, Barry Dingle <btdingle@gmail.com> wrote:

> Was the suggestion to change the Title to "WebRTC *Media* Forward Error
> Correction Requirements" - because there is no mention of Data Channel FEC
> in this draft - accepted?
>
> If not, why not?
>
> Barry Dingle
> = = =
>
> On Tue, Mar 10, 2015 at 11:36 AM, Justin Uberti <juberti@google.com>
> wrote:
>
>> [fixing cc list]
>>
>> On Mon, Mar 9, 2015 at 4:36 PM, Justin Uberti <juberti@google.com> wrote:
>>
>>> This version incorporates the feedback from Honolulu, as well as several
>>> comments from the list.
>>>
>>> The Honolulu feedback suggested removing support for redundant audio
>>> with codecs like PCMU, but I did get a request for this on the list, so I
>>> left it in.
>>>
>>> On Sun, Mar 8, 2015 at 9:47 PM, <internet-drafts@ietf.org> wrote:
>>>
>>>>
>>>> A New Internet-Draft is available from the on-line Internet-Drafts
>>>> directories.
>>>>  This draft is a work item of the Real-Time Communication in
>>>> WEB-browsers Working Group of the IETF.
>>>>
>>>>         Title           : WebRTC Forward Error Correction Requirements
>>>>         Author          : Justin Uberti
>>>>         Filename        : draft-ietf-rtcweb-fec-01.txt
>>>>         Pages           : 7
>>>>         Date            : 2015-03-08
>>>>
>>>> Abstract:
>>>>    This document provides information and requirements for how Forward
>>>>    Error Correction (FEC) should be used by WebRTC applications.
>>>>
>>>>
>>>> The IETF datatracker status page for this draft is:
>>>> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/
>>>>
>>>> There's also a htmlized version available at:
>>>> http://tools.ietf.org/html/draft-ietf-rtcweb-fec-01
>>>>
>>>> A diff from the previous version is available at:
>>>> http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-fec-01
>>>>
>>>>
>>>> Please note that it may take a couple of minutes from the time of
>>>> submission
>>>> until the htmlized version and diff are available at tools.ietf.org.
>>>>
>>>> Internet-Drafts are also available by anonymous FTP at:
>>>> ftp://ftp.ietf.org/internet-drafts/
>>>>
>>>> _______________________________________________
>>>> rtcweb mailing list
>>>> rtcweb@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>
>>>
>>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>

--001a113f29bc2f31b50510e8ad9e
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Based on the discussion on the list, I elected to include =
a section on Application FEC:=C2=A0<a href=3D"https://tools.ietf.org/html/d=
raft-ietf-rtcweb-fec-01#section-6">https://tools.ietf.org/html/draft-ietf-r=
tcweb-fec-01#section-6</a></div><div class=3D"gmail_extra"><br><div class=
=3D"gmail_quote">On Mon, Mar 9, 2015 at 8:20 PM, Barry Dingle <span dir=3D"=
ltr">&lt;<a href=3D"mailto:btdingle@gmail.com" target=3D"_blank">btdingle@g=
mail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=
=3D"ltr"><div>Was the suggestion to change the Title to &quot;WebRTC <b>Med=
ia</b> Forward Error Correction Requirements&quot; - because there is no me=
ntion of Data Channel FEC in this draft - accepted? <br><br></div>If not, w=
hy not?<span class=3D"HOEnZb"><font color=3D"#888888"><br></font></span><di=
v class=3D"gmail_extra"><span class=3D"HOEnZb"><font color=3D"#888888"><br =
clear=3D"all"><div><div><div dir=3D"ltr">Barry Dingle<br></div></div></div>=
=3D =3D =3D <br></font></span><div><div class=3D"h5"><br><div class=3D"gmai=
l_quote">On Tue, Mar 10, 2015 at 11:36 AM, Justin Uberti <span dir=3D"ltr">=
&lt;<a href=3D"mailto:juberti@google.com" target=3D"_blank">juberti@google.=
com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr=
">[fixing cc list]</div><div><div><div class=3D"gmail_extra"><br><div class=
=3D"gmail_quote">On Mon, Mar 9, 2015 at 4:36 PM, Justin Uberti <span dir=3D=
"ltr">&lt;<a href=3D"mailto:juberti@google.com" target=3D"_blank">juberti@g=
oogle.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=
=3D"ltr">This version incorporates the feedback from Honolulu, as well as s=
everal comments from the list.<div><br></div><div>The Honolulu feedback sug=
gested removing support for redundant audio with codecs like PCMU, but I di=
d get a request for this on the list, so I left it in.</div></div><div><div=
><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Sun, Mar 8, 2=
015 at 9:47 PM,  <span dir=3D"ltr">&lt;<a href=3D"mailto:internet-drafts@ie=
tf.org" target=3D"_blank">internet-drafts@ietf.org</a>&gt;</span> wrote:<br=
><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1=
px #ccc solid;padding-left:1ex"><br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
=C2=A0This draft is a work item of the Real-Time Communication in WEB-brows=
ers Working Group of the IETF.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Title=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 WebRTC Forward Error Correction Requirements<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Author=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : Just=
in Uberti<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Filename=C2=A0 =C2=A0 =C2=A0 =C2=A0 : draft-iet=
f-rtcweb-fec-01.txt<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Pages=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 7<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Date=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :=
 2015-03-08<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0This document provides information and requirements for how Fo=
rward<br>
=C2=A0 =C2=A0Error Correction (FEC) should be used by WebRTC applications.<=
br>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/" target=
=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/</a><br>
<br>
There&#39;s also a htmlized version available at:<br>
<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-fec-01" target=3D"_=
blank">http://tools.ietf.org/html/draft-ietf-rtcweb-fec-01</a><br>
<br>
A diff from the previous version is available at:<br>
<a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-fec-01" tar=
get=3D"_blank">http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-fec-01<=
/a><br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp://ftp=
.ietf.org/internet-drafts/</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div>
</div></div></blockquote></div><br></div>
</div></div><br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div></div></div>
</blockquote></div><br></div>

--001a113f29bc2f31b50510e8ad9e--


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From: Barry Dingle <btdingle@gmail.com>
Date: Tue, 10 Mar 2015 17:43:38 +1100
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To: Justin Uberti <juberti@google.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-fec-01.txt
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Thanks Justin.

Barry Dingle



On Tue, Mar 10, 2015 at 4:46 PM, Justin Uberti <juberti@google.com> wrote:

> Based on the discussion on the list, I elected to include a section on
> Application FEC:
> https://tools.ietf.org/html/draft-ietf-rtcweb-fec-01#section-6
>
> On Mon, Mar 9, 2015 at 8:20 PM, Barry Dingle <btdingle@gmail.com> wrote:
>
>> Was the suggestion to change the Title to "WebRTC *Media* Forward Error
>> Correction Requirements" - because there is no mention of Data Channel FEC
>> in this draft - accepted?
>>
>> If not, why not?
>>
>> Barry Dingle
>> = = =
>>
>> On Tue, Mar 10, 2015 at 11:36 AM, Justin Uberti <juberti@google.com>
>> wrote:
>>
>>> [fixing cc list]
>>>
>>> On Mon, Mar 9, 2015 at 4:36 PM, Justin Uberti <juberti@google.com>
>>> wrote:
>>>
>>>> This version incorporates the feedback from Honolulu, as well as
>>>> several comments from the list.
>>>>
>>>> The Honolulu feedback suggested removing support for redundant audio
>>>> with codecs like PCMU, but I did get a request for this on the list, so I
>>>> left it in.
>>>>
>>>> On Sun, Mar 8, 2015 at 9:47 PM, <internet-drafts@ietf.org> wrote:
>>>>
>>>>>
>>>>> A New Internet-Draft is available from the on-line Internet-Drafts
>>>>> directories.
>>>>>  This draft is a work item of the Real-Time Communication in
>>>>> WEB-browsers Working Group of the IETF.
>>>>>
>>>>>         Title           : WebRTC Forward Error Correction Requirements
>>>>>         Author          : Justin Uberti
>>>>>         Filename        : draft-ietf-rtcweb-fec-01.txt
>>>>>         Pages           : 7
>>>>>         Date            : 2015-03-08
>>>>>
>>>>> Abstract:
>>>>>    This document provides information and requirements for how Forward
>>>>>    Error Correction (FEC) should be used by WebRTC applications.
>>>>>
>>>>>
>>>>> The IETF datatracker status page for this draft is:
>>>>> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/
>>>>>
>>>>> There's also a htmlized version available at:
>>>>> http://tools.ietf.org/html/draft-ietf-rtcweb-fec-01
>>>>>
>>>>> A diff from the previous version is available at:
>>>>> http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-fec-01
>>>>>
>>>>>
>>>>> Please note that it may take a couple of minutes from the time of
>>>>> submission
>>>>> until the htmlized version and diff are available at tools.ietf.org.
>>>>>
>>>>> Internet-Drafts are also available by anonymous FTP at:
>>>>> ftp://ftp.ietf.org/internet-drafts/
>>>>>
>>>>> _______________________________________________
>>>>> rtcweb mailing list
>>>>> rtcweb@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>>
>>>>
>>>>
>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>>
>>
>

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<div dir=3D"ltr">Thanks Justin.<br><div class=3D"gmail_extra"><br clear=3D"=
all"><div><div class=3D"gmail_signature"><div dir=3D"ltr">Barry Dingle<br><=
br><br></div></div></div>
<br><div class=3D"gmail_quote">On Tue, Mar 10, 2015 at 4:46 PM, Justin Uber=
ti <span dir=3D"ltr">&lt;<a href=3D"mailto:juberti@google.com" target=3D"_b=
lank">juberti@google.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmai=
l_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left=
:1ex"><div dir=3D"ltr">Based on the discussion on the list, I elected to in=
clude a section on Application FEC:=C2=A0<a href=3D"https://tools.ietf.org/=
html/draft-ietf-rtcweb-fec-01#section-6" target=3D"_blank">https://tools.ie=
tf.org/html/draft-ietf-rtcweb-fec-01#section-6</a></div><div class=3D"HOEnZ=
b"><div class=3D"h5"><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Mon, Mar 9, 2015 at 8:20 PM, Barry Dingle <span dir=3D"ltr">&lt;<a h=
ref=3D"mailto:btdingle@gmail.com" target=3D"_blank">btdingle@gmail.com</a>&=
gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 =
0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div>W=
as the suggestion to change the Title to &quot;WebRTC <b>Media</b> Forward =
Error Correction Requirements&quot; - because there is no mention of Data C=
hannel FEC in this draft - accepted? <br><br></div>If not, why not?<span><f=
ont color=3D"#888888"><br></font></span><div class=3D"gmail_extra"><span><f=
ont color=3D"#888888"><br clear=3D"all"><div><div><div dir=3D"ltr">Barry Di=
ngle<br></div></div></div>=3D =3D =3D <br></font></span><div><div><br><div =
class=3D"gmail_quote">On Tue, Mar 10, 2015 at 11:36 AM, Justin Uberti <span=
 dir=3D"ltr">&lt;<a href=3D"mailto:juberti@google.com" target=3D"_blank">ju=
berti@google.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote"=
 style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><d=
iv dir=3D"ltr">[fixing cc list]</div><div><div><div class=3D"gmail_extra"><=
br><div class=3D"gmail_quote">On Mon, Mar 9, 2015 at 4:36 PM, Justin Uberti=
 <span dir=3D"ltr">&lt;<a href=3D"mailto:juberti@google.com" target=3D"_bla=
nk">juberti@google.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex"><div dir=3D"ltr">This version incorporates the feedback from Honolulu, =
as well as several comments from the list.<div><br></div><div>The Honolulu =
feedback suggested removing support for redundant audio with codecs like PC=
MU, but I did get a request for this on the list, so I left it in.</div></d=
iv><div><div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On S=
un, Mar 8, 2015 at 9:47 PM,  <span dir=3D"ltr">&lt;<a href=3D"mailto:intern=
et-drafts@ietf.org" target=3D"_blank">internet-drafts@ietf.org</a>&gt;</spa=
n> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;b=
order-left:1px #ccc solid;padding-left:1ex"><br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
=C2=A0This draft is a work item of the Real-Time Communication in WEB-brows=
ers Working Group of the IETF.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Title=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 WebRTC Forward Error Correction Requirements<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Author=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : Just=
in Uberti<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Filename=C2=A0 =C2=A0 =C2=A0 =C2=A0 : draft-iet=
f-rtcweb-fec-01.txt<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Pages=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 7<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Date=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :=
 2015-03-08<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0This document provides information and requirements for how Fo=
rward<br>
=C2=A0 =C2=A0Error Correction (FEC) should be used by WebRTC applications.<=
br>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/" target=
=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/</a><br>
<br>
There&#39;s also a htmlized version available at:<br>
<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-fec-01" target=3D"_=
blank">http://tools.ietf.org/html/draft-ietf-rtcweb-fec-01</a><br>
<br>
A diff from the previous version is available at:<br>
<a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-fec-01" tar=
get=3D"_blank">http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-fec-01<=
/a><br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp://ftp=
.ietf.org/internet-drafts/</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div>
</div></div></blockquote></div><br></div>
</div></div><br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div></div></div>
</blockquote></div><br></div>
</div></div></blockquote></div><br></div></div>

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From nobody Tue Mar 10 06:40:18 2015
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To: "Hutton, Andrew" <andrew.hutton@unify.com>,  Alan Johnston <alan.b.johnston@gmail.com>, Benjamin Schwartz <bemasc@webrtc.org>
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Subject: Re: [rtcweb] RETURN-05
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Le 2015-03-10 09:39, Hutton, Andrew a écrit :
> I agree with Alan’s assessment and also believe that we should move
> forward with this draft.

+1

Simon


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Subject: Re: [rtcweb] RETURN-05
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On 3/9/2015 1:39 PM, Alan Johnston wrote:
> I think RETURN is essential for WebRTC to work well across enterprise=20
> borders.  We will increasingly see border TURN servers used by=20
> enterprises to manage and control WebRTC usage, most likely as=20
> additions to existing firewalls and SBCs.  Without RETURN, the usage=20
> of border TURN servers will result in loss of functionality in WebRTC=20
> applications that use a TURN server to provide features in addition to=20
> simple NAT traversal.  RETURN should work perfectly to allow both TURN=20
> servers to work, while still allowing end-to-end encrypted and=20
> authenticated media flows.
>
> I=92m not aware of any other solutions to the enterprise requirements=20
> identified in Section 3.3.5 of=20
> draft-ietf-rtcweb-use-cases-and-requirements.  I think we need to move=20
> forward with this draft in RTCWEB.

I agree.  I'll note that RETURN covers the auto-configuration portion of=20
3.3.5, but browsers (and other WebRTC-compatible devices by implication)=20
should handle manual configurations as well.  (Just a note, it doesn't=20
affect the spec or RETURN I believe.)

    It must be possible to configure the browsers used in the enterprise=20
with
    network specific STUN and TURN servers.  This should be possible to
    achieve by auto-configuration methods.

--=20
Randell Jesup -- rjesup a t mozilla d o t com


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 10 Mar 2015 16:26:31 +0100
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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2015-03-10 1:49 GMT+01:00 Eric Rescorla <ekr@rtfm.com>:
> #2 seems like the right answer to me.

+1. Other specs should then update to consider the virtual transport
when ICE is in use.


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


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To: "'Randell Jesup'" <randell-ietf@jesup.org>, <rtcweb@ietf.org>, "'Alan Johnston'" <alan.b.johnston@gmail.com>, "'Hutton, Andrew'" <andrew.hutton@unify.com>, "'Simon Perreault'" <sperreault@jive.com>
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I agree with all of you that this draft is very important to move =
forward
with.

As stated under Figure 2 in the draft:
   ...This TURN server might be used by an enterprise, ISP,
   or home network to enable WebRTC media flows that would otherwise be
   blocked by the firewall, or to improve quality of service on flows
   that pass through this TURN server. =20

I see this (in combination with auto discovery of the "Border TURN =
server")
as necessary
to finally get a proper solution to the NAT/firewall traversal problem =
that
has plagued=20
real-time communication since its occurrence in the IP world.

I hope for quick test implementations in the WebRTC browsers.=20

/Karl


-----Ursprungligt meddelande-----
Fr=E5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=F6r Randell Jesup
Skickat: den 10 mars 2015 14:46
Till: rtcweb@ietf.org
=C4mne: Re: [rtcweb] RETURN-05

On 3/9/2015 1:39 PM, Alan Johnston wrote:
> I think RETURN is essential for WebRTC to work well across enterprise=20
> borders.  We will increasingly see border TURN servers used by=20
> enterprises to manage and control WebRTC usage, most likely as=20
> additions to existing firewalls and SBCs.  Without RETURN, the usage=20
> of border TURN servers will result in loss of functionality in WebRTC=20
> applications that use a TURN server to provide features in addition to =

> simple NAT traversal.  RETURN should work perfectly to allow both TURN =

> servers to work, while still allowing end-to-end encrypted and=20
> authenticated media flows.
>
> I=92m not aware of any other solutions to the enterprise requirements=20
> identified in Section 3.3.5 of=20
> draft-ietf-rtcweb-use-cases-and-requirements.  I think we need to move =

> forward with this draft in RTCWEB.

I agree.  I'll note that RETURN covers the auto-configuration portion of
3.3.5, but browsers (and other WebRTC-compatible devices by implication)
should handle manual configurations as well.  (Just a note, it doesn't
affect the spec or RETURN I believe.)

    It must be possible to configure the browsers used in the enterprise
with
    network specific STUN and TURN servers.  This should be possible to
    achieve by auto-configuration methods.

--
Randell Jesup -- rjesup a t mozilla d o t com

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


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>, Cullen Jennings <fluffy@cisco.com>
Thread-Topic: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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Cc: Jonathan Lennox <jonathan@vidyo.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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Date: Wed, 11 Mar 2015 14:09:21 +0100
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [rtcweb] WGLC for draft-ietf-rtcweb-video-04
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On 2015-02-27 00:37, Sean Turner wrote:
> Dear WG,
> 
> This message initiates a WGLC (Working Group Last Call) for the "WebRTC Video Processing and Codec Requirements” draft:
> 
> http://datatracker.ietf.org/doc/draft-ietf-rtcweb-video/
> 
> Please review the draft in detail.  This WGLC will end on 13 March, 2015.
> 

Hi,

I have reviewed the draft and have some few comments.

1. Section 5: Spelling error "WebRTC Brower"

2. Section 6:

   SDP allows for codec-independent indication of preferred video
   resolutions using the mechanism described in [RFC6236]. If a
   recipient of video indicates a receiving resolution, the sender
   SHOULD accommodate this resolution, as the receiver may not be
   capable of handling higher resolutions.

I would note that this document does not indicate an expectation of a
WebRTC Browser or non-browser to actually support a=imageattr. I would
have expected a clear formulation using RFC 2119 words on the need to
support this.

I also note that JSEP-09 lists a=imageattr as an open issue.

I think this unclarity needs to be improved. Either be explicit about
JSEP defining the need for implementing the attribute, or define in this
document, and JSEP's open issue will be resolved.


3. Section 7:

Implementers should consider
   whether the use of variable bit rate video codecs are appropriate for
   their application based on [RFC6562].

I don't quite understand the reference to RFC6562 here. That discusses
that audio codecs that have a variable rate based on the audio input can
leak information about what is said. This is primarily done through
voicing classifications on the frame basis that reveals the spoken
words. I fail to understand how that applies to video.



4. Section 10.1: This list to be a bit bloated. I find no reference to
the following reference, which also ID-nits find.

  == Unused Reference: 'RFC4175' is defined on line 345, but no explicit
     reference was found in the text

  == Unused Reference: 'RFC4421' is defined on line 348, but no explicit
     reference was found in the text

  == Unused Reference: 'RFC5104' is defined on line 352, but no explicit
     reference was found in the text


Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 11 Mar 2015 14:42:45 +0100
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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2015-03-11 13:49 GMT+01:00 Christer Holmberg <christer.holmberg@ericsson.co=
m>:
>>To handle the issue raised by the OP, I think we should either
>
>>1) Fix the DTLS draft (and any others) to not be bound to a 5-tuple when
>> ICE is used
>
>>2) Update ICE-bis to say that any language about 5-tuples is invalidated =
by
>> ICE, and should be treated as a >virtual transport.
>
>
>
> When you say =E2=80=9Cany language=E2=80=9D, do you refer specifically to=
 DTLS, or to
> whatever protocol is used together with ICE?

Clearly he means "whatever protocol is used together with ICE".


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


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Date: Wed, 11 Mar 2015 10:10:15 -0400
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From: Roman Shpount <roman@telurix.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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--001a1141ca9e95809a051103d3c6
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On Wed, Mar 11, 2015 at 9:55 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> So, we assume that whatever protocol, existing and future, will work with
> a "virtual connection"?
>
> E.g. in case of TCP, you WILL have separate connections for each TCP
> candidate - each using a single 5-tuple. We can't say that, with ICE, a
> single TCP connection can suddenly span multiple 5-tuples (one for each
> candidate pair), can we?
>
>
I think there is some confusion between the underlying link protocol used
by candidates and the protocol running on top of ICE. I do not think TCP is
defined as a protocol that can operate on top of ICE. It is defined in RFC
6544 using framing from RFC 4571, as a protocol for underlying links.
Whatever protocol is defined on top of ICE should support operation on top
of packetized communication protocol provided by ICE and should define how
its packets can be demuxed from STUN packets used for ICE connectivity
checks. So far things like RTP, DTLS, and SCTP fit such requirements. New
things can be defined in the future. When they do, they should treat ICE a
virtual communication channel that provides unreliable packet transport
with no order guarantees which can span multiple 5-tuples.
_____________
Roman Shpount

--001a1141ca9e95809a051103d3c6
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Wed, Mar 11, 2015 at 9:55 AM, Christer Holmberg <span dir=3D"ltr">&=
lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chri=
ster.holmberg@ericsson.com</a>&gt;</span> wrote:<br></div></div><div class=
=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px =
0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-l=
eft-style:solid;padding-left:1ex">So, we assume that whatever protocol, exi=
sting and future, will work with a &quot;virtual connection&quot;?<br>
<br>
E.g. in case of TCP, you WILL have separate connections for each TCP candid=
ate - each using a single 5-tuple. We can&#39;t say that, with ICE, a singl=
e TCP connection can suddenly span multiple 5-tuples (one for each candidat=
e pair), can we?<br>
<br></blockquote><div><br></div><div>I think there is some confusion betwee=
n the underlying link protocol used by candidates and the protocol running =
on top of ICE. I do not think TCP is defined as a protocol that can operate=
 on top of ICE. It is defined in RFC 6544 using framing from RFC 4571, as a=
 protocol for underlying links. Whatever protocol is defined on top of ICE =
should support operation on top of packetized communication protocol provid=
ed by ICE and should define how its packets can be demuxed from STUN packet=
s used for ICE connectivity checks. So far things like RTP, DTLS, and SCTP =
fit such requirements. New things can be defined in the future. When they d=
o, they should treat ICE a virtual communication channel that provides unre=
liable packet transport with no order guarantees which can span multiple 5-=
tuples.</div><div><div class=3D"gmail_signature">_____________<br>Roman Shp=
ount</div></div><br><div>=C2=A0</div></div></div></div>

--001a1141ca9e95809a051103d3c6--


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 11 Mar 2015 15:14:28 +0100
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To: Roman Shpount <roman@telurix.com>
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Cc: Cullen Jennings <fluffy@cisco.com>, Jonathan Lennox <jonathan@vidyo.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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2015-03-11 15:10 GMT+01:00 Roman Shpount <roman@telurix.com>:
> I think there is some confusion between the underlying link protocol used=
 by
> candidates and the protocol running on top of ICE. I do not think TCP is
> defined as a protocol that can operate on top of ICE. It is defined in RF=
C
> 6544 using framing from RFC 4571, as a protocol for underlying links.
> Whatever protocol is defined on top of ICE should support operation on to=
p
> of packetized communication protocol provided by ICE and should define ho=
w
> its packets can be demuxed from STUN packets used for ICE connectivity
> checks. So far things like RTP, DTLS, and SCTP fit such requirements. New
> things can be defined in the future. When they do, they should treat ICE =
a
> virtual communication channel that provides unreliable packet transport w=
ith
> no order guarantees which can span multiple 5-tuples.

Exactly. Please Christer, don't go to layer 4 protocols. Neither IPv6
can run over an ICE initiated "virtual transport" ;)


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


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: =?utf-8?B?ScOxYWtpIEJheiBDYXN0aWxsbw==?= <ibc@aliax.net>, Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 11 Mar 2015 15:20:37 +0100
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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2015-03-11 15:19 GMT+01:00 Christer Holmberg <christer.holmberg@ericsson.co=
m>:
> I am not the one saying "whatever protocol" :)
>
> My point is that we need to be more clear about the scope - we can't just=
 say "Take whatever protocol and do a "5-tuple"/"virtual connection" search=
-replace." :)

"Whatever protocol suitable to run over a ICE initiated path"


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


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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From nobody Wed Mar 11 07:28:47 2015
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: =?utf-8?B?ScOxYWtpIEJheiBDYXN0aWxsbw==?= <ibc@aliax.net>
Thread-Topic: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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Date: Wed, 11 Mar 2015 14:28:40 +0000
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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From nobody Wed Mar 11 07:32:09 2015
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Date: Wed, 11 Mar 2015 10:32:02 -0400
Message-ID: <CAD5OKxs1grSqAG32mf__wtsjpo68jZmKonbd+EsJmYNsDHUbFQ@mail.gmail.com>
From: Roman Shpount <roman@telurix.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: Cullen Jennings <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, Jonathan Lennox <jonathan@vidyo.com>
Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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--90e6ba6147e27bd01f051104216d
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On Wed, Mar 11, 2015 at 10:24 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> I assume you mean SCTP-over-DTLS? Usage of "plain" SCTP with ICE is not
> defined, as far as I know.
>

You are correct.


>
> > New things can be defined in the future. When they do, they should treat
> ICE a virtual communication channel that
> > provides unreliable packet transport with no order guarantees which can
> span multiple 5-tuples.
>
> Then the scope of what we discuss now should not be "whatever protocol" -
> it should be the specific protocols we are discussing.
>
>
I think ICE-bis should define protocol requirements for the protocols that
can run on top of ICE, which includes:
1. Ability to run over unreliable packet based transport with no order
guarantees
2. Ability to demux with STUN packets
3. Not t make any assumption about IP addresses, ports, or other transport
level protocols attributes such as TOS.
_____________
Roman Shpount

--90e6ba6147e27bd01f051104216d
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On W=
ed, Mar 11, 2015 at 10:24 AM, Christer Holmberg <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.hol=
mberg@ericsson.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quot=
e" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-colo=
r:rgb(204,204,204);border-left-style:solid;padding-left:1ex">I assume you m=
ean SCTP-over-DTLS? Usage of &quot;plain&quot; SCTP with ICE is not defined=
, as far as I know.<br></blockquote><div><br></div><div>You are correct.</d=
iv><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0=
px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);borde=
r-left-style:solid;padding-left:1ex">
<span class=3D""><br>
&gt; New things can be defined in the future. When they do, they should tre=
at ICE a virtual communication channel that<br>
&gt; provides unreliable packet transport with no order guarantees which ca=
n span multiple 5-tuples.<br>
<br>
</span>Then the scope of what we discuss now should not be &quot;whatever p=
rotocol&quot; - it should be the specific protocols we are discussing.<br>
<br></blockquote><div><br></div><div>I think ICE-bis should define protocol=
 requirements for the protocols that can run on top of ICE, which includes:=
</div><div>1. Ability to run over unreliable packet based transport with no=
 order guarantees</div><div>2. Ability to demux with STUN packets</div><div=
>3. Not t make any assumption about IP addresses, ports, or other transport=
 level protocols attributes such as TOS.=C2=A0</div><div><div class=3D"gmai=
l_signature">_____________<br>Roman Shpount</div></div><div>=C2=A0</div></d=
iv></div></div>

--90e6ba6147e27bd01f051104216d--


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References: <54F74B02.1070902@jive.com> <CAD5OKxs8JYG3-Vvndi59ZrdPE7UTj22ozD4tcWTHgzWrHv=q7Q@mail.gmail.com> <54F756B2.60408@jive.com> <7594FB04B1934943A5C02806D1A2204B1D726AD8@ESESSMB209.ericsson.se> <CAD5OKxu7py3HbrFjxTDZS5ECFzx7vd=wpjve-gT6gWwksjEu+g@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D726B71@ESESSMB209.ericsson.se> <CABcZeBO1O6sA8MqvWkCDu3RPLz5-P2G65Us28i0baOavDnRT7Q@mail.gmail.com> <CAD5OKxuWCdgMR5Kxjv9BSwZ3Jm9kGXx9Pi-9FrfsnuQZ_91jAA@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D726DC1@ESESSMB209.ericsson.se> <CALiegfkipJhsy7-40+=d9xMUf4RJGdn3_fABL3NN2KuFNvS2BA@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D727570@ESESSMB209.ericsson.se> <CALiegfmfvz3NWSjcovGBytiOTbR6kFfyh0vx5cXoMJtytfGzRA@mail.gmail.com> <CAD5OKxsu3D0xHY-zYbDu1hyH_+4=3mWDvW2i98WCVZ+29BpKCw@mail.gmail.com> <CA5E97EE-99F8-44D8-B05B-C9EFDED1A9BB@vidyo.com> <2F467A7E-7A6C-4B1B-985A-0D9C089BE973@cisco.com> <CAOJ7v-1TjZOZ5G31vy_Gt73ADGLRay1RHVeMi=H6Q4=N1b6HLA@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D7367A0@ESESSMB209.ericsson.se> <CALiegfmyp=v6thk4eLz7nL1BHh2Qj7jmC84tdG7ufg8HPXsVKA@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D7369C9@ESESSMB209.ericsson.se> <CAD5OKxtCswToNzoZnnqJ5M66mjNjKJoA++WYNqN5155n+CWXsA@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D736AC0@ESESSMB209.ericsson.se> <CAD5OKxs1grSqAG32mf__wtsjpo68jZmKonbd+EsJmYNsDHUbFQ@mail.gmail.com>
From: Justin Uberti <juberti@google.com>
Date: Wed, 11 Mar 2015 08:33:47 -0700
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To: Roman Shpount <roman@telurix.com>
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Cc: Cullen Jennings <fluffy@cisco.com>, Jonathan Lennox <jonathan@vidyo.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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--20cf3011d8fb8363df051104ffe8
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On Wed, Mar 11, 2015 at 7:32 AM, Roman Shpount <roman@telurix.com> wrote:

> On Wed, Mar 11, 2015 at 10:24 AM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
>
>> I assume you mean SCTP-over-DTLS? Usage of "plain" SCTP with ICE is not
>> defined, as far as I know.
>>
>
> You are correct.
>
>
>>
>> > New things can be defined in the future. When they do, they should
>> treat ICE a virtual communication channel that
>> > provides unreliable packet transport with no order guarantees which can
>> span multiple 5-tuples.
>>
>> Then the scope of what we discuss now should not be "whatever protocol" -
>> it should be the specific protocols we are discussing.
>>
>>
> I think ICE-bis should define protocol requirements for the protocols that
> can run on top of ICE, which includes:
> 1. Ability to run over unreliable packet based transport with no order
> guarantees
> 2. Ability to demux with STUN packets
> 3. Not t make any assumption about IP addresses, ports, or other transport
> level protocols attributes such as TOS.
>
>
I think these are good criteria. Note that TCP would meet these criteria,
and I see no problem running TCP atop ICE (we used to do this in an old
version of our data channel code).

HTTP, on the other hand, would not meet criterion #1.

--20cf3011d8fb8363df051104ffe8
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Wed, Mar 11, 2015 at 7:32 AM, Roman Shpount <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.com</a>&=
gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 =
0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div c=
lass=3D"gmail_extra"><div class=3D"gmail_quote"><span class=3D"">On Wed, Ma=
r 11, 2015 at 10:24 AM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=3D"=
mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmberg@=
ericsson.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" sty=
le=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(=
204,204,204);border-left-style:solid;padding-left:1ex">I assume you mean SC=
TP-over-DTLS? Usage of &quot;plain&quot; SCTP with ICE is not defined, as f=
ar as I know.<br></blockquote><div><br></div></span><div>You are correct.</=
div><span class=3D""><div>=C2=A0</div><blockquote class=3D"gmail_quote" sty=
le=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(=
204,204,204);border-left-style:solid;padding-left:1ex">
<span><br>
&gt; New things can be defined in the future. When they do, they should tre=
at ICE a virtual communication channel that<br>
&gt; provides unreliable packet transport with no order guarantees which ca=
n span multiple 5-tuples.<br>
<br>
</span>Then the scope of what we discuss now should not be &quot;whatever p=
rotocol&quot; - it should be the specific protocols we are discussing.<br>
<br></blockquote><div><br></div></span><div>I think ICE-bis should define p=
rotocol requirements for the protocols that can run on top of ICE, which in=
cludes:</div><div>1. Ability to run over unreliable packet based transport =
with no order guarantees</div><div>2. Ability to demux with STUN packets</d=
iv><div>3. Not t make any assumption about IP addresses, ports, or other tr=
ansport level protocols attributes such as TOS.=C2=A0</div><div><div><span =
class=3D"HOEnZb"><font color=3D"#888888"><br></font></span></div></div></di=
v></div></div></blockquote><div><br></div><div>I think these are good crite=
ria. Note that TCP would meet these criteria, and I see no problem running =
TCP atop ICE (we used to do this in an old version of our data channel code=
).</div><div><br></div><div>HTTP, on the other hand, would not meet criteri=
on #1.</div></div></div></div>

--20cf3011d8fb8363df051104ffe8--


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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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On 03/11/2015 03:33 PM, Justin Uberti wrote:
>
>
> On Wed, Mar 11, 2015 at 7:32 AM, Roman Shpount <roman@telurix.com
> <mailto:roman@telurix.com>> wrote:
>
>     On Wed, Mar 11, 2015 at 10:24 AM, Christer Holmberg
>     <christer.holmberg@ericsson.com
>     <mailto:christer.holmberg@ericsson.com>> wrote:
>
>         I assume you mean SCTP-over-DTLS? Usage of "plain" SCTP with
>         ICE is not defined, as far as I know.
>
>
>     You are correct.
>     =20
>
>
>         > New things can be defined in the future. When they do, they
>         should treat ICE a virtual communication channel that
>         > provides unreliable packet transport with no order
>         guarantees which can span multiple 5-tuples.
>
>         Then the scope of what we discuss now should not be "whatever
>         protocol" - it should be the specific protocols we are discussi=
ng.
>
>
>     I think ICE-bis should define protocol requirements for the
>     protocols that can run on top of ICE, which includes:
>     1. Ability to run over unreliable packet based transport with no
>     order guarantees
>     2. Ability to demux with STUN packets
>     3. Not t make any assumption about IP addresses, ports, or other
>     transport level protocols attributes such as TOS.=20
>
>
> I think these are good criteria. Note that TCP would meet these
> criteria, and I see no problem running TCP atop ICE (we used to do
> this in an old version of our data channel code).
>
> HTTP, on the other hand, would not meet criterion #1.

I seem to remember a draft I wrote once upon a time (Feb 2011).... I
called it a "datagram service".

draft-alvestrand-dispatch-rtcweb-datagram-01

The ability to demux with STUN packets was expressed here as:

   The datagram service is not completely transparent; in particular, it
   is not possible to carry a datagram where the two highest bits of the
   first octet are zero and octet 5 to 8 contain the value 0x2112A442,
   since these datagrams are reserved for use of the STUN protocol (RFC
   5389 section 6).

It didn't seem to warrant a special doc at the time, given responses, so
I dropped it.

Note that straight-up TCP on top of this model would require special
work, since the TCP header checksum covers the address fields. A
TCP-like protocol with different checksums would be trivial to define.







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    <div class="moz-cite-prefix">On 03/11/2015 03:33 PM, Justin Uberti
      wrote:<br>
    </div>
    <blockquote
cite="mid:CAOJ7v-3YypG1s9KXOCA+Fo58SuVuUk5-thcSc0k3N2j=4ZmJoA@mail.gmail.com"
      type="cite">
      <div dir="ltr"><br>
        <div class="gmail_extra"><br>
          <div class="gmail_quote">On Wed, Mar 11, 2015 at 7:32 AM,
            Roman Shpount <span dir="ltr">&lt;<a moz-do-not-send="true"
                href="mailto:roman@telurix.com" target="_blank">roman@telurix.com</a>&gt;</span>
            wrote:<br>
            <blockquote class="gmail_quote" style="margin:0 0 0
              .8ex;border-left:1px #ccc solid;padding-left:1ex">
              <div dir="ltr">
                <div class="gmail_extra">
                  <div class="gmail_quote"><span class="">On Wed, Mar
                      11, 2015 at 10:24 AM, Christer Holmberg <span
                        dir="ltr">&lt;<a moz-do-not-send="true"
                          href="mailto:christer.holmberg@ericsson.com"
                          target="_blank">christer.holmberg@ericsson.com</a>&gt;</span>
                      wrote:<br>
                      <blockquote class="gmail_quote" style="margin:0px
                        0px 0px
0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">I
                        assume you mean SCTP-over-DTLS? Usage of "plain"
                        SCTP with ICE is not defined, as far as I know.<br>
                      </blockquote>
                      <div><br>
                      </div>
                    </span>
                    <div>You are correct.</div>
                    <span class="">
                      <div> </div>
                      <blockquote class="gmail_quote" style="margin:0px
                        0px 0px
0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><span><br>
                          &gt; New things can be defined in the future.
                          When they do, they should treat ICE a virtual
                          communication channel that<br>
                          &gt; provides unreliable packet transport with
                          no order guarantees which can span multiple
                          5-tuples.<br>
                          <br>
                        </span>Then the scope of what we discuss now
                        should not be "whatever protocol" - it should be
                        the specific protocols we are discussing.<br>
                        <br>
                      </blockquote>
                      <div><br>
                      </div>
                    </span>
                    <div>I think ICE-bis should define protocol
                      requirements for the protocols that can run on top
                      of ICE, which includes:</div>
                    <div>1. Ability to run over unreliable packet based
                      transport with no order guarantees</div>
                    <div>2. Ability to demux with STUN packets</div>
                    <div>3. Not t make any assumption about IP
                      addresses, ports, or other transport level
                      protocols attributes such as TOS. </div>
                    <div>
                      <div><span class="HOEnZb"><font color="#888888"><br>
                          </font></span></div>
                    </div>
                  </div>
                </div>
              </div>
            </blockquote>
            <div><br>
            </div>
            <div>I think these are good criteria. Note that TCP would
              meet these criteria, and I see no problem running TCP atop
              ICE (we used to do this in an old version of our data
              channel code).</div>
            <div><br>
            </div>
            <div>HTTP, on the other hand, would not meet criterion #1.</div>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    I seem to remember a draft I wrote once upon a time (Feb 2011).... I
    called it a "datagram service".<br>
    <br>
    draft-alvestrand-dispatch-rtcweb-datagram-01<br>
    <br>
    The ability to demux with STUN packets was expressed here as:<br>
    <br>
       The datagram service is not completely transparent; in
    particular, it<br>
       is not possible to carry a datagram where the two highest bits of
    the<br>
       first octet are zero and octet 5 to 8 contain the value
    0x2112A442,<br>
       since these datagrams are reserved for use of the STUN protocol
    (RFC<br>
       5389 section 6).<br>
    <br>
    It didn't seem to warrant a special doc at the time, given
    responses, so I dropped it.<br>
    <br>
    Note that straight-up TCP on top of this model would require special
    work, since the TCP header checksum covers the address fields. A
    TCP-like protocol with different checksums would be trivial to
    define.<br>
    <br>
    <br>
    <br>
    <br>
    <br>
    <br>
  </body>
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>, Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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To: Ted Hardie <ted.ietf@gmail.com>
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Cc: "Cullen Jennings \(fluffy\)" <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Call for adoption of draft-alvestrand-rtcweb-gateways
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Hi,

2015-03-09 18:41 GMT+01:00 Ted Hardie <ted.ietf@gmail.com>:

> On Mon, Mar 9, 2015 at 2:07 AM, Jos=C3=A9 Luis Mill=C3=A1n <jmillan@aliax=
.net>
> wrote:
>
>> Is there any new about the adoption of the draft?
>>
>>
> =E2=80=8BHowdy,
>
> It was adopted, but we have not determined when it should be put in the
> timeline.
> During the "Doc review" portion of the upcoming meeting, we plan to
> discuss our
> overall timeline, and can cover this then.
>

Thanks a lot for the update.


>
> regards,
>
> Ted Hardie
>
>
>
>> Thanks
>>
>>
>> --
>> Jos=C3=A9 Luis Mill=C3=A1n
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>


--=20
Jos=C3=A9 Luis Mill=C3=A1n

--089e013d173a0f0d4b05111554ab
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<div dir=3D"ltr">Hi,<br><div class=3D"gmail_extra"><br><div class=3D"gmail_=
quote">2015-03-09 18:41 GMT+01:00 Ted Hardie <span dir=3D"ltr">&lt;<a href=
=3D"mailto:ted.ietf@gmail.com" target=3D"_blank" onclick=3D"window.open(&#3=
9;https://mail.google.com/mail/?view=3Dcm&amp;tf=3D1&amp;to=3Dted.ietf@gmai=
l.com&amp;cc=3D&amp;bcc=3D&amp;su=3D&amp;body=3D&#39;,&#39;_blank&#39;);ret=
urn false;">ted.ietf@gmail.com</a>&gt;</span>:<br><blockquote class=3D"gmai=
l_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left=
:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><span class=3D"">On Mon, =
Mar 9, 2015 at 2:07 AM, Jos=C3=A9 Luis Mill=C3=A1n <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:jmillan@aliax.net" target=3D"_blank" onclick=3D"window.open=
(&#39;https://mail.google.com/mail/?view=3Dcm&amp;tf=3D1&amp;to=3Djmillan@a=
liax.net&amp;cc=3D&amp;bcc=3D&amp;su=3D&amp;body=3D&#39;,&#39;_blank&#39;);=
return false;">jmillan@aliax.net</a>&gt;</span> wrote:<br></span><div class=
=3D"gmail_quote"><span class=3D""><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=
=3D"ltr"><div class=3D"gmail_quote"><span></span><div>Is there any new abou=
t the adoption of the draft?</div><div><br></div></div></div></blockquote><=
/span><div><br><div style=3D"font-family:tahoma,sans-serif;font-size:small"=
>=E2=80=8BHowdy,<br><br></div><div style=3D"font-family:tahoma,sans-serif;f=
ont-size:small">It was adopted, but we have not determined when it should b=
e put in the timeline.<br></div><div style=3D"font-family:tahoma,sans-serif=
;font-size:small">During the &quot;Doc review&quot; portion of the upcoming=
 meeting, we plan to discuss our<br></div><div style=3D"font-family:tahoma,=
sans-serif;font-size:small">overall timeline, and can cover this then.<br><=
/div></div></div></div></div></blockquote><div><br></div><div>Thanks a lot =
for the update.</div><div>=C2=A0</div><blockquote class=3D"gmail_quote" sty=
le=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div d=
ir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div><div =
style=3D"font-family:tahoma,sans-serif;font-size:small"><br></div><div styl=
e=3D"font-family:tahoma,sans-serif;font-size:small">regards,<br><br></div><=
div style=3D"font-family:tahoma,sans-serif;font-size:small">Ted Hardie<br><=
/div><br>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 =
0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span class=3D""><div d=
ir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div></div=
><div>Thanks</div></div><span><font color=3D"#888888"><br clear=3D"all"><di=
v><br></div>-- <br><div>Jos=C3=A9 Luis Mill=C3=A1n</div>
</font></span></div></div>
<br></span><span class=3D"">_______________________________________________=
<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank" onclick=3D"window.open=
(&#39;https://mail.google.com/mail/?view=3Dcm&amp;tf=3D1&amp;to=3Drtcweb@ie=
tf.org&amp;cc=3D&amp;bcc=3D&amp;su=3D&amp;body=3D&#39;,&#39;_blank&#39;);re=
turn false;">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></span></blockquote></div><br></div></div>
</blockquote></div><br><br clear=3D"all"><div><br></div>-- <br><div class=
=3D"gmail_signature">Jos=C3=A9 Luis Mill=C3=A1n</div>
</div></div>

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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Date: Thu, 12 Mar 2015 10:48:30 -0400
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From: Roman Shpount <roman@telurix.com>
To: Justin Uberti <juberti@google.com>
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Cc: Cullen Jennings <fluffy@cisco.com>, Jonathan Lennox <jonathan@vidyo.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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--047d7b1117bf3a80060511187aa9
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On Wed, Mar 11, 2015 at 11:33 AM, Justin Uberti <juberti@google.com> wrote:

>
>
> On Wed, Mar 11, 2015 at 7:32 AM, Roman Shpount <roman@telurix.com> wrote:
>
>> On Wed, Mar 11, 2015 at 10:24 AM, Christer Holmberg <
>> christer.holmberg@ericsson.com> wrote:
>>
>>> I assume you mean SCTP-over-DTLS? Usage of "plain" SCTP with ICE is not
>>> defined, as far as I know.
>>>
>>
>> You are correct.
>>
>>
>>>
>>> > New things can be defined in the future. When they do, they should
>>> treat ICE a virtual communication channel that
>>> > provides unreliable packet transport with no order guarantees which
>>> can span multiple 5-tuples.
>>>
>>> Then the scope of what we discuss now should not be "whatever protocol"
>>> - it should be the specific protocols we are discussing.
>>>
>>>
>> I think ICE-bis should define protocol requirements for the protocols
>> that can run on top of ICE, which includes:
>> 1. Ability to run over unreliable packet based transport with no order
>> guarantees
>> 2. Ability to demux with STUN packets
>> 3. Not t make any assumption about IP addresses, ports, or other
>> transport level protocols attributes such as TOS.
>>
>>
> I think these are good criteria. Note that TCP would meet these criteria,
> and I see no problem running TCP atop ICE (we used to do this in an old
> version of our data channel code).
>
> I think for TCP to meet this criteria, usage of TCP over ICE would still
need to be defined somewhere. In particular the use of address, port, and
checksum in such use case would need to be specified somewhere. Until this
is done, TCP-over-ICE would be under-defined and not quite usable.
_____________
Roman Shpount

--047d7b1117bf3a80060511187aa9
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On W=
ed, Mar 11, 2015 at 11:33 AM, Justin Uberti <span dir=3D"ltr">&lt;<a href=
=3D"mailto:juberti@google.com" target=3D"_blank">juberti@google.com</a>&gt;=
</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px=
 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-=
left-style:solid;padding-left:1ex"><div dir=3D"ltr"><br><div class=3D"gmail=
_extra"><br><div class=3D"gmail_quote"><div><div class=3D"h5">On Wed, Mar 1=
1, 2015 at 7:32 AM, Roman Shpount <span dir=3D"ltr">&lt;<a href=3D"mailto:r=
oman@telurix.com" target=3D"_blank">roman@telurix.com</a>&gt;</span> wrote:=
<br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;bor=
der-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:sol=
id;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><div class=
=3D"gmail_quote"><span>On Wed, Mar 11, 2015 at 10:24 AM, Christer Holmberg =
<span dir=3D"ltr">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" tar=
get=3D"_blank">christer.holmberg@ericsson.com</a>&gt;</span> wrote:<br><blo=
ckquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left=
-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;paddi=
ng-left:1ex">I assume you mean SCTP-over-DTLS? Usage of &quot;plain&quot; S=
CTP with ICE is not defined, as far as I know.<br></blockquote><div><br></d=
iv></span><div>You are correct.</div><span><div>=C2=A0</div><blockquote cla=
ss=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;=
border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex=
">
<span><br>
&gt; New things can be defined in the future. When they do, they should tre=
at ICE a virtual communication channel that<br>
&gt; provides unreliable packet transport with no order guarantees which ca=
n span multiple 5-tuples.<br>
<br>
</span>Then the scope of what we discuss now should not be &quot;whatever p=
rotocol&quot; - it should be the specific protocols we are discussing.<br>
<br></blockquote><div><br></div></span><div>I think ICE-bis should define p=
rotocol requirements for the protocols that can run on top of ICE, which in=
cludes:</div><div>1. Ability to run over unreliable packet based transport =
with no order guarantees</div><div>2. Ability to demux with STUN packets</d=
iv><div>3. Not t make any assumption about IP addresses, ports, or other tr=
ansport level protocols attributes such as TOS.=C2=A0</div><div><div><span>=
<font color=3D"#888888"><br></font></span></div></div></div></div></div></b=
lockquote><div><br></div></div></div><div>I think these are good criteria. =
Note that TCP would meet these criteria, and I see no problem running TCP a=
top ICE (we used to do this in an old version of our data channel code).</d=
iv><div><br></div></div></div></div></blockquote></div></div><div class=3D"=
gmail_extra">I think for TCP to meet this criteria, usage of TCP over ICE w=
ould still need to be defined somewhere. In particular the use of address, =
port, and checksum in such use case would need to be specified somewhere. U=
ntil this is done, TCP-over-ICE would be under-defined and not quite usable=
.</div><div class=3D"gmail_extra"><div><div class=3D"gmail_signature">_____=
________<br>Roman Shpount</div></div><div><br></div></div></div>

--047d7b1117bf3a80060511187aa9--


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References: <54F74B02.1070902@jive.com> <CAD5OKxs8JYG3-Vvndi59ZrdPE7UTj22ozD4tcWTHgzWrHv=q7Q@mail.gmail.com> <54F756B2.60408@jive.com> <7594FB04B1934943A5C02806D1A2204B1D726AD8@ESESSMB209.ericsson.se> <CAD5OKxu7py3HbrFjxTDZS5ECFzx7vd=wpjve-gT6gWwksjEu+g@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D726B71@ESESSMB209.ericsson.se> <CABcZeBO1O6sA8MqvWkCDu3RPLz5-P2G65Us28i0baOavDnRT7Q@mail.gmail.com> <CAD5OKxuWCdgMR5Kxjv9BSwZ3Jm9kGXx9Pi-9FrfsnuQZ_91jAA@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D726DC1@ESESSMB209.ericsson.se> <CALiegfkipJhsy7-40+=d9xMUf4RJGdn3_fABL3NN2KuFNvS2BA@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D727570@ESESSMB209.ericsson.se> <CALiegfmfvz3NWSjcovGBytiOTbR6kFfyh0vx5cXoMJtytfGzRA@mail.gmail.com> <CAD5OKxsu3D0xHY-zYbDu1hyH_+4=3mWDvW2i98WCVZ+29BpKCw@mail.gmail.com> <CA5E97EE-99F8-44D8-B05B-C9EFDED1A9BB@vidyo.com> <2F467A7E-7A6C-4B1B-985A-0D9C089BE973@cisco.com> <CAOJ7v-1TjZOZ5G31vy_Gt73ADGLRay1RHVeMi=H6Q4=N1b6HLA@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D7367A0@ESESSMB209.ericsson.se> <CALiegfmyp=v6thk4eLz7nL1BHh2Qj7jmC84tdG7ufg8HPXsVKA@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D7369C9@ESESSMB209.ericsson.se> <CAD5OKxtCswToNzoZnnqJ5M66mjNjKJoA++WYNqN5155n+CWXsA@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D736AC0@ESESSMB209.ericsson.se> <CAD5OKxs1grSqAG32mf__wtsjpo68jZmKonbd+EsJmYNsDHUbFQ@mail.gmail.com> <CAOJ7v-3YypG1s9KXOCA+Fo58SuVuUk5-thcSc0k3N2j=4ZmJoA@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D737A76@ESESSMB209.ericsson.se>
Date: Thu, 12 Mar 2015 10:53:06 -0400
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From: Roman Shpount <roman@telurix.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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On Thu, Mar 12, 2015 at 6:12 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> I don't think a TCP connection can span over multiple 5-tuples - each TCP
> connection will be bound to one 5-tuple.
>

The TCP use case over ICE should be defined in order to be usable. Doing so
is fairly straight forward but was not needed this far.

Perhaps the protocol running on top of TCP can switch between different TCP
> connections, though. For example, would it be possible to span a TLS
> connection over multiple TCP connections?
>
>
TLS (vs DTLS) cannot run on top of ICE since it is not a protocol which can
run on top of unreliable packet based transport with no order guarantees.
It would require a stream based transport to run below it in order to
operate. If someone defines TCP over ICE, that would make a good underlying
stream protocol to run below TLS. Once again, no one needed this so far.
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On T=
hu, Mar 12, 2015 at 6:12 AM, Christer Holmberg <span dir=3D"ltr">&lt;<a hre=
f=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holm=
berg@ericsson.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote=
" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color=
:rgb(204,204,204);border-left-style:solid;padding-left:1ex">I don&#39;t thi=
nk a TCP connection can span over multiple 5-tuples - each TCP connection w=
ill be bound to one 5-tuple.<br></blockquote><div><br></div><div>The TCP us=
e case over ICE should be defined in order to be usable. Doing so is fairly=
 straight forward but was not needed this far.=C2=A0</div><div><br></div><b=
lockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-le=
ft-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;pad=
ding-left:1ex">Perhaps the protocol running on top of TCP can switch betwee=
n different TCP connections, though. For example, would it be possible to s=
pan a TLS connection over multiple TCP connections?<br>
<span class=3D""><br></span></blockquote><div><br></div><div>TLS (vs DTLS) =
cannot run on top of ICE since it is not a protocol which can run on top of=
=C2=A0<span style=3D"color:rgb(80,0,80);font-size:12.8000001907349px">unrel=
iable packet based transport with no order guarantees. It would require a s=
tream based transport to run below it in order to operate. If someone defin=
es TCP over ICE, that would make a good underlying stream protocol to run b=
elow TLS. Once again, no one needed this so far.</span></div><div><div clas=
s=3D"gmail_signature">_____________<br>Roman Shpount</div></div><div>=C2=A0=
</div></div></div></div>

--001a1141ca9eaef1310511188a0e--


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From: Justin Uberti <juberti@google.com>
Date: Thu, 12 Mar 2015 09:00:45 -0700
Message-ID: <CAOJ7v-1baW-jme7pApSFZc7aDXAmVm++p60-c9ZtjFxHSybf=g@mail.gmail.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: Cullen Jennings <fluffy@cisco.com>, Jonathan Lennox <jonathan@vidyo.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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On Thu, Mar 12, 2015 at 3:12 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> >>>> New things can be defined in the future. When they do, they should
> treat ICE a virtual communication channel that
> >>>> provides unreliable packet transport with no order guarantees which
> can span multiple 5-tuples.
> >>>
> >>> Then the scope of what we discuss now should not be "whatever
> protocol" - it should be the specific protocols we are discussing.
> >>
> >> I think ICE-bis should define protocol requirements for the protocols
> that can run on top of ICE, which includes:
> >> 1. Ability to run over unreliable packet based transport with no order
> guarantees
> >> 2. Ability to demux with STUN packets
> >> 3. Not make any assumption about IP addresses, ports, or other
> transport level protocols attributes such as TOS.
> >
> > I think these are good criteria. Note that TCP would meet these
> criteria, and I see no problem running TCP atop ICE (we used to do this in
> an old version of our data channel code).
>
> I don't think a TCP connection can span over multiple 5-tuples - each TCP
> connection will be bound to one 5-tuple.
>

I don't agree. SCTP can be tunneled over UDP, as we know, so why not TCP?
The ports in such a tunnel scenario are just as unnecessary as in SCTP over
UDP.

>
> Perhaps the protocol running on top of TCP can switch between different
> TCP connections, though. For example, would it be possible to span a TLS
> connection over multiple TCP connections?
>
> > HTTP, on the other hand, would not meet criterion #1.
>
> Agree.
>
> Regards,
>
> Christer
>

--001a1140a146c8f03e0511197df6
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Thu, Mar 12, 2015 at 3:12 AM, Christer Holmberg <span dir=3D"ltr">&l=
t;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chris=
ter.holmberg@ericsson.com</a>&gt;</span> wrote:<br><blockquote class=3D"gma=
il_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-lef=
t:1ex">Hi,<br>
<span class=3D""><br>
&gt;&gt;&gt;&gt; New things can be defined in the future. When they do, the=
y should treat ICE a virtual communication channel that<br>
&gt;&gt;&gt;&gt; provides unreliable packet transport with no order guarant=
ees which can span multiple 5-tuples.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Then the scope of what we discuss now should not be &quot;what=
ever protocol&quot; - it should be the specific protocols we are discussing=
.<br>
&gt;&gt;<br>
&gt;&gt; I think ICE-bis should define protocol requirements for the protoc=
ols that can run on top of ICE, which includes:<br>
&gt;&gt; 1. Ability to run over unreliable packet based transport with no o=
rder guarantees<br>
&gt;&gt; 2. Ability to demux with STUN packets<br>
</span>&gt;&gt; 3. Not make any assumption about IP addresses, ports, or ot=
her transport level protocols attributes such as TOS.=C2=A0<br>
<span class=3D"">&gt;<br>
&gt; I think these are good criteria. Note that TCP would meet these criter=
ia, and I see no problem running TCP atop ICE (we used to do this in an old=
 version of our data channel code).<br>
<br>
</span>I don&#39;t think a TCP connection can span over multiple 5-tuples -=
 each TCP connection will be bound to one 5-tuple.<br></blockquote><div><br=
></div><div>I don&#39;t agree. SCTP can be tunneled over UDP, as we know, s=
o why not TCP? The ports in such a tunnel scenario are just as unnecessary =
as in SCTP over UDP.=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"=
margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
Perhaps the protocol running on top of TCP can switch between different TCP=
 connections, though. For example, would it be possible to span a TLS conne=
ction over multiple TCP connections?<br>
<span class=3D""><br>
&gt; HTTP, on the other hand, would not meet criterion #1.<br>
<br>
</span>Agree.<br>
<br>
Regards,<br>
<br>
Christer<br>
</blockquote></div><br></div></div>

--001a1140a146c8f03e0511197df6--


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From: Harald Alvestrand <harald@alvestrand.no>
Date: Fri, 13 Mar 2015 10:55:16 +0000
To: Justin Uberti <juberti@google.com>, Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: Cullen Jennings <fluffy@cisco.com>, Jonathan Lennox <jonathan@vidyo.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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It's the checksum.

Den 12. mars 2015 16.00.45 WET, skrev Justin Uberti <juberti@google.com>:
>On Thu, Mar 12, 2015 at 3:12 AM, Christer Holmberg <
>christer.holmberg@ericsson.com> wrote:
>
>> Hi,
>>
>> >>>> New things can be defined in the future. When they do, they
>should
>> treat ICE a virtual communication channel that
>> >>>> provides unreliable packet transport with no order guarantees
>which
>> can span multiple 5-tuples.
>> >>>
>> >>> Then the scope of what we discuss now should not be "whatever
>> protocol" - it should be the specific protocols we are discussing.
>> >>
>> >> I think ICE-bis should define protocol requirements for the
>protocols
>> that can run on top of ICE, which includes:
>> >> 1. Ability to run over unreliable packet based transport with no
>order
>> guarantees
>> >> 2. Ability to demux with STUN packets
>> >> 3. Not make any assumption about IP addresses, ports, or other
>> transport level protocols attributes such as TOS.
>> >
>> > I think these are good criteria. Note that TCP would meet these
>> criteria, and I see no problem running TCP atop ICE (we used to do
>this in
>> an old version of our data channel code).
>>
>> I don't think a TCP connection can span over multiple 5-tuples - each
>TCP
>> connection will be bound to one 5-tuple.
>>
>
>I don't agree. SCTP can be tunneled over UDP, as we know, so why not
>TCP?
>The ports in such a tunnel scenario are just as unnecessary as in SCTP
>over
>UDP.
>
>>
>> Perhaps the protocol running on top of TCP can switch between
>different
>> TCP connections, though. For example, would it be possible to span a
>TLS
>> connection over multiple TCP connections?
>>
>> > HTTP, on the other hand, would not meet criterion #1.
>>
>> Agree.
>>
>> Regards,
>>
>> Christer
>>
>
>
>------------------------------------------------------------------------
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

-- 
Sent from my Android device with K-9 Mail. Please excuse my brevity.
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<html><head></head><body>It&#39;s the checksum.<br><br><div class="gmail_quote">Den 12. mars 2015 16.00.45 WET, skrev Justin Uberti &lt;juberti@google.com&gt;:<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div dir="ltr"><br /><div class="gmail_extra"><br /><div class="gmail_quote">On Thu, Mar 12, 2015 at 3:12 AM, Christer Holmberg <span dir="ltr">&lt;<a href="mailto:christer.holmberg@ericsson.com" target="_blank">christer.holmberg@ericsson.com</a>&gt;</span> wrote:<br /><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi,<br />
<span class=""><br />
&gt;&gt;&gt;&gt; New things can be defined in the future. When they do, they should treat ICE a virtual communication channel that<br />
&gt;&gt;&gt;&gt; provides unreliable packet transport with no order guarantees which can span multiple 5-tuples.<br />
&gt;&gt;&gt;<br />
&gt;&gt;&gt; Then the scope of what we discuss now should not be &quot;whatever protocol&quot; - it should be the specific protocols we are discussing.<br />
&gt;&gt;<br />
&gt;&gt; I think ICE-bis should define protocol requirements for the protocols that can run on top of ICE, which includes:<br />
&gt;&gt; 1. Ability to run over unreliable packet based transport with no order guarantees<br />
&gt;&gt; 2. Ability to demux with STUN packets<br />
</span>&gt;&gt; 3. Not make any assumption about IP addresses, ports, or other transport level protocols attributes such as TOS.Â <br />
<span class="">&gt;<br />
&gt; I think these are good criteria. Note that TCP would meet these criteria, and I see no problem running TCP atop ICE (we used to do this in an old version of our data channel code).<br />
<br />
</span>I don&#39;t think a TCP connection can span over multiple 5-tuples - each TCP connection will be bound to one 5-tuple.<br /></blockquote><div><br /></div><div>I don&#39;t agree. SCTP can be tunneled over UDP, as we know, so why not TCP? The ports in such a tunnel scenario are just as unnecessary as in SCTP over UDP.Â </div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br />
Perhaps the protocol running on top of TCP can switch between different TCP connections, though. For example, would it be possible to span a TLS connection over multiple TCP connections?<br />
<span class=""><br />
&gt; HTTP, on the other hand, would not meet criterion #1.<br />
<br />
</span>Agree.<br />
<br />
Regards,<br />
<br />
Christer<br />
</blockquote></div><br /></div></div>
<p style="margin-top: 2.5em; margin-bottom: 1em; border-bottom: 1px solid #000"></p><pre class="k9mail"><hr /><br />rtcweb mailing list<br />rtcweb@ietf.org<br /><a href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a><br /></pre></blockquote></div><br>
-- <br>
Sent from my Android device with K-9 Mail. Please excuse my brevity.</body></html>
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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From nobody Fri Mar 13 04:06:04 2015
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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From nobody Fri Mar 13 04:07:10 2015
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: Cullen Jennings <fluffy@cisco.com>, Jonathan Lennox <jonathan@vidyo.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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Guys, this thread is getting surreal :)
On 13 Mar 2015 12:02, "Christer Holmberg" <christer.holmberg@ericsson.com>
wrote:

> Hi Justin,
>
> >>>>>> New things can be defined in the future. When they do, they should
> treat ICE a virtual communication channel that
> >>>>>> provides unreliable packet transport with no order guarantees which
> can span multiple 5-tuples.
> >>>>>
> >>>>> Then the scope of what we discuss now should not be "whatever
> protocol" - it should be the specific protocols we are discussing.
> >>>>
> >>>> I think ICE-bis should define protocol requirements for the protocols
> that can run on top of ICE, which includes:
> >>>> 1. Ability to run over unreliable packet based transport with no
> order guarantees
> >>>> 2. Ability to demux with STUN packets
> >>>> 3. Not make any assumption about IP addresses, ports, or other
> transport level protocols attributes such as TOS.
> >>>
> >>> I think these are good criteria. Note that TCP would meet these
> criteria, and I see no problem running TCP atop ICE (we used to do this in
> an old version of our data channel code).
> >>
> >> I don't think a TCP connection can span over multiple 5-tuples - each
> TCP connection will be bound to one 5-tuple.
> >
> > I don't agree. SCTP can be tunneled over UDP, as we know, so why not
> TCP? The ports in such a tunnel scenario are just as unnecessary as in SCTP
> over UDP.
>
> We need to be clear about what we mean.
>
> When I say "TCP", I mean plain TCP :)
>
> If we talk about TCP-over-UDP, we should say "TCP-over-UDP" :)
>
> Regards,
>
> Christer
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<p dir=3D"ltr">Guys, this thread is getting surreal :)</p>
<div class=3D"gmail_quote">On 13 Mar 2015 12:02, &quot;Christer Holmberg&qu=
ot; &lt;<a href=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg=
@ericsson.com</a>&gt; wrote:<br type=3D"attribution"><blockquote class=3D"g=
mail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-l=
eft:1ex">Hi Justin,<br>
<br>
&gt;&gt;&gt;&gt;&gt;&gt; New things can be defined in the future. When they=
 do, they should treat ICE a virtual communication channel that<br>
&gt;&gt;&gt;&gt;&gt;&gt; provides unreliable packet transport with no order=
 guarantees which can span multiple 5-tuples.<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; Then the scope of what we discuss now should not be &q=
uot;whatever protocol&quot; - it should be the specific protocols we are di=
scussing.<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; I think ICE-bis should define protocol requirements for th=
e protocols that can run on top of ICE, which includes:<br>
&gt;&gt;&gt;&gt; 1. Ability to run over unreliable packet based transport w=
ith no order guarantees<br>
&gt;&gt;&gt;&gt; 2. Ability to demux with STUN packets<br>
&gt;&gt;&gt;&gt; 3. Not make any assumption about IP addresses, ports, or o=
ther transport level protocols attributes such as TOS.=C2=A0<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; I think these are good criteria. Note that TCP would meet thes=
e criteria, and I see no problem running TCP atop ICE (we used to do this i=
n an old version of our data channel code).<br>
&gt;&gt;<br>
&gt;&gt; I don&#39;t think a TCP connection can span over multiple 5-tuples=
 - each TCP connection will be bound to one 5-tuple.<br>
&gt;<br>
&gt; I don&#39;t agree. SCTP can be tunneled over UDP, as we know, so why n=
ot TCP? The ports in such a tunnel scenario are just as unnecessary as in S=
CTP over UDP.=C2=A0<br>
<br>
We need to be clear about what we mean.<br>
<br>
When I say &quot;TCP&quot;, I mean plain TCP :)<br>
<br>
If we talk about TCP-over-UDP, we should say &quot;TCP-over-UDP&quot; :)<br=
>
<br>
Regards,<br>
<br>
Christer<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div>

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: =?utf-8?B?ScOxYWtpIEJheiBDYXN0aWxsbw==?= <ibc@aliax.net>
Thread-Topic: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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Le 2015-03-13 07:09, Christer Holmberg a écrit :
> BTW, is this going to be discussed in Dallas?

It should. Maybe during the ICE-bis slot in MMUSIC?

Simon


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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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--089e01229aaaa1602305112dc422
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On Fri, Mar 13, 2015 at 4:05 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> > TLS (vs DTLS) cannot run on top of ICE since it is not a protocol which
> can run on top of unreliable packet based transport with no order
> guarantees. It would require a stream
> > based transport to run below it in order to operate. If someone defines
> TCP over ICE, that would make a good underlying stream protocol to run
> below TLS. Once again, no one
> > needed this so far.
>
> Even if we had TCP over ICE, I still don't know whether we would be able
> to run TLS on top. Because, in addition to reliability, doesn't TLS also
> rely on the ordering provided by TCP? We would need to make sure the
> ordering is maintained if endpoints use multiple TCP 5-tuples.
>
>
A few points:
- Anything we are say "runs over ICE" is going to be tunneled over UDP or
TCP, whichever ICE chooses to use as its underlying datagram transport. So
the notion of running "plain TCP" over ICE is nonsensical.

- Assuming we do run TCP over ICE, we can certainly run TLS on top. Since
TCP is just treating the underlying ICE transport as a datagram transport,
TCP retains all of its reliability and in-order properties. So TLS just
sees that it is running over TCP, and doesn't care that it is ICE at the
lowest layer. Note that this stacking of TLS/TCP/ICE is different from the
case of running TLS over ICE-TCP, since ICE-TCP does not guarantee
reliability or ordering.

- Harald's point about the TCP checksum is true; if we were truly
standardizing TCP-over-ICE, we would have to define an appropriate way to
compute the TCP checksum given that said checksum makes assumptions about
IP particulars. This would not be hard though.

- Lastly, this is clearly a source of massive confusion, so I agree we
should discuss it in detail in Dallas. I would be happy to help prepare
presentation material if needed.

--089e01229aaaa1602305112dc422
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Fri, Mar 13, 2015 at 4:05 AM, Christer Holmberg <span dir=3D"ltr">&l=
t;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chris=
ter.holmberg@ericsson.com</a>&gt;</span> wrote:<br><blockquote class=3D"gma=
il_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-lef=
t:1ex">Hi,<br>
<span class=3D""><br>
&gt; TLS (vs DTLS) cannot run on top of ICE since it is not a protocol whic=
h can run on top of=C2=A0unreliable packet based transport with no order gu=
arantees. It would require a stream<br>
&gt; based transport to run below it in order to operate. If someone define=
s TCP over ICE, that would make a good underlying stream protocol to run be=
low TLS. Once again, no one<br>
&gt; needed this so far.<br>
<br>
</span>Even if we had TCP over ICE, I still don&#39;t know whether we would=
 be able to run TLS on top. Because, in addition to reliability, doesn&#39;=
t TLS also rely on the ordering provided by TCP? We would need to make sure=
 the ordering is maintained if endpoints use multiple TCP 5-tuples.<br>
<br></blockquote><div><br></div><div>A few points:<br></div><div>- Anything=
 we are say &quot;runs over ICE&quot; is going to be tunneled over UDP or T=
CP, whichever ICE chooses to use as its underlying datagram transport. So t=
he notion of running &quot;plain TCP&quot; over ICE is nonsensical.</div><d=
iv><br></div><div>- Assuming we do run TCP over ICE, we can certainly run T=
LS on top. Since TCP is just treating the underlying ICE transport as a dat=
agram transport, TCP retains all of its reliability and in-order properties=
. So TLS just sees that it is running over TCP, and doesn&#39;t care that i=
t is ICE at the lowest layer. Note that this stacking of TLS/TCP/ICE is dif=
ferent from the case of running TLS over ICE-TCP, since ICE-TCP does not gu=
arantee reliability or ordering.</div><div><br></div><div>- Harald&#39;s po=
int about the TCP checksum is true; if we were truly standardizing TCP-over=
-ICE, we would have to define an appropriate way to compute the TCP checksu=
m given that said checksum makes assumptions about IP particulars. This wou=
ld not be hard though.</div><div><br></div><div>- Lastly, this is clearly a=
 source of massive confusion, so I agree we should discuss it in detail in =
Dallas. I would be happy to help prepare presentation material if needed.</=
div><div>=C2=A0</div></div></div></div>

--089e01229aaaa1602305112dc422--


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On 03/13/2015 04:12 PM, Justin Uberti wrote:
>
>
> On Fri, Mar 13, 2015 at 4:05 AM, Christer Holmberg
> <christer.holmberg@ericsson.com
> <mailto:christer.holmberg@ericsson.com>> wrote:
>
>     Hi,
>
>     > TLS (vs DTLS) cannot run on top of ICE since it is not a
>     protocol which can run on top of unreliable packet based transport
>     with no order guarantees. It would require a stream
>     > based transport to run below it in order to operate. If someone
>     defines TCP over ICE, that would make a good underlying stream
>     protocol to run below TLS. Once again, no one
>     > needed this so far.
>
>     Even if we had TCP over ICE, I still don't know whether we would
>     be able to run TLS on top. Because, in addition to reliability,
>     doesn't TLS also rely on the ordering provided by TCP? We would
>     need to make sure the ordering is maintained if endpoints use
>     multiple TCP 5-tuples.
>
>
> A few points:
> - Anything we are say "runs over ICE" is going to be tunneled over UDP
> or TCP, whichever ICE chooses to use as its underlying datagram
> transport. So the notion of running "plain TCP" over ICE is nonsensical=
=2E
>
> - Assuming we do run TCP over ICE, we can certainly run TLS on top.
> Since TCP is just treating the underlying ICE transport as a datagram
> transport, TCP retains all of its reliability and in-order properties.
> So TLS just sees that it is running over TCP, and doesn't care that it
> is ICE at the lowest layer. Note that this stacking of TLS/TCP/ICE is
> different from the case of running TLS over ICE-TCP, since ICE-TCP
> does not guarantee reliability or ordering.
>
> - Harald's point about the TCP checksum is true; if we were truly
> standardizing TCP-over-ICE, we would have to define an appropriate way
> to compute the TCP checksum given that said checksum makes assumptions
> about IP particulars. This would not be hard though.
>
> - Lastly, this is clearly a source of massive confusion, so I agree we
> should discuss it in detail in Dallas. I would be happy to help
> prepare presentation material if needed.

Can we define an use case for which there is a compelling need for
<whatever it is we're discussing> before spending too much meeting time
on it?

It's clearly very simple to define a protocol that is like TCP except
that the checksum doesn't include the addresses, and where the two
highest bits of the first byte aren't 00. It's not interoperable with
standard TCP, it can't take advantage of TCP accelerator hardware or be
detected by TCP-sensitive middle boxes (unless tweaked). But it's simple
to define.

It's also clearly very simple to run this protocol over ICE.

But why?

And is this scenario relevant for RTCWEB, which is trying to finish its
current set of options?


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    <div class="moz-cite-prefix">On 03/13/2015 04:12 PM, Justin Uberti
      wrote:<br>
    </div>
    <blockquote
cite="mid:CAOJ7v-3bCxVP9fNuRFp_sVBXh4msnF1=fVZrefE0jejMzY8VQQ@mail.gmail.com"
      type="cite">
      <div dir="ltr"><br>
        <div class="gmail_extra"><br>
          <div class="gmail_quote">On Fri, Mar 13, 2015 at 4:05 AM,
            Christer Holmberg <span dir="ltr">&lt;<a
                moz-do-not-send="true"
                href="mailto:christer.holmberg@ericsson.com"
                target="_blank">christer.holmberg@ericsson.com</a>&gt;</span>
            wrote:<br>
            <blockquote class="gmail_quote" style="margin:0 0 0
              .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi,<br>
              <span class=""><br>
                &gt; TLS (vs DTLS) cannot run on top of ICE since it is
                not a protocol which can run on top of unreliable packet
                based transport with no order guarantees. It would
                require a stream<br>
                &gt; based transport to run below it in order to
                operate. If someone defines TCP over ICE, that would
                make a good underlying stream protocol to run below TLS.
                Once again, no one<br>
                &gt; needed this so far.<br>
                <br>
              </span>Even if we had TCP over ICE, I still don't know
              whether we would be able to run TLS on top. Because, in
              addition to reliability, doesn't TLS also rely on the
              ordering provided by TCP? We would need to make sure the
              ordering is maintained if endpoints use multiple TCP
              5-tuples.<br>
              <br>
            </blockquote>
            <div><br>
            </div>
            <div>A few points:<br>
            </div>
            <div>- Anything we are say "runs over ICE" is going to be
              tunneled over UDP or TCP, whichever ICE chooses to use as
              its underlying datagram transport. So the notion of
              running "plain TCP" over ICE is nonsensical.</div>
            <div><br>
            </div>
            <div>- Assuming we do run TCP over ICE, we can certainly run
              TLS on top. Since TCP is just treating the underlying ICE
              transport as a datagram transport, TCP retains all of its
              reliability and in-order properties. So TLS just sees that
              it is running over TCP, and doesn't care that it is ICE at
              the lowest layer. Note that this stacking of TLS/TCP/ICE
              is different from the case of running TLS over ICE-TCP,
              since ICE-TCP does not guarantee reliability or ordering.</div>
            <div><br>
            </div>
            <div>- Harald's point about the TCP checksum is true; if we
              were truly standardizing TCP-over-ICE, we would have to
              define an appropriate way to compute the TCP checksum
              given that said checksum makes assumptions about IP
              particulars. This would not be hard though.</div>
            <div><br>
            </div>
            <div>- Lastly, this is clearly a source of massive
              confusion, so I agree we should discuss it in detail in
              Dallas. I would be happy to help prepare presentation
              material if needed.</div>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    Can we define an use case for which there is a compelling need for
    &lt;whatever it is we're discussing&gt; before spending too much
    meeting time on it?<br>
    <br>
    It's clearly very simple to define a protocol that is like TCP
    except that the checksum doesn't include the addresses, and where
    the two highest bits of the first byte aren't 00. It's not
    interoperable with standard TCP, it can't take advantage of TCP
    accelerator hardware or be detected by TCP-sensitive middle boxes
    (unless tweaked). But it's simple to define.<br>
    <br>
    It's also clearly very simple to run this protocol over ICE.<br>
    <br>
    But why?<br>
    <br>
    And is this scenario relevant for RTCWEB, which is trying to finish
    its current set of options?<br>
    <br>
  </body>
</html>

--------------020403020107010504070804--


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From: Justin Uberti <juberti@google.com>
Date: Fri, 13 Mar 2015 11:50:55 -0700
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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On Fri, Mar 13, 2015 at 9:46 AM, Harald Alvestrand <harald@alvestrand.no>
wrote:

>  On 03/13/2015 04:12 PM, Justin Uberti wrote:
>
>
>
> On Fri, Mar 13, 2015 at 4:05 AM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
>
>> Hi,
>>
>> > TLS (vs DTLS) cannot run on top of ICE since it is not a protocol which
>> can run on top of unreliable packet based transport with no order
>> guarantees. It would require a stream
>> > based transport to run below it in order to operate. If someone defines
>> TCP over ICE, that would make a good underlying stream protocol to run
>> below TLS. Once again, no one
>> > needed this so far.
>>
>> Even if we had TCP over ICE, I still don't know whether we would be able
>> to run TLS on top. Because, in addition to reliability, doesn't TLS also
>> rely on the ordering provided by TCP? We would need to make sure the
>> ordering is maintained if endpoints use multiple TCP 5-tuples.
>>
>>
>  A few points:
>  - Anything we are say "runs over ICE" is going to be tunneled over UDP
> or TCP, whichever ICE chooses to use as its underlying datagram transport.
> So the notion of running "plain TCP" over ICE is nonsensical.
>
>  - Assuming we do run TCP over ICE, we can certainly run TLS on top.
> Since TCP is just treating the underlying ICE transport as a datagram
> transport, TCP retains all of its reliability and in-order properties. So
> TLS just sees that it is running over TCP, and doesn't care that it is ICE
> at the lowest layer. Note that this stacking of TLS/TCP/ICE is different
> from the case of running TLS over ICE-TCP, since ICE-TCP does not guarantee
> reliability or ordering.
>
>  - Harald's point about the TCP checksum is true; if we were truly
> standardizing TCP-over-ICE, we would have to define an appropriate way to
> compute the TCP checksum given that said checksum makes assumptions about
> IP particulars. This would not be hard though.
>
>  - Lastly, this is clearly a source of massive confusion, so I agree we
> should discuss it in detail in Dallas. I would be happy to help prepare
> presentation material if needed.
>
>
> Can we define an use case for which there is a compelling need for
> <whatever it is we're discussing> before spending too much meeting time on
> it?
>
> It's clearly very simple to define a protocol that is like TCP except that
> the checksum doesn't include the addresses, and where the two highest bits
> of the first byte aren't 00. It's not interoperable with standard TCP, it
> can't take advantage of TCP accelerator hardware or be detected by
> TCP-sensitive middle boxes (unless tweaked). But it's simple to define.
>
> It's also clearly very simple to run this protocol over ICE.
>
> But why?
>
> And is this scenario relevant for RTCWEB, which is trying to finish its
> current set of options?
>

To be clear, I am not advocating going off and doing work to make TCP run
over ICE. My point was merely that essentially any protocol that runs over
IP or some other datagram transport can be made to run over ICE with zero
or minimal modifications; this is not well understood; and so this should
be explicitly codified in ICE-bis.

--001a1141bc5a3292f505112ffce8
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Fri, Mar 13, 2015 at 9:46 AM, Harald Alvestrand <span dir=3D"ltr">&l=
t;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestra=
nd.no</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000"><div><div class=3D"h5">
    <div>On 03/13/2015 04:12 PM, Justin Uberti
      wrote:<br>
    </div>
    <blockquote type=3D"cite">
      <div dir=3D"ltr"><br>
        <div class=3D"gmail_extra"><br>
          <div class=3D"gmail_quote">On Fri, Mar 13, 2015 at 4:05 AM,
            Christer Holmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:chris=
ter.holmberg@ericsson.com" target=3D"_blank">christer.holmberg@ericsson.com=
</a>&gt;</span>
            wrote:<br>
            <blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bo=
rder-left:1px #ccc solid;padding-left:1ex">Hi,<br>
              <span><br>
                &gt; TLS (vs DTLS) cannot run on top of ICE since it is
                not a protocol which can run on top of=C2=A0unreliable pack=
et
                based transport with no order guarantees. It would
                require a stream<br>
                &gt; based transport to run below it in order to
                operate. If someone defines TCP over ICE, that would
                make a good underlying stream protocol to run below TLS.
                Once again, no one<br>
                &gt; needed this so far.<br>
                <br>
              </span>Even if we had TCP over ICE, I still don&#39;t know
              whether we would be able to run TLS on top. Because, in
              addition to reliability, doesn&#39;t TLS also rely on the
              ordering provided by TCP? We would need to make sure the
              ordering is maintained if endpoints use multiple TCP
              5-tuples.<br>
              <br>
            </blockquote>
            <div><br>
            </div>
            <div>A few points:<br>
            </div>
            <div>- Anything we are say &quot;runs over ICE&quot; is going t=
o be
              tunneled over UDP or TCP, whichever ICE chooses to use as
              its underlying datagram transport. So the notion of
              running &quot;plain TCP&quot; over ICE is nonsensical.</div>
            <div><br>
            </div>
            <div>- Assuming we do run TCP over ICE, we can certainly run
              TLS on top. Since TCP is just treating the underlying ICE
              transport as a datagram transport, TCP retains all of its
              reliability and in-order properties. So TLS just sees that
              it is running over TCP, and doesn&#39;t care that it is ICE a=
t
              the lowest layer. Note that this stacking of TLS/TCP/ICE
              is different from the case of running TLS over ICE-TCP,
              since ICE-TCP does not guarantee reliability or ordering.</di=
v>
            <div><br>
            </div>
            <div>- Harald&#39;s point about the TCP checksum is true; if we
              were truly standardizing TCP-over-ICE, we would have to
              define an appropriate way to compute the TCP checksum
              given that said checksum makes assumptions about IP
              particulars. This would not be hard though.</div>
            <div><br>
            </div>
            <div>- Lastly, this is clearly a source of massive
              confusion, so I agree we should discuss it in detail in
              Dallas. I would be happy to help prepare presentation
              material if needed.</div>
          </div>
        </div>
      </div>
    </blockquote>
    <br></div></div>
    Can we define an use case for which there is a compelling need for
    &lt;whatever it is we&#39;re discussing&gt; before spending too much
    meeting time on it?<br>
    <br>
    It&#39;s clearly very simple to define a protocol that is like TCP
    except that the checksum doesn&#39;t include the addresses, and where
    the two highest bits of the first byte aren&#39;t 00. It&#39;s not
    interoperable with standard TCP, it can&#39;t take advantage of TCP
    accelerator hardware or be detected by TCP-sensitive middle boxes
    (unless tweaked). But it&#39;s simple to define.<br>
    <br>
    It&#39;s also clearly very simple to run this protocol over ICE.<br>
    <br>
    But why?<br>
    <br>
    And is this scenario relevant for RTCWEB, which is trying to finish
    its current set of options?<br></div></blockquote><div><br></div><div>T=
o be clear, I am not advocating going off and doing work to make TCP run ov=
er ICE. My point was merely that essentially any protocol that runs over IP=
 or some other datagram transport can be made to run over ICE with zero or =
minimal modifications; this is not well understood; and so this should be e=
xplicitly codified in ICE-bis.</div></div></div></div>

--001a1141bc5a3292f505112ffce8--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>, Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
Thread-Index: AQHQVqbHfYSbCv5RRE+VguSZdJaU4Z0MmygAgAACNQCAAB9EuP//8M8AgAASNNj//+/xAIAADFaAgAAlg8SAABtygIAAeWFLgAAf7wCAAAYfgIAAVgeAgAZK9ICAAInMAIACfsqQ///+kYAAAkFGAP//9aSA///silCAABmMAIAAEUGA//63n9CAAs9YAP/+nTMwAGFnfwAAATXyAAAEVRiA///dC6A=
Date: Fri, 13 Mar 2015 19:58:04 +0000
Message-ID: <7594FB04B1934943A5C02806D1A2204B1D73A2E9@ESESSMB209.ericsson.se>
References: <54F74B02.1070902@jive.com> <CA5E97EE-99F8-44D8-B05B-C9EFDED1A9BB@vidyo.com> <2F467A7E-7A6C-4B1B-985A-0D9C089BE973@cisco.com> <CAOJ7v-1TjZOZ5G31vy_Gt73ADGLRay1RHVeMi=H6Q4=N1b6HLA@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D7367A0@ESESSMB209.ericsson.se> <CALiegfmyp=v6thk4eLz7nL1BHh2Qj7jmC84tdG7ufg8HPXsVKA@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D7369C9@ESESSMB209.ericsson.se> <CAD5OKxtCswToNzoZnnqJ5M66mjNjKJoA++WYNqN5155n+CWXsA@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D736AC0@ESESSMB209.ericsson.se> <CAD5OKxs1grSqAG32mf__wtsjpo68jZmKonbd+EsJmYNsDHUbFQ@mail.gmail.com> <CAOJ7v-3YypG1s9KXOCA+Fo58SuVuUk5-thcSc0k3N2j=4ZmJoA@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D737A76@ESESSMB209.ericsson.se> <CAD5OKxs+OEDp9pYrZHw237PfsNunao=PSC89dRhWiFcMwEQUXg@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D739611@ESESSMB209.ericsson.se> <CAOJ7v-3bCxVP9fNuRFp_sVBXh4msnF1=fVZrefE0jejMzY8VQQ@mail.gmail.com> <5503147C.3060506@alvestrand.no> <CAOJ7v-23_H2jjgG6GrOd1MPv9M-iDbzsdF=L9e5xJsEbH=21PA@mail.gmail.com>
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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--_000_7594FB04B1934943A5C02806D1A2204B1D73A2E9ESESSMB209erics_--


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A new Request for Comments is now available in online RFC libraries.

        
        RFC 7478

        Title:      Web Real-Time Communication Use Cases 
                    and Requirements 
        Author:     C. Holmberg, S. Hakansson, G. Eriksson
        Status:     Informational
        Stream:     IETF
        Date:       March 2015
        Mailbox:    christer.holmberg@ericsson.com, 
                    stefan.lk.hakansson@ericsson.com, 
                    goran.ap.eriksson@ericsson.com
        Pages:      28
        Characters: 64824
        Updates/Obsoletes/SeeAlso:   None

        I-D Tag:    draft-ietf-rtcweb-use-cases-and-requirements-16.txt

        URL:        https://www.rfc-editor.org/info/rfc7478

This document describes web-based real-time communication use cases.
Requirements on the browser functionality are derived from the use
cases.

This document was developed in an initial phase of the work with
rather minor updates at later stages.  It has not really served as a
tool in deciding features or scope for the WG's efforts so far.  It
is being published to record the early conclusions of the WG.  It
will not be used as a set of rigid guidelines that specifications and
implementations will be held to in the future.

This document is a product of the Real-Time Communication in WEB-browsers Working Group of the IETF.


INFORMATIONAL: This memo provides information for the Internet community.
It does not specify an Internet standard of any kind. Distribution of
this memo is unlimited.

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Date: Fri, 13 Mar 2015 16:43:56 -0400
From: Simon Perreault <sperreault@jive.com>
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To: Christer Holmberg <christer.holmberg@ericsson.com>,  Justin Uberti <juberti@google.com>, Harald Alvestrand <harald@alvestrand.no>
References: <54F74B02.1070902@jive.com> <7594FB04B1934943A5C02806D1A2204B1D7367A0@ESESSMB209.ericsson.se> <CALiegfmyp=v6thk4eLz7nL1BHh2Qj7jmC84tdG7ufg8HPXsVKA@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D7369C9@ESESSMB209.ericsson.se> <CAD5OKxtCswToNzoZnnqJ5M66mjNjKJoA++WYNqN5155n+CWXsA@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D736AC0@ESESSMB209.ericsson.se> <CAD5OKxs1grSqAG32mf__wtsjpo68jZmKonbd+EsJmYNsDHUbFQ@mail.gmail.com> <CAOJ7v-3YypG1s9KXOCA+Fo58SuVuUk5-thcSc0k3N2j=4ZmJoA@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D737A76@ESESSMB209.ericsson.se> <CAD5OKxs+OEDp9pYrZHw237PfsNunao=PSC89dRhWiFcMwEQUXg@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D739611@ESESSMB209.ericsson.se> <CAOJ7v-3bCxVP9fNuRFp_sVBXh4msnF1=fVZrefE0jejMzY8VQQ@mail.gmail.com> <5503147C.3060506@alvestrand.no> <CAOJ7v-23_H2jjgG6GrOd1MPv9M-iDbzsdF=L9e5xJsEbH=21PA@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D73A2E9@ESESSMB209.ericsson.se>
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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Le 2015-03-13 15:58, Christer Holmberg a écrit :
> The thing that brought this whole thing up was DTLS.
> 
> So, unless people see a need for something else, maybe we should limit
> the scope to DTLS, and describe how a DTLS connection can span multiple
> candidate pairs (read: multiple 5-tuples)?
> 
> Otherwise we may end up spending lots of time for something which nobody
> really need…

+1000

What we need is to explain why text in DTLS-SRTP that specifically talks
about binding to a specific 5-tuple applies differently with ICE.

Simon


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From: Roman Shpount <roman@telurix.com>
To: Simon Perreault <sperreault@jive.com>
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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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--90e6ba6147e2c5b5be051131bc13
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On Fri, Mar 13, 2015 at 4:43 PM, Simon Perreault <sperreault@jive.com>
wrote:

> What we need is to explain why text in DTLS-SRTP that specifically talks
> about binding to a specific 5-tuple applies differently with ICE.
>
>
We need text that explains that RTP/RTCP, SRTP/SRTCP, DTLS and DTLS-SRTP
should not bind to a specific 5-tuple when running on top of ICE. We would
also need to explain the concept of the virtual transport channel created
by ICE. We will need to specify how specific 5-tuples get associated with
the same virtual transport. Anything else is of no immediate value. I
believe, the most appropriate place for this would be ICE-bis. One way of
doing this is to introduce the concept of "ICE compatible transports", list
the requirements, and specify transports currently supported as examples.
_____________
Roman Shpount

--90e6ba6147e2c5b5be051131bc13
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Fri, Mar 13, 2015 at 4:43 PM, Simon Perreault <span dir=3D"ltr">&lt=
;<a href=3D"mailto:sperreault@jive.com" target=3D"_blank">sperreault@jive.c=
om</a>&gt;</span> wrote:<br></div></div><div class=3D"gmail_quote"><blockqu=
ote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-wid=
th:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-l=
eft:1ex">What we need is to explain why text in DTLS-SRTP that specifically=
 talks<br>
about binding to a specific 5-tuple applies differently with ICE.<br>
<span class=3D""><font color=3D"#888888"><br></font></span></blockquote><di=
v><br></div><div>We need text that explains that RTP/RTCP, SRTP/SRTCP, DTLS=
 and DTLS-SRTP should not bind to a specific 5-tuple when running on top of=
 ICE. We would also need to explain the concept of the virtual transport ch=
annel created by ICE. We will need to specify how specific 5-tuples get ass=
ociated with the same virtual transport. Anything else is of no immediate v=
alue. I believe, the most appropriate place for this would be ICE-bis. One =
way of doing this is to introduce the concept of &quot;ICE compatible trans=
ports&quot;, list the requirements, and specify transports currently suppor=
ted as examples.</div><div><div class=3D"gmail_signature">_____________<br>=
Roman Shpount</div></div><div>=C2=A0</div></div></div></div>

--90e6ba6147e2c5b5be051131bc13--


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2015-03-13 20:58 GMT+01:00 Christer Holmberg <christer.holmberg@ericsson.co=
m>:
> Otherwise we may end up spending lots of time for something which nobody
> really need=E2=80=A6

You started the "TCP over ICE" subject... XD

Agreed. ICE is already well defined for RTP and RTCP usages. The only
"remaining" problem is that the DTLS-SRTP RFC talks about 5-tuples
while it is clear that the virtual transport (to be) defined by ICE
also fits DTLS(-SRTP) requirements.


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


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Subject: Re: [rtcweb] DTLS, DTLS-SRTP, and 5-tuples
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On 3/11/15 10:24 AM, Christer Holmberg wrote:
> Hi,
>
>>> So, we assume that whatever protocol, existing and future, will work with a "virtual connection"?
>>>
>>> E.g. in case of TCP, you WILL have separate connections for each TCP candidate - each using a single 5-tuple. We can't say >>that, with ICE, a single TCP connection can suddenly span multiple 5-tuples (one for each candidate pair), can we?
>>
>> I think there is some confusion between the underlying link protocol used by candidates and the protocol running on top > of ICE. I do not think TCP is defined as a protocol that can operate on top of ICE. It is defined in RFC 6544 using framing
>> from RFC 4571, as a protocol for underlying links. Whatever protocol is defined on top of ICE should support operation on > top of packetized communication protocol provided by ICE and should define how its packets can be demuxed from STUN > packets used for ICE connectivity checks. So far things like RTP, DTLS, and SCTP fit such requirements.
>
> I assume you mean SCTP-over-DTLS? Usage of "plain" SCTP with ICE is not defined, as far as I know.

Yes. This means that it is DTLS, not SCTP that is running over ICE, and 
that must be prepared for dealing with multiple 5-tuples. DTLS should 
make this transparent to SCTP.

>> New things can be defined in the future. When they do, they should treat ICE a virtual communication channel that
>> provides unreliable packet transport with no order guarantees which can span multiple 5-tuples.
>
> Then the scope of what we discuss now should not be "whatever protocol" - it should be the specific protocols we are discussing.

AFAIK RTP and DTLS are the only things that have been considered. 
Possibly there are other things that could successfully run directly 
over ICE.

	Thanks,
	Paul

> Regards,
>
> Christer
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
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Subject: [rtcweb] Comments on draft-ietf-rtcweb-constraints-registry-02.txt
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Hi,

I had a quick review of draft-ietf-rtcweb-constraints-registry-02. So
some few comments.

1. Shouldn't the name of the registry be "WebRTC Constrainable
Properties"? My understanding was that we are using WebRTC rather than
RTCWeb as the name for the overarching thing.

2. Is there a good reason why not a outer limit on the property name
can't be set, even if it is large. Would not for example 256 characters
be sufficient?

3. I find no requirements on the reference. Should there be guidance on
this to the expert reviewer. For example are references that are secret
and can't even be given to the expert reviewer allowed? Are references
under a pay-wall okay, as long as the reference is provided the expert
reviewer? I think the expectation should be made clear. I think their
are good points to allow almost anyone to register, however we might
want to make clear the expectancy that the expert reviewer is provided
with one copy, possibly under NDA if really required.

Cheers

Magnus Westerlund

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Subject: [rtcweb] Comments on draft-ietf-rtcweb-overview-13
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Hi,

I have reviewed the overview document and have some comments.


1. Section 2.2:

   o  A Javascript API specification, done in the W3C
      [W3C.WD-webrtc-20120209][W3C.WD-mediacapture-streams-20120628]

To me it looks these references needs a bit of an update.

2. Section 2.2:
 Together, these two specifications aim to provide an environment
   where Javascript embedded in any page,

I think the sentence would be more correct if "two specifications" would
be changed to "two sets of two specifications" as neither is a single
document.

3. Section 2.2:
   o  A WebRTC gateway is a WebRTC-compatible endpoint that mediates
      media traffic to non-WebRTC entities.

I have some difficulties to accept endpoint with the current
formulation. To me an gateway is an middlebox that performs
transformations that makes an non-WebRTC entity look like a
WebRTC-compatible endpoint.

4. Section 2.4:

   API:  Application Programming Interface - a specification of a set of
      calls and events, usually tied to a programming language or an
      abstract formal specification such as WebIDL, with its defined
      semantics.

Missing reference to WebIDL

5. Section 2.5:
   Media path:  The path that media data follows from one WebRTC
      endpoint to another.

I get the impression that there are an implicit "network path" that this
one defines, or?

6. Section 7.

   3.  When a new codec is specified, and the SDP for the new codec is
       specified in the MMUSIC WG, no other standardization should be
       required for it to be possible to use that in the web browsers.

I think the reference here to MMUSIC is wrong. Only in very unusual
cases would one need to go to MMUSIC to define new SDP parts to
introduce any new codec. I would suggest to simply remove "in the MMUSIC
WG".

7. Section 13.2:

Is really the 2011 WD the right reference for HTML5?


In large this document is in pretty good shape. And I think the main
question is when to say that these documents are really the basic set of
WebRTC and now we shall publish it.


Cheers

Magnus Westerlund

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Subject: [rtcweb] Comments on draft-ietf-rtcweb-transports-08
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Hi,

I have reviewed draft-ietf-rtcweb-transports-08 and have some comments.

1. Section 1:

   This document describes requirements that apply to all WebRTC
   devices.  When there are requirements that apply only to WebRTC
   browsers, this is called out by using the word "browser".

Shouldn't this be a reference to overview and rather grab the relevant
terms from there?

2. Section 3.1:
   o  UDP.  This is the protocol assumed by most protocol elements
      described.

   o  TCP.  This is used for HTTP/WebSockets, as well as for TURN/SSL
      and ICE-TCP.

References to UDP and TCP?

3. Section 3.3:

   When a client gathers all IPv6 addresses on a host, and both
   temporary addresses and permanent addresses of the same scope are
   present, the client SHOULD discard the permanent addresses before
   forming pairs.

I think the last two words are wrong in this statement. Isn't the point
to actually discard the local candidates with permanent addresses where
a temporary address candidate has been created for the same interface?

4. Section 3.5:

WebRTC implementations MUST support multiplexing of DTLS and RTP over
   the same port pair, as described in the DTLS_SRTP specification
   [RFC5764], section 5.1.2.

DTLS_SRTP should be DTLS-SRTP

5. Section 4.1:
(Question: Does there need
   to be an API knob to turn off DSCP markings?)

So the thinking here would be that one like to apply the other
prioritization, but like to avoid any fear that any DSCP marking other
than Best Effort can cause issues on the network path.

I am personally quite uncertain if it is needed, but if it is to exist,
I think the know should be such that one force the DSCP setting to best
effort for all transport flows.

But, I think it is time this open issue is resolved.

6.  Section 4.1:

All packets arrying data from the SCTP association supporting the
   data channels MUST use a single DSCP code point.

s/arrying/carrying

7. Section 3.5:

   The setup protocol for WebRTC data channels is described in
   [I-D.ietf-rtcweb-data-protocol].

I am a bit surprised over the none RFC2119 requirement here and that the
data-protocol is an informative reference. Looking in the data-channel
document, it is also not normatively requiring the data-protocol
documents implementation. Overview is requiring that implementation, but
I don't think the formulation here correspond to the requirement level
in overview. Can we please align?


Cheers

Magnus Westerlund

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Subject: [rtcweb] Comments on draft-ietf-rtcweb-alpn-01
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Hi,

I have reviewed draft-ietf-rtcweb-alpn-01 and have a comment:

1. Section 5:

I think the IANA section could be more clearly separate the two entries
so that it is clearer that there are two being registered.


Cheers

Magnus Westerlund

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On 3/11/15 08:09, Magnus Westerlund wrote:
> I have reviewed the draft and have some few comments.
>
> 1. Section 5: Spelling error "WebRTC Brower"

Thanks.

>
> 2. Section 6:
>
>     SDP allows for codec-independent indication of preferred video
>     resolutions using the mechanism described in [RFC6236]. If a
>     recipient of video indicates a receiving resolution, the sender
>     SHOULD accommodate this resolution, as the receiver may not be
>     capable of handling higher resolutions.
>
> I would note that this document does not indicate an expectation of a
> WebRTC Browser or non-browser to actually support a=imageattr. I would
> have expected a clear formulation using RFC 2119 words on the need to
> support this.
>
> I also note that JSEP-09 lists a=imageattr as an open issue.
>
> I think this unclarity needs to be improved. Either be explicit about
> JSEP defining the need for implementing the attribute, or define in this
> document, and JSEP's open issue will be resolved.

Thanks for calling attention to this. I agree that the current treatment 
is a bit weak. I can also speak, from an implementation perspective, to 
the fact that we have run into circumstances (generally on lower-end 
mobile devices) where honoring imageattr is important. On the other 
hand, I do know that you can easily end up with encoding situations 
where the sending device has limited or no control over the resolution 
being encoded. I think MAY send and SHOULD honor is the right strength 
(and is in keeping with what the document currently says).

So I propose the following, which isn't a change in content as much as a 
change in clarity:

    SDP allows for codec-independent indication of preferred video
    resolutions
    using the mechanism described in {{RFC6236}}. WebRTC endpoints MAY
    send an
    "a=imageattr" attribute to indicate the maximum resolution they wish to
    receive. Senders SHOULD interpret and honor this attribute by
    limiting the
    encoded resolution to the indicated maximum size, as the receiver
    may not be
    capable of handling higher resolutions.


> 3. Section 7:
>
> Implementers should consider
>     whether the use of variable bit rate video codecs are appropriate for
>     their application based on [RFC6562].
>
> I don't quite understand the reference to RFC6562 here. That discusses
> that audio codecs that have a variable rate based on the audio input can
> leak information about what is said. This is primarily done through
> voicing classifications on the frame basis that reveals the spoken
> words. I fail to understand how that applies to video.

There's an equivalent issue with video, in that variable bit rates can 
reveal the degree of motion in the frame. For something like a 
videoconference, this might not be a major issue. For something more 
like a surveillance camera trained on a typically static scene, this can 
allow passive attackers to deduce whether activity is taking place in 
the space covered by the camera.

I agree that 6562 isn't an ideal treatment for this topic, but neither 
do I think I should flesh this section out to be a complete "6562 for 
video."

So, I propose a brief summary of the issue: "Implementors should 
consider whether the use of variable bit rate video codecs are 
appropriate for their application, keeping in mind that the degree of 
inter-frame change (and, by inference, the amount of motion in the 
frame) may be deduced by an eavesdropper based on the video stream's bit 
rate."


> 4. Section 10.1: This list to be a bit bloated. I find no reference to
> the following reference, which also ID-nits find.
>
>    == Unused Reference: 'RFC4175' is defined on line 345, but no explicit
>       reference was found in the text
>
>    == Unused Reference: 'RFC4421' is defined on line 348, but no explicit
>       reference was found in the text
>
>    == Unused Reference: 'RFC5104' is defined on line 352, but no explicit
>       reference was found in the text
>

Thanks. I've removed these.

/a

--------------060600020906030805070405
Content-Type: text/html; charset=utf-8
Content-Transfer-Encoding: 7bit

<html>
  <head>
    <meta content="text/html; charset=utf-8" http-equiv="Content-Type">
  </head>
  <body text="#000000" bgcolor="#FFFFFF">
    <div class="moz-cite-prefix">On 3/11/15 08:09, Magnus Westerlund
      wrote:<br>
    </div>
    <blockquote cite="mid:55003E81.1020107@ericsson.com" type="cite">
      <pre wrap="">
I have reviewed the draft and have some few comments.

1. Section 5: Spelling error "WebRTC Brower"</pre>
    </blockquote>
    <br>
    Thanks.<br>
    <br>
    <blockquote cite="mid:55003E81.1020107@ericsson.com" type="cite">
      <pre wrap="">

2. Section 6:

   SDP allows for codec-independent indication of preferred video
   resolutions using the mechanism described in [RFC6236]. If a
   recipient of video indicates a receiving resolution, the sender
   SHOULD accommodate this resolution, as the receiver may not be
   capable of handling higher resolutions.

I would note that this document does not indicate an expectation of a
WebRTC Browser or non-browser to actually support a=imageattr. I would
have expected a clear formulation using RFC 2119 words on the need to
support this.

I also note that JSEP-09 lists a=imageattr as an open issue.

I think this unclarity needs to be improved. Either be explicit about
JSEP defining the need for implementing the attribute, or define in this
document, and JSEP's open issue will be resolved.</pre>
    </blockquote>
    <br>
    Thanks for calling attention to this. I agree that the current
    treatment is a bit weak. I can also speak, from an implementation
    perspective, to the fact that we have run into circumstances
    (generally on lower-end mobile devices) where honoring imageattr is
    important. On the other hand, I do know that you can easily end up
    with encoding situations where the sending device has limited or no
    control over the resolution being encoded. I think MAY send and
    SHOULD honor is the right strength (and is in keeping with what the
    document currently says).<br>
    <br>
    So I propose the following, which isn't a change in content as much
    as a change in clarity:<br>
    <br>
    <blockquote>SDP allows for codec-independent indication of preferred
      video resolutions<br>
      using the mechanism described in {{RFC6236}}. WebRTC endpoints MAY
      send an<br>
      "a=imageattr" attribute to indicate the maximum resolution they
      wish to<br>
      receive. Senders SHOULD interpret and honor this attribute by
      limiting the<br>
      encoded resolution to the indicated maximum size, as the receiver
      may not be<br>
      capable of handling higher resolutions.<br>
    </blockquote>
    <br>
    <blockquote cite="mid:55003E81.1020107@ericsson.com" type="cite">
      <pre wrap="">
3. Section 7:

Implementers should consider
   whether the use of variable bit rate video codecs are appropriate for
   their application based on [RFC6562].

I don't quite understand the reference to RFC6562 here. That discusses
that audio codecs that have a variable rate based on the audio input can
leak information about what is said. This is primarily done through
voicing classifications on the frame basis that reveals the spoken
words. I fail to understand how that applies to video.</pre>
    </blockquote>
    <br>
    There's an equivalent issue with video, in that variable bit rates
    can reveal the degree of motion in the frame. For something like a
    videoconference, this might not be a major issue. For something more
    like a surveillance camera trained on a typically static scene, this
    can allow passive attackers to deduce whether activity is taking
    place in the space covered by the camera.<br>
    <br>
    I agree that 6562 isn't an ideal treatment for this topic, but
    neither do I think I should flesh this section out to be a complete
    "6562 for video."<br>
    <br>
    So, I propose a brief summary of the issue: "Implementors should
    consider whether the use of variable bit rate video codecs are
    appropriate for their application, keeping in mind that the degree
    of inter-frame change (and, by inference, the amount of motion in
    the frame) may be deduced by an eavesdropper based on the video
    stream's bit rate."<br>
    <br>
    <br>
    <blockquote cite="mid:55003E81.1020107@ericsson.com" type="cite">
      <pre wrap="">
4. Section 10.1: This list to be a bit bloated. I find no reference to
the following reference, which also ID-nits find.

  == Unused Reference: 'RFC4175' is defined on line 345, but no explicit
     reference was found in the text

  == Unused Reference: 'RFC4421' is defined on line 348, but no explicit
     reference was found in the text

  == Unused Reference: 'RFC5104' is defined on line 352, but no explicit
     reference was found in the text

</pre>
    </blockquote>
    <br>
    Thanks. I've removed these.<br>
    <br>
    /a<br>
  </body>
</html>

--------------060600020906030805070405--


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On 17 March 2015 at 07:38, Magnus Westerlund
<magnus.westerlund@ericsson.com> wrote:
> I have reviewed draft-ietf-rtcweb-alpn-01 and have a comment:
>
> 1. Section 5:
>
> I think the IANA section could be more clearly separate the two entries
> so that it is clearer that there are two being registered.

I've played with the formatting a little to (hopefully) help avoid this.

I found more problems than you did :)  Some duplication, some
potential confusion about the distinction between media and data.
I've updated my copy:
<http://martinthomson.github.io/drafts/draft-ietf-rtcweb-alpn.html>
We can discuss what to do with that at the meeting.


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Subject: Re: [rtcweb] WGLC for draft-ietf-rtcweb-video-04
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Hi,

Please see inline comments on this.

On 2015-03-17 17:55, Adam Roach wrote:
> On 3/11/15 08:09, Magnus Westerlund wrote:
>>
>> 2. Section 6:
>>
>>    SDP allows for codec-independent indication of preferred video
>>    resolutions using the mechanism described in [RFC6236]. If a
>>    recipient of video indicates a receiving resolution, the sender
>>    SHOULD accommodate this resolution, as the receiver may not be
>>    capable of handling higher resolutions.
>>
>> I would note that this document does not indicate an expectation of a
>> WebRTC Browser or non-browser to actually support a=imageattr. I would
>> have expected a clear formulation using RFC 2119 words on the need to
>> support this.
>>
>> I also note that JSEP-09 lists a=imageattr as an open issue.
>>
>> I think this unclarity needs to be improved. Either be explicit about
>> JSEP defining the need for implementing the attribute, or define in this
>> document, and JSEP's open issue will be resolved.
> 
> Thanks for calling attention to this. I agree that the current treatment
> is a bit weak. I can also speak, from an implementation perspective, to
> the fact that we have run into circumstances (generally on lower-end
> mobile devices) where honoring imageattr is important. On the other
> hand, I do know that you can easily end up with encoding situations
> where the sending device has limited or no control over the resolution
> being encoded. I think MAY send and SHOULD honor is the right strength
> (and is in keeping with what the document currently says).
> 
> So I propose the following, which isn't a change in content as much as a
> change in clarity:
> 
>     SDP allows for codec-independent indication of preferred video
>     resolutions
>     using the mechanism described in {{RFC6236}}. WebRTC endpoints MAY
>     send an
>     "a=imageattr" attribute to indicate the maximum resolution they wish to
>     receive. Senders SHOULD interpret and honor this attribute by
>     limiting the
>     encoded resolution to the indicated maximum size, as the receiver
>     may not be
>     capable of handling higher resolutions.
> 
> 

Two comments, do we actually want to mandate the implementation support
of a=imageattr?

I think the usage rules are fairly reasonable. I do wonder if we want to
be explicit about the SHOULD exception is really for cases where the
encoder can't be changed, for example due to lacking APIs to affect a
change on a webcam etc.

>> 3. Section 7:
>>
>> Implementers should consider
>>    whether the use of variable bit rate video codecs are appropriate for
>>    their application based on [RFC6562].
>>
>> I don't quite understand the reference to RFC6562 here. That discusses
>> that audio codecs that have a variable rate based on the audio input can
>> leak information about what is said. This is primarily done through
>> voicing classifications on the frame basis that reveals the spoken
>> words. I fail to understand how that applies to video.
> 
> There's an equivalent issue with video, in that variable bit rates can
> reveal the degree of motion in the frame. For something like a
> videoconference, this might not be a major issue. For something more
> like a surveillance camera trained on a typically static scene, this can
> allow passive attackers to deduce whether activity is taking place in
> the space covered by the camera.
> 
> I agree that 6562 isn't an ideal treatment for this topic, but neither
> do I think I should flesh this section out to be a complete "6562 for
> video."
> 
> So, I propose a brief summary of the issue: "Implementors should
> consider whether the use of variable bit rate video codecs are
> appropriate for their application, keeping in mind that the degree of
> inter-frame change (and, by inference, the amount of motion in the
> frame) may be deduced by an eavesdropper based on the video stream's bit
> rate."
> 
> 

Thanks, I hadn't considered that implication of the issue for video. I
think your proposal is appropriate. I note that hiding this behaviour
for the current video codecs is going to be very expensive in terms of
bits. The recommendation for mitigations that are in RFC6562 is not
really realistic to apply.


Cheers

Magnus Westerlund

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Hi,

I have reviewed draft-ietf-rtcweb-fec-01 and have some comments.

1. Title:

My impression of the document from the title is that this is only
requirements and no normative specification of FEC for WebRTC. Thus, I
would suggest changing the title. I think one can simply remove
"requirements" by changing the order, i.e. "Forward Error Correction in
WebRTC".

2. Section 3.1:
   This approach, as described in [RFC5956], Section 4.3, sends FEC
   packets as an independent SSRC-multiplexed stream, with its own SSRC
   and payload type.  While by far the most flexible, each FEC packet
   will have its own IP+UDP+RTP+FEC header, leading to additional
   overhead of the FEC stream.

I think this section must have more description of what "Separate FEC
stream" really is. The "FEC Grouping for SSRC-Multiplexed RTP Streams"
reference also appear to restrictive. Based on the transport documents
there actually exists an requirement for supporting fully unbundle
configuration. That is not really matching what Section 4.3 describes.

>From my perspective there are two sets of orthogonal questions around
FEC streams. First the relation between the source flow(s) and the
repair flows, which may be 1 to 1, 1 to M, or N to 1, or N to M. Then we
have the other question of how they are transmitted in relation to the
source flow. Within the same RTP session, on different RTP session,
outside of the RTP session. And if one has many source flows, then one
can also discuss if there is N source RTP sessions and M repair sessions
or 1 source RTP session and 1 repair session.

3. Section 3.4:

I think you want to discuss FEC information as a "redundant" encoding also.


4. Section 4.1:
   When using constant-bitrate codecs, e.g.  PCMU, use of [RFC2198]
   redundant encoding MAY be used, but note that this will result in a
   potentially significant bitrate increase, and that suddenly
   increasing bitrate to deal with losses from congestion may actually
   make things worse.

   Because of the lower packet rate of audio encodings, usually a single
   packet per frame, use of a separate FEC stream comes with a higher
   overhead than other mechanisms, and therefore is NOT RECOMMENDED.

I think this fails to discuss the quite reasonable application of XOR
FEC as a redundant encoding information. That way one can achieve a
bitrate increase that is ~1/N of the source flow bit-rate by including
one XOR packet every N packets covering those N packets. Sure that only
gives single packet loss protection, but might be appropriate anyway.

5. Section 4.2:

I think this more clearer needs to reference and possibly discuss the
signalling context it is operating in, i.e. JSEP.

6. Section 5.1:
   Note that this only allows the FEC stream to protect a single primary
   stream.  Support for protecting multiple primary streams with a
   single FEC stream is complicated by WebRTC's 1-m-line-per-stream
   policy and requires further study.

For efficient application of FEC this is the way one have to go.
Unfortunately that also requires FEC codes that are more complex than
XOR and also more likely to be IPR encumbered. But if ones goal is to
have low delay, be almost independent of loss pattern and have
protection for multi packet loss, then aggregating together all the data
sent in a relatively short intervals (<50 ms) and protect them together
is the way to go.

Unfortunately I don't think we have all the pieces in place here in IETF
to perform such a protection, especially in the context of BUNDLE as
well as the possibility to fully unbundle them. It mostly an exercise to
put the pieces and the necessary identification required for aggregating
flows into a common source block and signal that, and make it clear
where the protection data is sent that is missing.

I was part of doing this in 3GPP Multimedia Broadcast/Multicast Service
(MBMS), but that system performs the FEC operation on UDP Flows as a way
to handle both RTP/RTCP as well as MIKEY messages with the keys to
media. It is also a streaming service with more relaxed timing
requirements.

If you are interested to see how that is specified please see Section
8.2.2 of TS 26.346:
http://www.3gpp.org/ftp/Specs/archive/26_series/26.346/26346-c40.zip

I personally think this is the direction to go to achieve efficient FEC
usage for the real-time media, especially when you have multi-stream
situations. However, I can't contribute the hours needed to make rapid
enough progress on this. Thus, I guess this needs to remain a potential
second phase of the FEC work for WebRTC.

7. Section 6.

I think this section must discuss the fact that the data channel have
several possibilities for dealing with reliability. First one can use a
reliable stream, that will do retransmissions. One can use semi-reliable
stream to not needless continue to try, or the application can select to
repeat the data itself over an unreliable stream. I think the last
should be recommended against as it will waste resources unnecessarily.

8. Section 7:

What about using RFC 2198 and [I-D.ietf-payload-flexible-fec-scheme]? Or
isn't that supported at all by [I-D.ietf-payload-flexible-fec-scheme]?
(Yes, I will review that spec, but haven't gotten to it yet).

9. Section 9.

I think there is a point of discussing how FEC configuration can cause
significant waste of bandwidth. Making even a encoded stream with modest
bandwidth needs into a bandwidth consuming monster simply by influencing
the usage of FEC and its parameters. Thus, I think this document needs
to discuss the issues with possible interference from malicious
applications with the signalling and the need for a WebRTC
implementation to actually control the application and usage of FEC
itself and not rely on the signalling parameters.


Cheers

Magnus Westerlund

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Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
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Thanks for the review!
Skipping nits....

On 03/17/2015 03:33 PM, Magnus Westerlund wrote:
> Hi,
>
> I have reviewed draft-ietf-rtcweb-transports-08 and have some comments.
>
> 1. Section 1:
>
>     This document describes requirements that apply to all WebRTC
>     devices.  When there are requirements that apply only to WebRTC
>     browsers, this is called out by using the word "browser".
>
> Shouldn't this be a reference to overview and rather grab the relevant
> terms from there?

Good point.

>
> 2. Section 3.1:
>     o  UDP.  This is the protocol assumed by most protocol elements
>        described.
>
>     o  TCP.  This is used for HTTP/WebSockets, as well as for TURN/SSL
>        and ICE-TCP.
>
> References to UDP and TCP?

What would you suggest? RFC 791 is not the best TCP reference any more...

>
> 3. Section 3.3:
>
>     When a client gathers all IPv6 addresses on a host, and both
>     temporary addresses and permanent addresses of the same scope are
>     present, the client SHOULD discard the permanent addresses before
>     forming pairs.
>
> I think the last two words are wrong in this statement. Isn't the point
> to actually discard the local candidates with permanent addresses where
> a temporary address candidate has been created for the same interface?

"forming pairs"?

I don't understand how your comment relates to those 2 words, so I'm not 
sure what you want me to change.

The point is that if we get the same connectivity with a temporary 
address, we shouldn't expose a permanent address to the JS layer at all.

In the case of multiple interfaces, some of which have temporary 
addresses and some of which have only permanent addresses, I'm not sure 
what the right thing is. Commentary?

>
> 4. Section 3.5:
>
> WebRTC implementations MUST support multiplexing of DTLS and RTP over
>     the same port pair, as described in the DTLS_SRTP specification
>     [RFC5764], section 5.1.2.
>
> DTLS_SRTP should be DTLS-SRTP

Yup.
>
> 5. Section 4.1:
> (Question: Does there need
>     to be an API knob to turn off DSCP markings?)
>
> So the thinking here would be that one like to apply the other
> prioritization, but like to avoid any fear that any DSCP marking other
> than Best Effort can cause issues on the network path.
>
> I am personally quite uncertain if it is needed, but if it is to exist,
> I think the know should be such that one force the DSCP setting to best
> effort for all transport flows.
>
> But, I think it is time this open issue is resolved.

That makes two of us being uncertain :-) Anyone else?
>
> 6.  Section 4.1:
>
> All packets arrying data from the SCTP association supporting the
>     data channels MUST use a single DSCP code point.
>
> s/arrying/carrying
>
> 7. Section 3.5:
>
>     The setup protocol for WebRTC data channels is described in
>     [I-D.ietf-rtcweb-data-protocol].
>
> I am a bit surprised over the none RFC2119 requirement here and that the
> data-protocol is an informative reference. Looking in the data-channel
> document, it is also not normatively requiring the data-protocol
> documents implementation. Overview is requiring that implementation, but
> I don't think the formulation here correspond to the requirement level
> in overview. Can we please align?

Suggested fix: "The setup protocol for WebRTC data channels described in 
[I-D....] MUST be supported".
And data-protocol moved to normative references.

Any objections to this?

>
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> FÃ¤rÃ¶gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Hi,

Please see inline.

On 2015-03-18 14:02, Harald Alvestrand wrote:
> 
> On 03/17/2015 03:33 PM, Magnus Westerlund wrote:
>> Hi,
>>
>> I have reviewed draft-ietf-rtcweb-transports-08 and have some comments.
>>
>> 1. Section 1:
>>
>>     This document describes requirements that apply to all WebRTC
>>     devices.  When there are requirements that apply only to WebRTC
>>     browsers, this is called out by using the word "browser".
>>
>> Shouldn't this be a reference to overview and rather grab the relevant
>> terms from there?
> 
> Good point.
> 
>>
>> 2. Section 3.1:
>>     o  UDP.  This is the protocol assumed by most protocol elements
>>        described.
>>
>>     o  TCP.  This is used for HTTP/WebSockets, as well as for TURN/SSL
>>        and ICE-TCP.
>>
>> References to UDP and TCP?
> 
> What would you suggest? RFC 791 is not the best TCP reference any more...

I assume you mean RFC 793.

A possible alternative is RFC 7414.

The TSV directorate had a discussion about this, and for a general TCP
reference it is quite fine to continue use RFC 793.

> 
>>
>> 3. Section 3.3:
>>
>>     When a client gathers all IPv6 addresses on a host, and both
>>     temporary addresses and permanent addresses of the same scope are
>>     present, the client SHOULD discard the permanent addresses before
>>     forming pairs.
>>
>> I think the last two words are wrong in this statement. Isn't the point
>> to actually discard the local candidates with permanent addresses where
>> a temporary address candidate has been created for the same interface?
> 
> "forming pairs"?
> 
> I don't understand how your comment relates to those 2 words, so I'm not
> sure what you want me to change.

Well, if the purpose is to not use the permanent address, then I don't
see how one can create candidates even with these addressed.

My reaction is that forming pairs is done later in the process, when the
gathering is already completed. Thus, pruning should happen during the
gathering phase.

> 
> The point is that if we get the same connectivity with a temporary
> address, we shouldn't expose a permanent address to the JS layer at all.

Fully agree with this purpose.

> 
> In the case of multiple interfaces, some of which have temporary
> addresses and some of which have only permanent addresses, I'm not sure
> what the right thing is. Commentary?

I think one would have to expose the permanent address if that is the
only thing one can get from that interface.

The one alternative I could think of, is that the gathering of any
permanent address would depend on the privacy setting of the WebRTC
endpoint.


>>
>> 5. Section 4.1:
>> (Question: Does there need
>>     to be an API knob to turn off DSCP markings?)
>>
>> So the thinking here would be that one like to apply the other
>> prioritization, but like to avoid any fear that any DSCP marking other
>> than Best Effort can cause issues on the network path.
>>
>> I am personally quite uncertain if it is needed, but if it is to exist,
>> I think the know should be such that one force the DSCP setting to best
>> effort for all transport flows.
>>
>> But, I think it is time this open issue is resolved.
> 
> That makes two of us being uncertain :-) Anyone else?
>>
>> 6.  Section 4.1:
>>
>> All packets arrying data from the SCTP association supporting the
>>     data channels MUST use a single DSCP code point.
>>
>> s/arrying/carrying
>>
>> 7. Section 3.5:
>>
>>     The setup protocol for WebRTC data channels is described in
>>     [I-D.ietf-rtcweb-data-protocol].
>>
>> I am a bit surprised over the none RFC2119 requirement here and that the
>> data-protocol is an informative reference. Looking in the data-channel
>> document, it is also not normatively requiring the data-protocol
>> documents implementation. Overview is requiring that implementation, but
>> I don't think the formulation here correspond to the requirement level
>> in overview. Can we please align?
> 
> Suggested fix: "The setup protocol for WebRTC data channels described in
> [I-D....] MUST be supported".
> And data-protocol moved to normative references.
> 
> Any objections to this?

Sounds good to me.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
FÃ¤rÃ¶gatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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From nobody Wed Mar 18 08:42:32 2015
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From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Comments on draft-ietf-rtcweb-transports-08
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On 03/18/2015 04:31 PM, Magnus Westerlund wrote:
> Hi,
>
> Please see inline.
>
> On 2015-03-18 14:02, Harald Alvestrand wrote:
>> On 03/17/2015 03:33 PM, Magnus Westerlund wrote:
>>> Hi,
>>>
>>> I have reviewed draft-ietf-rtcweb-transports-08 and have some comments.
>>>
>>> 1. Section 1:
>>>
>>>      This document describes requirements that apply to all WebRTC
>>>      devices.  When there are requirements that apply only to WebRTC
>>>      browsers, this is called out by using the word "browser".
>>>
>>> Shouldn't this be a reference to overview and rather grab the relevant
>>> terms from there?
>> Good point.
>>
>>> 2. Section 3.1:
>>>      o  UDP.  This is the protocol assumed by most protocol elements
>>>         described.
>>>
>>>      o  TCP.  This is used for HTTP/WebSockets, as well as for TURN/SSL
>>>         and ICE-TCP.
>>>
>>> References to UDP and TCP?
>> What would you suggest? RFC 791 is not the best TCP reference any more...
> I assume you mean RFC 793.
>
> A possible alternative is RFC 7414.
>
> The TSV directorate had a discussion about this, and for a general TCP
> reference it is quite fine to continue use RFC 793.
>
>>> 3. Section 3.3:
>>>
>>>      When a client gathers all IPv6 addresses on a host, and both
>>>      temporary addresses and permanent addresses of the same scope are
>>>      present, the client SHOULD discard the permanent addresses before
>>>      forming pairs.
>>>
>>> I think the last two words are wrong in this statement. Isn't the point
>>> to actually discard the local candidates with permanent addresses where
>>> a temporary address candidate has been created for the same interface?
>> "forming pairs"?
>>
>> I don't understand how your comment relates to those 2 words, so I'm not
>> sure what you want me to change.
> Well, if the purpose is to not use the permanent address, then I don't
> see how one can create candidates even with these addressed.
>
> My reaction is that forming pairs is done later in the process, when the
> gathering is already completed. Thus, pruning should happen during the
> gathering phase.

<lightbulb switching on>

aha - in the case where you're doing a createOffer, the addresses must 
be discarded before you do the createOffer, which is long before the 
time at which you form pairs (after setRemoteDescription) - so the text 
"before forming pairs" is misleading.

I'll replace the text with "before exposing addresses to the application".

>
>> The point is that if we get the same connectivity with a temporary
>> address, we shouldn't expose a permanent address to the JS layer at all.
> Fully agree with this purpose.
>
>> In the case of multiple interfaces, some of which have temporary
>> addresses and some of which have only permanent addresses, I'm not sure
>> what the right thing is. Commentary?
> I think one would have to expose the permanent address if that is the
> only thing one can get from that interface.
>
> The one alternative I could think of, is that the gathering of any
> permanent address would depend on the privacy setting of the WebRTC
> endpoint.

Saying that would require introducing the term "privacy setting", which 
we so far have not done.

(snipping stuff that's OK with me)


From nobody Thu Mar 19 02:54:51 2015
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This is a multi-part message in MIME format.
--------------030305020101070904060307
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Speaking with my W3C WG chair hat on:

An item has come up that relates to the IETF draft 
"draft-ietf-rtcweb-security-arch".

The issue is this requirement:

    Implementations MUST obtain explicit user consent prior to providing
    access to the camera and/or microphone.  Implementations MUST at
    minimum support the following two permissions models for HTTPS
    origins.

    o  Requests for one-time camera/microphone access.
    o  Requests for permanent access.

.....

    API Requirement:  The API MUST provide a mechanism for the requesting
       JS to indicate which of these forms of permissions it is
       requesting.  This allows the browser client to know what sort of
       user interface experience to provide to the user, including what
       permissions to request from the user and hence what to enforce
       later.  For instance, browsers might display a non-invasive door
       hanger ("some features of this site may not work..." when asking
       for long-term permissions) but a more invasive UI ("here is your
       own video") for single-call permissions.  The API MAY grant weaker
       permissions than the JS asked for if the user chooses to authorize
       only those permissions, but if it intends to grant stronger ones
       it SHOULD display the appropriate UI for those permissions and
       MUST clearly indicate what permissions are being requested.


At the moment, the W3C Media Capture TF doesn't have a mechanism for the 
JS to indicate whether it desires one-time access or permanent access. 
There hasn't been a request for this from app developers, and none of 
the implementations so far have added such a feature.

Having a requirement specification and an API specification that don't 
match is obviously not a Good Thing. There are two solutions that 
suggest themselves:

- Delete the requirement
- Add a mechanism

Discussion on the public-media-capture list did not provide a consensus 
that one of these alternatives is clearly superior to the other, and the 
suggestion was made that face-to-face discussion would be a reasonable 
way to move forward.

Since the security-arch document is an IETF document, and the IETF 
meeting is next week, it seems reasonable to ask for some agenda time 
next week for this issue.

Can this be accomodated?

Harald


--------------030305020101070904060307
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<html>
  <head>

    <meta http-equiv="content-type" content="text/html; charset=utf-8">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    Speaking with my W3C WG chair hat on:<br>
    <br>
    An item has come up that relates to the IETF draft
    "draft-ietf-rtcweb-security-arch".<br>
    <br>
    The issue is this requirement:<br>
    <br>
    Â Â  Implementations MUST obtain explicit user consent prior to
    providing<br>
    Â Â  access to the camera and/or microphone.Â  Implementations MUST at<br>
    Â Â  minimum support the following two permissions models for HTTPS<br>
    Â Â  origins.<br>
    <br>
    Â Â  oÂ  Requests for one-time camera/microphone access.<br>
    Â Â  oÂ  Requests for permanent access.<br>
    <br>
    .....<br>
    <br>
    <pre class="" style="font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><font face="arial, helvetica, sans-serif">   API Requirement:  The API MUST provide a mechanism for the requesting
      JS to indicate which of these forms of permissions it is
      requesting.  This allows the browser client to know what sort of
      user interface experience to provide to the user, including what
      permissions to request from the user and hence what to enforce
      later.  For instance, browsers might display a non-invasive door
      hanger ("some features of this site may not work..." when asking
      for long-term permissions) but a more invasive UI ("here is your
      own video") for single-call permissions.  The API MAY grant weaker
      permissions than the JS asked for if the user chooses to authorize
      only those permissions, but if it intends to grant stronger ones
      it SHOULD display the appropriate UI for those permissions and
      MUST clearly indicate what permissions are being requested.</font></pre>
    <br>
    At the moment, the W3C Media Capture TF doesn't have a mechanism for
    the JS to indicate whether it desires one-time access or permanent
    access. There hasn't been a request for this from app developers,
    and none of the implementations so far have added such a feature.<br>
    <br>
    Having a requirement specification and an API specification that
    don't match is obviously not a Good Thing. There are two solutions
    that suggest themselves:<br>
    <br>
    - Delete the requirement<br>
    - Add a mechanism<br>
    <br>
    Discussion on the public-media-capture list did not provide a
    consensus that one of these alternatives is clearly superior to the
    other, and the suggestion was made that face-to-face discussion
    would be a reasonable way to move forward.<br>
    <br>
    Since the security-arch document is an IETF document, and the IETF
    meeting is next week, it seems reasonable to ask for some agenda
    time next week for this issue.<br>
    <br>
    Can this be accomodated?<br>
    <br>
    Harald<br>
    <br>
  </body>
</html>

--------------030305020101070904060307--


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A suggested PR for addressing Magnus' comments is here:

https://github.com/rtcweb-wg/rtcweb-transport/pull/3

For nits, please make comments on the PR. For substantive issues, please 
send to the list.


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Subject: Re: [rtcweb] Agenda time request: Security API requirements for persistent permissions
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We have added this topic to the working copy of the agenda, and it will be
in the first day.  We've allocated 15 minutes for it at the moment, but we
can probably accommodate more on day two if need be.

regards,

Ted

On Thu, Mar 19, 2015 at 2:54 AM, Harald Alvestrand <harald@alvestrand.no>
wrote:

>  Speaking with my W3C WG chair hat on:
>
> An item has come up that relates to the IETF draft
> "draft-ietf-rtcweb-security-arch".
>
> The issue is this requirement:
>
>    Implementations MUST obtain explicit user consent prior to providing
>    access to the camera and/or microphone.  Implementations MUST at
>    minimum support the following two permissions models for HTTPS
>    origins.
>
>    o  Requests for one-time camera/microphone access.
>    o  Requests for permanent access.
>
> .....
>
>    API Requirement:  The API MUST provide a mechanism for the requesting
>       JS to indicate which of these forms of permissions it is
>       requesting.  This allows the browser client to know what sort of
>       user interface experience to provide to the user, including what
>       permissions to request from the user and hence what to enforce
>       later.  For instance, browsers might display a non-invasive door
>       hanger ("some features of this site may not work..." when asking
>       for long-term permissions) but a more invasive UI ("here is your
>       own video") for single-call permissions.  The API MAY grant weaker
>       permissions than the JS asked for if the user chooses to authorize
>       only those permissions, but if it intends to grant stronger ones
>       it SHOULD display the appropriate UI for those permissions and
>       MUST clearly indicate what permissions are being requested.
>
>
> At the moment, the W3C Media Capture TF doesn't have a mechanism for the
> JS to indicate whether it desires one-time access or permanent access.
> There hasn't been a request for this from app developers, and none of the
> implementations so far have added such a feature.
>
> Having a requirement specification and an API specification that don't
> match is obviously not a Good Thing. There are two solutions that suggest
> themselves:
>
> - Delete the requirement
> - Add a mechanism
>
> Discussion on the public-media-capture list did not provide a consensus
> that one of these alternatives is clearly superior to the other, and the
> suggestion was made that face-to-face discussion would be a reasonable way
> to move forward.
>
> Since the security-arch document is an IETF document, and the IETF meeting
> is next week, it seems reasonable to ask for some agenda time next week for
> this issue.
>
> Can this be accomodated?
>
> Harald
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:tahoma,s=
ans-serif;font-size:small">We have added this topic to the working copy of =
the agenda, and it will be in the first day.=C2=A0 We&#39;ve allocated 15 m=
inutes for it at the moment, but we can probably accommodate more on day tw=
o if need be.<br><br></div><div class=3D"gmail_default" style=3D"font-famil=
y:tahoma,sans-serif;font-size:small">regards,<br><br></div><div class=3D"gm=
ail_default" style=3D"font-family:tahoma,sans-serif;font-size:small">Ted<br=
></div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On T=
hu, Mar 19, 2015 at 2:54 AM, Harald Alvestrand <span dir=3D"ltr">&lt;<a hre=
f=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestrand.no</a=
>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 =
0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
 =20

   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000">
    Speaking with my W3C WG chair hat on:<br>
    <br>
    An item has come up that relates to the IETF draft
    &quot;draft-ietf-rtcweb-security-arch&quot;.<br>
    <br>
    The issue is this requirement:<br>
    <br>
    =C2=A0=C2=A0 Implementations MUST obtain explicit user consent prior to
    providing<br>
    =C2=A0=C2=A0 access to the camera and/or microphone.=C2=A0 Implementati=
ons MUST at<br>
    =C2=A0=C2=A0 minimum support the following two permissions models for H=
TTPS<br>
    =C2=A0=C2=A0 origins.<br>
    <br>
    =C2=A0=C2=A0 o=C2=A0 Requests for one-time camera/microphone access.<br=
>
    =C2=A0=C2=A0 o=C2=A0 Requests for permanent access.<br>
    <br>
    .....<br>
    <br>
    <pre style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(=
0,0,0)"><font face=3D"arial, helvetica, sans-serif">   API Requirement:  Th=
e API MUST provide a mechanism for the requesting
      JS to indicate which of these forms of permissions it is
      requesting.  This allows the browser client to know what sort of
      user interface experience to provide to the user, including what
      permissions to request from the user and hence what to enforce
      later.  For instance, browsers might display a non-invasive door
      hanger (&quot;some features of this site may not work...&quot; when a=
sking
      for long-term permissions) but a more invasive UI (&quot;here is your
      own video&quot;) for single-call permissions.  The API MAY grant weak=
er
      permissions than the JS asked for if the user chooses to authorize
      only those permissions, but if it intends to grant stronger ones
      it SHOULD display the appropriate UI for those permissions and
      MUST clearly indicate what permissions are being requested.</font></p=
re>
    <br>
    At the moment, the W3C Media Capture TF doesn&#39;t have a mechanism fo=
r
    the JS to indicate whether it desires one-time access or permanent
    access. There hasn&#39;t been a request for this from app developers,
    and none of the implementations so far have added such a feature.<br>
    <br>
    Having a requirement specification and an API specification that
    don&#39;t match is obviously not a Good Thing. There are two solutions
    that suggest themselves:<br>
    <br>
    - Delete the requirement<br>
    - Add a mechanism<br>
    <br>
    Discussion on the public-media-capture list did not provide a
    consensus that one of these alternatives is clearly superior to the
    other, and the suggestion was made that face-to-face discussion
    would be a reasonable way to move forward.<br>
    <br>
    Since the security-arch document is an IETF document, and the IETF
    meeting is next week, it seems reasonable to ask for some agenda
    time next week for this issue.<br>
    <br>
    Can this be accomodated?<span class=3D"HOEnZb"><font color=3D"#888888">=
<br>
    <br>
    Harald<br>
    <br>
  </font></span></div>

<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Draft RTCWeb Agenda IETF92 
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IETF 92 - Dallas, Texas (v2)
RTCWEB

Chairs:  Cullen Jennings, Sean Turner, Ted Hardie

Wednesday, March 25, 2015 1300-1500 (Gold)

Administrivia:  5 minutes
Chair Doc Review: (Chairs) 15 minutes

W3C Security Issue (HTA+EKR) 15 min

draft-ietf-rtcweb-jsep-09: (EKR) 55 minutes
draft-ietf-rtcweb-fec-00: (Uberti) 30 minutes

Thursday, March 26, 2015 1740-1840 (International):

WGLC resolutions: (Chairs) 0 to 20 minutes
draft-ietf-rtcweb-transports: (HTA) 20 minutes
draft-nandakumar-rtcweb-sdp-07: (Nandakumar) 10 minutes
draft-schwartz-rtcweb-return-04: (Schwartz) 10 minutes


Drafts we have=20

draft-ietf-rtcweb-audio-07.txt
draft-ietf-rtcweb-audio-codecs-for-interop-01.txt
draft-ietf-rtcweb-constraints-registry-01.txt
draft-ietf-rtcweb-data-channel-13.txt (in the RFC editor queue)
draft-ietf-rtcweb-data-protocol-09.txt (in the RFC editor queue)
draft-ietf-rtcweb-fec-00.txt
draft-ietf-rtcweb-overview-13.txt
draft-ietf-rtcweb-rtp-usage-22.txt
draft-ietf-rtcweb-stun-consent-freshness-11.txt (pending write-up)
draft-ietf-rtcweb-transports-07.txt
draft-ietf-rtcweb-video-03.txt


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A new Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

    Title         : WebRTC Video Processing and Codec Requirements

    Author(s)     : A. Roach
    Filename      : draft-ietf-rtcweb-video
    Pages         : 9 
    Date          : 2015-03-19 
    
   This specification provides the requirements and considerations for
   WebRTC applications to send and receive video across a network.  It
   specifies the video processing that is required, as well as video
   codecs and their parameters.

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Subject: Re: [rtcweb] Agenda time request: Security API requirements for persistent permissions
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On 19/03/15 17:49, Ted Hardie wrote:
> We have added this topic to the working copy of the agenda, and it will be
> in the first day.  We've allocated 15 minutes for it at the moment, but we
> can probably accommodate more on day two if need be.

Good to see this discussed. I think I have a conflict or I'd be
hoping to be there. If I'm not, I guess I'd be delighted if that
discussion also includes revocation of permissions.

Some folks might consider that the way to handle revocation of
that is to e.g. revoke all camera/mic permissions for all origins
(as is done with e.g. "private data" on some browsers). I think
that approach is very privacy unfriendly and hope that it's not
considered acceptable for WebRTC. The effect of the privacy
unfriendly approach is that a user cannot revoke permissions for
any origin so long as there is one origin for whom they do not
want to revoke permissions. The net of that is that the user will
not revoke permissions, so the feature at that point may as well
not have been present.

I think that is a real problem. I am not asserting that providing
per-origin revocation interfaces is a perfect answer, but I think
it is defensible as being a lot better.

Cheers,
S.


> 
> regards,
> 
> Ted
> 
> On Thu, Mar 19, 2015 at 2:54 AM, Harald Alvestrand <harald@alvestrand.no>
> wrote:
> 
>>  Speaking with my W3C WG chair hat on:
>>
>> An item has come up that relates to the IETF draft
>> "draft-ietf-rtcweb-security-arch".
>>
>> The issue is this requirement:
>>
>>    Implementations MUST obtain explicit user consent prior to providing
>>    access to the camera and/or microphone.  Implementations MUST at
>>    minimum support the following two permissions models for HTTPS
>>    origins.
>>
>>    o  Requests for one-time camera/microphone access.
>>    o  Requests for permanent access.
>>
>> .....
>>
>>    API Requirement:  The API MUST provide a mechanism for the requesting
>>       JS to indicate which of these forms of permissions it is
>>       requesting.  This allows the browser client to know what sort of
>>       user interface experience to provide to the user, including what
>>       permissions to request from the user and hence what to enforce
>>       later.  For instance, browsers might display a non-invasive door
>>       hanger ("some features of this site may not work..." when asking
>>       for long-term permissions) but a more invasive UI ("here is your
>>       own video") for single-call permissions.  The API MAY grant weaker
>>       permissions than the JS asked for if the user chooses to authorize
>>       only those permissions, but if it intends to grant stronger ones
>>       it SHOULD display the appropriate UI for those permissions and
>>       MUST clearly indicate what permissions are being requested.
>>
>>
>> At the moment, the W3C Media Capture TF doesn't have a mechanism for the
>> JS to indicate whether it desires one-time access or permanent access.
>> There hasn't been a request for this from app developers, and none of the
>> implementations so far have added such a feature.
>>
>> Having a requirement specification and an API specification that don't
>> match is obviously not a Good Thing. There are two solutions that suggest
>> themselves:
>>
>> - Delete the requirement
>> - Add a mechanism
>>
>> Discussion on the public-media-capture list did not provide a consensus
>> that one of these alternatives is clearly superior to the other, and the
>> suggestion was made that face-to-face discussion would be a reasonable way
>> to move forward.
>>
>> Since the security-arch document is an IETF document, and the IETF meeting
>> is next week, it seems reasonable to ask for some agenda time next week for
>> this issue.
>>
>> Can this be accomodated?
>>
>> Harald
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
> 
> 
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 


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From nobody Thu Mar 19 17:03:54 2015
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--089e011771f39f5c290511ad0b63
Content-Type: text/plain; charset=UTF-8

On Thu, Mar 19, 2015 at 12:28 PM, Stephen Farrell <stephen.farrell@cs.tcd.ie
> wrote:

>
>
> On 19/03/15 17:49, Ted Hardie wrote:
> > We have added this topic to the working copy of the agenda, and it will
> be
> > in the first day.  We've allocated 15 minutes for it at the moment, but
> we
> > can probably accommodate more on day two if need be.
>
> Good to see this discussed. I think I have a conflict or I'd be
> hoping to be there. If I'm not, I guess I'd be delighted if that
> discussion also includes revocation of permissions.
>
> Some folks might consider that the way to handle revocation of
> that is to e.g. revoke all camera/mic permissions for all origins
> (as is done with e.g. "private data" on some browsers). I think
> that approach is very privacy unfriendly and hope that it's not
> considered acceptable for WebRTC. The effect of the privacy
> unfriendly approach is that a user cannot revoke permissions for
> any origin so long as there is one origin for whom they do not
> want to revoke permissions. The net of that is that the user will
> not revoke permissions, so the feature at that point may as well
> not have been present.
>
> I think that is a real problem. I am not asserting that providing
> per-origin revocation interfaces is a perfect answer, but I think
> it is defensible as being a lot better.
>

FWIW I believe that both Chrome and Firefox allow per-origin revocation

-Ekr

Cheers,
> S.
>
>
> >
> > regards,
> >
> > Ted
> >
> > On Thu, Mar 19, 2015 at 2:54 AM, Harald Alvestrand <harald@alvestrand.no
> >
> > wrote:
> >
> >>  Speaking with my W3C WG chair hat on:
> >>
> >> An item has come up that relates to the IETF draft
> >> "draft-ietf-rtcweb-security-arch".
> >>
> >> The issue is this requirement:
> >>
> >>    Implementations MUST obtain explicit user consent prior to providing
> >>    access to the camera and/or microphone.  Implementations MUST at
> >>    minimum support the following two permissions models for HTTPS
> >>    origins.
> >>
> >>    o  Requests for one-time camera/microphone access.
> >>    o  Requests for permanent access.
> >>
> >> .....
> >>
> >>    API Requirement:  The API MUST provide a mechanism for the requesting
> >>       JS to indicate which of these forms of permissions it is
> >>       requesting.  This allows the browser client to know what sort of
> >>       user interface experience to provide to the user, including what
> >>       permissions to request from the user and hence what to enforce
> >>       later.  For instance, browsers might display a non-invasive door
> >>       hanger ("some features of this site may not work..." when asking
> >>       for long-term permissions) but a more invasive UI ("here is your
> >>       own video") for single-call permissions.  The API MAY grant weaker
> >>       permissions than the JS asked for if the user chooses to authorize
> >>       only those permissions, but if it intends to grant stronger ones
> >>       it SHOULD display the appropriate UI for those permissions and
> >>       MUST clearly indicate what permissions are being requested.
> >>
> >>
> >> At the moment, the W3C Media Capture TF doesn't have a mechanism for the
> >> JS to indicate whether it desires one-time access or permanent access.
> >> There hasn't been a request for this from app developers, and none of
> the
> >> implementations so far have added such a feature.
> >>
> >> Having a requirement specification and an API specification that don't
> >> match is obviously not a Good Thing. There are two solutions that
> suggest
> >> themselves:
> >>
> >> - Delete the requirement
> >> - Add a mechanism
> >>
> >> Discussion on the public-media-capture list did not provide a consensus
> >> that one of these alternatives is clearly superior to the other, and the
> >> suggestion was made that face-to-face discussion would be a reasonable
> way
> >> to move forward.
> >>
> >> Since the security-arch document is an IETF document, and the IETF
> meeting
> >> is next week, it seems reasonable to ask for some agenda time next week
> for
> >> this issue.
> >>
> >> Can this be accomodated?
> >>
> >> Harald
> >>
> >>
> >> _______________________________________________
> >> rtcweb mailing list
> >> rtcweb@ietf.org
> >> https://www.ietf.org/mailman/listinfo/rtcweb
> >>
> >>
> >
> >
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--089e011771f39f5c290511ad0b63
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Thu, Mar 19, 2015 at 12:28 PM, Stephen Farrell <span dir=3D"ltr">&lt=
;<a href=3D"mailto:stephen.farrell@cs.tcd.ie" target=3D"_blank">stephen.far=
rell@cs.tcd.ie</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" s=
tyle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><spa=
n class=3D""><br>
<br>
On 19/03/15 17:49, Ted Hardie wrote:<br>
&gt; We have added this topic to the working copy of the agenda, and it wil=
l be<br>
&gt; in the first day.=C2=A0 We&#39;ve allocated 15 minutes for it at the m=
oment, but we<br>
&gt; can probably accommodate more on day two if need be.<br>
<br>
</span>Good to see this discussed. I think I have a conflict or I&#39;d be<=
br>
hoping to be there. If I&#39;m not, I guess I&#39;d be delighted if that<br=
>
discussion also includes revocation of permissions.<br>
<br>
Some folks might consider that the way to handle revocation of<br>
that is to e.g. revoke all camera/mic permissions for all origins<br>
(as is done with e.g. &quot;private data&quot; on some browsers). I think<b=
r>
that approach is very privacy unfriendly and hope that it&#39;s not<br>
considered acceptable for WebRTC. The effect of the privacy<br>
unfriendly approach is that a user cannot revoke permissions for<br>
any origin so long as there is one origin for whom they do not<br>
want to revoke permissions. The net of that is that the user will<br>
not revoke permissions, so the feature at that point may as well<br>
not have been present.<br>
<br>
I think that is a real problem. I am not asserting that providing<br>
per-origin revocation interfaces is a perfect answer, but I think<br>
it is defensible as being a lot better.<br></blockquote><div><br></div><div=
>FWIW I believe that both Chrome and Firefox allow per-origin revocation</d=
iv><div><br></div><div>-Ekr</div><div><br></div><blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex">
Cheers,<br>
<div class=3D"HOEnZb"><div class=3D"h5">S.<br>
<br>
<br>
&gt;<br>
&gt; regards,<br>
&gt;<br>
&gt; Ted<br>
&gt;<br>
&gt; On Thu, Mar 19, 2015 at 2:54 AM, Harald Alvestrand &lt;<a href=3D"mail=
to:harald@alvestrand.no">harald@alvestrand.no</a>&gt;<br>
&gt; wrote:<br>
&gt;<br>
&gt;&gt;=C2=A0 Speaking with my W3C WG chair hat on:<br>
&gt;&gt;<br>
&gt;&gt; An item has come up that relates to the IETF draft<br>
&gt;&gt; &quot;draft-ietf-rtcweb-security-arch&quot;.<br>
&gt;&gt;<br>
&gt;&gt; The issue is this requirement:<br>
&gt;&gt;<br>
&gt;&gt;=C2=A0 =C2=A0 Implementations MUST obtain explicit user consent pri=
or to providing<br>
&gt;&gt;=C2=A0 =C2=A0 access to the camera and/or microphone.=C2=A0 Impleme=
ntations MUST at<br>
&gt;&gt;=C2=A0 =C2=A0 minimum support the following two permissions models =
for HTTPS<br>
&gt;&gt;=C2=A0 =C2=A0 origins.<br>
&gt;&gt;<br>
&gt;&gt;=C2=A0 =C2=A0 o=C2=A0 Requests for one-time camera/microphone acces=
s.<br>
&gt;&gt;=C2=A0 =C2=A0 o=C2=A0 Requests for permanent access.<br>
&gt;&gt;<br>
&gt;&gt; .....<br>
&gt;&gt;<br>
&gt;&gt;=C2=A0 =C2=A0 API Requirement:=C2=A0 The API MUST provide a mechani=
sm for the requesting<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0JS to indicate which of these forms of p=
ermissions it is<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0requesting.=C2=A0 This allows the browse=
r client to know what sort of<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0user interface experience to provide to =
the user, including what<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0permissions to request from the user and=
 hence what to enforce<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0later.=C2=A0 For instance, browsers migh=
t display a non-invasive door<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0hanger (&quot;some features of this site=
 may not work...&quot; when asking<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0for long-term permissions) but a more in=
vasive UI (&quot;here is your<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0own video&quot;) for single-call permiss=
ions.=C2=A0 The API MAY grant weaker<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0permissions than the JS asked for if the=
 user chooses to authorize<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0only those permissions, but if it intend=
s to grant stronger ones<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0it SHOULD display the appropriate UI for=
 those permissions and<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0MUST clearly indicate what permissions a=
re being requested.<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; At the moment, the W3C Media Capture TF doesn&#39;t have a mechani=
sm for the<br>
&gt;&gt; JS to indicate whether it desires one-time access or permanent acc=
ess.<br>
&gt;&gt; There hasn&#39;t been a request for this from app developers, and =
none of the<br>
&gt;&gt; implementations so far have added such a feature.<br>
&gt;&gt;<br>
&gt;&gt; Having a requirement specification and an API specification that d=
on&#39;t<br>
&gt;&gt; match is obviously not a Good Thing. There are two solutions that =
suggest<br>
&gt;&gt; themselves:<br>
&gt;&gt;<br>
&gt;&gt; - Delete the requirement<br>
&gt;&gt; - Add a mechanism<br>
&gt;&gt;<br>
&gt;&gt; Discussion on the public-media-capture list did not provide a cons=
ensus<br>
&gt;&gt; that one of these alternatives is clearly superior to the other, a=
nd the<br>
&gt;&gt; suggestion was made that face-to-face discussion would be a reason=
able way<br>
&gt;&gt; to move forward.<br>
&gt;&gt;<br>
&gt;&gt; Since the security-arch document is an IETF document, and the IETF=
 meeting<br>
&gt;&gt; is next week, it seems reasonable to ask for some agenda time next=
 week for<br>
&gt;&gt; this issue.<br>
&gt;&gt;<br>
&gt;&gt; Can this be accomodated?<br>
&gt;&gt;<br>
&gt;&gt; Harald<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; _______________________________________________<br>
&gt;&gt; rtcweb mailing list<br>
&gt;&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D=
"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div></div>

--089e011771f39f5c290511ad0b63--


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On Mar 19, 2015 5:04 PM, "Eric Rescorla" <ekr@rtfm.com> wrote:
> FWIW I believe that both Chrome and Firefox allow per-origin revocation

...Both without actually visiting that origin. You can thank Alissa Cooper
for that.

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<p dir="ltr"><br>
On Mar 19, 2015 5:04 PM, &quot;Eric Rescorla&quot; &lt;<a href="mailto:ekr@rtfm.com">ekr@rtfm.com</a>&gt; wrote:<br>
&gt; FWIW I believe that both Chrome and Firefox allow per-origin revocation</p>
<p dir="ltr">...Both without actually visiting that origin. You can thank Alissa Cooper for that.<br>
</p>

--e89a8f838e098357d10511af145b--


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Subject: [rtcweb] Homework for Dallas: SetLocal/SetRemote
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Folks,

The new JSEP draft contains new extensive text on processing
Local and Remote descriptions when set, starting at:

http://rtcweb-wg.github.io/jsep/#rfc.section.5.4

It would be really good if people could read these sections before
Dallas. The editors intend to proceed in the direction indicated here
unless people express unhappiness soon.

-Ekr

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<div dir=3D"ltr">Folks,<div><br></div><div>The new JSEP draft contains new =
extensive text on processing</div><div>Local and Remote descriptions when s=
et, starting at:</div><div><br></div><div><a href=3D"http://rtcweb-wg.githu=
b.io/jsep/#rfc.section.5.4">http://rtcweb-wg.github.io/jsep/#rfc.section.5.=
4</a><br></div><div><br></div><div>It would be really good if people could =
read these sections before</div><div>Dallas. The editors intend to proceed =
in the direction indicated here</div><div>unless people express unhappiness=
 soon.</div><div><br></div><div>-Ekr</div><div><br></div></div>

--047d7bf10b1c30578b0511baa525--


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One quick thing: per RFC 5245 (
http://tools.ietf.org/html/rfc5245#section-15.3) "ice-lite" is a session
level attribute only.

Why is it a media level attribute in section 5.6.2 (
http://rtcweb-wg.github.io/jsep/#rfc.section.5.6.2)? Have this been
redefined somewhere?

On the separate note, I do think ice-lite should be a media level
attribute, and can provide use cases for that, if necessary, but this is
against the current ICE specification and as far as I know was not changed.

_____________
Roman Shpount

On Fri, Mar 20, 2015 at 12:16 PM, Eric Rescorla <ekr@rtfm.com> wrote:

> Folks,
>
> The new JSEP draft contains new extensive text on processing
> Local and Remote descriptions when set, starting at:
>
> http://rtcweb-wg.github.io/jsep/#rfc.section.5.4
>
> It would be really good if people could read these sections before
> Dallas. The editors intend to proceed in the direction indicated here
> unless people express unhappiness soon.
>
> -Ekr
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr">One quick thing: per RFC 5245 (<a href=3D"http://tools.iet=
f.org/html/rfc5245#section-15.3">http://tools.ietf.org/html/rfc5245#section=
-15.3</a>) &quot;ice-lite&quot; is a session level attribute only.<div><br>=
</div><div>Why is it a media level attribute in section 5.6.2 (<a href=3D"h=
ttp://rtcweb-wg.github.io/jsep/#rfc.section.5.6.2">http://rtcweb-wg.github.=
io/jsep/#rfc.section.5.6.2</a>)? Have this been redefined somewhere?</div><=
div><br></div><div>On the separate note, I do think ice-lite should be a me=
dia level attribute, and can provide use cases for that, if necessary, but =
this is against the current ICE specification and as far as I know was not =
changed.</div></div><div class=3D"gmail_extra"><br clear=3D"all"><div><div =
class=3D"gmail_signature">_____________<br>Roman Shpount</div></div>
<br><div class=3D"gmail_quote">On Fri, Mar 20, 2015 at 12:16 PM, Eric Resco=
rla <span dir=3D"ltr">&lt;<a href=3D"mailto:ekr@rtfm.com" target=3D"_blank"=
>ekr@rtfm.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" st=
yle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div =
dir=3D"ltr">Folks,<div><br></div><div>The new JSEP draft contains new exten=
sive text on processing</div><div>Local and Remote descriptions when set, s=
tarting at:</div><div><br></div><div><a href=3D"http://rtcweb-wg.github.io/=
jsep/#rfc.section.5.4" target=3D"_blank">http://rtcweb-wg.github.io/jsep/#r=
fc.section.5.4</a><br></div><div><br></div><div>It would be really good if =
people could read these sections before</div><div>Dallas. The editors inten=
d to proceed in the direction indicated here</div><div>unless people expres=
s unhappiness soon.</div><div><br></div><div>-Ekr</div><div><br></div></div=
>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--90e6ba1ef2ea06173d0511bb0ba6--


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From: Bernard Aboba <bernard.aboba@gmail.com>
Date: Fri, 20 Mar 2015 17:02:14 -0700
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Looking through the document, I still see a TODO item in Appendix A.

Also, in reviewing discussion on the document, I noticed some  issues
brought up that don't seem to have been addressed in this or other docs,
such as the recommended hash algorithms for certificate fingerprints:
http://www.ietf.org/mail-archive/web/rtcweb/current/msg12660.html



On Thu, Mar 19, 2015 at 12:36 PM, Sean Turner <turners@ieca.com> wrote:

> spt has requested publication of draft-ietf-rtcweb-security-arch-11 as
> Proposed Standard on behalf of the RTCWEB working group.
>
> Please verify the document's state at
> http://datatracker.ietf.org/doc/draft-ietf-rtcweb-security-arch/
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">Looking through the document, I still see a TODO item in A=
ppendix A. =C2=A0<div><br></div><div>Also, in reviewing discussion on the d=
ocument, I noticed some =C2=A0issues brought up that don&#39;t seem to have=
 been addressed in this or other docs, such as the recommended hash algorit=
hms for certificate fingerprints:=C2=A0<div><a href=3D"http://www.ietf.org/=
mail-archive/web/rtcweb/current/msg12660.html">http://www.ietf.org/mail-arc=
hive/web/rtcweb/current/msg12660.html</a></div><div><br></div><div><br></di=
v></div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On =
Thu, Mar 19, 2015 at 12:36 PM, Sean Turner <span dir=3D"ltr">&lt;<a href=3D=
"mailto:turners@ieca.com" target=3D"_blank">turners@ieca.com</a>&gt;</span>=
 wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bor=
der-left:1px #ccc solid;padding-left:1ex">spt has requested publication of =
draft-ietf-rtcweb-security-arch-11 as Proposed Standard on behalf of the RT=
CWEB working group.<br>
<br>
Please verify the document&#39;s state at <a href=3D"http://datatracker.iet=
f.org/doc/draft-ietf-rtcweb-security-arch/" target=3D"_blank">http://datatr=
acker.ietf.org/doc/draft-ietf-rtcweb-security-arch/</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div>

--047d7b86d8b2677c5c0511c12615--


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The draft asks=20

 (Question: Does there need
  to be an API knob to turn off DSCP markings?)

I think that effectively there is - if the API requests them all as =
"normal",  (aka low in latest qos draft), they all get set to DF which =
is the same as what would happen if this were "turned off"



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From: Cullen Jennings <fluffy@iii.ca>
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Subject: [rtcweb] Irrelevant comments on draft-ietf-rtcweb-audio
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Two small things ...

The draft has=20

   the clients SHOULD ensure that
   the speaker-to-microphone gain is below unity at all frequencies to
   avoid instability

I have no idea how the average browser could implement that. If it's not =
implementable, we might want to remove it.=20

I think I know what you are trying to get at with=20

   For
   clients that do not control the audio capture and playback hardware,
   it is RECOMMENDED to support echo cancellation between devices
   running at slightly different sampling rates, such as when a webcam
   is used for microphone.

but I'm not really sure what that paragraph means. It might need a bit =
of clarification. They can't run a different rates for a long time with =
some Q overflowing or underflowing so somewhere the rates are =
normalized. And I'm not sure that client control or not changes this - =
even if the client controls them, they can still be based off different =
clock sources.=20

But that said, if there is no desire to change these I don't care =
because I don't think changing the two above points will impact =
implementations in any real way.=20




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> On Mar 18, 2015, at 5:12 AM, Magnus Westerlund =
<magnus.westerlund@ericsson.com> wrote:
>=20
>=20
> Two comments, do we actually want to mandate the implementation =
support
> of a=3Dimageattr?
>=20
> I think the usage rules are fairly reasonable. I do wonder if we want =
to
> be explicit about the SHOULD exception is really for cases where the
> encoder can't be changed, for example due to lacking APIs to affect a
> change on a webcam etc.

We have to have some way for the application to limit the size of the =
sent video. This can be different for each codec but I would hope it =
would imageattr for new codecs going forward. I think we do have to =
mandate the browser pick a resolution that fits inside the envelope =
provides and I think that is can be done on even the low end mobile =
devices adam mentioned. Without this there is no way to know things will =
work.  If something sends 4k video to low end device, it probably can't =
decode it. If a device is transcoding between different video codecs, it =
may also need to constrain the resolution to fit inside a given box. =
It's not possible to build systems that are guarantee to work without a =
way to tell the browser a limit on the video size and be sure the sender =
will not exceed that limit.=20

So I think the SHOULD needs to be a MUST.=20





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> On Mar 17, 2015, at 6:54 AM, Magnus Westerlund =
<magnus.westerlund@ericsson.com> wrote:
>=20
> Hi,
>=20
> I had a quick review of draft-ietf-rtcweb-constraints-registry-02. So
> some few comments.
>=20
> 1. Shouldn't the name of the registry be "WebRTC Constrainable
> Properties"? My understanding was that we are using WebRTC rather than
> RTCWeb as the name for the overarching thing.

I agree

>=20
> 2. Is there a good reason why not a outer limit on the property name
> can't be set, even if it is large. Would not for example 256 =
characters
> be sufficient?

not sure it will matter much but seems like a fine change. These names =
only really occur in JS code and specs - they won't show in in SDP. =20

>=20
> 3. I find no requirements on the reference. Should there be guidance =
on
> this to the expert reviewer. For example are references that are =
secret
> and can't even be given to the expert reviewer allowed? Are references
> under a pay-wall okay, as long as the reference is provided the expert
> reviewer? I think the expectation should be made clear. I think their
> are good points to allow almost anyone to register, however we might
> want to make clear the expectancy that the expert reviewer is provided
> with one copy, possibly under NDA if really required.

Excellent point but seems like something best handled in =
draft-leiba-cotton-iana-5226bis to update the "Specification Required" =
text. Given part of the job of the expert is to catch duplicates,  I =
think that things have to be available to expert, and all future =
experts, and all people that want to review the experts decision =
including any IETF participant but this is seems like a very =
controversial topic that should be solved IETF wide and not juste here.=20=


>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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Hash: SHA1

Hi Cullen,

Thanks for the feedback. See below.

On 22/03/15 10:38 AM, Cullen Jennings wrote:
> The draft has
> 
> the clients SHOULD ensure that the speaker-to-microphone gain is 
> below unity at all frequencies to avoid instability
> 
> I have no idea how the average browser could implement that. If
> it's not implementable, we might want to remove it.

I guess this would have been best phrased using the 6919 SHOULD
CONSIDER, but I agree that it's probably not too practical (you can do
it, but it'd be about the same complexity as a simple echo canceller).

> I think I know what you are trying to get at with
> 
> For clients that do not control the audio capture and playback 
> hardware, it is RECOMMENDED to support echo cancellation between 
> devices running at slightly different sampling rates, such as when
> a webcam is used for microphone.
> 
> but I'm not really sure what that paragraph means. It might need a 
> bit of clarification. They can't run a different rates for a long 
> time with some Q overflowing or underflowing so somewhere the
> rates are normalized. And I'm not sure that client control or not
> changes this - even if the client controls them, they can still be
> based off different clock sources.

What I'm trying to say here is that unless you actually designed the
capture/playback hardware (e.g. when you design a phone yourself),
then you may be stuck with the capture and playback running on
different clocks. That's a very common situation on PCs and while the
jitter buffer will usually cope relatively easily (unless the clock
drift is very large), a basic echo canceller will not. So the
recommendation is that if you're in this situation (and every browser
is), you should use an echo canceller that's able to cope with clock
drift.

> But that said, if there is no desire to change these I don't care 
> because I don't think changing the two above points will impact 
> implementations in any real way.

Well, I have no objection to removing the first one. About the second
one, I don't mind whether it's SHOULD, MAY, SHOULD CONSIDER or OUGHT
TO, but I'd like to keep some text that acts as "best practice to
avoid bad echo problems".

Cheers,

	Jean-Marc
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On 2015-03-22 08:45, Cullen Jennings wrote:
> The draft asks 
> 
> (Question: Does there need to be an API knob to turn off DSCP
> markings?)
> 
> I think that effectively there is - if the API requests them all as
"normal", (aka low in latest qos draft), they all get set to DF which is
the same as what would happen if this were "turned off"

I interpreted the question here as being able to not set the DSCP bits,
but still have local endpoint priority scheduling.

See the discussion in my comments on the transport documents.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
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----------------------------------------------------------------------
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From nobody Sun Mar 22 16:47:21 2015
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Hello all,

Interactive Connectivity Establishment (ICE) is a standard that allows 
endpoints to establish connectivity with each other across the Internet, 
even in the presence of NATs and firewalls, and has become prevalent 
with the deployment of WebRTC. However, its inner workings are not 
always well understood.

We could have come up with a simpler, easier-to-understand protocol, but 
we ran out of time, so Justin, Pal, Brandon and I have decided to simply 
explain how the existing protocol works.

Come join us at 11:00 in State where we attempt to shed light on this 
fundamental, yet mysterious protocol.

See you then!

Pal, Brandon, Justin and Emil


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More on timing:

The tutorial is taking place at 11:00am tomorrow morning (Monday) in State.

Emil

On 22.03.15 18:47, Emil Ivov wrote:
> Hello all,
>
> Interactive Connectivity Establishment (ICE) is a standard that allows
> endpoints to establish connectivity with each other across the Internet,
> even in the presence of NATs and firewalls, and has become prevalent
> with the deployment of WebRTC. However, its inner workings are not
> always well understood.
>
> We could have come up with a simpler, easier-to-understand protocol, but
> we ran out of time, so Justin, Pal, Brandon and I have decided to simply
> explain how the existing protocol works.
>
> Come join us at 11:00 in State where we attempt to shed light on this
> fundamental, yet mysterious protocol.
>
> See you then!
>
> Pal, Brandon, Justin and Emil
>

-- 
https://jitsi.org


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Hi,

I have reviewed this draft and have the following comments:

1. Section 3:

WebRTC clients are REQUIRED to implement the following audio codecs:

I think this should use the Overview definitions. Thus, I think this
applies to an "WebRTC endpoint".



I was tempted to bring up an issue I was in the rough a year ago, but I
guess I will let the issue be.


Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
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Ericsson AB                 | Phone  +46 10 7148287
FÃ¤rÃ¶gatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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Hello
> 3. I find no requirements on the reference. Should there be guidance on
> this to the expert reviewer. For example are references that are secret
> and can't even be given to the expert reviewer allowed? Are references
> under a pay-wall okay, as long as the reference is provided the expert
> reviewer? I think the expectation should be made clear. I think their
> are good points to allow almost anyone to register, however we might
> want to make clear the expectancy that the expert reviewer is provided
> with one copy, possibly under NDA if really required.
> Excellent point but seems like something best handled in draft-leiba-cotton-iana-5226bis to update the "Specification Required" text. Given part of the job of the expert is to catch duplicates,  I think that things have to be available to expert, and all future experts, and all people that want to review the experts decision including any IETF participant but this is seems like a very controversial topic that should be solved IETF wide and not juste here.
[CNG] Another approach would be to define a namespace for private 
registrations where there's less scrutiny and keep the job of the expert 
to review publicly available documents. Its the approach that was taken 
for megaco [RFC5615] and I think has worked fairly well.

Christian


>


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Inline:

On Wed, Mar 18, 2015 at 3:13 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> Hi,
>
> I have reviewed draft-ietf-rtcweb-fec-01 and have some comments.
>
> 1. Title:
>
> My impression of the document from the title is that this is only
> requirements and no normative specification of FEC for WebRTC. Thus, I
> would suggest changing the title. I think one can simply remove
> "requirements" by changing the order, i.e. "Forward Error Correction in
> WebRTC".
>

OK.

>
> 2. Section 3.1:
>    This approach, as described in [RFC5956], Section 4.3, sends FEC
>    packets as an independent SSRC-multiplexed stream, with its own SSRC
>    and payload type.  While by far the most flexible, each FEC packet
>    will have its own IP+UDP+RTP+FEC header, leading to additional
>    overhead of the FEC stream.
>
> I think this section must have more description of what "Separate FEC
> stream" really is. The "FEC Grouping for SSRC-Multiplexed RTP Streams"
> reference also appear to restrictive. Based on the transport documents
> there actually exists an requirement for supporting fully unbundle
> configuration. That is not really matching what Section 4.3 describes.
>

Even when BUNDLE is not used, unified plan still indicates that the FEC
stream will be sent as a SSRC-muxed stream. I think the text in 3.1 is
accurate, although this could be spelled out better.

>
> >From my perspective there are two sets of orthogonal questions around
> FEC streams. First the relation between the source flow(s) and the
> repair flows, which may be 1 to 1, 1 to M, or N to 1, or N to M. Then we
> have the other question of how they are transmitted in relation to the
> source flow. Within the same RTP session, on different RTP session,
> outside of the RTP session. And if one has many source flows, then one
> can also discuss if there is N source RTP sessions and M repair sessions
> or 1 source RTP session and 1 repair session.
>

The salient issue is that there is no current way with unified to have a
FEC stream as its own m= line. This will require some thinking to resolve.


>
> 3. Section 3.4:
>
> I think you want to discuss FEC information as a "redundant" encoding also.
>

There is no section 3.4 - are you suggesting clarifying 3.2's use of
"Redundant Encoding" to refer to 2198?

>
>
> 4. Section 4.1:
>    When using constant-bitrate codecs, e.g.  PCMU, use of [RFC2198]
>    redundant encoding MAY be used, but note that this will result in a
>    potentially significant bitrate increase, and that suddenly
>    increasing bitrate to deal with losses from congestion may actually
>    make things worse.
>
>    Because of the lower packet rate of audio encodings, usually a single
>    packet per frame, use of a separate FEC stream comes with a higher
>    overhead than other mechanisms, and therefore is NOT RECOMMENDED.
>
> I think this fails to discuss the quite reasonable application of XOR
> FEC as a redundant encoding information. That way one can achieve a
> bitrate increase that is ~1/N of the source flow bit-rate by including
> one XOR packet every N packets covering those N packets. Sure that only
> gives single packet loss protection, but might be appropriate anyway.
>

I'll bring this up on Wednesday. I didn't get the sense from Honolulu that
there was support for doing any nontrivial FEC for audio.


>
> 5. Section 4.2:
>
> I think this more clearer needs to reference and possibly discuss the
> signalling context it is operating in, i.e. JSEP.
>

OK.


>
> 6. Section 5.1:
>    Note that this only allows the FEC stream to protect a single primary
>    stream.  Support for protecting multiple primary streams with a
>    single FEC stream is complicated by WebRTC's 1-m-line-per-stream
>    policy and requires further study.
>
> For efficient application of FEC this is the way one have to go.
> Unfortunately that also requires FEC codes that are more complex than
> XOR and also more likely to be IPR encumbered. But if ones goal is to
> have low delay, be almost independent of loss pattern and have
> protection for multi packet loss, then aggregating together all the data
> sent in a relatively short intervals (<50 ms) and protect them together
> is the way to go.
>
> Unfortunately I don't think we have all the pieces in place here in IETF
> to perform such a protection, especially in the context of BUNDLE as
> well as the possibility to fully unbundle them. It mostly an exercise to
> put the pieces and the necessary identification required for aggregating
> flows into a common source block and signal that, and make it clear
> where the protection data is sent that is missing.
>
> I was part of doing this in 3GPP Multimedia Broadcast/Multicast Service
> (MBMS), but that system performs the FEC operation on UDP Flows as a way
> to handle both RTP/RTCP as well as MIKEY messages with the keys to
> media. It is also a streaming service with more relaxed timing
> requirements.
>
> If you are interested to see how that is specified please see Section
> 8.2.2 of TS 26.346:
> http://www.3gpp.org/ftp/Specs/archive/26_series/26.346/26346-c40.zip
>
> I personally think this is the direction to go to achieve efficient FEC
> usage for the real-time media, especially when you have multi-stream
> situations. However, I can't contribute the hours needed to make rapid
> enough progress on this. Thus, I guess this needs to remain a potential
> second phase of the FEC work for WebRTC.
>

Agreed.

>
> 7. Section 6.
>
> I think this section must discuss the fact that the data channel have
> several possibilities for dealing with reliability. First one can use a
> reliable stream, that will do retransmissions. One can use semi-reliable
> stream to not needless continue to try, or the application can select to
> repeat the data itself over an unreliable stream. I think the last
> should be recommended against as it will waste resources unnecessarily.
>

If the application wants to send redundant data as an application-level FEC
mechanism, that seems fine to me. The SCTP CC will prevent the application
from harming others.

>
> 8. Section 7:
>
> What about using RFC 2198 and [I-D.ietf-payload-flexible-fec-scheme]? Or
> isn't that supported at all by [I-D.ietf-payload-flexible-fec-scheme]?
> (Yes, I will review that spec, but haven't gotten to it yet).
>

Do you mean using these two in conjunction? I am not sure there is any
advantage to doing so, but open to discussing this.

>
> 9. Section 9.
>
> I think there is a point of discussing how FEC configuration can cause
> significant waste of bandwidth. Making even a encoded stream with modest
> bandwidth needs into a bandwidth consuming monster simply by influencing
> the usage of FEC and its parameters. Thus, I think this document needs
> to discuss the issues with possible interference from malicious
> applications with the signalling and the need for a WebRTC
> implementation to actually control the application and usage of FEC
> itself and not rely on the signalling parameters.
>

True, but it will still be capped by the congestion control, so I don't
think there is a real-world attack here.

>
>

--001a1140ca3862152f05120201a0
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Inline:<br><div class=3D"gmail_extra"><br><div class=3D"gm=
ail_quote">On Wed, Mar 18, 2015 at 3:13 AM, Magnus Westerlund <span dir=3D"=
ltr">&lt;<a href=3D"mailto:magnus.westerlund@ericsson.com" target=3D"_blank=
">magnus.westerlund@ericsson.com</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex">Hi,<br>
<br>
I have reviewed draft-ietf-rtcweb-fec-01 and have some comments.<br>
<br>
1. Title:<br>
<br>
My impression of the document from the title is that this is only<br>
requirements and no normative specification of FEC for WebRTC. Thus, I<br>
would suggest changing the title. I think one can simply remove<br>
&quot;requirements&quot; by changing the order, i.e. &quot;Forward Error Co=
rrection in<br>
WebRTC&quot;.<br></blockquote><div><br></div><div>OK.=C2=A0</div><blockquot=
e class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc sol=
id;padding-left:1ex">
<br>
2. Section 3.1:<br>
=C2=A0 =C2=A0This approach, as described in [RFC5956], Section 4.3, sends F=
EC<br>
=C2=A0 =C2=A0packets as an independent SSRC-multiplexed stream, with its ow=
n SSRC<br>
=C2=A0 =C2=A0and payload type.=C2=A0 While by far the most flexible, each F=
EC packet<br>
=C2=A0 =C2=A0will have its own IP+UDP+RTP+FEC header, leading to additional=
<br>
=C2=A0 =C2=A0overhead of the FEC stream.<br>
<br>
I think this section must have more description of what &quot;Separate FEC<=
br>
stream&quot; really is. The &quot;FEC Grouping for SSRC-Multiplexed RTP Str=
eams&quot;<br>
reference also appear to restrictive. Based on the transport documents<br>
there actually exists an requirement for supporting fully unbundle<br>
configuration. That is not really matching what Section 4.3 describes.<br><=
/blockquote><div><br></div><div>Even when BUNDLE is not used, unified plan =
still indicates that the FEC stream will be sent as a SSRC-muxed stream. I =
think the text in 3.1 is accurate, although this could be spelled out bette=
r.</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border=
-left:1px #ccc solid;padding-left:1ex">
<br>
&gt;From my perspective there are two sets of orthogonal questions around<b=
r>
FEC streams. First the relation between the source flow(s) and the<br>
repair flows, which may be 1 to 1, 1 to M, or N to 1, or N to M. Then we<br=
>
have the other question of how they are transmitted in relation to the<br>
source flow. Within the same RTP session, on different RTP session,<br>
outside of the RTP session. And if one has many source flows, then one<br>
can also discuss if there is N source RTP sessions and M repair sessions<br=
>
or 1 source RTP session and 1 repair session.<br></blockquote><div><br></di=
v><div>The salient issue is that there is no current way with unified to ha=
ve a FEC stream as its own m=3D line. This will require some thinking to re=
solve.</div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
3. Section 3.4:<br>
<br>
I think you want to discuss FEC information as a &quot;redundant&quot; enco=
ding also.<br></blockquote><div><br></div><div>There is no section 3.4 - ar=
e you suggesting clarifying 3.2&#39;s use of &quot;Redundant Encoding&quot;=
 to refer to 2198?=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"ma=
rgin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
<br>
4. Section 4.1:<br>
=C2=A0 =C2=A0When using constant-bitrate codecs, e.g.=C2=A0 PCMU, use of [R=
FC2198]<br>
=C2=A0 =C2=A0redundant encoding MAY be used, but note that this will result=
 in a<br>
=C2=A0 =C2=A0potentially significant bitrate increase, and that suddenly<br=
>
=C2=A0 =C2=A0increasing bitrate to deal with losses from congestion may act=
ually<br>
=C2=A0 =C2=A0make things worse.<br>
<br>
=C2=A0 =C2=A0Because of the lower packet rate of audio encodings, usually a=
 single<br>
=C2=A0 =C2=A0packet per frame, use of a separate FEC stream comes with a hi=
gher<br>
=C2=A0 =C2=A0overhead than other mechanisms, and therefore is NOT RECOMMEND=
ED.<br>
<br>
I think this fails to discuss the quite reasonable application of XOR<br>
FEC as a redundant encoding information. That way one can achieve a<br>
bitrate increase that is ~1/N of the source flow bit-rate by including<br>
one XOR packet every N packets covering those N packets. Sure that only<br>
gives single packet loss protection, but might be appropriate anyway.<br></=
blockquote><div><br></div><div>I&#39;ll bring this up on Wednesday. I didn&=
#39;t get the sense from Honolulu that there was support for doing any nont=
rivial FEC for audio.</div><div>=C2=A0<br></div><blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex">
<br>
5. Section 4.2:<br>
<br>
I think this more clearer needs to reference and possibly discuss the<br>
signalling context it is operating in, i.e. JSEP.<br></blockquote><div><br>=
</div><div>OK.</div><div>=C2=A0</div><blockquote class=3D"gmail_quote" styl=
e=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
6. Section 5.1:<br>
=C2=A0 =C2=A0Note that this only allows the FEC stream to protect a single =
primary<br>
=C2=A0 =C2=A0stream.=C2=A0 Support for protecting multiple primary streams =
with a<br>
=C2=A0 =C2=A0single FEC stream is complicated by WebRTC&#39;s 1-m-line-per-=
stream<br>
=C2=A0 =C2=A0policy and requires further study.<br>
<br>
For efficient application of FEC this is the way one have to go.<br>
Unfortunately that also requires FEC codes that are more complex than<br>
XOR and also more likely to be IPR encumbered. But if ones goal is to<br>
have low delay, be almost independent of loss pattern and have<br>
protection for multi packet loss, then aggregating together all the data<br=
>
sent in a relatively short intervals (&lt;50 ms) and protect them together<=
br>
is the way to go.<br>
<br>
Unfortunately I don&#39;t think we have all the pieces in place here in IET=
F<br>
to perform such a protection, especially in the context of BUNDLE as<br>
well as the possibility to fully unbundle them. It mostly an exercise to<br=
>
put the pieces and the necessary identification required for aggregating<br=
>
flows into a common source block and signal that, and make it clear<br>
where the protection data is sent that is missing.<br>
<br>
I was part of doing this in 3GPP Multimedia Broadcast/Multicast Service<br>
(MBMS), but that system performs the FEC operation on UDP Flows as a way<br=
>
to handle both RTP/RTCP as well as MIKEY messages with the keys to<br>
media. It is also a streaming service with more relaxed timing<br>
requirements.<br>
<br>
If you are interested to see how that is specified please see Section<br>
8.2.2 of TS 26.346:<br>
<a href=3D"http://www.3gpp.org/ftp/Specs/archive/26_series/26.346/26346-c40=
.zip" target=3D"_blank">http://www.3gpp.org/ftp/Specs/archive/26_series/26.=
346/26346-c40.zip</a><br>
<br>
I personally think this is the direction to go to achieve efficient FEC<br>
usage for the real-time media, especially when you have multi-stream<br>
situations. However, I can&#39;t contribute the hours needed to make rapid<=
br>
enough progress on this. Thus, I guess this needs to remain a potential<br>
second phase of the FEC work for WebRTC.<br></blockquote><div><br></div><di=
v>Agreed.=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 =
0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
7. Section 6.<br>
<br>
I think this section must discuss the fact that the data channel have<br>
several possibilities for dealing with reliability. First one can use a<br>
reliable stream, that will do retransmissions. One can use semi-reliable<br=
>
stream to not needless continue to try, or the application can select to<br=
>
repeat the data itself over an unreliable stream. I think the last<br>
should be recommended against as it will waste resources unnecessarily.<br>=
</blockquote><div><br></div><div>If the application wants to send redundant=
 data as an application-level FEC mechanism, that seems fine to me. The SCT=
P CC will prevent the application from harming others.=C2=A0</div><blockquo=
te class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc so=
lid;padding-left:1ex">
<br>
8. Section 7:<br>
<br>
What about using RFC 2198 and [I-D.ietf-payload-flexible-fec-scheme]? Or<br=
>
isn&#39;t that supported at all by [I-D.ietf-payload-flexible-fec-scheme]?<=
br>
(Yes, I will review that spec, but haven&#39;t gotten to it yet).<br></bloc=
kquote><div><br></div><div>Do you mean using these two in conjunction? I am=
 not sure there is any advantage to doing so, but open to discussing this.=
=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bo=
rder-left:1px #ccc solid;padding-left:1ex">
<br>
9. Section 9.<br>
<br>
I think there is a point of discussing how FEC configuration can cause<br>
significant waste of bandwidth. Making even a encoded stream with modest<br=
>
bandwidth needs into a bandwidth consuming monster simply by influencing<br=
>
the usage of FEC and its parameters. Thus, I think this document needs<br>
to discuss the issues with possible interference from malicious<br>
applications with the signalling and the need for a WebRTC<br>
implementation to actually control the application and usage of FEC<br>
itself and not rely on the signalling parameters.<br></blockquote><div><br>=
</div><div>True, but it will still be capped by the congestion control, so =
I don&#39;t think there is a real-world attack here. =C2=A0</div><blockquot=
e class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc sol=
id;padding-left:1ex"><br></blockquote></div></div></div>

--001a1140ca3862152f05120201a0--


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--bcaec51b1ea118da4c05120d3cb1
Content-Type: text/plain; charset=UTF-8

Few quick initial comments.

Section 3.2 -
[Suhas]
   a. The state machine captures successful scenarios alone. Would it be
worthwhile to capture error states ?

    b. The state themselves are not defined anywhere. The State names
across Section 3.2 and Section 5.4 are not uniform.


Section 5.4 says

   - Next, the SessionDescription is parsed into a data structure, as
   described in the Section 5.6
   <https://tools.ietf.org/id/draft-ietf-rtcweb-jsep-09.html#sec.parsing-a-desc>section
   below. If parsing fails for any reason, processing MUST stop and an error
   MUST be returned.

[Suhas]     Should we add a note to say that the remote offer MUST be
rejected by the JSEP application when the application of Answer fails.


Section 5.6

The "c=" line MUST be parsed and stored.
[Suhas]   Should we say anything out connection address being Unicast vs
Multicast.

For non-attribute (non-"a=") lines, their sequencing, syntax, and
semantics, are checked, as mentioned above. The following lines are not
meaningful in the JSEP context and MAY be discarded once they have been
checked.

   - TODO

[Suhas]. These can be added here u=, e=, p=, r=, z=, k=

[OPEN ISSUE: For example, because session-level bandwidth is ambiguous when
multiple media streams are present, a "b=" line at session level is not
useful and its value SHOULD be ignored. [OPEN ISSUE: is this WG consensus?
Are there other non-a= lines that we need to do more than just syntactical
validation, e.g. v=?]

[Suhas] For b= handling, I agree that having them specified at the media
level makes sense. The draft should however say if any of RFC4566 or
RFC3556 b= attributes are missing at the media level, what needs to be the
expected behavior


Section 5.6.2
The "b=" line, if present, MUST be parsed as specified in [RFC4566]
<https://tools.ietf.org/id/draft-ietf-rtcweb-jsep-09.html#RFC4566>, Section
5.8, and the bwtype and bandwidth values stored.
[Suhas]  We also need to add RFC3556 references here.
Also we need to add ice-options for media section.

Section 5.6.3
[Suhas]    "Semantics verification" is not defined and the intent behind
the text in this section is unclear.

Section 5.9
[Suhas]  Is underspecified (WIP)


Just a crazy thought .. Would it help in defining a registry of SDP
attributes for RTCWeb usage.. This helps today's and future SDP attributes
that gets defined for RTCWeb usage to go to a single place for reference.
Also makes it easier to add new attributes to the registry in the future.


Thanks
Suhas

--bcaec51b1ea118da4c05120d3cb1
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div><br></div><div>Few quick initial comments.=C2=A0</div=
><div>=C2=A0</div><div><font face=3D"verdana, sans-serif">Section 3.2 -</fo=
nt></div><div><span style=3D"color:rgb(0,0,0);font-family:verdana,sans-seri=
f;font-size:13.3333330154419px">[Suhas]</span><font face=3D"verdana, sans-s=
erif">=C2=A0</font></div><div><font face=3D"verdana, sans-serif">=C2=A0=C2=
=A0 a. The state machine captures successful scenarios alone. Would it be w=
orthwhile to capture error states ?</font></div><div><font face=3D"verdana,=
 sans-serif">=C2=A0</font></div><div><font face=3D"verdana, sans-serif">=C2=
=A0 =C2=A0 b. The state themselves are not defined anywhere. The State name=
s across Section 3.2 and Section 5.4 are not uniform.</font></div><div><fon=
t face=3D"verdana, sans-serif"><br></font></div><div><font face=3D"verdana,=
 sans-serif"><br></font></div><div><font face=3D"verdana, sans-serif">Secti=
on 5.4 says</font></div><ul style=3D"color:rgb(0,0,0);font-size:13.33333301=
54419px"><li style=3D"margin-left:2em;margin-right:2em"><font face=3D"verda=
na, sans-serif">Next, the SessionDescription is parsed into a data structur=
e, as described in the=C2=A0<a href=3D"https://tools.ietf.org/id/draft-ietf=
-rtcweb-jsep-09.html#sec.parsing-a-desc" style=3D"text-decoration:none">Sec=
tion 5.6</a>section below. If parsing fails for any reason, processing MUST=
 stop and an error MUST be returned.</font></li></ul><div><span style=3D"co=
lor:rgb(0,0,0);font-family:verdana,sans-serif;font-size:13.3333330154419px"=
>[Suhas]</span><font color=3D"#000000" face=3D"verdana, sans-serif"><span s=
tyle=3D"font-size:13.3333330154419px">=C2=A0 =C2=A0 =C2=A0Should we add a n=
ote to say that the remote offer MUST be rejected by the JSEP application w=
hen the application of Answer fails.</span></font></div><div><font face=3D"=
verdana, sans-serif"><br></font></div><div><font face=3D"verdana, sans-seri=
f"><br></font></div><div><font face=3D"verdana, sans-serif">Section 5.6</fo=
nt></div><div><font face=3D"verdana, sans-serif">=C2=A0</font></div><div><s=
pan style=3D"color:rgb(0,0,0);font-size:13.3333330154419px"><font face=3D"v=
erdana, sans-serif">The &quot;c=3D&quot; line MUST be parsed and stored.</f=
ont></span></div><div><span style=3D"color:rgb(0,0,0);font-size:13.33333301=
54419px"><font face=3D"verdana, sans-serif">[Suhas] =C2=A0 Should we say an=
ything out connection address being Unicast vs Multicast.</font></span></di=
v><div><span style=3D"color:rgb(0,0,0);font-size:13.3333330154419px"><font =
face=3D"verdana, sans-serif"><br></font></span></div><div><span style=3D"co=
lor:rgb(0,0,0);font-size:13.3333330154419px"><font face=3D"verdana, sans-se=
rif">For non-attribute (non-&quot;a=3D&quot;) lines, their sequencing, synt=
ax, and semantics, are checked, as mentioned above. The following lines are=
 not meaningful in the JSEP context and MAY be discarded once they have bee=
n checked.</font></span></div><div><ul class=3D"" style=3D"list-style-type:=
none;color:rgb(0,0,0);font-size:13.3333330154419px"><li style=3D"margin-lef=
t:2em;margin-right:2em;margin-top:0.5em"><font face=3D"verdana, sans-serif"=
>TODO</font></li></ul></div><div><span style=3D"color:rgb(0,0,0);font-size:=
13.3333330154419px"><font face=3D"verdana, sans-serif">[Suhas]. These can b=
e added here u=3D, e=3D, p=3D, r=3D, z=3D, k=3D</font></span></div><div><sp=
an style=3D"color:rgb(0,0,0);font-size:13.3333330154419px"><font face=3D"ve=
rdana, sans-serif"><br></font></span></div><div><span style=3D"color:rgb(0,=
0,0);font-size:13.3333330154419px"><font face=3D"verdana, sans-serif">[OPEN=
 ISSUE: For example, because session-level bandwidth is ambiguous when mult=
iple media streams are present, a &quot;b=3D&quot; line at session level is=
 not useful and its value SHOULD be ignored. [OPEN ISSUE: is this WG consen=
sus? Are there other non-a=3D lines that we need to do more than just synta=
ctical validation, e.g. v=3D?]</font></span></div><div><font face=3D"verdan=
a, sans-serif"><br></font></div><div><font face=3D"verdana, sans-serif">[Su=
has] For b=3D handling, I agree that having them specified at the media lev=
el makes sense. The draft should however say if any of RFC4566 or RFC3556 b=
=3D attributes are missing at the media level, what needs to be the expecte=
d behavior</font></div><div><font face=3D"verdana, sans-serif"><br></font><=
/div><div><font face=3D"verdana, sans-serif"><br></font></div><div><font fa=
ce=3D"verdana, sans-serif">Section 5.6.2</font></div><div><font face=3D"ver=
dana, sans-serif"><span style=3D"color:rgb(0,0,0);font-size:13.333333015441=
9px">The &quot;b=3D&quot; line, if present, MUST be parsed as specified in=
=C2=A0</span><a href=3D"https://tools.ietf.org/id/draft-ietf-rtcweb-jsep-09=
.html#RFC4566" style=3D"font-size:13.3333330154419px;text-decoration:none">=
[RFC4566]</a><span style=3D"color:rgb(0,0,0);font-size:13.3333330154419px">=
, Section 5.8, and the bwtype and bandwidth values stored.</span></font></d=
iv><div><span style=3D"color:rgb(0,0,0);font-size:13.3333330154419px"><font=
 face=3D"verdana, sans-serif">[Suhas] =C2=A0We also need to add RFC3556 ref=
erences here.</font></span></div><div><span style=3D"color:rgb(0,0,0);font-=
size:13.3333330154419px"><font face=3D"verdana, sans-serif">Also we need to=
 add ice-options for media section.=C2=A0</font></span></div><div><span sty=
le=3D"color:rgb(0,0,0);font-size:13.3333330154419px"><font face=3D"verdana,=
 sans-serif"><br></font></span></div><div><span style=3D"color:rgb(0,0,0);f=
ont-size:13.3333330154419px"><font face=3D"verdana, sans-serif">Section 5.6=
.3</font></span></div><div><span style=3D"color:rgb(0,0,0);font-family:verd=
ana,sans-serif;font-size:13.3333330154419px">[Suhas]</span><span style=3D"c=
olor:rgb(0,0,0);font-size:13.3333330154419px"><font face=3D"verdana, sans-s=
erif">=C2=A0 =C2=A0 &quot;Semantics verification&quot; is not defined and t=
he intent behind the text in this section is unclear.</font></span></div><d=
iv><span style=3D"color:rgb(0,0,0);font-size:13.3333330154419px"><font face=
=3D"verdana, sans-serif"><br></font></span></div><div><font color=3D"#00000=
0" face=3D"verdana, sans-serif"><span style=3D"font-size:13.3333330154419px=
">Section 5.9</span></font></div><div><span style=3D"color:rgb(0,0,0);font-=
family:verdana,sans-serif;font-size:13.3333330154419px">[Suhas]</span><font=
 color=3D"#000000" face=3D"verdana, sans-serif"><span style=3D"font-size:13=
.3333330154419px">=C2=A0 Is underspecified (WIP)</span></font></div><div><f=
ont color=3D"#000000" face=3D"verdana, sans-serif"><span style=3D"font-size=
:13.3333330154419px"><br></span></font></div><div><font color=3D"#000000" f=
ace=3D"verdana, sans-serif"><span style=3D"font-size:13.3333330154419px"><b=
r></span></font></div><div><div><font face=3D"verdana, sans-serif">Just a c=
razy thought .. Would it help in defining a registry of SDP attributes for =
RTCWeb usage.. This helps today&#39;s and future SDP attributes that gets d=
efined for RTCWeb usage to go to a single place for reference. Also makes i=
t easier to add new attributes to the registry in the future.</font></div><=
/div><div><font face=3D"verdana, sans-serif"><br></font></div><div><br></di=
v><div><font color=3D"#000000" face=3D"verdana, sans-serif"><span style=3D"=
font-size:13.3333330154419px">Thanks</span></font></div><div><font color=3D=
"#000000" face=3D"verdana, sans-serif"><span style=3D"font-size:13.33333301=
54419px">Suhas</span></font></div></div>

--bcaec51b1ea118da4c05120d3cb1--


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Hi,

Removing the parts where there is nothing further to discuss.

On 2015-03-24 00:24, Justin Uberti wrote:
> On Wed, Mar 18, 2015 at 3:13 AM, Magnus Westerlund
> <magnus.westerlund@ericsson.com <mailto:magnus.westerlund@ericsson.com>>
> wrote:
> 
>     2. Section 3.1:
>        This approach, as described in [RFC5956], Section 4.3, sends FEC
>        packets as an independent SSRC-multiplexed stream, with its own SSRC
>        and payload type.  While by far the most flexible, each FEC packet
>        will have its own IP+UDP+RTP+FEC header, leading to additional
>        overhead of the FEC stream.
> 
>     I think this section must have more description of what "Separate FEC
>     stream" really is. The "FEC Grouping for SSRC-Multiplexed RTP Streams"
>     reference also appear to restrictive. Based on the transport documents
>     there actually exists an requirement for supporting fully unbundle
>     configuration. That is not really matching what Section 4.3 describes.
> 
> 
> Even when BUNDLE is not used, unified plan still indicates that the FEC
> stream will be sent as a SSRC-muxed stream. I think the text in 3.1 is
> accurate, although this could be spelled out better.

I am not fully convinced. I would think that if you completely unbundle
that would imply that you can have separate RTP sessions for FEC. The
reason I raise this is that if one actually have flow based QoS you
might actually want to operate the FEC stream on a higher drop
probability than the main media stream.

I note that it doesn't appear that we have any documentation going into
a document that we intended to publish that discusses this issue.

https://datatracker.ietf.org/doc/draft-ietf-avtcore-multi-media-rtp-session/

does in Section 7.2 have a short discussion of this.



> 
> 
>     >From my perspective there are two sets of orthogonal questions around
>     FEC streams. First the relation between the source flow(s) and the
>     repair flows, which may be 1 to 1, 1 to M, or N to 1, or N to M. Then we
>     have the other question of how they are transmitted in relation to the
>     source flow. Within the same RTP session, on different RTP session,
>     outside of the RTP session. And if one has many source flows, then one
>     can also discuss if there is N source RTP sessions and M repair sessions
>     or 1 source RTP session and 1 repair session.
> 
> 
> The salient issue is that there is no current way with unified to have a
> FEC stream as its own m= line. This will require some thinking to resolve.
>  
> 

What encodes unified really. Bundle encodes certain properties as long
as you actually do bundling. However, if I don't include bundle
attributes then we what prevents anyone from having a different m= line
with the FEC payload format and bind it to a media description using the
a=Group: FEC-FR to bind them together.



> 
>     3. Section 3.4:
> 
>     I think you want to discuss FEC information as a "redundant"
>     encoding also.
> 
> 
> There is no section 3.4 - are you suggesting clarifying 3.2's use of
> "Redundant Encoding" to refer to 2198? 

It was suggestion for a new section. What I mean is that one rune XOR
based but include it in the RTP session using RFC2198. OR at least
discuss why you excluded this possible inclusion.
> 
> 
> 
>     4. Section 4.1:
>        When using constant-bitrate codecs, e.g.  PCMU, use of [RFC2198]
>        redundant encoding MAY be used, but note that this will result in a
>        potentially significant bitrate increase, and that suddenly
>        increasing bitrate to deal with losses from congestion may actually
>        make things worse.
> 
>        Because of the lower packet rate of audio encodings, usually a single
>        packet per frame, use of a separate FEC stream comes with a higher
>        overhead than other mechanisms, and therefore is NOT RECOMMENDED.
> 
>     I think this fails to discuss the quite reasonable application of XOR
>     FEC as a redundant encoding information. That way one can achieve a
>     bitrate increase that is ~1/N of the source flow bit-rate by including
>     one XOR packet every N packets covering those N packets. Sure that only
>     gives single packet loss protection, but might be appropriate anyway.
> 
> 
> I'll bring this up on Wednesday. I didn't get the sense from Honolulu
> that there was support for doing any nontrivial FEC for audio.

Good.

> 
>     7. Section 6.
> 
>     I think this section must discuss the fact that the data channel have
>     several possibilities for dealing with reliability. First one can use a
>     reliable stream, that will do retransmissions. One can use semi-reliable
>     stream to not needless continue to try, or the application can select to
>     repeat the data itself over an unreliable stream. I think the last
>     should be recommended against as it will waste resources unnecessarily.
> 
> 
> If the application wants to send redundant data as an application-level
> FEC mechanism, that seems fine to me. The SCTP CC will prevent the
> application from harming others. 

Sure, this might not be that dangerous, but I think it is worth spending
some few words pointing out that the Datachannel actually have several
modes to achieve full or partial reliability and don't really need to
use FEC.

> 
> 
>     8. Section 7:
> 
>     What about using RFC 2198 and [I-D.ietf-payload-flexible-fec-scheme]? Or
>     isn't that supported at all by [I-D.ietf-payload-flexible-fec-scheme]?
>     (Yes, I will review that spec, but haven't gotten to it yet).
> 
> 
> Do you mean using these two in conjunction? I am not sure there is any
> advantage to doing so, but open to discussing this. 

Yes, are there any interest to use them in conjunction?

> 
> 
>     9. Section 9.
> 
>     I think there is a point of discussing how FEC configuration can cause
>     significant waste of bandwidth. Making even a encoded stream with modest
>     bandwidth needs into a bandwidth consuming monster simply by influencing
>     the usage of FEC and its parameters. Thus, I think this document needs
>     to discuss the issues with possible interference from malicious
>     applications with the signalling and the need for a WebRTC
>     implementation to actually control the application and usage of FEC
>     itself and not rely on the signalling parameters.
> 
> 
> True, but it will still be capped by the congestion control, so I don't
> think there is a real-world attack here.  
> 

Yes, it might not hurt the external world particular much. It will waste
some resources and reduced the user experience.

Cheers

Magnus Westerlund

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Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
FÃ¤rÃ¶gatan 6                 | Mobile +46 73 0949079
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Hi,

About using AS at session level. Making it an error may cause
interoperability problems with other systems.

 

Note that 3GPP TS22114
(http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.08.00_60/ts_126
114v120800p.pdf ) in section A.6 uses AS at the session level and not CT for
total bandwidth

 

The IMTC SIP parity WG took the same approach in the SIP Video Profile Best
Practices


 

Roni Even

 

 


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div.WordSection1
	{page:WordSection1;}
--></style><!--[if gte mso 9]><xml>
<o:shapedefaults v:ext=3D"edit" spidmax=3D"1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext=3D"edit">
<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>Hi,<o:p></o:p></p><p class=3DMsoNormal>About using AS =
at session level. Making it an error may cause interoperability problems =
with other systems.<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Note that =
3GPP TS22114 (<a =
href=3D"http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.08.00=
_60/ts_126114v120800p.pdf">http://www.etsi.org/deliver/etsi_ts/126100_126=
199/126114/12.08.00_60/ts_126114v120800p.pdf</a> ) in section A.6 uses =
AS at the session level and not CT for total bandwidth<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>The IMTC SIP =
parity WG took the same approach in the SIP Video Profile Best =
Practices<span style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif";color:black'><o:p></o:p></span></p><table =
class=3DMsoNormalTable border=3D1 cellspacing=3D0 cellpadding=3D0 =
style=3D'border-collapse:collapse;border:none'><tr =
style=3D'height:6.25pt'><td width=3D255 valign=3Dtop =
style=3D'width:191.25pt;border:none;padding:0in 5.4pt 0in =
5.4pt;height:6.25pt'><p class=3DMsoNormal =
style=3D'text-autospace:none'><span style=3D'color:black'> =
<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
style=3D'color:black'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span style=3D'color:black'>Roni =
Even<o:p></o:p></span></p></td></tr></table><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></body></html>
------=_NextPart_000_0185_01D06743.4E14CC40--


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Date: Wed, 25 Mar 2015 14:36:11 -0500
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [rtcweb] Comments on draft-ietf-rtcweb-security-arch-11
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Hi,

I have reviewed draft-ietf-rtcweb-security-arch-11:

1. Section 5.2: The screen sharing Open issue must go away. I am fine
with the general direction proposed yesterday.

2. Section 5.4:

I wonder if there are actually needs to be a consideration on the WebRTC
implementation to have a user knob that forces it into TURN only mode,
so that one can't use the browser to get the local IP. Otherwise there
is no way for the user to preserve their IP addresses against a
malicious JS that like to find a browser hiding behind a proxy or other
address hiding mechanism.

3. Section 5.5:

   All implementations MUST implement DTLS 1.0, with the cipher suite
   TLS_ECDHE_RSA_WITH_AES_128_CBC_SHA and the DTLS-SRTP protection
   profile SRTP_AES128_CM_HMAC_SHA1_80.  Implementations SHOULD
   implement DTLS 1.2 with the TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256
   cipher suite.  Implementations SHOULD favor cipher suites which
   support PFS over non-PFS cipher suites and GCM over CBC cipher
   suites.  [[OPEN ISSUE: Should we require ECDSA?  Waiting for WG
   Consensus.]]

This open issue must be resolved.

4. Section 5.5:


      *  The "security characteristics" MUST include an indication as to
         whether the cryptographic keys were delivered out-of-band (from
         a server) or were generated as a result of a pairwise
         negotiation.

Are we actually allowing any method that would allow out-of-band
provisioning. If not should this be removed.

5. Section 5.5:

      *  The "security characteristics" MUST indicate the cryptographic
         algorithms in use (For example: "AES-CBC" or "Null Cipher".)
         However, if Null ciphers are used, that MUST be presented to
         the user at the top-level UI.

I thought Null cipher was forbidden, does it makes sense to keep that
example?

6. Section 5.6.4.2:


   identity-attribute  = "identity:" identity-assertion
                         [ SP identity-extension
                           *(";" [ SP ] identity-extension) ]
   identity-assertion  = base64
   base64              = 1*(ALPHA / DIGIT / "+" / "/" / "=" )
   identity-extension  = extension-att-name [ "=" extension-att-value ]
   extension-att-name  = token
   extension-att-value = 1*(%x01-09 / %x0b-0c / %x0e-3a / %x3c-ff)
                         ; byte-string from [RFC4566] omitting ";"

These two lines requires references for base64 and token:

identity-assertion  = base64
extension-att-name  = token

7. Section 6.1:
   While this document favors DTLS-SRTP, it permits a variety of
   communications security mechanisms and thus the level of
   communications security actually provided varies considerably.

I think this is also a remnant of the time before we restricted the
keying to DTLS-SRTP for SRTP?

8. Section 6.1:

In order to protect against malicious content JavaScript, that
   JavaScript MUST NOT be allowed to have direct access to---or perform
   computations with---DTLS keys.

There are decorations in the text that are not standard ASCII.

9. Section 6.3:

   [I-D.ietf-rtcweb-rtp-usage] Section 13 documents a number of
   potential RTCP-based DoS attacks and countermeasures.

Those security considerations in RTP usage are not only RTCP based,
there is both discussion of the VBR threat as well as header extension
encryption. I wonder if the reference to this documents security issues
needs to be expanded.

10. Section 6.4.1:

   In order to prevent this attack, we require that all signatures be
   tied to a specific origin ("rtcweb://...") which cannot be produced
   by content JavaScript.

I have seen no indication that this scheme is being registered. Because,
it looks like it needs to be registered to ensure that it doesn't start
being used in another context.

11. Section 8:

S/Magnus Westerland/Magnus Westerlund


12. Reference section, please update.

   [I-D.ietf-avtcore-6222bis]
              Begen, A., Perkins, C., Wing, D., and E. Rescorla,
              "Guidelines for Choosing RTP Control Protocol (RTCP)
              Canonical Names (CNAMEs)", draft-ietf-avtcore-6222bis-06
              (work in progress), July 2013.

This is RFC 7022.

   [I-D.muthu-behave-consent-freshness]
              Perumal, M., Wing, D., R, R., and T. Reddy, "STUN Usage
              for Consent Freshness", draft-muthu-behave-consent-
              freshness-04 (work in progress), July 2013.

This is draft-ietf-rtcweb-stun-consent-freshness-11

13. Appendix A:

[[TODO: These still need some cleanup.]]

Please perform what ever this todo means and then remove it.


Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
FÃ¤rÃ¶gatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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--001a11413ace2619760512223f01
Content-Type: text/plain; charset=UTF-8

On 25 March 2015 at 14:33, Roni Even <ron.even.tlv@gmail.com> wrote:

> About using AS at session level. Making it an error may cause
> interoperability problems with other systems.
>

Would it be a major problem in that situation if the b=AS were ignored?

I think that we could also require that b=CT be ignored on the media level
too.  If it turns out that it has significant usage, that is.

--001a11413ace2619760512223f01
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">=
On 25 March 2015 at 14:33, Roni Even <span dir=3D"ltr">&lt;<a href=3D"mailt=
o:ron.even.tlv@gmail.com" target=3D"_blank">ron.even.tlv@gmail.com</a>&gt;<=
/span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex"><p class=3D"MsoNormal">Abou=
t using AS at session level. Making it an error may cause interoperability =
problems with other systems.<u></u><u></u></p></blockquote></div><br></div>=
<div class=3D"gmail_extra">Would it be a major problem in that situation if=
 the b=3DAS were ignored?<br><br></div><div class=3D"gmail_extra">I think t=
hat we could also require that b=3DCT be ignored on the media level too.=C2=
=A0 If it turns out that it has significant usage, that is.<br></div></div>

--001a11413ace2619760512223f01--


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I=92ve uploaded mt=92s slides to the meeting material site:
http://www.ietf.org/proceedings/92/slides/slides-92-rtcweb-6.pdf.

spt=


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From: Justin Uberti <juberti@google.com>
Date: Thu, 26 Mar 2015 00:24:43 -0500
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Subject: [rtcweb] Endpoints that don't support RTCP
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This is described in rtp-usage, and the need to limit bitrate is explicitly
called out:

7.2 <https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-22#section-7.2>.
Congestion Control Interoperability and Legacy Systems

   There are legacy RTP implementations that do not implement RTCP, and
   hence do not provide any congestion feedback.  Congestion control
   cannot be performed with these end-points.  WebRTC Endpoints that
   need to interwork with such end-points MUST limit their transmission
   to a low rate, equivalent to a VoIP call using a low bandwidth codec,
   that is unlikely to cause any significant congestion.


Note however that this is incompatible with this section that mandates
use of circuit breakers:


7.1 <https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-22#section-7.1>.
Boundary Conditions and Circuit Breakers

   WebRTC Endpoints MUST implement the RTP circuit breaker algorithm
   that is described in [I-D.ietf-avtcore-rtp-circuit-breakers
<https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-22#ref-I-D.ietf-avtcore-rtp-circuit-breakers>].
The
   RTP circuit breaker is designed to enable applications to recognise
   and react to situations of extreme network congestion.  However,
   since the RTP circuit breaker might not be triggered until congestion
   becomes extreme, it cannot be considered a substitute for congestion
   control, and applications MUST also implement congestion control to
   allow them to adapt to changes in network capacity.  Any future RTP
   congestion control algorithms are expected to operate within the
   envelope allowed by the circuit breaker.


I think we need to clarify exactly how WebRTC endpoints need to deal
with legacy gear that doesn't send RTCP (which might just be to say
"no").

--001a11408e50e8eeb105122a3c0c
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">This is described in rtp-usage, and the need to limit bitr=
ate is explicitly called out:<div><br></div><div><pre class=3D"" style=3D"f=
ont-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><span class=
=3D"" style=3D"line-height:0pt;display:inline;font-size:1em;font-weight:bol=
d"><h3 style=3D"line-height:0pt;display:inline;font-size:1em"><a class=3D""=
 name=3D"section-7.2" href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb=
-rtp-usage-22#section-7.2" style=3D"color:black;text-decoration:none">7.2</=
a>.  Congestion Control Interoperability and Legacy Systems</h3></span>

   There are legacy RTP implementations that do not implement RTCP, and
   hence do not provide any congestion feedback.  Congestion control
   cannot be performed with these end-points.  WebRTC Endpoints that
   need to interwork with such end-points MUST limit their transmission
   to a low rate, equivalent to a VoIP call using a low bandwidth codec,
   that is unlikely to cause any significant congestion.</pre><pre class=3D=
"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)=
"><br></pre><pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bo=
ttom:0px;color:rgb(0,0,0)"><span style=3D"color:rgb(34,34,34);font-family:a=
rial,sans-serif;font-size:small;white-space:normal">Note however that this =
is incompatible with this section that mandates use of circuit breakers:</s=
pan><br></pre><pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-=
bottom:0px;color:rgb(0,0,0)"><span style=3D"color:rgb(34,34,34);font-family=
:arial,sans-serif;font-size:small;white-space:normal"><br></span></pre><pre=
 class=3D"" style=3D"margin-top:0px;margin-bottom:0px"><pre class=3D"" styl=
e=3D"color:rgb(0,0,0);font-size:1em;margin-top:0px;margin-bottom:0px"><span=
 class=3D"" style=3D"line-height:0pt;display:inline;font-size:1em;font-weig=
ht:bold"><h3 style=3D"line-height:0pt;display:inline;font-size:1em"><a clas=
s=3D"" name=3D"section-7.1" href=3D"https://tools.ietf.org/html/draft-ietf-=
rtcweb-rtp-usage-22#section-7.1" style=3D"color:black;text-decoration:none"=
>7.1</a>.  Boundary Conditions and Circuit Breakers</h3></span>

   WebRTC Endpoints MUST implement the RTP circuit breaker algorithm
   that is described in [<a href=3D"https://tools.ietf.org/html/draft-ietf-=
rtcweb-rtp-usage-22#ref-I-D.ietf-avtcore-rtp-circuit-breakers">I-D.ietf-avt=
core-rtp-circuit-breakers</a>].  The
   RTP circuit breaker is designed to enable applications to recognise
   and react to situations of extreme network congestion.  However,
   since the RTP circuit breaker might not be triggered until congestion
   becomes extreme, it cannot be considered a substitute for congestion
   control, and applications MUST also implement congestion control to
   allow them to adapt to changes in network capacity.  Any future RTP
   congestion control algorithms are expected to operate within the
   envelope allowed by the circuit breaker.</pre><pre class=3D"" style=3D"c=
olor:rgb(0,0,0);font-size:1em;margin-top:0px;margin-bottom:0px"><br></pre><=
pre class=3D"" style=3D"color:rgb(0,0,0);font-size:1em;margin-top:0px;margi=
n-bottom:0px"><span style=3D"color:rgb(34,34,34);font-family:arial,sans-ser=
if;font-size:small;white-space:normal">I think we need to clarify exactly h=
ow WebRTC endpoints need to deal with legacy gear that doesn&#39;t send RTC=
P (which might just be to say &quot;no&quot;).</span></pre><div style=3D"co=
lor:rgb(0,0,0);font-size:1em"><span style=3D"color:rgb(34,34,34);font-famil=
y:arial,sans-serif;font-size:small;white-space:normal"><br></span></div><pr=
e class=3D"" style=3D"color:rgb(0,0,0);font-size:1em;margin-top:0px;margin-=
bottom:0px"><br></pre><pre class=3D"" style=3D"color:rgb(0,0,0);font-size:1=
em;margin-top:0px;margin-bottom:0px"><br></pre></pre><pre class=3D"" style=
=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br></=
pre><pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px=
;color:rgb(0,0,0)"><br></pre></div></div>

--001a11408e50e8eeb105122a3c0c--


From nobody Thu Mar 26 07:19:02 2015
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From: Colin Perkins <csp@csperkins.org>
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Date: Thu, 26 Mar 2015 09:18:41 -0500
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Subject: Re: [rtcweb] Endpoints that don't support RTCP
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--Apple-Mail=_B9BFBBD8-8EE0-4B2F-BDEA-2CAB12C61D38
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Can you suggest text that you think would be clearer?

Colin



On 26 Mar 2015, at 00:24, Justin Uberti <juberti@google.com> wrote:
> This is described in rtp-usage, and the need to limit bitrate is =
explicitly called out:
>=20
> 7.2.  Congestion Control Interoperability and Legacy Systems
>=20
>    There are legacy RTP implementations that do not implement RTCP, =
and
>    hence do not provide any congestion feedback.  Congestion control
>    cannot be performed with these end-points.  WebRTC Endpoints that
>    need to interwork with such end-points MUST limit their =
transmission
>    to a low rate, equivalent to a VoIP call using a low bandwidth =
codec,
>    that is unlikely to cause any significant congestion.
>=20
> Note however that this is incompatible with this section that mandates =
use of circuit breakers:
>=20
> 7.1.  Boundary Conditions and Circuit Breakers
>=20
>    WebRTC Endpoints MUST implement the RTP circuit breaker algorithm
>    that is described in [I-D.ietf-avtcore-rtp-circuit-breakers].  The
>    RTP circuit breaker is designed to enable applications to recognise
>    and react to situations of extreme network congestion.  However,
>    since the RTP circuit breaker might not be triggered until =
congestion
>    becomes extreme, it cannot be considered a substitute for =
congestion
>    control, and applications MUST also implement congestion control to
>    allow them to adapt to changes in network capacity.  Any future RTP
>    congestion control algorithms are expected to operate within the
>    envelope allowed by the circuit breaker.
>=20
> I think we need to clarify exactly how WebRTC endpoints need to deal =
with legacy gear that doesn't send RTCP (which might just be to say =
"no").
>=20
>=20
>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



--=20
Colin Perkins
https://csperkins.org/





--Apple-Mail=_B9BFBBD8-8EE0-4B2F-BDEA-2CAB12C61D38
Content-Transfer-Encoding: quoted-printable
Content-Type: text/html;
	charset=us-ascii

<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dus-ascii"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space;"><div>Can you suggest text that you think would be =
clearer?</div><div><br></div><div>Colin</div><div><br></div><div><br></div=
><div><br><div><div>On 26 Mar 2015, at 00:24, Justin Uberti &lt;<a =
href=3D"mailto:juberti@google.com">juberti@google.com</a>&gt; =
wrote:</div><blockquote type=3D"cite"><div dir=3D"ltr">This is described =
in rtp-usage, and the need to limit bitrate is explicitly called =
out:<div><br></div><div><pre class=3D"" style=3D"font-size: 1em; =
margin-top: 0px; margin-bottom: 0px;"><span class=3D"" =
style=3D"line-height:0pt;display:inline;font-size:1em;font-weight:bold"><h=
3 style=3D"line-height:0pt;display:inline;font-size:1em"><a class=3D"" =
name=3D"section-7.2" =
href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-22#section=
-7.2" style=3D"text-decoration: none;">7.2</a>.  Congestion Control =
Interoperability and Legacy Systems</h3></span>

   There are legacy RTP implementations that do not implement RTCP, and
   hence do not provide any congestion feedback.  Congestion control
   cannot be performed with these end-points.  WebRTC Endpoints that
   need to interwork with such end-points MUST limit their transmission
   to a low rate, equivalent to a VoIP call using a low bandwidth codec,
   that is unlikely to cause any significant congestion.</pre><pre =
class=3D"" style=3D"font-size: 1em; margin-top: 0px; margin-bottom: =
0px;"><br></pre><pre class=3D"" style=3D"font-size: 1em; margin-top: =
0px; margin-bottom: 0px;"><span =
style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:small;=
white-space:normal">Note however that this is incompatible with this =
section that mandates use of circuit breakers:</span><br></pre><pre =
class=3D"" style=3D"font-size: 1em; margin-top: 0px; margin-bottom: =
0px;"><span =
style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:small;=
white-space:normal"><br></span></pre><pre class=3D"" =
style=3D"margin-top:0px;margin-bottom:0px"><pre class=3D"" =
style=3D"font-size: 1em; margin-top: 0px; margin-bottom: 0px;"><span =
class=3D"" =
style=3D"line-height:0pt;display:inline;font-size:1em;font-weight:bold"><h=
3 style=3D"line-height:0pt;display:inline;font-size:1em"><a class=3D"" =
name=3D"section-7.1" =
href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-22#section=
-7.1" style=3D"text-decoration: none;">7.1</a>.  Boundary Conditions and =
Circuit Breakers</h3></span>

   WebRTC Endpoints MUST implement the RTP circuit breaker algorithm
   that is described in [<a =
href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-22#ref-I-D=
.ietf-avtcore-rtp-circuit-breakers">I-D.ietf-avtcore-rtp-circuit-breakers<=
/a>].  The
   RTP circuit breaker is designed to enable applications to recognise
   and react to situations of extreme network congestion.  However,
   since the RTP circuit breaker might not be triggered until congestion
   becomes extreme, it cannot be considered a substitute for congestion
   control, and applications MUST also implement congestion control to
   allow them to adapt to changes in network capacity.  Any future RTP
   congestion control algorithms are expected to operate within the
   envelope allowed by the circuit breaker.</pre><pre class=3D"" =
style=3D"font-size: 1em; margin-top: 0px; margin-bottom: =
0px;"><br></pre><pre class=3D"" style=3D"font-size: 1em; margin-top: =
0px; margin-bottom: 0px;"><span =
style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:small;=
white-space:normal">I think we need to clarify exactly how WebRTC =
endpoints need to deal with legacy gear that doesn't send RTCP (which =
might just be to say "no").</span></pre><div style=3D"font-size: =
1em;"><span =
style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:small;=
white-space:normal"><br></span></div><pre class=3D"" style=3D"font-size: =
1em; margin-top: 0px; margin-bottom: 0px;"><br></pre><pre class=3D"" =
style=3D"font-size: 1em; margin-top: 0px; margin-bottom: =
0px;"><br></pre></pre><pre class=3D"" style=3D"font-size: 1em; =
margin-top: 0px; margin-bottom: 0px;"><br></pre><pre class=3D"" =
style=3D"font-size: 1em; margin-top: 0px; margin-bottom: =
0px;"><br></pre></div></div>
_______________________________________________<br>rtcweb mailing =
list<br><a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>https://www.ietf.or=
g/mailman/listinfo/rtcweb<br></blockquote></div><br><div =
apple-content-edited=3D"true">
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
border-spacing: 0px;"><div><br class=3D"Apple-interchange-newline"><br =
class=3D"khtml-block-placeholder"></div><div>--&nbsp;</div><div></div><div=
>Colin Perkins</div><div><a =
href=3D"https://csperkins.org/">https://csperkins.org/</a></div><div><br><=
/div></span><br class=3D"Apple-interchange-newline"><br =
class=3D"Apple-interchange-newline">
</div>
<br></div></body></html>=

--Apple-Mail=_B9BFBBD8-8EE0-4B2F-BDEA-2CAB12C61D38--


From nobody Thu Mar 26 07:23:33 2015
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From: Matthew Kaufman <matthew@matthew.at>
Date: Thu, 26 Mar 2015 07:23:25 -0700
To: Justin Uberti <juberti@google.com>
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Subject: Re: [rtcweb] Endpoints that don't support RTCP
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These are the legacy endpoints that don't do RTCP but do have ICE and DTLS-S=
RTP?

Matthew Kaufman

(Sent from my iPhone)

> On Mar 25, 2015, at 10:24 PM, Justin Uberti <juberti@google.com> wrote:
>=20
> This is described in rtp-usage, and the need to limit bitrate is explicitl=
y called out:
>=20
> 7.2.  Congestion Control Interoperability and Legacy Systems
>=20
>    There are legacy RTP implementations that do not implement RTCP, and
>    hence do not provide any congestion feedback.  Congestion control
>    cannot be performed with these end-points.  WebRTC Endpoints that
>    need to interwork with such end-points MUST limit their transmission
>    to a low rate, equivalent to a VoIP call using a low bandwidth codec,
>    that is unlikely to cause any significant congestion.
>=20
> Note however that this is incompatible with this section that mandates use=
 of circuit breakers:
>=20
> 7.1.  Boundary Conditions and Circuit Breakers
>=20
>    WebRTC Endpoints MUST implement the RTP circuit breaker algorithm
>    that is described in [I-D.ietf-avtcore-rtp-circuit-breakers].  The
>    RTP circuit breaker is designed to enable applications to recognise
>    and react to situations of extreme network congestion.  However,
>    since the RTP circuit breaker might not be triggered until congestion
>    becomes extreme, it cannot be considered a substitute for congestion
>    control, and applications MUST also implement congestion control to
>    allow them to adapt to changes in network capacity.  Any future RTP
>    congestion control algorithms are expected to operate within the
>    envelope allowed by the circuit breaker.
>=20
> I think we need to clarify exactly how WebRTC endpoints need to deal with l=
egacy gear that doesn't send RTCP (which might just be to say "no").
>=20
>=20
>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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<html><head><meta http-equiv=3D"content-type" content=3D"text/html; charset=3D=
utf-8"></head><body dir=3D"auto"><div>These are the legacy endpoints that do=
n't do RTCP but do have ICE and DTLS-SRTP?<br><br>Matthew Kaufman<div><br>(S=
ent from my iPhone)</div></div><div><br>On Mar 25, 2015, at 10:24 PM, Justin=
 Uberti &lt;<a href=3D"mailto:juberti@google.com">juberti@google.com</a>&gt;=
 wrote:<br><br></div><blockquote type=3D"cite"><div><div dir=3D"ltr">This is=
 described in rtp-usage, and the need to limit bitrate is explicitly called o=
ut:<div><br></div><div><pre class=3D"" style=3D"font-size:1em;margin-top:0px=
;margin-bottom:0px;color:rgb(0,0,0)"><span class=3D"" style=3D"line-height:0=
pt;display:inline;font-size:1em;font-weight:bold"><h3 style=3D"line-height:0=
pt;display:inline;font-size:1em"><a class=3D"" name=3D"section-7.2" href=3D"=
https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-22#section-7.2" styl=
e=3D"color:black;text-decoration:none">7.2</a>.  Congestion Control Interope=
rability and Legacy Systems</h3></span>

   There are legacy RTP implementations that do not implement RTCP, and
   hence do not provide any congestion feedback.  Congestion control
   cannot be performed with these end-points.  WebRTC Endpoints that
   need to interwork with such end-points MUST limit their transmission
   to a low rate, equivalent to a VoIP call using a low bandwidth codec,
   that is unlikely to cause any significant congestion.</pre><pre class=3D"=
" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">=
<br></pre><pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-botto=
m:0px;color:rgb(0,0,0)"><span style=3D"color:rgb(34,34,34);font-family:arial=
,sans-serif;font-size:small;white-space:normal">Note however that this is in=
compatible with this section that mandates use of circuit breakers:</span><b=
r></pre><pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:=
0px;color:rgb(0,0,0)"><span style=3D"color:rgb(34,34,34);font-family:arial,s=
ans-serif;font-size:small;white-space:normal"><br></span></pre><pre class=3D=
"" style=3D"margin-top:0px;margin-bottom:0px"><pre class=3D"" style=3D"color=
:rgb(0,0,0);font-size:1em;margin-top:0px;margin-bottom:0px"><span class=3D""=
 style=3D"line-height:0pt;display:inline;font-size:1em;font-weight:bold"><h3=
 style=3D"line-height:0pt;display:inline;font-size:1em"><a class=3D"" name=3D=
"section-7.1" href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usag=
e-22#section-7.1" style=3D"color:black;text-decoration:none">7.1</a>.  Bound=
ary Conditions and Circuit Breakers</h3></span>

   WebRTC Endpoints MUST implement the RTP circuit breaker algorithm
   that is described in [<a href=3D"https://tools.ietf.org/html/draft-ietf-r=
tcweb-rtp-usage-22#ref-I-D.ietf-avtcore-rtp-circuit-breakers">I-D.ietf-avtco=
re-rtp-circuit-breakers</a>].  The
   RTP circuit breaker is designed to enable applications to recognise
   and react to situations of extreme network congestion.  However,
   since the RTP circuit breaker might not be triggered until congestion
   becomes extreme, it cannot be considered a substitute for congestion
   control, and applications MUST also implement congestion control to
   allow them to adapt to changes in network capacity.  Any future RTP
   congestion control algorithms are expected to operate within the
   envelope allowed by the circuit breaker.</pre><pre class=3D"" style=3D"co=
lor:rgb(0,0,0);font-size:1em;margin-top:0px;margin-bottom:0px"><br></pre><pr=
e class=3D"" style=3D"color:rgb(0,0,0);font-size:1em;margin-top:0px;margin-b=
ottom:0px"><span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;f=
ont-size:small;white-space:normal">I think we need to clarify exactly how We=
bRTC endpoints need to deal with legacy gear that doesn't send RTCP (which m=
ight just be to say "no").</span></pre><div style=3D"color:rgb(0,0,0);font-s=
ize:1em"><span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;fon=
t-size:small;white-space:normal"><br></span></div><pre class=3D"" style=3D"c=
olor:rgb(0,0,0);font-size:1em;margin-top:0px;margin-bottom:0px"><br></pre><p=
re class=3D"" style=3D"color:rgb(0,0,0);font-size:1em;margin-top:0px;margin-=
bottom:0px"><br></pre></pre><pre class=3D"" style=3D"font-size:1em;margin-to=
p:0px;margin-bottom:0px;color:rgb(0,0,0)"><br></pre><pre class=3D"" style=3D=
"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br></pre>=
</div></div>
</div></blockquote><blockquote type=3D"cite"><div><span>____________________=
___________________________</span><br><span>rtcweb mailing list</span><br><s=
pan><a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br><span><=
a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org=
/mailman/listinfo/rtcweb</a></span><br></div></blockquote></body></html>=

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Right, I think mandating use of RTCP is the correct solution. e.g.

7.2 <https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-22#section-7.2>.
Congestion Control Interoperability and Legacy Systems

   All endpoints that wish to interwork with WebRTC MUST

   implement RTCP and provide congestion feedback via the defined RTCP

   reporting mechanisms.


   When interworking with legacy implementations that support RTCP using
   the RTP/AVP profile [RFC3551
<https://tools.ietf.org/html/rfc3551>], congestion feedback is
provided in
   RTCP RR packets every few seconds.  Implementations that have to
   interwork with such end-points MUST ensure that they keep within the
   RTP circuit breaker [I-D.ietf-avtcore-rtp-circuit-breakers
<https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-22#ref-I-D.ietf-avtcore-rtp-circuit-breakers>]
   constraints to limit the congestion they can cause.


On Thu, Mar 26, 2015 at 9:23 AM, Matthew Kaufman <matthew@matthew.at> wrote:

> These are the legacy endpoints that don't do RTCP but do have ICE and
> DTLS-SRTP?
>
> Matthew Kaufman
>
> (Sent from my iPhone)
>
> On Mar 25, 2015, at 10:24 PM, Justin Uberti <juberti@google.com> wrote:
>
> This is described in rtp-usage, and the need to limit bitrate is
> explicitly called out:
>
> 7.2 <https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-22#section-7.2>.  Congestion Control Interoperability and Legacy Systems
>
>    There are legacy RTP implementations that do not implement RTCP, and
>    hence do not provide any congestion feedback.  Congestion control
>    cannot be performed with these end-points.  WebRTC Endpoints that
>    need to interwork with such end-points MUST limit their transmission
>    to a low rate, equivalent to a VoIP call using a low bandwidth codec,
>    that is unlikely to cause any significant congestion.
>
>
> Note however that this is incompatible with this section that mandates use of circuit breakers:
>
>
> 7.1 <https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-22#section-7.1>.  Boundary Conditions and Circuit Breakers
>
>    WebRTC Endpoints MUST implement the RTP circuit breaker algorithm
>    that is described in [I-D.ietf-avtcore-rtp-circuit-breakers <https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-22#ref-I-D.ietf-avtcore-rtp-circuit-breakers>].  The
>    RTP circuit breaker is designed to enable applications to recognise
>    and react to situations of extreme network congestion.  However,
>    since the RTP circuit breaker might not be triggered until congestion
>    becomes extreme, it cannot be considered a substitute for congestion
>    control, and applications MUST also implement congestion control to
>    allow them to adapt to changes in network capacity.  Any future RTP
>    congestion control algorithms are expected to operate within the
>    envelope allowed by the circuit breaker.
>
>
> I think we need to clarify exactly how WebRTC endpoints need to deal with legacy gear that doesn't send RTCP (which might just be to say "no").
>
>
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr">Right, I think mandating use of RTCP is the correct soluti=
on. e.g.<div><br></div><div><pre style=3D"white-space:pre-wrap;font-size:1e=
m;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><span style=3D"line-he=
ight:0pt;font-size:1em;font-weight:bold"><h3 style=3D"line-height:0pt;displ=
ay:inline;font-size:1em"><a name=3D"14c548a2ce2ce9cd_section-7.2" href=3D"h=
ttps://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-22#section-7.2" targ=
et=3D"_blank" style=3D"color:black;text-decoration:none">7.2</a>.  Congesti=
on Control Interoperability and Legacy Systems</h3></span>

   All endpoints that wish to interwork with WebRTC MUST=C2=A0</pre><pre st=
yle=3D"white-space:pre-wrap;font-size:1em;margin-top:0px;margin-bottom:0px;=
color:rgb(0,0,0)">   implement RTCP and provide congestion feedback via the=
 defined RTCP</pre><pre style=3D"white-space:pre-wrap;font-size:1em;margin-=
top:0px;margin-bottom:0px;color:rgb(0,0,0)">   reporting mechanisms. </pre>=
<pre style=3D"white-space:pre-wrap;font-size:1em;margin-top:0px;margin-bott=
om:0px;color:rgb(0,0,0)"><br></pre><pre style=3D"white-space:pre-wrap;font-=
size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><pre class=3D""=
 style=3D"font-size:1em;margin-top:0px;margin-bottom:0px">   When interwork=
ing with legacy implementations that support RTCP using
   the RTP/AVP profile [<a href=3D"https://tools.ietf.org/html/rfc3551" tit=
le=3D"&quot;RTP Profile for Audio and Video Conferences with Minimal Contro=
l&quot;">RFC3551</a>], congestion feedback is provided in
   RTCP RR packets every few seconds.  Implementations that have to
   interwork with such end-points MUST ensure that they keep within the
   RTP circuit breaker [<a href=3D"https://tools.ietf.org/html/draft-ietf-r=
tcweb-rtp-usage-22#ref-I-D.ietf-avtcore-rtp-circuit-breakers">I-D.ietf-avtc=
ore-rtp-circuit-breakers</a>]
   constraints to limit the congestion they can cause.</pre></pre></div></d=
iv><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Thu, Mar 26=
, 2015 at 9:23 AM, Matthew Kaufman <span dir=3D"ltr">&lt;<a href=3D"mailto:=
matthew@matthew.at" target=3D"_blank">matthew@matthew.at</a>&gt;</span> wro=
te:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-=
left:1px #ccc solid;padding-left:1ex"><div dir=3D"auto"><div>These are the =
legacy endpoints that don&#39;t do RTCP but do have ICE and DTLS-SRTP?<br><=
br>Matthew Kaufman<div><br>(Sent from my iPhone)</div></div><div><div class=
=3D"h5"><div><br>On Mar 25, 2015, at 10:24 PM, Justin Uberti &lt;<a href=3D=
"mailto:juberti@google.com" target=3D"_blank">juberti@google.com</a>&gt; wr=
ote:<br><br></div><blockquote type=3D"cite"><div><div dir=3D"ltr">This is d=
escribed in rtp-usage, and the need to limit bitrate is explicitly called o=
ut:<div><br></div><div><pre style=3D"font-size:1em;margin-top:0px;margin-bo=
ttom:0px;color:rgb(0,0,0)"><span style=3D"line-height:0pt;display:inline;fo=
nt-size:1em;font-weight:bold"><h3 style=3D"line-height:0pt;display:inline;f=
ont-size:1em"><a name=3D"14c56776b7749c05_section-7.2" href=3D"https://tool=
s.ietf.org/html/draft-ietf-rtcweb-rtp-usage-22#section-7.2" style=3D"color:=
black;text-decoration:none" target=3D"_blank">7.2</a>.  Congestion Control =
Interoperability and Legacy Systems</h3></span>

   There are legacy RTP implementations that do not implement RTCP, and
   hence do not provide any congestion feedback.  Congestion control
   cannot be performed with these end-points.  WebRTC Endpoints that
   need to interwork with such end-points MUST limit their transmission
   to a low rate, equivalent to a VoIP call using a low bandwidth codec,
   that is unlikely to cause any significant congestion.</pre><pre style=3D=
"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br></pre=
><pre style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0=
,0)"><span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-s=
ize:small;white-space:normal">Note however that this is incompatible with t=
his section that mandates use of circuit breakers:</span><br></pre><pre sty=
le=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><spa=
n style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:small=
;white-space:normal"><br></span></pre><pre style=3D"margin-top:0px;margin-b=
ottom:0px"><pre style=3D"color:rgb(0,0,0);font-size:1em;margin-top:0px;marg=
in-bottom:0px"><span style=3D"line-height:0pt;display:inline;font-size:1em;=
font-weight:bold"><h3 style=3D"line-height:0pt;display:inline;font-size:1em=
"><a name=3D"14c56776b7749c05_section-7.1" href=3D"https://tools.ietf.org/h=
tml/draft-ietf-rtcweb-rtp-usage-22#section-7.1" style=3D"color:black;text-d=
ecoration:none" target=3D"_blank">7.1</a>.  Boundary Conditions and Circuit=
 Breakers</h3></span>

   WebRTC Endpoints MUST implement the RTP circuit breaker algorithm
   that is described in [<a href=3D"https://tools.ietf.org/html/draft-ietf-=
rtcweb-rtp-usage-22#ref-I-D.ietf-avtcore-rtp-circuit-breakers" target=3D"_b=
lank">I-D.ietf-avtcore-rtp-circuit-breakers</a>].  The
   RTP circuit breaker is designed to enable applications to recognise
   and react to situations of extreme network congestion.  However,
   since the RTP circuit breaker might not be triggered until congestion
   becomes extreme, it cannot be considered a substitute for congestion
   control, and applications MUST also implement congestion control to
   allow them to adapt to changes in network capacity.  Any future RTP
   congestion control algorithms are expected to operate within the
   envelope allowed by the circuit breaker.</pre><pre style=3D"color:rgb(0,=
0,0);font-size:1em;margin-top:0px;margin-bottom:0px"><br></pre><pre style=
=3D"color:rgb(0,0,0);font-size:1em;margin-top:0px;margin-bottom:0px"><span =
style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:small;w=
hite-space:normal">I think we need to clarify exactly how WebRTC endpoints =
need to deal with legacy gear that doesn&#39;t send RTCP (which might just =
be to say &quot;no&quot;).</span></pre><div style=3D"color:rgb(0,0,0);font-=
size:1em"><span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;f=
ont-size:small;white-space:normal"><br></span></div><pre style=3D"color:rgb=
(0,0,0);font-size:1em;margin-top:0px;margin-bottom:0px"><br></pre><pre styl=
e=3D"color:rgb(0,0,0);font-size:1em;margin-top:0px;margin-bottom:0px"><br><=
/pre></pre><pre style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;col=
or:rgb(0,0,0)"><br></pre><pre style=3D"font-size:1em;margin-top:0px;margin-=
bottom:0px;color:rgb(0,0,0)"><br></pre></div></div>
</div></blockquote></div></div><span class=3D""><blockquote type=3D"cite"><=
div><span>_______________________________________________</span><br><span>r=
tcweb mailing list</span><br><span><a href=3D"mailto:rtcweb@ietf.org" targe=
t=3D"_blank">rtcweb@ietf.org</a></span><br><span><a href=3D"https://www.iet=
f.org/mailman/listinfo/rtcweb" target=3D"_blank">https://www.ietf.org/mailm=
an/listinfo/rtcweb</a></span><br></div></blockquote></span></div></blockquo=
te></div><br></div>

--001a11c39b12819aff0512320278--


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Date: Thu, 26 Mar 2015 15:57:14 +0100
From: Lorenzo Miniero <lorenzo@meetecho.com>
To: Matthew Kaufman <matthew@matthew.at>
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Subject: Re: [rtcweb] Endpoints that don't support RTCP
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Well, a legacy device that doesn't support RTCP might get ICE and DTLS
support with the help of a WebRTC gateway.

Lorenzo



On Thu, 26 Mar 2015 07:23:25 -0700
Matthew Kaufman <matthew@matthew.at> wrote:

> These are the legacy endpoints that don't do RTCP but do have ICE and
> DTLS-SRTP?
> 
> Matthew Kaufman
> 
> (Sent from my iPhone)
> 
> > On Mar 25, 2015, at 10:24 PM, Justin Uberti <juberti@google.com>
> > wrote:
> > 
> > This is described in rtp-usage, and the need to limit bitrate is
> > explicitly called out:
> > 
> > 7.2.  Congestion Control Interoperability and Legacy Systems
> > 
> >    There are legacy RTP implementations that do not implement RTCP,
> > and hence do not provide any congestion feedback.  Congestion
> > control cannot be performed with these end-points.  WebRTC
> > Endpoints that need to interwork with such end-points MUST limit
> > their transmission to a low rate, equivalent to a VoIP call using a
> > low bandwidth codec, that is unlikely to cause any significant
> > congestion.
> > 
> > Note however that this is incompatible with this section that
> > mandates use of circuit breakers:
> > 
> > 7.1.  Boundary Conditions and Circuit Breakers
> > 
> >    WebRTC Endpoints MUST implement the RTP circuit breaker algorithm
> >    that is described in [I-D.ietf-avtcore-rtp-circuit-breakers].
> > The RTP circuit breaker is designed to enable applications to
> > recognise and react to situations of extreme network congestion.
> > However, since the RTP circuit breaker might not be triggered until
> > congestion becomes extreme, it cannot be considered a substitute
> > for congestion control, and applications MUST also implement
> > congestion control to allow them to adapt to changes in network
> > capacity.  Any future RTP congestion control algorithms are
> > expected to operate within the envelope allowed by the circuit
> > breaker.
> > 
> > I think we need to clarify exactly how WebRTC endpoints need to
> > deal with legacy gear that doesn't send RTCP (which might just be
> > to say "no").
> > 
> > 
> > 
> > 
> > 
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb


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From: Justin Uberti <juberti@google.com>
Date: Thu, 26 Mar 2015 10:01:16 -0500
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--001a1140a146d51e630512324abd
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Sure, although said gateway could certainly send some minimal RTCP as well.

On Thu, Mar 26, 2015 at 9:57 AM, Lorenzo Miniero <lorenzo@meetecho.com>
wrote:

> Well, a legacy device that doesn't support RTCP might get ICE and DTLS
> support with the help of a WebRTC gateway.
>
> Lorenzo
>
>
>
> On Thu, 26 Mar 2015 07:23:25 -0700
> Matthew Kaufman <matthew@matthew.at> wrote:
>
> > These are the legacy endpoints that don't do RTCP but do have ICE and
> > DTLS-SRTP?
> >
> > Matthew Kaufman
> >
> > (Sent from my iPhone)
> >
> > > On Mar 25, 2015, at 10:24 PM, Justin Uberti <juberti@google.com>
> > > wrote:
> > >
> > > This is described in rtp-usage, and the need to limit bitrate is
> > > explicitly called out:
> > >
> > > 7.2.  Congestion Control Interoperability and Legacy Systems
> > >
> > >    There are legacy RTP implementations that do not implement RTCP,
> > > and hence do not provide any congestion feedback.  Congestion
> > > control cannot be performed with these end-points.  WebRTC
> > > Endpoints that need to interwork with such end-points MUST limit
> > > their transmission to a low rate, equivalent to a VoIP call using a
> > > low bandwidth codec, that is unlikely to cause any significant
> > > congestion.
> > >
> > > Note however that this is incompatible with this section that
> > > mandates use of circuit breakers:
> > >
> > > 7.1.  Boundary Conditions and Circuit Breakers
> > >
> > >    WebRTC Endpoints MUST implement the RTP circuit breaker algorithm
> > >    that is described in [I-D.ietf-avtcore-rtp-circuit-breakers].
> > > The RTP circuit breaker is designed to enable applications to
> > > recognise and react to situations of extreme network congestion.
> > > However, since the RTP circuit breaker might not be triggered until
> > > congestion becomes extreme, it cannot be considered a substitute
> > > for congestion control, and applications MUST also implement
> > > congestion control to allow them to adapt to changes in network
> > > capacity.  Any future RTP congestion control algorithms are
> > > expected to operate within the envelope allowed by the circuit
> > > breaker.
> > >
> > > I think we need to clarify exactly how WebRTC endpoints need to
> > > deal with legacy gear that doesn't send RTCP (which might just be
> > > to say "no").
> > >
> > >
> > >
> > >
> > >
> > > _______________________________________________
> > > rtcweb mailing list
> > > rtcweb@ietf.org
> > > https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--001a1140a146d51e630512324abd
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Sure, although said gateway could certainly send some mini=
mal RTCP as well.</div><div class=3D"gmail_extra"><br><div class=3D"gmail_q=
uote">On Thu, Mar 26, 2015 at 9:57 AM, Lorenzo Miniero <span dir=3D"ltr">&l=
t;<a href=3D"mailto:lorenzo@meetecho.com" target=3D"_blank">lorenzo@meetech=
o.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Well, a legac=
y device that doesn&#39;t support RTCP might get ICE and DTLS<br>
support with the help of a WebRTC gateway.<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
Lorenzo<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
<br>
On Thu, 26 Mar 2015 07:23:25 -0700<br>
Matthew Kaufman &lt;<a href=3D"mailto:matthew@matthew.at">matthew@matthew.a=
t</a>&gt; wrote:<br>
<br>
&gt; These are the legacy endpoints that don&#39;t do RTCP but do have ICE =
and<br>
&gt; DTLS-SRTP?<br>
&gt;<br>
&gt; Matthew Kaufman<br>
&gt;<br>
&gt; (Sent from my iPhone)<br>
&gt;<br>
&gt; &gt; On Mar 25, 2015, at 10:24 PM, Justin Uberti &lt;<a href=3D"mailto=
:juberti@google.com">juberti@google.com</a>&gt;<br>
&gt; &gt; wrote:<br>
&gt; &gt;<br>
&gt; &gt; This is described in rtp-usage, and the need to limit bitrate is<=
br>
&gt; &gt; explicitly called out:<br>
&gt; &gt;<br>
&gt; &gt; 7.2.=C2=A0 Congestion Control Interoperability and Legacy Systems=
<br>
&gt; &gt;<br>
&gt; &gt;=C2=A0 =C2=A0 There are legacy RTP implementations that do not imp=
lement RTCP,<br>
&gt; &gt; and hence do not provide any congestion feedback.=C2=A0 Congestio=
n<br>
&gt; &gt; control cannot be performed with these end-points.=C2=A0 WebRTC<b=
r>
&gt; &gt; Endpoints that need to interwork with such end-points MUST limit<=
br>
&gt; &gt; their transmission to a low rate, equivalent to a VoIP call using=
 a<br>
&gt; &gt; low bandwidth codec, that is unlikely to cause any significant<br=
>
&gt; &gt; congestion.<br>
&gt; &gt;<br>
&gt; &gt; Note however that this is incompatible with this section that<br>
&gt; &gt; mandates use of circuit breakers:<br>
&gt; &gt;<br>
&gt; &gt; 7.1.=C2=A0 Boundary Conditions and Circuit Breakers<br>
&gt; &gt;<br>
&gt; &gt;=C2=A0 =C2=A0 WebRTC Endpoints MUST implement the RTP circuit brea=
ker algorithm<br>
&gt; &gt;=C2=A0 =C2=A0 that is described in [I-D.ietf-avtcore-rtp-circuit-b=
reakers].<br>
&gt; &gt; The RTP circuit breaker is designed to enable applications to<br>
&gt; &gt; recognise and react to situations of extreme network congestion.<=
br>
&gt; &gt; However, since the RTP circuit breaker might not be triggered unt=
il<br>
&gt; &gt; congestion becomes extreme, it cannot be considered a substitute<=
br>
&gt; &gt; for congestion control, and applications MUST also implement<br>
&gt; &gt; congestion control to allow them to adapt to changes in network<b=
r>
&gt; &gt; capacity.=C2=A0 Any future RTP congestion control algorithms are<=
br>
&gt; &gt; expected to operate within the envelope allowed by the circuit<br=
>
&gt; &gt; breaker.<br>
&gt; &gt;<br>
&gt; &gt; I think we need to clarify exactly how WebRTC endpoints need to<b=
r>
&gt; &gt; deal with legacy gear that doesn&#39;t send RTCP (which might jus=
t be<br>
&gt; &gt; to say &quot;no&quot;).<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; rtcweb mailing list<br>
&gt; &gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; &gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=
=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
</div></div></blockquote></div><br></div>

--001a1140a146d51e630512324abd--


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From: Cullen Jennings <fluffy@cisco.com>
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Date: Thu, 26 Mar 2015 10:20:09 -0500
To: rtcweb@ietf.org
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Subject: [rtcweb] draft-schwartz-rtcweb-return
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I'd like to point out that the combination of =
ietf-tram-turn-server-discovery and draft-schwartz-rtcweb-return allow =
any network you are connected to more or less MITM your media and do =
things like rate limit it, generate analytics on who you are talking to, =
force your traffic through an intermediary that is in a  different legal =
jurisdiction and so on.=20

They are also not clear on how the browser gets the credentials to use =
the discovered TURN server. This seems like a major lacking before we =
can significantly discuss this.=20

As we have seen from the google proxy deployments, enough revenue can be =
generated from this relaying info to pay for the relay. I'm not keen on =
that happening automatically with no user consent or awareness.=20

But I don't get how this will work for enterprise deployments - It's =
just very unclear how the JS would end with the appropriate set of TURN =
servers to use.=20




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From: Jonathan Lennox <jonathan@vidyo.com>
To: Matthew Kaufman <matthew@matthew.at>
Thread-Topic: [rtcweb] Endpoints that don't support RTCP
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Subject: Re: [rtcweb] Endpoints that don't support RTCP
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Your legacy gateway can know definitively (from SDP) whether the peer it=92=
s talking to will do ICE or DTLS-SRTP, but unfortunately devices that don=
=92t do RTCP also typically don=92t bother to indicate this fact in any way=
.

So assuming (as I think we should) that rtp-usage mandates RTCP support, a =
legacy gateway will need to figure out dynamically whether it can count on =
receiving RTCP from its media destination, or whether it needs to synthesiz=
e RTCP locally.  Perhaps some text can be written for the -gateways draft t=
hat can give some guidance on how a gateway can make this decision.

On Mar 26, 2015, at 9:23 AM, Matthew Kaufman <matthew@matthew.at<mailto:mat=
thew@matthew.at>> wrote:

These are the legacy endpoints that don't do RTCP but do have ICE and DTLS-=
SRTP?

Matthew Kaufman

(Sent from my iPhone)

On Mar 25, 2015, at 10:24 PM, Justin Uberti <juberti@google.com<mailto:jube=
rti@google.com>> wrote:

This is described in rtp-usage, and the need to limit bitrate is explicitly=
 called out:


7.2<https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-22#section-7.2>=
.  Congestion Control Interoperability and Legacy Systems


   There are legacy RTP implementations that do not implement RTCP, and
   hence do not provide any congestion feedback.  Congestion control
   cannot be performed with these end-points.  WebRTC Endpoints that
   need to interwork with such end-points MUST limit their transmission
   to a low rate, equivalent to a VoIP call using a low bandwidth codec,
   that is unlikely to cause any significant congestion.


Note however that this is incompatible with this section that mandates use =
of circuit breakers:


7.1<https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-22#section-7.1>=
.  Boundary Conditions and Circuit Breakers


   WebRTC Endpoints MUST implement the RTP circuit breaker algorithm
   that is described in [I-D.ietf-avtcore-rtp-circuit-breakers<https://tool=
s.ietf.org/html/draft-ietf-rtcweb-rtp-usage-22#ref-I-D.ietf-avtcore-rtp-cir=
cuit-breakers>].  The
   RTP circuit breaker is designed to enable applications to recognise
   and react to situations of extreme network congestion.  However,
   since the RTP circuit breaker might not be triggered until congestion
   becomes extreme, it cannot be considered a substitute for congestion
   control, and applications MUST also implement congestion control to
   allow them to adapt to changes in network capacity.  Any future RTP
   congestion control algorithms are expected to operate within the
   envelope allowed by the circuit breaker.


I think we need to clarify exactly how WebRTC endpoints need to deal with l=
egacy gear that doesn't send RTCP (which might just be to say "no").






_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb


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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3DWindows-1=
252">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space;">
<div>Your legacy gateway can know definitively (from SDP) whether the peer =
it=92s talking to will do ICE or DTLS-SRTP, but unfortunately devices that =
don=92t do RTCP also typically don=92t bother to indicate this fact in any =
way.</div>
<div><br>
</div>
<div>So assuming (as I think we should) that rtp-usage mandates RTCP suppor=
t, a legacy gateway will need to figure out dynamically whether it can coun=
t on receiving RTCP from its media destination, or whether it needs to synt=
hesize RTCP locally. &nbsp;Perhaps some
 text can be written for the -gateways draft that can give some guidance on=
 how a gateway can make this decision.</div>
<br>
<div>
<div>On Mar 26, 2015, at 9:23 AM, Matthew Kaufman &lt;<a href=3D"mailto:mat=
thew@matthew.at">matthew@matthew.at</a>&gt; wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">
<div dir=3D"auto">
<div>These are the legacy endpoints that don't do RTCP but do have ICE and =
DTLS-SRTP?<br>
<br>
Matthew Kaufman
<div><br>
(Sent from my iPhone)</div>
</div>
<div><br>
On Mar 25, 2015, at 10:24 PM, Justin Uberti &lt;<a href=3D"mailto:juberti@g=
oogle.com">juberti@google.com</a>&gt; wrote:<br>
<br>
</div>
<blockquote type=3D"cite">
<div dir=3D"ltr">This is described in rtp-usage, and the need to limit bitr=
ate is explicitly called out:
<div><br>
</div>
<div>
<pre class=3D"" style=3D"font-size: 1em; margin-top: 0px; margin-bottom: 0p=
x;"><span class=3D"" style=3D"line-height:0pt;display:inline;font-size:1em;=
font-weight:bold"><h3 style=3D"line-height:0pt;display:inline;font-size:1em=
"><a class=3D"" name=3D"section-7.2" href=3D"https://tools.ietf.org/html/dr=
aft-ietf-rtcweb-rtp-usage-22#section-7.2" style=3D"text-decoration: none;">=
7.2</a>.  Congestion Control Interoperability and Legacy Systems</h3></span=
>

   There are legacy RTP implementations that do not implement RTCP, and
   hence do not provide any congestion feedback.  Congestion control
   cannot be performed with these end-points.  WebRTC Endpoints that
   need to interwork with such end-points MUST limit their transmission
   to a low rate, equivalent to a VoIP call using a low bandwidth codec,
   that is unlikely to cause any significant congestion.</pre>
<pre class=3D"" style=3D"font-size: 1em; margin-top: 0px; margin-bottom: 0p=
x;"><br></pre>
<pre class=3D"" style=3D"font-size: 1em; margin-top: 0px; margin-bottom: 0p=
x;"><span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-si=
ze:small;white-space:normal">Note however that this is incompatible with th=
is section that mandates use of circuit breakers:</span><br></pre>
<pre class=3D"" style=3D"font-size: 1em; margin-top: 0px; margin-bottom: 0p=
x;"><span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-si=
ze:small;white-space:normal"><br></span></pre>
<pre class=3D"" style=3D"margin-top:0px;margin-bottom:0px"><pre class=3D"" =
style=3D"font-size: 1em; margin-top: 0px; margin-bottom: 0px;"><span class=
=3D"" style=3D"line-height:0pt;display:inline;font-size:1em;font-weight:bol=
d"><h3 style=3D"line-height:0pt;display:inline;font-size:1em"><a class=3D""=
 name=3D"section-7.1" href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb=
-rtp-usage-22#section-7.1" style=3D"text-decoration: none;">7.1</a>.  Bound=
ary Conditions and Circuit Breakers</h3></span>

   WebRTC Endpoints MUST implement the RTP circuit breaker algorithm
   that is described in [<a href=3D"https://tools.ietf.org/html/draft-ietf-=
rtcweb-rtp-usage-22#ref-I-D.ietf-avtcore-rtp-circuit-breakers">I-D.ietf-avt=
core-rtp-circuit-breakers</a>].  The
   RTP circuit breaker is designed to enable applications to recognise
   and react to situations of extreme network congestion.  However,
   since the RTP circuit breaker might not be triggered until congestion
   becomes extreme, it cannot be considered a substitute for congestion
   control, and applications MUST also implement congestion control to
   allow them to adapt to changes in network capacity.  Any future RTP
   congestion control algorithms are expected to operate within the
   envelope allowed by the circuit breaker.</pre><pre class=3D"" style=3D"f=
ont-size: 1em; margin-top: 0px; margin-bottom: 0px;"><br></pre><pre class=
=3D"" style=3D"font-size: 1em; margin-top: 0px; margin-bottom: 0px;"><span =
style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:small;w=
hite-space:normal">I think we need to clarify exactly how WebRTC endpoints =
need to deal with legacy gear that doesn't send RTCP (which might just be t=
o say &quot;no&quot;).</span></pre><div style=3D"font-size: 1em;"><span sty=
le=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:small;whit=
e-space:normal"><br></span></div><pre class=3D"" style=3D"font-size: 1em; m=
argin-top: 0px; margin-bottom: 0px;"><br></pre><pre class=3D"" style=3D"fon=
t-size: 1em; margin-top: 0px; margin-bottom: 0px;"><br></pre></pre>
<pre class=3D"" style=3D"font-size: 1em; margin-top: 0px; margin-bottom: 0p=
x;"><br></pre>
<pre class=3D"" style=3D"font-size: 1em; margin-top: 0px; margin-bottom: 0p=
x;"><br></pre>
</div>
</div>
</blockquote>
<blockquote type=3D"cite"><span>___________________________________________=
____</span><br>
<span>rtcweb mailing list</span><br>
<span><a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br>
<span><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.=
ietf.org/mailman/listinfo/rtcweb</a></span><br>
</blockquote>
</div>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
https://www.ietf.org/mailman/listinfo/rtcweb<br>
</blockquote>
</div>
<br>
</body>
</html>

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On Thu, Mar 26, 2015 at 11:20 AM, Cullen Jennings <fluffy@cisco.com> wrote:

> I'd like to point out that the combination of
> ietf-tram-turn-server-discovery and draft-schwartz-rtcweb-return allow any
> network you are connected to more or less MITM your media and do things
> like rate limit it, generate analytics on who you are talking to, force
> your traffic through an intermediary that is in a  different legal
> jurisdiction and so on.
>

This is true on any network, right?  If you can do NAT, you can do all of
those things.


> They are also not clear on how the browser gets the credentials to use the
> discovered TURN server. This seems like a major lacking before we can
> significantly discuss this.
>

I agree that turn-server-discovery should address this.


> As we have seen from the google proxy deployments, enough revenue can be
> generated from this relaying info to pay for the relay. I'm not keen on
> that happening automatically with no user consent or awareness.
>

It's already happening automatically, since 1999, thanks to WPAD, which
allows the network to automatically configure proxies for all web traffic
with no user intervention on all major browsers.

But I don't get how this will work for enterprise deployments - It's just
> very unclear how the JS would end with the appropriate set of TURN servers
> to use.
>

It would be helpful if you could point to text that you think is unclear.


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

--001a1147d3ea894c4d051232fc6e
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On T=
hu, Mar 26, 2015 at 11:20 AM, Cullen Jennings <span dir=3D"ltr">&lt;<a href=
=3D"mailto:fluffy@cisco.com" target=3D"_blank">fluffy@cisco.com</a>&gt;</sp=
an> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px=
 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left=
-style:solid;padding-left:1ex">I&#39;d like to point out that the combinati=
on of ietf-tram-turn-server-discovery and draft-schwartz-rtcweb-return allo=
w any network you are connected to more or less MITM your media and do thin=
gs like rate limit it, generate analytics on who you are talking to, force =
your traffic through an intermediary that is in a=C2=A0 different legal jur=
isdiction and so on.<br></blockquote><div><br></div><div>This is true on an=
y network, right?=C2=A0 If you can do NAT, you can do all of those things.<=
/div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px=
 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);bor=
der-left-style:solid;padding-left:1ex">
They are also not clear on how the browser gets the credentials to use the =
discovered TURN server. This seems like a major lacking before we can signi=
ficantly discuss this.<br></blockquote><div><br></div><div>I agree that tur=
n-server-discovery should address this.</div><div>=C2=A0</div><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1p=
x;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1=
ex">
As we have seen from the google proxy deployments, enough revenue can be ge=
nerated from this relaying info to pay for the relay. I&#39;m not keen on t=
hat happening automatically with no user consent or awareness.<br></blockqu=
ote><div><br></div><div>It&#39;s already happening automatically, since 199=
9, thanks to WPAD, which allows the network to automatically configure prox=
ies for all web traffic with no user intervention on all major browsers.</d=
iv><div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px=
 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-=
left-style:solid;padding-left:1ex">
But I don&#39;t get how this will work for enterprise deployments - It&#39;=
s just very unclear how the JS would end with the appropriate set of TURN s=
ervers to use.<br></blockquote><div><br></div><div>It would be helpful if y=
ou could point to text that you think is unclear.</div><div>=C2=A0</div><bl=
ockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-lef=
t-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padd=
ing-left:1ex">
<br>
<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div></div>

--001a1147d3ea894c4d051232fc6e--


From nobody Thu Mar 26 09:24:12 2015
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Date: Thu, 26 Mar 2015 11:24:08 -0500
Message-ID: <CAKhHsXGgNasqyjFG_gd2LQZry2VrOw_ktk8kFkxF=+pXZf-9nw@mail.gmail.com>
From: Alan Johnston <alan.b.johnston@gmail.com>
To: Cullen Jennings <fluffy@cisco.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] draft-schwartz-rtcweb-return
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--20cf3033449d0024ef0512337226
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Cullen,

I think the answer is obvious about credentials.  If an enterprise is
forcing users to use an enterprise TURN server, then it is up to the
enterprise to manage the credentials.  There is nothing new here.

As for your MitM comments, that isn't true unless you are saying that
DTLS-SRTP does not have any protection against MitM attacks.

- Alan -

On Thu, Mar 26, 2015 at 10:20 AM, Cullen Jennings <fluffy@cisco.com> wrote:

> I'd like to point out that the combination of
> ietf-tram-turn-server-discovery and draft-schwartz-rtcweb-return allow any
> network you are connected to more or less MITM your media and do things
> like rate limit it, generate analytics on who you are talking to, force
> your traffic through an intermediary that is in a  different legal
> jurisdiction and so on.
>
> They are also not clear on how the browser gets the credentials to use the
> discovered TURN server. This seems like a major lacking before we can
> significantly discuss this.
>
> As we have seen from the google proxy deployments, enough revenue can be
> generated from this relaying info to pay for the relay. I'm not keen on
> that happening automatically with no user consent or awareness.
>
> But I don't get how this will work for enterprise deployments - It's just
> very unclear how the JS would end with the appropriate set of TURN servers
> to use.
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--20cf3033449d0024ef0512337226
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<div dir=3D"ltr">Cullen,<div><br></div><div>I think the answer is obvious a=
bout credentials.=C2=A0 If an enterprise is forcing users to use an enterpr=
ise TURN server, then it is up to the enterprise to manage the credentials.=
=C2=A0 There is nothing new here.</div><div><br></div><div>As for your MitM=
 comments, that isn&#39;t true unless you are saying that DTLS-SRTP does no=
t have any protection against MitM attacks.</div><div><br></div><div>- Alan=
 -<br><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Thu, Mar=
 26, 2015 at 10:20 AM, Cullen Jennings <span dir=3D"ltr">&lt;<a href=3D"mai=
lto:fluffy@cisco.com" target=3D"_blank">fluffy@cisco.com</a>&gt;</span> wro=
te:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-=
left:1px #ccc solid;padding-left:1ex">I&#39;d like to point out that the co=
mbination of ietf-tram-turn-server-discovery and draft-schwartz-rtcweb-retu=
rn allow any network you are connected to more or less MITM your media and =
do things like rate limit it, generate analytics on who you are talking to,=
 force your traffic through an intermediary that is in a=C2=A0 different le=
gal jurisdiction and so on.<br>
<br>
They are also not clear on how the browser gets the credentials to use the =
discovered TURN server. This seems like a major lacking before we can signi=
ficantly discuss this.<br>
<br>
As we have seen from the google proxy deployments, enough revenue can be ge=
nerated from this relaying info to pay for the relay. I&#39;m not keen on t=
hat happening automatically with no user consent or awareness.<br>
<br>
But I don&#39;t get how this will work for enterprise deployments - It&#39;=
s just very unclear how the JS would end with the appropriate set of TURN s=
ervers to use.<br>
<br>
<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div></div></div>

--20cf3033449d0024ef0512337226--


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From: Justin Uberti <juberti@google.com>
Date: Thu, 26 Mar 2015 11:26:59 -0500
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To: Benjamin Schwartz <bemasc@webrtc.org>
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Cc: Cullen Jennings <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] draft-schwartz-rtcweb-return
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--90e6ba614aa260ede90512337d72
Content-Type: text/plain; charset=UTF-8

On Thu, Mar 26, 2015 at 10:51 AM, Benjamin Schwartz <bemasc@webrtc.org>
wrote:

> On Thu, Mar 26, 2015 at 11:20 AM, Cullen Jennings <fluffy@cisco.com>
> wrote:
>
>> I'd like to point out that the combination of
>> ietf-tram-turn-server-discovery and draft-schwartz-rtcweb-return allow any
>> network you are connected to more or less MITM your media and do things
>> like rate limit it, generate analytics on who you are talking to, force
>> your traffic through an intermediary that is in a  different legal
>> jurisdiction and so on.
>>
>
> This is true on any network, right?  If you can do NAT, you can do all of
> those things.
>
>
>> They are also not clear on how the browser gets the credentials to use
>> the discovered TURN server. This seems like a major lacking before we can
>> significantly discuss this.
>>
>
> I agree that turn-server-discovery should address this.
>

This is an extant problem for HTTP proxies, not unique to this solution
(and hasn't been a blocking issue there)

>
>
>> As we have seen from the google proxy deployments, enough revenue can be
>> generated from this relaying info to pay for the relay. I'm not keen on
>> that happening automatically with no user consent or awareness.
>
>
> It's already happening automatically, since 1999, thanks to WPAD, which
> allows the network to automatically configure proxies for all web traffic
> with no user intervention on all major browsers.
>

I don't buy the revenue argument in this context. There is no cleartext
data to analyze.

>
> But I don't get how this will work for enterprise deployments - It's just
>> very unclear how the JS would end with the appropriate set of TURN servers
>> to use.
>>
>
> It would be helpful if you could point to text that you think is unclear.
>

JS doesn't get these TURN servers, the browser does and applies them the
same way it does HTTPS/SOCKS proxies.

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

--90e6ba614aa260ede90512337d72
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Thu, Mar 26, 2015 at 10:51 AM, Benjamin Schwartz <span dir=3D"ltr">&=
lt;<a href=3D"mailto:bemasc@webrtc.org" target=3D"_blank">bemasc@webrtc.org=
</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin=
:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><=
div class=3D"gmail_extra"><div class=3D"gmail_quote"><span class=3D"">On Th=
u, Mar 26, 2015 at 11:20 AM, Cullen Jennings <span dir=3D"ltr">&lt;<a href=
=3D"mailto:fluffy@cisco.com" target=3D"_blank">fluffy@cisco.com</a>&gt;</sp=
an> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px=
 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left=
-style:solid;padding-left:1ex">I&#39;d like to point out that the combinati=
on of ietf-tram-turn-server-discovery and draft-schwartz-rtcweb-return allo=
w any network you are connected to more or less MITM your media and do thin=
gs like rate limit it, generate analytics on who you are talking to, force =
your traffic through an intermediary that is in a=C2=A0 different legal jur=
isdiction and so on.<br></blockquote><div><br></div></span><div>This is tru=
e on any network, right?=C2=A0 If you can do NAT, you can do all of those t=
hings.</div><span class=3D""><div>=C2=A0</div><blockquote class=3D"gmail_qu=
ote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-co=
lor:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
They are also not clear on how the browser gets the credentials to use the =
discovered TURN server. This seems like a major lacking before we can signi=
ficantly discuss this.<br></blockquote><div><br></div></span><div>I agree t=
hat turn-server-discovery should address this.</div></div></div></div></blo=
ckquote><div><br></div><div>This is an extant problem for HTTP proxies, not=
 unique to this solution (and hasn&#39;t been a blocking issue there)</div>=
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra">=
<div class=3D"gmail_quote"><span class=3D""><div>=C2=A0</div><blockquote cl=
ass=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px=
;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1e=
x">
As we have seen from the google proxy deployments, enough revenue can be ge=
nerated from this relaying info to pay for the relay. I&#39;m not keen on t=
hat happening automatically with no user consent or awareness.=C2=A0</block=
quote></span></div></div></div></blockquote><blockquote class=3D"gmail_quot=
e" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">=
<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><spa=
n class=3D""><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px =
0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-=
style:solid;padding-left:1ex"></blockquote><div><br></div></span><div>It&#3=
9;s already happening automatically, since 1999, thanks to WPAD, which allo=
ws the network to automatically configure proxies for all web traffic with =
no user intervention on all major browsers.</div></div></div></div></blockq=
uote><div><br></div><div>I don&#39;t buy the revenue argument in this conte=
xt. There is no cleartext data to analyze.=C2=A0</div><blockquote class=3D"=
gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-=
left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_q=
uote"><span class=3D""><div><br></div><blockquote class=3D"gmail_quote" sty=
le=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(=
204,204,204);border-left-style:solid;padding-left:1ex">
But I don&#39;t get how this will work for enterprise deployments - It&#39;=
s just very unclear how the JS would end with the appropriate set of TURN s=
ervers to use.<br></blockquote><div><br></div></span><div>It would be helpf=
ul if you could point to text that you think is unclear.</div></div></div><=
/div></blockquote><div><br></div><div>JS doesn&#39;t get these TURN servers=
, the browser does and applies them the same way it does HTTPS/SOCKS proxie=
s.=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;=
border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D=
"gmail_extra"><div class=3D"gmail_quote"><span class=3D""><div>=C2=A0</div>=
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<br>
<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></span></div><br></div></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

--90e6ba614aa260ede90512337d72--


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From: Simon Perreault <sperreault@jive.com>
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Subject: Re: [rtcweb] Endpoints that don't support RTCP
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Le 2015-03-26 09:41, Justin Uberti a écrit :
> I think mandating use of RTCP is the correct solution.

That's an easy one. +1

Simon


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From: "Prashanth Patil (praspati)" <praspati@cisco.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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On 3/26/15, 8:20 AM, "Cullen Jennings (fluffy)" <fluffy@cisco.com> wrote:

>I'd like to point out that the combination of
>ietf-tram-turn-server-discovery and draft-schwartz-rtcweb-return allow
>any network you are connected to more or less MITM your media and do
>things like rate limit it, generate analytics on who you are talking to,
>force your traffic through an intermediary that is in a  different legal
>jurisdiction and so on.
>
>They are also not clear on how the browser gets the credentials to use
>the discovered TURN server. This seems like a major lacking before we can
>significantly discuss this.

The client should use the REALM attribute (returned in the 401 error
response) to determine the credentials to use for authentication.


-Prashanth

>=20
>
>As we have seen from the google proxy deployments, enough revenue can be
>generated from this relaying info to pay for the relay. I'm not keen on
>that happening automatically with no user consent or awareness.
>
>But I don't get how this will work for enterprise deployments - It's just
>very unclear how the JS would end with the appropriate set of TURN
>servers to use.=20
>
>
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb


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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Jonathan Lennox <jonathan@vidyo.com>, Matthew Kaufman <matthew@matthew.at>
Thread-Topic: [rtcweb] Endpoints that don't support RTCP
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--_000_D139D9B94ABB5mzanatyciscocom_
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I agree this should be handled in the gateways draft. Other drafts that onl=
y deal with WebRTC endpoints should just expect RTCP as a given just like t=
hey expect ICE and DTLS.

For the WebRTC endpoint side of the gateway, it must obey the RTP circuit b=
reaker in both send and receive directions, and synthesize RTCP on behalf o=
f the legacy side when necessary to avoid tripping the breaker in the recei=
ve direction.

Mo

On 3/26/15, 11:25 AM, Jonathan Lennox <jonathan@vidyo.com<mailto:jonathan@v=
idyo.com>> wrote:

Your legacy gateway can know definitively (from SDP) whether the peer it=92=
s talking to will do ICE or DTLS-SRTP, but unfortunately devices that don=
=92t do RTCP also typically don=92t bother to indicate this fact in any way=
.

So assuming (as I think we should) that rtp-usage mandates RTCP support, a =
legacy gateway will need to figure out dynamically whether it can count on =
receiving RTCP from its media destination, or whether it needs to synthesiz=
e RTCP locally.  Perhaps some text can be written for the -gateways draft t=
hat can give some guidance on how a gateway can make this decision.

On Mar 26, 2015, at 9:23 AM, Matthew Kaufman <matthew@matthew.at<mailto:mat=
thew@matthew.at>> wrote:

These are the legacy endpoints that don't do RTCP but do have ICE and DTLS-=
SRTP?

Matthew Kaufman

(Sent from my iPhone)

On Mar 25, 2015, at 10:24 PM, Justin Uberti <juberti@google.com<mailto:jube=
rti@google.com>> wrote:

This is described in rtp-usage, and the need to limit bitrate is explicitly=
 called out:


7.2<https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-22#section-7.2>=
.  Congestion Control Interoperability and Legacy Systems


   There are legacy RTP implementations that do not implement RTCP, and
   hence do not provide any congestion feedback.  Congestion control
   cannot be performed with these end-points.  WebRTC Endpoints that
   need to interwork with such end-points MUST limit their transmission
   to a low rate, equivalent to a VoIP call using a low bandwidth codec,
   that is unlikely to cause any significant congestion.


Note however that this is incompatible with this section that mandates use =
of circuit breakers:


7.1<https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-22#section-7.1>=
.  Boundary Conditions and Circuit Breakers


   WebRTC Endpoints MUST implement the RTP circuit breaker algorithm
   that is described in [I-D.ietf-avtcore-rtp-circuit-breakers<https://tool=
s.ietf.org/html/draft-ietf-rtcweb-rtp-usage-22#ref-I-D.ietf-avtcore-rtp-cir=
cuit-breakers>].  The
   RTP circuit breaker is designed to enable applications to recognise
   and react to situations of extreme network congestion.  However,
   since the RTP circuit breaker might not be triggered until congestion
   becomes extreme, it cannot be considered a substitute for congestion
   control, and applications MUST also implement congestion control to
   allow them to adapt to changes in network capacity.  Any future RTP
   congestion control algorithms are expected to operate within the
   envelope allowed by the circuit breaker.


I think we need to clarify exactly how WebRTC endpoints need to deal with l=
egacy gear that doesn't send RTCP (which might just be to say "no").






_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb


--_000_D139D9B94ABB5mzanatyciscocom_
Content-Type: text/html; charset="Windows-1252"
Content-ID: <43B61A3B51F61E45BD2EF9EE824F9F67@emea.cisco.com>
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3DWindows-1=
252">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; color: rgb(0, 0, 0); font-size: 12px; font-fami=
ly: Arial, sans-serif;">
<div>I agree this should be handled in the gateways draft. Other drafts tha=
t only deal with WebRTC endpoints should just expect RTCP as a given just l=
ike they expect ICE and DTLS.</div>
<div><br>
</div>
<div>For the WebRTC endpoint side of the gateway, it must obey the RTP circ=
uit breaker in both send and receive directions, and synthesize RTCP on beh=
alf of the legacy side when necessary to avoid tripping the breaker in the =
receive direction.</div>
<div><br>
</div>
<div>Mo</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div>
<div>On 3/26/15, 11:25 AM, Jonathan Lennox &lt;<a href=3D"mailto:jonathan@v=
idyo.com">jonathan@vidyo.com</a>&gt; wrote:</div>
</div>
<div><br>
</div>
<div>
<div style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line=
-break: after-white-space;">
<div>Your legacy gateway can know definitively (from SDP) whether the peer =
it=92s talking to will do ICE or DTLS-SRTP, but unfortunately devices that =
don=92t do RTCP also typically don=92t bother to indicate this fact in any =
way.</div>
<div><br>
</div>
<div>So assuming (as I think we should) that rtp-usage mandates RTCP suppor=
t, a legacy gateway will need to figure out dynamically whether it can coun=
t on receiving RTCP from its media destination, or whether it needs to synt=
hesize RTCP locally. &nbsp;Perhaps some
 text can be written for the -gateways draft that can give some guidance on=
 how a gateway can make this decision.</div>
<br>
<div>
<div>On Mar 26, 2015, at 9:23 AM, Matthew Kaufman &lt;<a href=3D"mailto:mat=
thew@matthew.at">matthew@matthew.at</a>&gt; wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">
<div dir=3D"auto">
<div>These are the legacy endpoints that don't do RTCP but do have ICE and =
DTLS-SRTP?<br>
<br>
Matthew Kaufman
<div><br>
(Sent from my iPhone)</div>
</div>
<div><br>
On Mar 25, 2015, at 10:24 PM, Justin Uberti &lt;<a href=3D"mailto:juberti@g=
oogle.com">juberti@google.com</a>&gt; wrote:<br>
<br>
</div>
<blockquote type=3D"cite">
<div dir=3D"ltr">This is described in rtp-usage, and the need to limit bitr=
ate is explicitly called out:
<div><br>
</div>
<div>
<pre class=3D"" style=3D"font-size: 1em; margin-top: 0px; margin-bottom: 0p=
x;"><span class=3D"" style=3D"line-height:0pt;display:inline;font-size:1em;=
font-weight:bold"><h3 style=3D"line-height:0pt;display:inline;font-size:1em=
"><a class=3D"" name=3D"section-7.2" href=3D"https://tools.ietf.org/html/dr=
aft-ietf-rtcweb-rtp-usage-22#section-7.2" style=3D"text-decoration: none;">=
7.2</a>.  Congestion Control Interoperability and Legacy Systems</h3></span=
>

   There are legacy RTP implementations that do not implement RTCP, and
   hence do not provide any congestion feedback.  Congestion control
   cannot be performed with these end-points.  WebRTC Endpoints that
   need to interwork with such end-points MUST limit their transmission
   to a low rate, equivalent to a VoIP call using a low bandwidth codec,
   that is unlikely to cause any significant congestion.</pre>
<pre class=3D"" style=3D"font-size: 1em; margin-top: 0px; margin-bottom: 0p=
x;"><br></pre>
<pre class=3D"" style=3D"font-size: 1em; margin-top: 0px; margin-bottom: 0p=
x;"><span style=3D"color: rgb(34, 34, 34); font-family: arial, sans-serif; =
font-size: small; white-space: normal;">Note however that this is incompati=
ble with this section that mandates use of circuit breakers:</span><br></pr=
e>
<pre class=3D"" style=3D"font-size: 1em; margin-top: 0px; margin-bottom: 0p=
x;"><span style=3D"color: rgb(34, 34, 34); font-family: arial, sans-serif; =
font-size: small; white-space: normal;"><br></span></pre>
<pre class=3D"" style=3D"margin-top:0px;margin-bottom:0px"><pre class=3D"" =
style=3D"font-size: 1em; margin-top: 0px; margin-bottom: 0px;"><span class=
=3D"" style=3D"line-height:0pt;display:inline;font-size:1em;font-weight:bol=
d"><h3 style=3D"line-height:0pt;display:inline;font-size:1em"><a class=3D""=
 name=3D"section-7.1" href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb=
-rtp-usage-22#section-7.1" style=3D"text-decoration: none;">7.1</a>.  Bound=
ary Conditions and Circuit Breakers</h3></span>

   WebRTC Endpoints MUST implement the RTP circuit breaker algorithm
   that is described in [<a href=3D"https://tools.ietf.org/html/draft-ietf-=
rtcweb-rtp-usage-22#ref-I-D.ietf-avtcore-rtp-circuit-breakers">I-D.ietf-avt=
core-rtp-circuit-breakers</a>].  The
   RTP circuit breaker is designed to enable applications to recognise
   and react to situations of extreme network congestion.  However,
   since the RTP circuit breaker might not be triggered until congestion
   becomes extreme, it cannot be considered a substitute for congestion
   control, and applications MUST also implement congestion control to
   allow them to adapt to changes in network capacity.  Any future RTP
   congestion control algorithms are expected to operate within the
   envelope allowed by the circuit breaker.</pre><pre class=3D"" style=3D"f=
ont-size: 1em; margin-top: 0px; margin-bottom: 0px;"><br></pre><pre class=
=3D"" style=3D"font-size: 1em; margin-top: 0px; margin-bottom: 0px;"><span =
style=3D"color: rgb(34, 34, 34); font-family: arial, sans-serif; font-size:=
 small; white-space: normal;">I think we need to clarify exactly how WebRTC=
 endpoints need to deal with legacy gear that doesn't send RTCP (which migh=
t just be to say &quot;no&quot;).</span></pre><div style=3D"font-size: 1em;=
"><span style=3D"color: rgb(34, 34, 34); font-family: arial, sans-serif; fo=
nt-size: small; white-space: normal;"><br></span></div><pre class=3D"" styl=
e=3D"font-size: 1em; margin-top: 0px; margin-bottom: 0px;"><br></pre><pre c=
lass=3D"" style=3D"font-size: 1em; margin-top: 0px; margin-bottom: 0px;"><b=
r></pre></pre>
<pre class=3D"" style=3D"font-size: 1em; margin-top: 0px; margin-bottom: 0p=
x;"><br></pre>
<pre class=3D"" style=3D"font-size: 1em; margin-top: 0px; margin-bottom: 0p=
x;"><br></pre>
</div>
</div>
</blockquote>
<blockquote type=3D"cite"><span>___________________________________________=
____</span><br>
<span>rtcweb mailing list</span><br>
<span><a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br>
<span><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.=
ietf.org/mailman/listinfo/rtcweb</a></span><br>
</blockquote>
</div>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.o=
rg/mailman/listinfo/rtcweb</a><br>
</blockquote>
</div>
<br>
</div>
</div>
</span>
</body>
</html>

--_000_D139D9B94ABB5mzanatyciscocom_--


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From: "Hutton, Andrew" <andrew.hutton@unify.com>
To: Cullen Jennings <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] draft-schwartz-rtcweb-return
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On: 26 March 2015 10:20 Cullen Jennings Wrote:
>=20
> I'd like to point out that the combination of ietf-tram-turn-server-
> discovery and draft-schwartz-rtcweb-return allow any network you are
> connected to more or less MITM your media and do things like rate limit
> it, generate analytics on who you are talking to, force your traffic
> through an intermediary that is in a  different legal jurisdiction and
> so on.
>=20

I don't see any new issues that are caused by this draft it is just suggest=
ing that the same is done for WebRTC Media as is already done for HTTP(S) b=
ut using a TURN proxy which is optimized to handle media.=20

We set the requirement in RFC 7478 (Requirement F20) requires that WebRTC h=
andles this scenario and this solution is the only one we have on the table=
 for this and provides a solution to the privacy concerns (IP Address leaks=
) with regard to deploying a network specific proxy.


>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Hi,

I have reviewed draft-ietf-rtcweb-jsep-09 and have some comments.

1. Section 3.4.4:

   JSEP applications typically inform the browser to begin ICE gathering
   via the information supplied to setLocalDescription, as this is where
   the app specifies the number of media streams, and thereby ICE
   components, for which to gather candidates.  However, to accelerate
   cases where the application knows the number of ICE components to use
   ahead of time, it may ask the browser to gather a pool of potential
   ICE candidates to help ensure rapid media setup.

I wonder why this is not using the number of added media stream tracks
to the peerconnection as an indication to how big draw on the pool there
will be?

2. Section 4.1.1:

This section defines: balanced, max-compat and max-bundle. The Transport
draft defines in Section 4.1 the following:

   A sending implementation MUST be able to support the following
   configurations:

   o  multiplex all media and data on a single 5-tuple (fully bundled)

   o  send each media stream on its own 5-tuple and data on its own
      5-tuple (fully unbundled)

I am missing the policy related to the last of these bullets (fully
unbundled).

3. Section 4.1.2:

The
   exact handling of subsequent offer generation is detailed in
   Section 5.2.2. below.

I suggest that you remove the below. Section 5.2.2 is quite far away.

4. Section 5.1.1:

I think you can remove R-16 for now. If people are missing
specifications, then lets add them at the point when we realize that we
are missing it.

5. Section 5.1.3:

   o  The profile in any "m=" line in any answer MUST exactly match the
      profile provided in the offer.

I think I understand the reasons for this and works fine for the remote
peer getting a offer, and then producing an answer. This works less well
if checked on the offerer. Because, then what was offered, is not the
same that is received in the answer. I think this must be clarified to
apply to the one generating an answer and being a JSEP endpoint.

6. Section 5.2.1

   o  An "a=setup" line, as specified in [RFC4145], Section 4, and
      clarified for use in DTLS-SRTP scenarios in [RFC5763], Section 5.
      The role value in the offer MUST be "actpass".

So this is correct if the setup is for the DTLS. But, what is one has
another protocol that uses a=setup? I think you should scope this rule
to anything that uses DTLS.

7. Section 5.2.1:

   o  An "a=ssrc" line, as specified in [RFC5576], Section 4.1,
      indicating the SSRC to be used for sending media, along with the
      mandatory "cname" source attribute, as specified in Section 6.1,
      indicating the CNAME for the source.  The CNAME must be generated
      in accordance with [RFC7022].  [OPEN ISSUE: How are CNAMEs
      specified for MSTs?  Are they randomly generated for each
      MediaStream?  If so, can two MediaStreams be synced?  See:
      https://github.com/rtcweb-wg/jsep/issues/4]

So, to my understanding all MSTs in the same MS MUST have the same
CNAME. All MST from one endpoint will have the same CNAME. But, for
multi-party centralized conferencing the endpoint will receive multiple
different MS. Each MS can have a different CNAME. So an offer or answer
from a MCU can contain multiple CNAMEs.

8. Section 5.2.1:

For simplicity, if both RTX and FEC
      are supported, the FEC SSRC MUST be the same as the RTX SSRC.

I wonder if this will actually work. The issue I see is that one have an
SSRC with only RTX or repair packets, a missing sequence number
indicates that you lost one packet of the type used. If you mix RTX and
FEC then a missing packet can be either. Thus, the right response to
this loss can't be determined. In addition if one have a mechanism that
requires concurrent sequence numbers then it will also not work. I am
also uncertain how this interact with mechanisms for determining when to
flush the FEC source block.

SSRCs aren't particular expensive, so I think you rather have the FEC
and RTX on separate SSRCs.

9. Section 5.2.1:
   Once all m= sections have been generated, a session-level "a=group"
   attribute MUST be added as specified in [RFC5888].  This attribute
   MUST have semantics "BUNDLE", and MUST include the mid identifiers of
   each m= section.  The effect of this is that the browser offers all
   m= sections as one BUNDLE group.  However, whether the m= sections
   are bundle-only or not depends on the BUNDLE policy.

This text is not supporting a policy of fully unbundle.

10. Section 5.2.2:

   o  If any MediaStreamTracks have been removed, either through the
      removeStream method or by removing them from an added MediaStream,
      their m= sections MUST be marked as recvonly by changing the value
      of the [RFC3264] directional attribute to "a=recvonly".

Should "removeStream" be "removeTrack" or should both be mentioned?

11. Section 5.2.2:
   The "a=group:BUNDLE" attribute MUST include the mid identifiers
   specified in the BUNDLE group in the most recent answer, minus any m=
   sections that have been marked as rejected, plus any newly added or
   re-enabled m= sections.  In other words, the BUNDLE attribute must
   contain all m= sections that were previously bundled, as long as they
   are still alive, as well as any new m= sections.

This appears to need to be reformulate to deal with sessions where
bundle has been rejected completely.

12. I am quite unclear how the a=candidate processing is intended to
work. The GenerateOffer can't fill it in as the gathering is defined to
happen when you do setLocal. So, are there empty a=candidates or are
they set by the setLocal.


13. Section 5.6.2:

  o  The "b=" line, if present, MUST be parsed as specified in
      [RFC4566], Section 5.8, and the bwtype and bandwidth values
      stored.

There can be multiple b= lines with different modifiers.

14. Section 6.

Proto needs to be on this list I think. If one are doing munching for
legacy systems.

15. Page 53:

The definition of telephone-event appears to miss a=fmtp attribute.

Cheers


Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
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FÃ¤rÃ¶gatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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Subject: [rtcweb] On transports and bundles
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If we factor out all of the W3C API conversations from what was said 
today, I think the only real point of contention turns on this 
statement; not because it's wrong, but because it's incomplete (and 
creates ambiguity as a result):

    A receiving implementation MUST be able to receive media and data in
    all these configurations.

If we're going to make this statement, then we need to also say 
something about receiving bundle configurations that are more advanced 
than the two that we must be able to send.

I would argue that anything short of saying that we MUST support any 
valid description as described by 
draft-ietf-mmusic-sdp-bundle-negotiation would be creating a profile of 
bundle, and I believe that would be harmful. To that end, I think we 
either say that implementations MUST be able to handle any bundle 
configuration that is valid per the bundle draft, or we omit the 
statement I cite above and leave it to bundle to make statements around 
what it means to implement the protocol it defines.

I slightly prefer the latter approach.

/a


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Subject: [rtcweb] RFC 5763 Section 6.7.1 (was DTLS, DTLS-SRTP, and 5-tuples)
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RFC 5763 Section 6.7.1 explains much of what we've been talking about on
this thread.
6.7.1 <http://tools.ietf.org/html/rfc5763#section-6.7.1>. ICE Interaction
   Interactive Connectivity Establishment (ICE), as specified in
   [RFC5245 <http://tools.ietf.org/html/rfc5245>], provides a methodology
of allowing participants in
   multimedia sessions to verify mutual connectivity.  When ICE is being
   used, the ICE connectivity checks are performed before the DTLS
   handshake begins.  Note that if aggressive nomination mode is used,
   multiple candidate pairs may be marked valid before ICE finally
   converges on a single candidate pair.  Implementations MUST treat all
   ICE candidate pairs associated with a single component as part of the
   same DTLS association.  Thus, there will be only one DTLS handshake
   even if there are multiple valid candidate pairs.  Note that this may
   mean adjusting the endpoint IP addresses if the selected candidate
   pair shifts, just as if the DTLS packets were an ordinary media
   stream.

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<div dir=3D"ltr"><div>RFC 5763 Section 6.7.1 explains much of what we&#39;v=
e been=C2=A0talking about on this thread.=C2=A0<br></div><span class=3D"h4"=
><h4><a name=3D"section-6.7.1" href=3D"http://tools.ietf.org/html/rfc5763#s=
ection-6.7.1">6.7.1</a>.  ICE Interaction</h4></span><h4><br>=C2=A0=C2=A0 I=
nteractive Connectivity Establishment (ICE), as specified in<br>=C2=A0=C2=
=A0 [<a title=3D"&quot;Interactive Connectivity Establishment (ICE): A Prot=
ocol for Network Address Translator (NAT) Traversal for Offer/Answer Protoc=
ols&quot;" href=3D"http://tools.ietf.org/html/rfc5245"><font color=3D"#0066=
cc">RFC5245</font></a>], provides a methodology of allowing participants in=
<br>=C2=A0=C2=A0 multimedia sessions to verify mutual connectivity.=C2=A0 W=
hen ICE is being<br>=C2=A0=C2=A0 used, the ICE connectivity checks are perf=
ormed before the DTLS<br>=C2=A0=C2=A0 handshake begins.=C2=A0 Note that if =
aggressive nomination mode is used,<br>=C2=A0=C2=A0 multiple candidate pair=
s may be marked valid before ICE finally<br>=C2=A0=C2=A0 converges on a sin=
gle candidate pair.=C2=A0 Implementations MUST treat all<br>=C2=A0=C2=A0 IC=
E candidate pairs associated with a single component as part of the<br>=C2=
=A0=C2=A0 same DTLS association.=C2=A0 Thus, there will be only one DTLS ha=
ndshake<br>=C2=A0=C2=A0 even if there are multiple valid candidate pairs.=
=C2=A0 Note that this may<br>=C2=A0=C2=A0 mean adjusting the endpoint IP ad=
dresses if the selected candidate<br>=C2=A0=C2=A0 pair shifts, just as if t=
he DTLS packets were an ordinary media<br>=C2=A0=C2=A0 stream.<br></h4></di=
v>

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From: Peter Thatcher <pthatcher@google.com>
Date: Thu, 26 Mar 2015 16:51:40 -0700
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Subject: Re: [rtcweb] On transports and bundles
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I agree that the fundamental question is about what SDP the browser
accepts.  I would state the question thus: does SetLocalDescription and
SetRemoteDescription allow multiple BUNDLE groups?

In the spirit of "let's just finish 1.0", I'd be happy to throw multiple
BUNDLE groups out of 1.0.  But I realize I might be in the minority.



On Thu, Mar 26, 2015 at 4:38 PM, Adam Roach <adam@nostrum.com> wrote:

> If we factor out all of the W3C API conversations from what was said
> today, I think the only real point of contention turns on this statement;
> not because it's wrong, but because it's incomplete (and creates ambiguity
> as a result):
>
>    A receiving implementation MUST be able to receive media and data in
>    all these configurations.
>
> If we're going to make this statement, then we need to also say something
> about receiving bundle configurations that are more advanced than the two
> that we must be able to send.
>
> I would argue that anything short of saying that we MUST support any valid
> description as described by draft-ietf-mmusic-sdp-bundle-negotiation
> would be creating a profile of bundle, and I believe that would be harmful.
> To that end, I think we either say that implementations MUST be able to
> handle any bundle configuration that is valid per the bundle draft, or we
> omit the statement I cite above and leave it to bundle to make statements
> around what it means to implement the protocol it defines.
>
> I slightly prefer the latter approach.
>
> /a
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif">I agree that the fundamental question is about what SDP=
 the browser accepts.=C2=A0 I would state the question thus: does SetLocalD=
escription and SetRemoteDescription allow multiple BUNDLE groups? =C2=A0</d=
iv><div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-s=
erif"><br></div><div class=3D"gmail_default" style=3D"font-family:arial,hel=
vetica,sans-serif">In the spirit of &quot;let&#39;s just finish 1.0&quot;, =
I&#39;d be happy to throw multiple BUNDLE groups out of 1.0.=C2=A0 But I re=
alize I might be in the minority.</div><div class=3D"gmail_default" style=
=3D"font-family:arial,helvetica,sans-serif"><br></div><div class=3D"gmail_d=
efault" style=3D"font-family:arial,helvetica,sans-serif"><br></div></div><d=
iv class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Thu, Mar 26, 201=
5 at 4:38 PM, Adam Roach <span dir=3D"ltr">&lt;<a href=3D"mailto:adam@nostr=
um.com" target=3D"_blank">adam@nostrum.com</a>&gt;</span> wrote:<br><blockq=
uote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc =
solid;padding-left:1ex">If we factor out all of the W3C API conversations f=
rom what was said today, I think the only real point of contention turns on=
 this statement; not because it&#39;s wrong, but because it&#39;s incomplet=
e (and creates ambiguity as a result):<br>
<br>
=C2=A0 =C2=A0A receiving implementation MUST be able to receive media and d=
ata in<br>
=C2=A0 =C2=A0all these configurations.<br>
<br>
If we&#39;re going to make this statement, then we need to also say somethi=
ng about receiving bundle configurations that are more advanced than the tw=
o that we must be able to send.<br>
<br>
I would argue that anything short of saying that we MUST support any valid =
description as described by draft-ietf-mmusic-sdp-bundle-<u></u>negotiation=
 would be creating a profile of bundle, and I believe that would be harmful=
. To that end, I think we either say that implementations MUST be able to h=
andle any bundle configuration that is valid per the bundle draft, or we om=
it the statement I cite above and leave it to bundle to make statements aro=
und what it means to implement the protocol it defines.<br>
<br>
I slightly prefer the latter approach.<br>
<br>
/a<br>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote></div><br></div>

--001a1134a2b0dee046051239b476--


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From: Roman Shpount <roman@telurix.com>
To: Bernard Aboba <bernard.aboba@gmail.com>
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Subject: Re: [rtcweb] RFC 5763 Section 6.7.1 (was DTLS, DTLS-SRTP, and 5-tuples)
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On Thu, Mar 26, 2015 at 7:41 PM, Bernard Aboba <bernard.aboba@gmail.com>
wrote:

> RFC 5763 Section 6.7.1 explains much of what we've been talking about on
> this thread.
> 6.7.1 <http://tools.ietf.org/html/rfc5763#section-6.7.1>. ICE Interaction
>    Interactive Connectivity Establishment (ICE), as specified in
>    [RFC5245 <http://tools.ietf.org/html/rfc5245>], provides a methodology
> of allowing participants in
>    multimedia sessions to verify mutual connectivity.  When ICE is being
>    used, the ICE connectivity checks are performed before the DTLS
>    handshake begins.  Note that if aggressive nomination mode is used,
>    multiple candidate pairs may be marked valid before ICE finally
>    converges on a single candidate pair.  Implementations MUST treat all
>    ICE candidate pairs associated with a single component as part of the
>    same DTLS association.  Thus, there will be only one DTLS handshake
>    even if there are multiple valid candidate pairs.  Note that this may
>    mean adjusting the endpoint IP addresses if the selected candidate
>    pair shifts, just as if the DTLS packets were an ordinary media
>    stream.
>
There is still a questions that needs to be resolved in relation to this:

What is the transport parameter change that requires a new DTLS handshake?
In particular, would ICE restart require a new DTLS handshake?

One option is that DTLS handshake is triggered by any ufrag changes,
including ICE restart. The problem with this option is that if some ICE
candidates are reused, the same candidate pair can end up into two
different associations at the same time.

Another option is that DTLS handshake is only triggered if fingerprint pair
changes. There is a problem with this, since DTLS Client hello can be
received before the answer with the new fingerprint.
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On T=
hu, Mar 26, 2015 at 7:41 PM, Bernard Aboba <span dir=3D"ltr">&lt;<a href=3D=
"mailto:bernard.aboba@gmail.com" target=3D"_blank">bernard.aboba@gmail.com<=
/a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:=
0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);=
border-left-style:solid;padding-left:1ex"><div dir=3D"ltr"><div>RFC 5763 Se=
ction 6.7.1 explains much of what we&#39;ve been=C2=A0talking about on this=
 thread.=C2=A0<br></div><span><h4><a name=3D"14c5876f5639babc_section-6.7.1=
" href=3D"http://tools.ietf.org/html/rfc5763#section-6.7.1" target=3D"_blan=
k">6.7.1</a>.  ICE Interaction</h4></span><h4><br>=C2=A0=C2=A0 Interactive =
Connectivity Establishment (ICE), as specified in<br>=C2=A0=C2=A0 [<a title=
=3D"&quot;Interactive Connectivity Establishment (ICE): A Protocol for Netw=
ork Address Translator (NAT) Traversal for Offer/Answer Protocols&quot;" hr=
ef=3D"http://tools.ietf.org/html/rfc5245" target=3D"_blank"><font color=3D"=
#0066cc">RFC5245</font></a>], provides a methodology of allowing participan=
ts in<br>=C2=A0=C2=A0 multimedia sessions to verify mutual connectivity.=C2=
=A0 When ICE is being<br>=C2=A0=C2=A0 used, the ICE connectivity checks are=
 performed before the DTLS<br>=C2=A0=C2=A0 handshake begins.=C2=A0 Note tha=
t if aggressive nomination mode is used,<br>=C2=A0=C2=A0 multiple candidate=
 pairs may be marked valid before ICE finally<br>=C2=A0=C2=A0 converges on =
a single candidate pair.=C2=A0 Implementations MUST treat all<br>=C2=A0=C2=
=A0 ICE candidate pairs associated with a single component as part of the<b=
r>=C2=A0=C2=A0 same DTLS association.=C2=A0 Thus, there will be only one DT=
LS handshake<br>=C2=A0=C2=A0 even if there are multiple valid candidate pai=
rs.=C2=A0 Note that this may<br>=C2=A0=C2=A0 mean adjusting the endpoint IP=
 addresses if the selected candidate<br>=C2=A0=C2=A0 pair shifts, just as i=
f the DTLS packets were an ordinary media<br>=C2=A0=C2=A0 stream.<br></h4><=
/div>
</blockquote></div>There is still a questions that needs to be resolved in =
relation to this:</div><div class=3D"gmail_extra"><br></div><div class=3D"g=
mail_extra">What is the transport parameter change that requires a new DTLS=
 handshake? In particular, would ICE restart require a new DTLS handshake?<=
/div><div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">One op=
tion is that DTLS handshake is triggered by any ufrag changes, including IC=
E restart. The problem with this option is that if some ICE candidates are =
reused, the same candidate pair can end up into two different associations =
at the same time.</div><div class=3D"gmail_extra"><br></div><div class=3D"g=
mail_extra">Another option is that DTLS handshake is only triggered if fing=
erprint pair changes. There is a problem with this, since DTLS Client hello=
 can be received before the answer with the new fingerprint.</div><div clas=
s=3D"gmail_extra"><div><div class=3D"gmail_signature">_____________<br>Roma=
n Shpount</div></div><div><br></div></div></div>

--001a1140f04aa828c2051239d40a--


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Subject: Re: [rtcweb] On transports and bundles
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On 3/26/15 18:51, Peter Thatcher wrote:
> I agree that the fundamental question is about what SDP the browser 
> accepts.  I would state the question thus: does SetLocalDescription 
> and SetRemoteDescription allow multiple BUNDLE groups?
>
> In the spirit of "let's just finish 1.0", I'd be happy to throw 
> multiple BUNDLE groups out of 1.0.

The issue I have with that is that you're saying "we're doing bundle, 
*except*..." -- which means you're making a profile of it, making it 
only partially compatible with the protocol as specified. That's bad for 
interop with more advanced devices, and it's bad for forwards 
compatibility with whatever we do with later APIs.

/a

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<html>
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    <meta content="text/html; charset=utf-8" http-equiv="Content-Type">
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  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">On 3/26/15 18:51, Peter Thatcher wrote:<br>
    </div>
    <blockquote
cite="mid:CAJrXDUGiG15M-ba6_Dcv4tzjNZguED488FUaFSii5bJJ79hHrQ@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_default"
          style="font-family:arial,helvetica,sans-serif">I agree that
          the fundamental question is about what SDP the browser
          accepts.Â  I would state the question thus: does
          SetLocalDescription and SetRemoteDescription allow multiple
          BUNDLE groups? Â </div>
        <div class="gmail_default"
          style="font-family:arial,helvetica,sans-serif"><br>
        </div>
        <div class="gmail_default"
          style="font-family:arial,helvetica,sans-serif">In the spirit
          of "let's just finish 1.0", I'd be happy to throw multiple
          BUNDLE groups out of 1.0.</div>
      </div>
    </blockquote>
    <br>
    The issue I have with that is that you're saying "we're doing
    bundle, *except*..." -- which means you're making a profile of it,
    making it only partially compatible with the protocol as specified.
    That's bad for interop with more advanced devices, and it's bad for
    forwards compatibility with whatever we do with later APIs.<br>
    <br>
    /a<br>
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On Thu, Mar 26, 2015 at 7:46 PM, Adam Roach <adam@nostrum.com> wrote:

>  On 3/26/15 18:51, Peter Thatcher wrote:
>
>  I agree that the fundamental question is about what SDP the browser
> accepts.  I would state the question thus: does SetLocalDescription and
> SetRemoteDescription allow multiple BUNDLE groups?
>
>  In the spirit of "let's just finish 1.0", I'd be happy to throw multiple
> BUNDLE groups out of 1.0.
>
>
> The issue I have with that is that you're saying "we're doing bundle,
> *except*..." -- which means you're making a profile of it, making it only
> partially compatible with the protocol as specified. That's bad for inter=
op
> with more advanced devices, and it's bad for forwards compatibility with
> whatever we do with later APIs.
>

I believe what Martin said Firefox does right now is still interoperable
and forwards compatible:  unbundle the bundle groups other than the first.
 =E2=80=8B

=E2=80=8BIt's just not optimal for particular advanced use cases which can =
be
addresses separately after 1.0.  And I'm OK with that.



>
> /a
>

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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif"><br></div><div class=3D"gmail_extra"><br><div class=3D"=
gmail_quote">On Thu, Mar 26, 2015 at 7:46 PM, Adam Roach <span dir=3D"ltr">=
&lt;<a href=3D"mailto:adam@nostrum.com" target=3D"_blank">adam@nostrum.com<=
/a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:=
0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000"><span class=3D"">
    <div>On 3/26/15 18:51, Peter Thatcher wrote:<br>
    </div>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">
        <div style=3D"font-family:arial,helvetica,sans-serif">I agree that
          the fundamental question is about what SDP the browser
          accepts.=C2=A0 I would state the question thus: does
          SetLocalDescription and SetRemoteDescription allow multiple
          BUNDLE groups? =C2=A0</div>
        <div style=3D"font-family:arial,helvetica,sans-serif"><br>
        </div>
        <div style=3D"font-family:arial,helvetica,sans-serif">In the spirit
          of &quot;let&#39;s just finish 1.0&quot;, I&#39;d be happy to thr=
ow multiple
          BUNDLE groups out of 1.0.</div>
      </div>
    </blockquote>
    <br></span>
    The issue I have with that is that you&#39;re saying &quot;we&#39;re do=
ing
    bundle, *except*...&quot; -- which means you&#39;re making a profile of=
 it,
    making it only partially compatible with the protocol as specified.
    That&#39;s bad for interop with more advanced devices, and it&#39;s bad=
 for
    forwards compatibility with whatever we do with later APIs.</div></bloc=
kquote><div><div class=3D"gmail_default" style=3D"font-family:arial,helveti=
ca,sans-serif;display:inline"><br></div></div><div><div class=3D"gmail_defa=
ult" style=3D"font-family:arial,helvetica,sans-serif;display:inline">I beli=
eve what Martin said Firefox does right now is still interoperable and forw=
ards compatible: =C2=A0unbundle the bundle groups other than the first. =C2=
=A0=E2=80=8B</div>=C2=A0<div class=3D"gmail_default" style=3D"font-family:a=
rial,helvetica,sans-serif;display:inline">=E2=80=8BIt&#39;s just not optima=
l for particular advanced use cases which can be addresses separately after=
 1.0.=C2=A0 And I&#39;m OK with that.</div><br></div><div><div class=3D"gma=
il_default" style=3D"font-family:arial,helvetica,sans-serif;display:inline"=
><br></div></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex"><div bgcolor=3D"#FFFFFF" te=
xt=3D"#000000"><span class=3D"HOEnZb"><font color=3D"#888888"><br>
    <br>
    /a<br>
  </font></span></div>

</blockquote></div><br></div></div>

--089e013a2b2cbc831f05123d8f17--


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Date: Fri, 27 Mar 2015 06:57:52 -0500
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From: Simon Perreault <sperreault@jive.com>
To: Bernard Aboba <bernard.aboba@gmail.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] RFC 5763 Section 6.7.1 (was DTLS, DTLS-SRTP, and 5-tuples)
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Wow. So many emails wasted. Thanks a lot!

</thread>

Simon
Le 2015-03-26 18:42, "Bernard Aboba" <bernard.aboba@gmail.com> a =C3=A9crit=
 :

> RFC 5763 Section 6.7.1 explains much of what we've been talking about on
> this thread.
> 6.7.1 <http://tools.ietf.org/html/rfc5763#section-6.7.1>. ICE Interaction
>    Interactive Connectivity Establishment (ICE), as specified in
>    [RFC5245 <http://tools.ietf.org/html/rfc5245>], provides a methodology
> of allowing participants in
>    multimedia sessions to verify mutual connectivity.  When ICE is being
>    used, the ICE connectivity checks are performed before the DTLS
>    handshake begins.  Note that if aggressive nomination mode is used,
>    multiple candidate pairs may be marked valid before ICE finally
>    converges on a single candidate pair.  Implementations MUST treat all
>    ICE candidate pairs associated with a single component as part of the
>    same DTLS association.  Thus, there will be only one DTLS handshake
>    even if there are multiple valid candidate pairs.  Note that this may
>    mean adjusting the endpoint IP addresses if the selected candidate
>    pair shifts, just as if the DTLS packets were an ordinary media
>    stream.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<p dir=3D"ltr">Wow. So many emails wasted. Thanks a lot!</p>
<p dir=3D"ltr">&lt;/thread&gt;</p>
<p dir=3D"ltr">Simon</p>
<div class=3D"gmail_quote">Le 2015-03-26 18:42, &quot;Bernard Aboba&quot; &=
lt;<a href=3D"mailto:bernard.aboba@gmail.com">bernard.aboba@gmail.com</a>&g=
t; a =C3=A9crit :<br type=3D"attribution"><blockquote class=3D"gmail_quote"=
 style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><d=
iv dir=3D"ltr"><div>RFC 5763 Section 6.7.1 explains much of what we&#39;ve =
been=C2=A0talking about on this thread.=C2=A0<br></div><span><h4><a name=3D=
"14c5876dcdbb5612_section-6.7.1" href=3D"http://tools.ietf.org/html/rfc5763=
#section-6.7.1" target=3D"_blank">6.7.1</a>.  ICE Interaction</h4></span><h=
4><br>=C2=A0=C2=A0 Interactive Connectivity Establishment (ICE), as specifi=
ed in<br>=C2=A0=C2=A0 [<a title=3D"&quot;Interactive Connectivity Establish=
ment (ICE): A Protocol for Network Address Translator (NAT) Traversal for O=
ffer/Answer Protocols&quot;" href=3D"http://tools.ietf.org/html/rfc5245" ta=
rget=3D"_blank"><font color=3D"#0066cc">RFC5245</font></a>], provides a met=
hodology of allowing participants in<br>=C2=A0=C2=A0 multimedia sessions to=
 verify mutual connectivity.=C2=A0 When ICE is being<br>=C2=A0=C2=A0 used, =
the ICE connectivity checks are performed before the DTLS<br>=C2=A0=C2=A0 h=
andshake begins.=C2=A0 Note that if aggressive nomination mode is used,<br>=
=C2=A0=C2=A0 multiple candidate pairs may be marked valid before ICE finall=
y<br>=C2=A0=C2=A0 converges on a single candidate pair.=C2=A0 Implementatio=
ns MUST treat all<br>=C2=A0=C2=A0 ICE candidate pairs associated with a sin=
gle component as part of the<br>=C2=A0=C2=A0 same DTLS association.=C2=A0 T=
hus, there will be only one DTLS handshake<br>=C2=A0=C2=A0 even if there ar=
e multiple valid candidate pairs.=C2=A0 Note that this may<br>=C2=A0=C2=A0 =
mean adjusting the endpoint IP addresses if the selected candidate<br>=C2=
=A0=C2=A0 pair shifts, just as if the DTLS packets were an ordinary media<b=
r>=C2=A0=C2=A0 stream.<br></h4></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div>

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From: "Roni Even" <ron.even.tlv@gmail.com>
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Subject: Re: [rtcweb] Usage of b= at session level - JSEP
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=20

=20

On 25 March 2015 at 14:33, Roni Even <ron.even.tlv@gmail.com> wrote:

About using AS at session level. Making it an error may cause =
interoperability problems with other systems.

=20

Would it be a major problem in that situation if the b=3DAS were =
ignored?

[Roni Even] b=3DAS at session level can be ignored (as any SDP =
attribute) but the problem will be with what it will do to the total =
bandwidth used. The session level limits the total for all media levels =
in the session, while when you sum the b=3D values in the media levels =
they can be higher than the b=3D at the session level.

I think that we could also require that b=3DCT be ignored on the media =
level too.  If it turns out that it has significant usage, that is.

[Roni Even] I am not aware of usage of CT, as you can see from the =
documents I had in my email (3GPP and IMTC SIP parity) they use AS and =
TIAS


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D'><o:p>&nbsp;</o:p></span></p><div =
style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt'><div><div><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><p =
class=3DMsoNormal>On 25 March 2015 at 14:33, Roni Even &lt;<a =
href=3D"mailto:ron.even.tlv@gmail.com" =
target=3D"_blank">ron.even.tlv@gmail.com</a>&gt; wrote:<o:p></o:p></p><p =
class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>About using =
AS at session level. Making it an error may cause interoperability =
problems with other systems.<o:p></o:p></p></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'>Would it be a major problem in that =
situation if the b=3DAS were ignored?<o:p></o:p></p><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><b><i><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>[Roni Even] b=3DAS at session level can be ignored (as any SDP =
attribute) but the problem will be with what it will do to the total =
bandwidth used. The session level limits the total for all media levels =
in the session, while when you sum the b=3D values in the media levels =
they can be higher than the b=3D at the session =
level.</span></i></b><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p></o:p></span></p></div><div><p class=3DMsoNormal>I think that we =
could also require that b=3DCT be ignored on the media level too.&nbsp; =
If it turns out that it has significant usage, that is.<o:p></o:p></p><p =
class=3DMsoNormal><b><i><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>[Roni Even] I am not aware of usage of CT, as you can see from the =
documents I had in my email (3GPP and IMTC SIP parity) they use AS and =
TIAS</span></i></b><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p></o:p></span></p></div></div></div></div></body></html>
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From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] On transports and bundles
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On 03/27/2015 12:38 AM, Adam Roach wrote:
> If we factor out all of the W3C API conversations from what was said
> today, I think the only real point of contention turns on this
> statement; not because it's wrong, but because it's incomplete (and
> creates ambiguity as a result):
>
>    A receiving implementation MUST be able to receive media and data in=

>    all these configurations.
>
> If we're going to make this statement, then we need to also say
> something about receiving bundle configurations that are more advanced
> than the two that we must be able to send.
>
> I would argue that anything short of saying that we MUST support any
> valid description as described by
> draft-ietf-mmusic-sdp-bundle-negotiation would be creating a profile
> of bundle, and I believe that would be harmful. To that end, I think
> we either say that implementations MUST be able to handle any bundle
> configuration that is valid per the bundle draft, or we omit the
> statement I cite above and leave it to bundle to make statements
> around what it means to implement the protocol it defines.

"Support any valid configuration" can be read two ways:

A - The bundling proposal MUST result in the bundling the offerer suggest=
ed
B - The bundling proposal MUST result in a state that is legal by BUNDLE
rules

I think we're all in agreement that we want to mandate support in sense B=
=2E

When I wrote the text, I was intending sense A for the fully-bundled and
fully-unbundled cases.

I don't think this creates a profile of BUNDLE.
I think we agree on the state we want, it's just a matter of getting the
text to say that...




>
> I slightly prefer the latter approach.
>
> /a
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


--=20
Surveillance is pervasive. Go Dark.



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On 3/27/15 10:41, Harald Alvestrand wrote:
> "Support any valid configuration" can be read two ways:
>
> A - The bundling proposal MUST result in the bundling the offerer suggested
> B - The bundling proposal MUST result in a state that is legal by BUNDLE
> rules
>
> I think we're all in agreement that we want to mandate support in sense B.

Yes, what I'm asking for is B.

> When I wrote the text, I was intending sense A for the fully-bundled and
> fully-unbundled cases.
>
> I don't think this creates a profile of BUNDLE.
> I think we agree on the state we want, it's just a matter of getting the
> text to say that...

Right. I agree that what you've described in this email makes sense, and 
we just need to turn that sentiment into document prose.

/a


From nobody Fri Mar 27 09:22:43 2015
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From: Peter Thatcher <pthatcher@google.com>
Date: Fri, 27 Mar 2015 11:21:59 -0500
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--001a113cd4128d9ed30512478a5a
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On Fri, Mar 27, 2015 at 11:03 AM, Adam Roach <adam@nostrum.com> wrote:

> On 3/27/15 10:41, Harald Alvestrand wrote:
>
>> "Support any valid configuration" can be read two ways:
>>
>> A - The bundling proposal MUST result in the bundling the offerer
>> suggested
>> B - The bundling proposal MUST result in a state that is legal by BUNDLE
>> rules
>>
>> I think we're all in agreement that we want to mandate support in sense B.
>>
>
> Yes, what I'm asking for is B.


I believe B is the same as allowing the browser to unbundle the bundle
groups other than the first.  Is that correct?

Would it also allow a browser to say "BUNDLE everything or BUNDLE nothing"
by unbundling everything if it sees SDP with more than one bundle group?



>
>
>  When I wrote the text, I was intending sense A for the fully-bundled and
>> fully-unbundled cases.
>>
>> I don't think this creates a profile of BUNDLE.
>> I think we agree on the state we want, it's just a matter of getting the
>> text to say that...
>>
>
> Right. I agree that what you've described in this email makes sense, and
> we just need to turn that sentiment into document prose.
>
>
> /a
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--001a113cd4128d9ed30512478a5a
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif"><br></div><div class=3D"gmail_extra"><br><div class=3D"=
gmail_quote">On Fri, Mar 27, 2015 at 11:03 AM, Adam Roach <span dir=3D"ltr"=
>&lt;<a href=3D"mailto:adam@nostrum.com" target=3D"_blank">adam@nostrum.com=
</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin=
:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204)=
;border-left-style:solid;padding-left:1ex"><span class=3D"">On 3/27/15 10:4=
1, Harald Alvestrand wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
&quot;Support any valid configuration&quot; can be read two ways:<br>
<br>
A - The bundling proposal MUST result in the bundling the offerer suggested=
<br>
B - The bundling proposal MUST result in a state that is legal by BUNDLE<br=
>
rules<br>
<br>
I think we&#39;re all in agreement that we want to mandate support in sense=
 B.<br>
</blockquote>
<br></span>
Yes, what I&#39;m asking for is B.</blockquote><div><br></div><div><div cla=
ss=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-serif">I bel=
ieve B is the same as allowing the browser to=C2=A0<span style=3D"font-size=
:12.8000001907349px">unbundle the bundle groups other than the first.=C2=A0=
 Is that correct?</span></div><div class=3D"gmail_default" style=3D"font-fa=
mily:arial,helvetica,sans-serif"><br></div><div class=3D"gmail_default" sty=
le=3D"font-family:arial,helvetica,sans-serif">Would it also allow a browser=
 to say &quot;BUNDLE everything or BUNDLE nothing&quot; by unbundling every=
thing if it sees SDP with more than one bundle group?</div><br></div><div>=
=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0=
.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-s=
tyle:solid;padding-left:1ex"><span class=3D""><br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
When I wrote the text, I was intending sense A for the fully-bundled and<br=
>
fully-unbundled cases.<br>
<br>
I don&#39;t think this creates a profile of BUNDLE.<br>
I think we agree on the state we want, it&#39;s just a matter of getting th=
e<br>
text to say that...<br>
</blockquote>
<br></span>
Right. I agree that what you&#39;ve described in this email makes sense, an=
d we just need to turn that sentiment into document prose.<div class=3D""><=
div class=3D"h5"><br>
<br>
/a<br>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div></div>

--001a113cd4128d9ed30512478a5a--


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On Mar 26, 2015, at 6:41 PM, Bernard Aboba <bernard.aboba@gmail.com<mailto:=
bernard.aboba@gmail.com>> wrote:

RFC 5763 Section 6.7.1 explains much of what we've been talking about on th=
is thread.
6.7.1<http://tools.ietf.org/html/rfc5763#section-6.7.1>. ICE Interaction

   Interactive Connectivity Establishment (ICE), as specified in
   [RFC5245<http://tools.ietf.org/html/rfc5245>], provides a methodology of=
 allowing participants in
   multimedia sessions to verify mutual connectivity.  When ICE is being
   used, the ICE connectivity checks are performed before the DTLS
   handshake begins.  Note that if aggressive nomination mode is used,
   multiple candidate pairs may be marked valid before ICE finally
   converges on a single candidate pair.  Implementations MUST treat all
   ICE candidate pairs associated with a single component as part of the
   same DTLS association.  Thus, there will be only one DTLS handshake
   even if there are multiple valid candidate pairs.  Note that this may
   mean adjusting the endpoint IP addresses if the selected candidate
   pair shifts, just as if the DTLS packets were an ordinary media
   stream
(Adding MMUSIC.)

The one recommendation I=92d make for this problem is that one of the DTLS-=
SCTP drafts (probably ietf-mmusic-sctp-sdp, I guess?) should repeat this te=
xt from 5763.



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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3DWindows-1=
252">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space;">
<br>
<div>
<div>On Mar 26, 2015, at 6:41 PM, Bernard Aboba &lt;<a href=3D"mailto:berna=
rd.aboba@gmail.com">bernard.aboba@gmail.com</a>&gt; wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">
<div dir=3D"ltr">
<div>RFC 5763 Section 6.7.1 explains much of what we've been&nbsp;talking a=
bout on this thread.&nbsp;<br>
</div>
<span class=3D"h4">
<h4><a name=3D"section-6.7.1" href=3D"http://tools.ietf.org/html/rfc5763#se=
ction-6.7.1">6.7.1</a>. ICE Interaction</h4>
</span>
<h4><br>
&nbsp;&nbsp; Interactive Connectivity Establishment (ICE), as specified in<=
br>
&nbsp;&nbsp; [<a title=3D"&quot;Interactive Connectivity Establishment (ICE=
): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answ=
er Protocols&quot;" href=3D"http://tools.ietf.org/html/rfc5245"><font color=
=3D"#0066cc">RFC5245</font></a>], provides a methodology of
 allowing participants in<br>
&nbsp;&nbsp; multimedia sessions to verify mutual connectivity.&nbsp; When =
ICE is being<br>
&nbsp;&nbsp; used, the ICE connectivity checks are performed before the DTL=
S<br>
&nbsp;&nbsp; handshake begins.&nbsp; Note that if aggressive nomination mod=
e is used,<br>
&nbsp;&nbsp; multiple candidate pairs may be marked valid before ICE finall=
y<br>
&nbsp;&nbsp; converges on a single candidate pair.&nbsp; Implementations MU=
ST treat all<br>
&nbsp;&nbsp; ICE candidate pairs associated with a single component as part=
 of the<br>
&nbsp;&nbsp; same DTLS association.&nbsp; Thus, there will be only one DTLS=
 handshake<br>
&nbsp;&nbsp; even if there are multiple valid candidate pairs.&nbsp; Note t=
hat this may<br>
&nbsp;&nbsp; mean adjusting the endpoint IP addresses if the selected candi=
date<br>
&nbsp;&nbsp; pair shifts, just as if the DTLS packets were an ordinary medi=
a<br>
&nbsp;&nbsp; stream<br>
</h4>
</div>
</blockquote>
</div>
<div>(Adding MMUSIC.)</div>
<div><br>
</div>
<div>The one recommendation I=92d make for this problem is that one of the =
DTLS-SCTP drafts (probably ietf-mmusic-sctp-sdp, I guess?) should repeat th=
is text from 5763.&nbsp;</div>
<div><br>
</div>
<br>
</body>
</html>

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Date: Mon, 30 Mar 2015 10:45:17 +0100
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Subject: Re: [rtcweb] Endpoints that don't support RTCP
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I see several statements in support of this, so I'll make the change and =
submit rtp-usage-23.

Colin



On 26 Mar 2015, at 14:41, Justin Uberti <juberti@google.com> wrote:

> Right, I think mandating use of RTCP is the correct solution. e.g.
>=20
> 7.2.  Congestion Control Interoperability and Legacy Systems
>=20
>    All endpoints that wish to interwork with WebRTC MUST=20
>    implement RTCP and provide congestion feedback via the defined RTCP
>    reporting mechanisms.=20
>=20
>    When interworking with legacy implementations that support RTCP =
using
>    the RTP/AVP profile [RFC3551], congestion feedback is provided in
>    RTCP RR packets every few seconds.  Implementations that have to
>    interwork with such end-points MUST ensure that they keep within =
the
>    RTP circuit breaker [I-D.ietf-avtcore-rtp-circuit-breakers]
>    constraints to limit the congestion they can cause.
>=20
> On Thu, Mar 26, 2015 at 9:23 AM, Matthew Kaufman <matthew@matthew.at> =
wrote:
> These are the legacy endpoints that don't do RTCP but do have ICE and =
DTLS-SRTP?
>=20
> Matthew Kaufman
>=20
> (Sent from my iPhone)
>=20
> On Mar 25, 2015, at 10:24 PM, Justin Uberti <juberti@google.com> =
wrote:
>=20
>> This is described in rtp-usage, and the need to limit bitrate is =
explicitly called out:
>>=20
>> 7.2.  Congestion Control Interoperability and Legacy Systems
>>=20
>>    There are legacy RTP implementations that do not implement RTCP, =
and
>>    hence do not provide any congestion feedback.  Congestion control
>>    cannot be performed with these end-points.  WebRTC Endpoints that
>>    need to interwork with such end-points MUST limit their =
transmission
>>    to a low rate, equivalent to a VoIP call using a low bandwidth =
codec,
>>    that is unlikely to cause any significant congestion.
>>=20
>> Note however that this is incompatible with this section that =
mandates use of circuit breakers:
>>=20
>> 7.1.  Boundary Conditions and Circuit Breakers
>>=20
>>    WebRTC Endpoints MUST implement the RTP circuit breaker algorithm
>>    that is described in [I-D.ietf-avtcore-rtp-circuit-breakers].  The
>>    RTP circuit breaker is designed to enable applications to =
recognise
>>    and react to situations of extreme network congestion.  However,
>>    since the RTP circuit breaker might not be triggered until =
congestion
>>    becomes extreme, it cannot be considered a substitute for =
congestion
>>    control, and applications MUST also implement congestion control =
to
>>    allow them to adapt to changes in network capacity.  Any future =
RTP
>>    congestion control algorithms are expected to operate within the
>>    envelope allowed by the circuit breaker.
>>=20
>> I think we need to clarify exactly how WebRTC endpoints need to deal =
with legacy gear that doesn't send RTCP (which might just be to say =
"no").
>>=20
>>=20
>>=20
>>=20
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



--=20
Colin Perkins
https://csperkins.org/





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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dus-ascii"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;">I see =
several statements in support of this, so I'll make the change and =
submit =
rtp-usage-23.<div><br></div><div>Colin</div><div><br></div><div><br></div>=
<div><br><div><div>On 26 Mar 2015, at 14:41, Justin Uberti &lt;<a =
href=3D"mailto:juberti@google.com">juberti@google.com</a>&gt; =
wrote:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><div dir=3D"ltr">Right, I think mandating use of RTCP is =
the correct solution. e.g.<div><br></div><div><pre style=3D"white-space: =
pre-wrap; font-size: 1em; margin-top: 0px; margin-bottom: 0px;"><span =
style=3D"line-height:0pt;font-size:1em;font-weight:bold"><h3 =
style=3D"line-height:0pt;display:inline;font-size:1em"><a =
name=3D"14c548a2ce2ce9cd_section-7.2" =
href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-22#section=
-7.2" target=3D"_blank" style=3D"text-decoration: none;">7.2</a>.  =
Congestion Control Interoperability and Legacy Systems</h3></span>

   All endpoints that wish to interwork with WebRTC MUST&nbsp;</pre><pre =
style=3D"white-space: pre-wrap; font-size: 1em; margin-top: 0px; =
margin-bottom: 0px;">   implement RTCP and provide congestion feedback =
via the defined RTCP</pre><pre style=3D"white-space: pre-wrap; =
font-size: 1em; margin-top: 0px; margin-bottom: 0px;">   reporting =
mechanisms. </pre><pre style=3D"white-space: pre-wrap; font-size: 1em; =
margin-top: 0px; margin-bottom: 0px;"><br></pre><pre style=3D"white-space:=
 pre-wrap; font-size: 1em; margin-top: 0px; margin-bottom: 0px;"><pre =
class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px">   =
When interworking with legacy implementations that support RTCP using
   the RTP/AVP profile [<a href=3D"https://tools.ietf.org/html/rfc3551" =
title=3D"&quot;RTP Profile for Audio and Video Conferences with Minimal =
Control&quot;">RFC3551</a>], congestion feedback is provided in
   RTCP RR packets every few seconds.  Implementations that have to
   interwork with such end-points MUST ensure that they keep within the
   RTP circuit breaker [<a =
href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-22#ref-I-D=
.ietf-avtcore-rtp-circuit-breakers">I-D.ietf-avtcore-rtp-circuit-breakers<=
/a>]
   constraints to limit the congestion they can =
cause.</pre></pre></div></div><div class=3D"gmail_extra"><br><div =
class=3D"gmail_quote">On Thu, Mar 26, 2015 at 9:23 AM, Matthew Kaufman =
<span dir=3D"ltr">&lt;<a href=3D"mailto:matthew@matthew.at" =
target=3D"_blank">matthew@matthew.at</a>&gt;</span> =
wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 =
.8ex;border-left:1px #ccc solid;padding-left:1ex"><div =
dir=3D"auto"><div>These are the legacy endpoints that don't do RTCP but =
do have ICE and DTLS-SRTP?<br><br>Matthew Kaufman<div><br>(Sent from my =
iPhone)</div></div><div><div class=3D"h5"><div><br>On Mar 25, 2015, at =
10:24 PM, Justin Uberti &lt;<a href=3D"mailto:juberti@google.com" =
target=3D"_blank">juberti@google.com</a>&gt; =
wrote:<br><br></div><blockquote type=3D"cite"><div dir=3D"ltr">This is =
described in rtp-usage, and the need to limit bitrate is explicitly =
called out:<div><br></div><div><pre style=3D"font-size: 1em; margin-top: =
0px; margin-bottom: 0px;"><span =
style=3D"line-height:0pt;display:inline;font-size:1em;font-weight:bold"><h=
3 style=3D"line-height:0pt;display:inline;font-size:1em"><a =
name=3D"14c56776b7749c05_section-7.2" =
href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-22#section=
-7.2" style=3D"text-decoration: none;" target=3D"_blank">7.2</a>.  =
Congestion Control Interoperability and Legacy Systems</h3></span>

   There are legacy RTP implementations that do not implement RTCP, and
   hence do not provide any congestion feedback.  Congestion control
   cannot be performed with these end-points.  WebRTC Endpoints that
   need to interwork with such end-points MUST limit their transmission
   to a low rate, equivalent to a VoIP call using a low bandwidth codec,
   that is unlikely to cause any significant congestion.</pre><pre =
style=3D"font-size: 1em; margin-top: 0px; margin-bottom: =
0px;"><br></pre><pre style=3D"font-size: 1em; margin-top: 0px; =
margin-bottom: 0px;"><span =
style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:small;=
white-space:normal">Note however that this is incompatible with this =
section that mandates use of circuit breakers:</span><br></pre><pre =
style=3D"font-size: 1em; margin-top: 0px; margin-bottom: 0px;"><span =
style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:small;=
white-space:normal"><br></span></pre><pre =
style=3D"margin-top:0px;margin-bottom:0px"><pre style=3D"font-size: 1em; =
margin-top: 0px; margin-bottom: 0px;"><span =
style=3D"line-height:0pt;display:inline;font-size:1em;font-weight:bold"><h=
3 style=3D"line-height:0pt;display:inline;font-size:1em"><a =
name=3D"14c56776b7749c05_section-7.1" =
href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-22#section=
-7.1" style=3D"text-decoration: none;" target=3D"_blank">7.1</a>.  =
Boundary Conditions and Circuit Breakers</h3></span>

   WebRTC Endpoints MUST implement the RTP circuit breaker algorithm
   that is described in [<a =
href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-22#ref-I-D=
.ietf-avtcore-rtp-circuit-breakers" =
target=3D"_blank">I-D.ietf-avtcore-rtp-circuit-breakers</a>].  The
   RTP circuit breaker is designed to enable applications to recognise
   and react to situations of extreme network congestion.  However,
   since the RTP circuit breaker might not be triggered until congestion
   becomes extreme, it cannot be considered a substitute for congestion
   control, and applications MUST also implement congestion control to
   allow them to adapt to changes in network capacity.  Any future RTP
   congestion control algorithms are expected to operate within the
   envelope allowed by the circuit breaker.</pre><pre style=3D"font-size: =
1em; margin-top: 0px; margin-bottom: 0px;"><br></pre><pre =
style=3D"font-size: 1em; margin-top: 0px; margin-bottom: 0px;"><span =
style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:small;=
white-space:normal">I think we need to clarify exactly how WebRTC =
endpoints need to deal with legacy gear that doesn't send RTCP (which =
might just be to say "no").</span></pre><div style=3D"font-size: =
1em;"><span =
style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:small;=
white-space:normal"><br></span></div><pre style=3D"font-size: 1em; =
margin-top: 0px; margin-bottom: 0px;"><br></pre><pre style=3D"font-size: =
1em; margin-top: 0px; margin-bottom: 0px;"><br></pre></pre><pre =
style=3D"font-size: 1em; margin-top: 0px; margin-bottom: =
0px;"><br></pre><pre style=3D"font-size: 1em; margin-top: 0px; =
margin-bottom: 0px;"><br></pre></div></div>
</blockquote></div></div><span class=3D""><blockquote =
type=3D"cite"><span>_______________________________________________</span>=
<br><span>rtcweb mailing list</span><br><span><a =
href=3D"mailto:rtcweb@ietf.org" =
target=3D"_blank">rtcweb@ietf.org</a></span><br><span><a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" =
target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a></span><=
br></blockquote></span></div></blockquote></div><br></div>
_______________________________________________<br>rtcweb mailing =
list<br><a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>https://www.ietf.or=
g/mailman/listinfo/rtcweb<br></blockquote></div><br><div =
apple-content-edited=3D"true">
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
color: rgb(0, 0, 0); font-family: 'Lucida Sans Typewriter'; font-style: =
normal; font-variant: normal; font-weight: normal; letter-spacing: =
normal; line-height: normal; orphans: 2; text-align: -webkit-auto; =
text-indent: 0px; text-transform: none; white-space: normal; widows: 2; =
word-spacing: 0px; border-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-stroke-width: =
0px;"><div><br class=3D"Apple-interchange-newline"><br =
class=3D"khtml-block-placeholder"></div><div>--&nbsp;</div><div></div><div=
>Colin Perkins</div><div><a =
href=3D"https://csperkins.org/">https://csperkins.org/</a></div><div><br><=
/div></span><br class=3D"Apple-interchange-newline"><br =
class=3D"Apple-interchange-newline">
</div>
<br></div></body></html>=

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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : Web Real-Time Communication (WebRTC): Media Transport and Use of RTP
        Authors         : Colin Perkins
                          Magnus Westerlund
                          Joerg Ott
	Filename        : draft-ietf-rtcweb-rtp-usage-23.txt
	Pages           : 45
	Date            : 2015-03-30

Abstract:
   The Web Real-Time Communication (WebRTC) framework provides support
   for direct interactive rich communication using audio, video, text,
   collaboration, games, etc.  between two peers' web-browsers.  This
   memo describes the media transport aspects of the WebRTC framework.
   It specifies how the Real-time Transport Protocol (RTP) is used in
   the WebRTC context, and gives requirements for which RTP features,
   profiles, and extensions need to be supported.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-rtp-usage/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-23

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-rtp-usage-23


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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


From nobody Mon Mar 30 06:13:45 2015
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Subject: Re: [rtcweb] Endpoints that don't support RTCP
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Den 26. mars 2015 20:52, skrev Mo Zanaty (mzanaty):
> I agree this should be handled in the gateways draft. Other drafts that
> only deal with WebRTC endpoints should just expect RTCP as a given just
> like they expect ICE and DTLS.
> 
> For the WebRTC endpoint side of the gateway, it must obey the RTP
> circuit breaker in both send and receive directions, and synthesize RTCP
> on behalf of the legacy side when necessary to avoid tripping the
> breaker in the receive direction.

This seems indeed to be a place where the gateways draft is the right
place for such advice :-)

Would it be appropriate to say something like (after the text saying
that a gateway needs to do ICE and DTLS-SRTP):

A gateway that passes on RTP needs to monitor the RTCP sent by the
non-WebRTC endpoint to see that it is adequate for congestion control
feedback, and if it is not, start synthesizing it.

In particular, if the non-WebRTC endpoint sends no RTCP, the gateway
needs to start sending syntehsized RTCP feedback before the circuit
breaker mechanism <ref> triggers, and keep on sending RTCP feedback at
appropriate intervals.




From nobody Mon Mar 30 06:25:49 2015
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Subject: Re: [rtcweb] Endpoints that don't support RTCP
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--90e6ba61465a4b722e0512816aae
Content-Type: text/plain; charset=UTF-8

On Mon, Mar 30, 2015 at 9:13 AM, Harald Alvestrand <harald@alvestrand.no>
wrote:

> Would it be appropriate to say something like (after the text saying
> that a gateway needs to do ICE and DTLS-SRTP):
>
> A gateway that passes on RTP needs to monitor the RTCP sent by the
> non-WebRTC endpoint to see that it is adequate for congestion control
> feedback, and if it is not, start synthesizing it.
>
> In particular, if the non-WebRTC endpoint sends no RTCP, the gateway
> needs to start sending syntehsized RTCP feedback before the circuit
> breaker mechanism <ref> triggers, and keep on sending RTCP feedback at
> appropriate intervals.
>

It should also be fine for an RTP relay to unconditionally generate RTCP
receiver reports using its own SSRC. *If* it does not already do that,
*then* it must do it as you suggest. Otherwise, no modification is
necessary.

Simon

--90e6ba61465a4b722e0512816aae
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">=
On Mon, Mar 30, 2015 at 9:13 AM, Harald Alvestrand <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestrand.n=
o</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div id=3D":nj" c=
lass=3D"a3s" style=3D"overflow:hidden">Would it be appropriate to say somet=
hing like (after the text saying<br>
that a gateway needs to do ICE and DTLS-SRTP):<br>
<br>
A gateway that passes on RTP needs to monitor the RTCP sent by the<br>
non-WebRTC endpoint to see that it is adequate for congestion control<br>
feedback, and if it is not, start synthesizing it.<br>
<br>
In particular, if the non-WebRTC endpoint sends no RTCP, the gateway<br>
needs to start sending syntehsized RTCP feedback before the circuit<br>
breaker mechanism &lt;ref&gt; triggers, and keep on sending RTCP feedback a=
t<br>
appropriate intervals.</div></blockquote></div><br>It should also be fine f=
or an RTP relay to unconditionally generate RTCP receiver reports using its=
 own SSRC. *If* it does not already do that, *then* it must do it as you su=
ggest. Otherwise, no modification is necessary.</div><div class=3D"gmail_ex=
tra"><br></div><div class=3D"gmail_extra">Simon</div></div>

--90e6ba61465a4b722e0512816aae--


From nobody Mon Mar 30 09:09:53 2015
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Subject: [rtcweb] SDP lines (JSEP Issue 27)
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Back in Hawaii, I took an action item to look over the various SDP lines 
to analyze their applicability to JSEP [1]. tl;dr: I think what we have 
in the document right now is correct and sufficient, pending the outcome 
of the b= conversation.

----------

I did my own analysis of SDP without having recently re-read the JSEP 
document, with the intention of providing a sanity check. The only thing 
I say below that isn't already covered by 4566 is k= (which is already 
governed by the "no SDES" normative language) and documenting the 
various fields that are of no interest for JSEP use cases (s=, i=, u=, 
e=, p=, z=, and t=).

v=
   MUST set to 0. MUST verify that value is 0 on receipt, treat as 
malformed if not.

o=, c=, m=, a=
   Encoded and used as described in SDP. There is no reason to surface 
the 'o=' username to the application.

s=
   MUST send, MUST verify presence on receipt, treat as malformed if 
not. I see no reason to have API surface to set or read this.

i=, u=, e=, p=, z=, r=
   SHOULD NOT send, MUST ignore. MUST treat as malformed if not in 
correct order.

t=
   MUST include exactly one, of the form "t=0 0". MUST verify that at 
least one t= line is present on receipt (and is syntactically valid), 
but otherwise ignored.

b=
   Ask Magnus.

k=
   MUST NOT send. If present at session level, MUST reject all media 
lines. If present at media level, MUST reject that media line.


In comparing these recommendations with JSEP section 5.2.1, I note that 
they are congruent with the existing language, modulo the ongoing 
conversation around "b=" (which will presumably reach its own 
conclusions). JSEP currently doesn't mention the ordinality of t= lines, 
but this seems pretty harmless.

Section 5.6 in its present form appears to cover the various "malformed" 
cases I describe above.

____
[1] https://github.com/rtcweb-wg/jsep/issues/27#issuecomment-62786027

/a


From nobody Mon Mar 30 12:18:49 2015
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> On Mar 30, 2015, at 10:09 AM, Adam Roach <adam@nostrum.com> wrote:
>=20
> i=3D, u=3D, e=3D, p=3D, z=3D, r=3D
>  SHOULD NOT send, MUST ignore.

I don't think MUST ignore is correct. First to be clear, I'm not saying =
the browser needs to do anything with the info in theses lines.=20

I think MAY ignore or not typically used by a webrtc browser might be a =
better way to say it. For example, if rtcweb device was taking a =
streaming video session via webrtc protocols, it might make sense for it =
to use the i=3D or u=3D particularly if the SDP had originally come from =
an RTSP stream that was gatewayed to WebRTC. Similarly if a browser =
reported the full SDP it had received in some stats interface, I would =
expect it not to ignore the lines but actually report what it was it =
received.=20

To be clear, I'm not saying the browser needs to do anything with the =
info in theses lines - it's fine to ignore them. But that's very =
different from all webrtc things must ignore them. (and I agree with =
SHOULD NOT send part)=20=


From nobody Mon Mar 30 12:36:22 2015
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Subject: Re: [rtcweb] SDP lines (JSEP Issue 27)
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On 3/30/15 14:18, Cullen Jennings wrote:
>> On Mar 30, 2015, at 10:09 AM, Adam Roach <adam@nostrum.com> wrote:
>>
>> i=, u=, e=, p=, z=, r=
>>   SHOULD NOT send, MUST ignore.
> I don't think MUST ignore is correct. First to be clear, I'm not saying the browser needs to do anything with the info in theses lines.
>
> I think MAY ignore or not typically used by a webrtc browser might be a better way to say it. For example, if rtcweb device was taking a streaming video session via webrtc protocols, it might make sense for it to use the i= or u= particularly if the SDP had originally come from an RTSP stream that was gatewayed to WebRTC. Similarly if a browser reported the full SDP it had received in some stats interface, I would expect it not to ignore the lines but actually report what it was it received.
>
> To be clear, I'm not saying the browser needs to do anything with the info in theses lines - it's fine to ignore them. But that's very different from all webrtc things must ignore them. (and I agree with SHOULD NOT send part)

I think we're in agreement about the principle, but might be at cross 
purposes regarding the venue: I agree that non-browsers may well have 
reasons to use these fields, but I don't think this document has much 
bearing on what those implementations do; at least, not as it currently 
describes itself.

 From the introduction section, JSEP "describes how the W3C WEBRTC 
RTCPeerConnection interface is used to control the setup, management and 
teardown of a multimedia session." So, in the context of *this* 
document, it would *appear* that we're in agreement that the proper 
normative behavior is to ignore the values in these lines. So I can see 
why we might be quibbling about "SHOULD" versus "MUST," but your 
objection seems much broader than that.

If our difference of opinion here arises from differing views about 
intended scope for the JSEP document, then we need to start a broader 
conversation around, minimally, the introduction section of the current 
draft.

/a


From nobody Mon Mar 30 12:54:14 2015
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Justin Uberti <juberti@google.com>, Colin Perkins <csp@csperkins.org>
Thread-Topic: [Suspected Junk Mail] [rtcweb] Endpoints that don't support RTCP
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Talked to a few people and it seems that some of the PSTN gateway manufactu=
res do have ICE and DTLS-SRTP but have no plans to add RTCP to calls that a=
re running at G.711 bandwidth or less. Thought I support that IETF wish tha=
t RTCP was mandatory, I really doubt yet another document saying MUST will =
change anything.=20

My suggested changes to text below.=20



> On Mar 25, 2015, at 11:24 PM, Justin Uberti <juberti@google.com> wrote:
>=20
> This is described in rtp-usage, and the need to limit bitrate is explicit=
ly called out:
>=20
> 7.2.  Congestion Control Interoperability and Legacy Systems
>=20
>=20
>    There are legacy RTP implementations that do not implement RTCP, and
>    hence do not provide any congestion feedback.  Congestion control
>    cannot be performed with these end-points.  WebRTC Endpoints that
>    need to interwork with such end-points MUST limit their transmission
>    to a low rate, equivalent to a VoIP call using a low bandwidth codec,
>    that is unlikely to cause any significant congestion.

I'd might replace "low bandwidth" in above to G.711

>=20
>=20
> Note however that this is incompatible with this section that mandates us=
e of circuit breakers:
>=20
> 7.1.  Boundary Conditions and Circuit Breakers
>=20
>=20
>    WebRTC Endpoints MUST implement the RTP circuit breaker algorithm
>    that is described in [
> I-D.ietf-avtcore-rtp-circuit-breakers
> ].  The
>    RTP circuit breaker is designed to enable applications to recognise
>    and react to situations of extreme network congestion.  However,
>    since the RTP circuit breaker might not be triggered until congestion
>    becomes extreme, it cannot be considered a substitute for congestion
>    control, and applications MUST also implement congestion control to
>    allow them to adapt to changes in network capacity.  Any future RTP
>    congestion control algorithms are expected to operate within the
>    envelope allowed by the circuit breaker.

I'd prefix this paragraph with something like "When using protocols that pr=
ovide congestion feedback " then rest of para. And then something at end li=
ke "When using protocols that do not provide congestion feedback, the bandw=
idth used must be limited to the equivalent of a G.711 call as described si=
on section 7.2"


>=20
>=20
> I think we need to clarify exactly how WebRTC endpoints need to deal with=
 legacy gear that doesn't send RTCP (which might just be to say "no").
>=20
>=20
>=20

We did talk about this in the meetings that lead up to this text and I don'=
t see any reason to need to go back and reopen the consensus from back then=
. The text I'm proposing here is pretty much what's need for real world and=
 requiring RTCP puts you into this awful user experience failure mode where=
 the call is fine for awhile then just randomly disconnects after you have =
been talking for a bit. Users don't like it.=20

One of the points that was brought up in previous discussion about this is =
the following. Imagine a browser (called A)  does webrtc to a media gateway=
 (called B) that terminates ICE and DTLS-SRTP then does SIP / SRTP to a PST=
N gateway (called C). We can assume the PSTN GW supports RTCP.  A and B may=
 be in the one country but C is likely to be in the country of the PSTN num=
ber being dialed, not collocated with B. So you could make a gateway that t=
erminated the RTP media and only reported the packet loss from A to B. But =
the RTP from A to B could be perfect and still having huge problems with RT=
P from B to C. Really what A wants to see is the media quality from A to C.=
 So it is best if the gateway is more of a relay for the RTCP from C to A (=
and visa versa). But here's the catch, at the time B connects to C, it has =
no idea if C does RTCP or not and there is no way to find without waiting u=
ntil the call is well underway then noticing that it never saw any RTCP.=20





From nobody Mon Mar 30 12:56:57 2015
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On 30 March 2015 at 14:54, Cullen Jennings (fluffy) <fluffy@cisco.com> wrot=
e:
>
> Talked to a few people and it seems that some of the PSTN gateway manufac=
tures do have ICE and DTLS-SRTP but have no plans to add RTCP to calls that=
 are running at G.711 bandwidth or less. Thought I support that IETF wish t=
hat RTCP was mandatory, I really doubt yet another document saying MUST wil=
l change anything.

It will in this case, if browsers implement circuit breakers.  Unless
you like really short calls, that is.


From nobody Mon Mar 30 12:59:33 2015
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Subject: Re: [rtcweb] [Suspected Junk Mail] Endpoints that don't support RTCP
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On Mon, Mar 30, 2015 at 9:56 PM, Martin Thomson
<martin.thomson@gmail.com> wrote:
> On 30 March 2015 at 14:54, Cullen Jennings (fluffy) <fluffy@cisco.com> wr=
ote:
>>
>> Talked to a few people and it seems that some of the PSTN gateway manufa=
ctures do have ICE and DTLS-SRTP but have no plans to add RTCP to calls tha=
t are running at G.711 bandwidth or less. Thought I support that IETF wish =
that RTCP was mandatory, I really doubt yet another document saying MUST wi=
ll change anything.
>
> It will in this case, if browsers implement circuit breakers.  Unless
> you like really short calls, that is.

Because, obviously, having circuit breakers is more important than
having your call continue ;)

Emil


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Date: Mon, 30 Mar 2015 15:56:27 -0500
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Subject: Re: [rtcweb] [Suspected Junk Mail] Endpoints that don't support RTCP
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On 30 March 2015 at 14:59, Emil Ivov <emcho@jitsi.org> wrote:
> On Mon, Mar 30, 2015 at 9:56 PM, Martin Thomson
> <martin.thomson@gmail.com> wrote:
>> On 30 March 2015 at 14:54, Cullen Jennings (fluffy) <fluffy@cisco.com> w=
rote:
>>>
>>> Talked to a few people and it seems that some of the PSTN gateway manuf=
actures do have ICE and DTLS-SRTP but have no plans to add RTCP to calls th=
at are running at G.711 bandwidth or less. Thought I support that IETF wish=
 that RTCP was mandatory, I really doubt yet another document saying MUST w=
ill change anything.
>>
>> It will in this case, if browsers implement circuit breakers.  Unless
>> you like really short calls, that is.
>
> Because, obviously, having circuit breakers is more important than
> having your call continue ;)

Indeed.  Given the number of actual calls that don't contain RTCP,
allowing an exception at 100kbps or less (pick your own number, but
this seems about right to me) makes sense.  The alternative is to have
gateways synthesize bogus RTCP, but that only wastes bits.


From nobody Mon Mar 30 14:00:29 2015
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From: Emil Ivov <emcho@jitsi.org>
Date: Mon, 30 Mar 2015 23:00:04 +0200
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Subject: Re: [rtcweb] [Suspected Junk Mail] Endpoints that don't support RTCP
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On Mon, Mar 30, 2015 at 10:56 PM, Martin Thomson
<martin.thomson@gmail.com> wrote:
> On 30 March 2015 at 14:59, Emil Ivov <emcho@jitsi.org> wrote:
>> On Mon, Mar 30, 2015 at 9:56 PM, Martin Thomson
>> <martin.thomson@gmail.com> wrote:
>>> On 30 March 2015 at 14:54, Cullen Jennings (fluffy) <fluffy@cisco.com> =
wrote:
>>>>
>>>> Talked to a few people and it seems that some of the PSTN gateway manu=
factures do have ICE and DTLS-SRTP but have no plans to add RTCP to calls t=
hat are running at G.711 bandwidth or less. Thought I support that IETF wis=
h that RTCP was mandatory, I really doubt yet another document saying MUST =
will change anything.
>>>
>>> It will in this case, if browsers implement circuit breakers.  Unless
>>> you like really short calls, that is.
>>
>> Because, obviously, having circuit breakers is more important than
>> having your call continue ;)
>
> Indeed.  Given the number of actual calls that don't contain RTCP,
> allowing an exception at 100kbps or less (pick your own number, but
> this seems about right to me) makes sense.

Makes sense to me! (and of course, any number would make sense here
given how superfluous circuit breakers are in the presence of consent
checks)

Emil

--=20
https://jitsi.org


From nobody Mon Mar 30 14:06:20 2015
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Subject: Re: [rtcweb] [Suspected Junk Mail] Endpoints that don't support RTCP
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On 30 March 2015 at 16:00, Emil Ivov <emcho@jitsi.org> wrote:
> Makes sense to me! (and of course, any number would make sense here
> given how superfluous circuit breakers are in the presence of consent
> checks)


I chose 100 only because that includes G.711 and every other voice
codec.  Audio that goes above this is special enough that RTCP is
probably worthwhile; video that is in this region already sucks enough
that I'm inclined to limit any exclusion to audio.


From nobody Mon Mar 30 14:10:00 2015
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Martin Thomson <martin.thomson@gmail.com>
Thread-Topic: [rtcweb] [Suspected Junk Mail] Endpoints that don't support RTCP
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Subject: Re: [rtcweb] [Suspected Junk Mail] Endpoints that don't support RTCP
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> On Mar 30, 2015, at 3:06 PM, Martin Thomson <martin.thomson@gmail.com> wr=
ote:
>=20
> On 30 March 2015 at 16:00, Emil Ivov <emcho@jitsi.org> wrote:
>> Makes sense to me! (and of course, any number would make sense here
>> given how superfluous circuit breakers are in the presence of consent
>> checks)
>=20
>=20
> I chose 100 only because that includes G.711 and every other voice
> codec.  Audio that goes above this is special enough that RTCP is
> probably worthwhile; video that is in this region already sucks enough
> that I'm inclined to limit any exclusion to audio.

I agree that 100kbps is plenty for the exception. And I don't think that is=
 going to cause huge congestion problems because if it was going to cause b=
ig problems, we would have already seen them with the current usage of VoIP=
.=20



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On 30 March 2015 at 16:09, Cullen Jennings (fluffy) <fluffy@cisco.com> wrote:
>
>> On Mar 30, 2015, at 3:06 PM, Martin Thomson <martin.thomson@gmail.com> wrote:
>>
>> On 30 March 2015 at 16:00, Emil Ivov <emcho@jitsi.org> wrote:
>>> Makes sense to me! (and of course, any number would make sense here
>>> given how superfluous circuit breakers are in the presence of consent
>>> checks)
>>
>>
>> I chose 100 only because that includes G.711 and every other voice
>> codec.  Audio that goes above this is special enough that RTCP is
>> probably worthwhile; video that is in this region already sucks enough
>> that I'm inclined to limit any exclusion to audio.
>
> I agree that 100kbps is plenty for the exception. And I don't think that is going to cause huge congestion problems because if it was going to cause big problems, we would have already seen them with the current usage of VoIP.

Oh, the other reason is that I don't think that any browser is doing
anything to prevent ICE connectivity checks from being sent at this
rate.


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From: Justin Uberti <juberti@google.com>
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Subject: Re: [rtcweb] [Suspected Junk Mail] Endpoints that don't support RTCP
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--089e013a06062119f70512899bcc
Content-Type: text/plain; charset=UTF-8

In terms of how this is actually implemented, I think it would be simplest
to do something like: "if the circuit breaker fires, ignore it if it is for
an audio stream with a codec bitrate of 64 kbps or less".

I don't think trying to make video without RTCP work makes sense.


On Mon, Mar 30, 2015 at 2:12 PM, Martin Thomson <martin.thomson@gmail.com>
wrote:

> On 30 March 2015 at 16:09, Cullen Jennings (fluffy) <fluffy@cisco.com>
> wrote:
> >
> >> On Mar 30, 2015, at 3:06 PM, Martin Thomson <martin.thomson@gmail.com>
> wrote:
> >>
> >> On 30 March 2015 at 16:00, Emil Ivov <emcho@jitsi.org> wrote:
> >>> Makes sense to me! (and of course, any number would make sense here
> >>> given how superfluous circuit breakers are in the presence of consent
> >>> checks)
> >>
> >>
> >> I chose 100 only because that includes G.711 and every other voice
> >> codec.  Audio that goes above this is special enough that RTCP is
> >> probably worthwhile; video that is in this region already sucks enough
> >> that I'm inclined to limit any exclusion to audio.
> >
> > I agree that 100kbps is plenty for the exception. And I don't think that
> is going to cause huge congestion problems because if it was going to cause
> big problems, we would have already seen them with the current usage of
> VoIP.
>
> Oh, the other reason is that I don't think that any browser is doing
> anything to prevent ICE connectivity checks from being sent at this
> rate.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--089e013a06062119f70512899bcc
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">In terms of how this is actually implemented, I think it w=
ould be simplest to do something like: &quot;if the circuit breaker fires, =
ignore it if it is for an audio stream with a codec bitrate of 64 kbps or l=
ess&quot;.<div><br></div><div>I don&#39;t think trying to make video withou=
t RTCP work makes sense.<br><div><div><br></div></div></div></div><div clas=
s=3D"gmail_extra"><br><div class=3D"gmail_quote">On Mon, Mar 30, 2015 at 2:=
12 PM, Martin Thomson <span dir=3D"ltr">&lt;<a href=3D"mailto:martin.thomso=
n@gmail.com" target=3D"_blank">martin.thomson@gmail.com</a>&gt;</span> wrot=
e:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-l=
eft:1px #ccc solid;padding-left:1ex"><span class=3D"">On 30 March 2015 at 1=
6:09, Cullen Jennings (fluffy) &lt;<a href=3D"mailto:fluffy@cisco.com">fluf=
fy@cisco.com</a>&gt; wrote:<br>
&gt;<br>
&gt;&gt; On Mar 30, 2015, at 3:06 PM, Martin Thomson &lt;<a href=3D"mailto:=
martin.thomson@gmail.com">martin.thomson@gmail.com</a>&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt; On 30 March 2015 at 16:00, Emil Ivov &lt;<a href=3D"mailto:emcho@j=
itsi.org">emcho@jitsi.org</a>&gt; wrote:<br>
&gt;&gt;&gt; Makes sense to me! (and of course, any number would make sense=
 here<br>
&gt;&gt;&gt; given how superfluous circuit breakers are in the presence of =
consent<br>
&gt;&gt;&gt; checks)<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; I chose 100 only because that includes G.711 and every other voice=
<br>
&gt;&gt; codec.=C2=A0 Audio that goes above this is special enough that RTC=
P is<br>
&gt;&gt; probably worthwhile; video that is in this region already sucks en=
ough<br>
&gt;&gt; that I&#39;m inclined to limit any exclusion to audio.<br>
&gt;<br>
&gt; I agree that 100kbps is plenty for the exception. And I don&#39;t thin=
k that is going to cause huge congestion problems because if it was going t=
o cause big problems, we would have already seen them with the current usag=
e of VoIP.<br>
<br>
</span>Oh, the other reason is that I don&#39;t think that any browser is d=
oing<br>
anything to prevent ICE connectivity checks from being sent at this<br>
rate.<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

--089e013a06062119f70512899bcc--


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Subject: Re: [rtcweb] [Suspected Junk Mail] Endpoints that don't support RTCP
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On 30 March 2015 at 18:11, Justin Uberti <juberti@google.com> wrote:
> In terms of how this is actually implemented, I think it would be simplest
> to do something like: "if the circuit breaker fires, ignore it if it is for
> an audio stream with a codec bitrate of 64 kbps or less".

Probably.  We can't really use b= as input to a decision to turn off
circuit breakers, can we?

> I don't think trying to make video without RTCP work makes sense.

Yep.


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Subject: Re: [rtcweb] [Suspected Junk Mail] Endpoints that don't support RTCP
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--001a11c35492034f26051289d495
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On Mon, Mar 30, 2015 at 4:16 PM, Martin Thomson <martin.thomson@gmail.com>
wrote:

> On 30 March 2015 at 18:11, Justin Uberti <juberti@google.com> wrote:
> > In terms of how this is actually implemented, I think it would be
> simplest
> > to do something like: "if the circuit breaker fires, ignore it if it is
> for
> > an audio stream with a codec bitrate of 64 kbps or less".
>
> Probably.  We can't really use b= as input to a decision to turn off
> circuit breakers, can we?
>

If we are generating a 512 kbps Opus stream, we could conceivably make a
different decision than a plain PCMU stream. But as that is a pathological
case, ignoring circuit breakers for all audio streams would probably be
fine.

>
> > I don't think trying to make video without RTCP work makes sense.
>
> Yep.
>

--001a11c35492034f26051289d495
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Mon, Mar 30, 2015 at 4:16 PM, Martin Thomson <span dir=3D"ltr">&lt;<=
a href=3D"mailto:martin.thomson@gmail.com" target=3D"_blank">martin.thomson=
@gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span cl=
ass=3D"">On 30 March 2015 at 18:11, Justin Uberti &lt;<a href=3D"mailto:jub=
erti@google.com">juberti@google.com</a>&gt; wrote:<br>
&gt; In terms of how this is actually implemented, I think it would be simp=
lest<br>
&gt; to do something like: &quot;if the circuit breaker fires, ignore it if=
 it is for<br>
&gt; an audio stream with a codec bitrate of 64 kbps or less&quot;.<br>
<br>
</span>Probably.=C2=A0 We can&#39;t really use b=3D as input to a decision =
to turn off<br>
circuit breakers, can we?<br></blockquote><div><br></div><div>If we are gen=
erating a 512 kbps Opus stream, we could conceivably make a different decis=
ion than a plain PCMU stream. But as that is a pathological case, ignoring =
circuit breakers for all audio streams would probably be fine.</div><blockq=
uote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc =
solid;padding-left:1ex">
<span class=3D""><br>
&gt; I don&#39;t think trying to make video without RTCP work makes sense.<=
br>
<br>
</span>Yep.<br>
</blockquote></div><br></div></div>

--001a11c35492034f26051289d495--


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On 30 March 2015 at 16:27, Justin Uberti <juberti@google.com> wrote:
> If we are generating a 512 kbps Opus stream, we could conceivably make a
> different decision than a plain PCMU stream. But as that is a pathological
> case, ignoring circuit breakers for all audio streams would probably be
> fine.

I think that I'd be more inclined to use whatever signal we got
telling us that 512kbps was needed to also tell us that circuit
breakers is.  I don't know what that signal is right now, BTW, but we
we are going to be sending at that rate, we presumably need to be told
to do so somehow.  Again, I think that b= is the wrong hook.


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] [Suspected Junk Mail] Endpoints that don't support RTCP
Thread-Index: AQHQay5I9Bna8X1CCUOzBj+Wfz9AYp01+yWAgAAW2QA=
Date: Tue, 31 Mar 2015 00:33:03 +0000
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Subject: Re: [rtcweb] [Suspected Junk Mail] Endpoints that don't support RTCP
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> On Mar 30, 2015, at 5:11 PM, Justin Uberti <juberti@google.com> wrote:
>=20
> In terms of how this is actually implemented, I think it would be simples=
t to do something like: "if the circuit breaker fires, ignore it if it is f=
or an audio stream with a codec bitrate of 64 kbps or less".
>=20
> I don't think trying to make video without RTCP work makes sense.
>=20

That would be fine with me - or the < 100kbps formulation. Practically they=
 end up about the same so either works for me.=20



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From: Tim Panton <tim@phonefromhere.com>
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To: Emil Ivov <emcho@jitsi.org>
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Subject: Re: [rtcweb] [Suspected Junk Mail] Endpoints that don't support RTCP
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> On 30 Mar 2015, at 22:00, Emil Ivov <emcho@jitsi.org =
<mailto:emcho@jitsi.org>> wrote:
>=20
> On Mon, Mar 30, 2015 at 10:56 PM, Martin Thomson
> <martin.thomson@gmail.com <mailto:martin.thomson@gmail.com>> wrote:
>> On 30 March 2015 at 14:59, Emil Ivov <emcho@jitsi.org =
<mailto:emcho@jitsi.org>> wrote:
>>> On Mon, Mar 30, 2015 at 9:56 PM, Martin Thomson
>>> <martin.thomson@gmail.com <mailto:martin.thomson@gmail.com>> wrote:
>>>> On 30 March 2015 at 14:54, Cullen Jennings (fluffy) =
<fluffy@cisco.com <mailto:fluffy@cisco.com>> wrote:
>>>>>=20
>>>>> Talked to a few people and it seems that some of the PSTN gateway =
manufactures do have ICE and DTLS-SRTP but have no plans to add RTCP to =
calls that are running at G.711 bandwidth or less. Thought I support =
that IETF wish that RTCP was mandatory, I really doubt yet another =
document saying MUST will change anything.
>>>>=20
>>>> It will in this case, if browsers implement circuit breakers.  =
Unless
>>>> you like really short calls, that is.
>>>=20
>>> Because, obviously, having circuit breakers is more important than
>>> having your call continue ;)
>>=20
>> Indeed.  Given the number of actual calls that don't contain RTCP,
>> allowing an exception at 100kbps or less (pick your own number, but
>> this seems about right to me) makes sense.
>=20
> Makes sense to me! (and of course, any number would make sense here
> given how superfluous circuit breakers are in the presence of consent
> checks)

It looks to me as if Firefox nightly has stopped doing consent checks, =
so
we have a way to see how well that turns out - badly so far this morning =
;-(


T.

>=20
> Emil
>=20
> --=20
> https://jitsi.org <https://jitsi.org/>
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk <http://www.westhawk.co.uk/>




--Apple-Mail=_FBA1CDB7-4448-4881-B9CC-454FFBBC86B8
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dus-ascii"><meta http-equiv=3D"Content-Type" content=3D"text/html=
 charset=3Dus-ascii"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" =
class=3D""><br class=3D""><div class=3D""><blockquote type=3D"cite" =
class=3D""><div class=3D"">On 30 Mar 2015, at 22:00, Emil Ivov &lt;<a =
href=3D"mailto:emcho@jitsi.org" class=3D"">emcho@jitsi.org</a>&gt; =
wrote:</div><br class=3D"Apple-interchange-newline"><div class=3D"">On =
Mon, Mar 30, 2015 at 10:56 PM, Martin Thomson<br class=3D"">&lt;<a =
href=3D"mailto:martin.thomson@gmail.com" =
class=3D"">martin.thomson@gmail.com</a>&gt; wrote:<br =
class=3D""><blockquote type=3D"cite" class=3D"">On 30 March 2015 at =
14:59, Emil Ivov &lt;<a href=3D"mailto:emcho@jitsi.org" =
class=3D"">emcho@jitsi.org</a>&gt; wrote:<br class=3D""><blockquote =
type=3D"cite" class=3D"">On Mon, Mar 30, 2015 at 9:56 PM, Martin =
Thomson<br class=3D"">&lt;<a href=3D"mailto:martin.thomson@gmail.com" =
class=3D"">martin.thomson@gmail.com</a>&gt; wrote:<br =
class=3D""><blockquote type=3D"cite" class=3D"">On 30 March 2015 at =
14:54, Cullen Jennings (fluffy) &lt;<a href=3D"mailto:fluffy@cisco.com" =
class=3D"">fluffy@cisco.com</a>&gt; wrote:<br class=3D""><blockquote =
type=3D"cite" class=3D""><br class=3D"">Talked to a few people and it =
seems that some of the PSTN gateway manufactures do have ICE and =
DTLS-SRTP but have no plans to add RTCP to calls that are running at =
G.711 bandwidth or less. Thought I support that IETF wish that RTCP was =
mandatory, I really doubt yet another document saying MUST will change =
anything.<br class=3D""></blockquote><br class=3D"">It will in this =
case, if browsers implement circuit breakers. &nbsp;Unless<br =
class=3D"">you like really short calls, that is.<br =
class=3D""></blockquote><br class=3D"">Because, obviously, having =
circuit breakers is more important than<br class=3D"">having your call =
continue ;)<br class=3D""></blockquote><br class=3D"">Indeed. =
&nbsp;Given the number of actual calls that don't contain RTCP,<br =
class=3D"">allowing an exception at 100kbps or less (pick your own =
number, but<br class=3D"">this seems about right to me) makes sense.<br =
class=3D""></blockquote><br class=3D"">Makes sense to me! (and of =
course, any number would make sense here<br class=3D"">given how =
superfluous circuit breakers are in the presence of consent<br =
class=3D"">checks)<br class=3D""></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D"">It looks to me as if Firefox nightly =
has stopped doing consent checks, so</div><div class=3D"">we have a way =
to see how well that turns out - badly so far this morning ;-(</div><div =
class=3D""><br class=3D""></div><div class=3D""><br =
class=3D""></div>T.</div><div class=3D""><br class=3D""><blockquote =
type=3D"cite" class=3D""><div class=3D""><br class=3D"">Emil<br =
class=3D""><br class=3D"">-- <br class=3D""><a href=3D"https://jitsi.org" =
class=3D"">https://jitsi.org</a><br class=3D""><br =
class=3D"">_______________________________________________<br =
class=3D"">rtcweb mailing list<br class=3D""><a =
href=3D"mailto:rtcweb@ietf.org" class=3D"">rtcweb@ietf.org</a><br =
class=3D"">https://www.ietf.org/mailman/listinfo/rtcweb<br =
class=3D""></div></blockquote></div><br class=3D""><div =
apple-content-edited=3D"true" class=3D"">
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; =
font-style: normal; font-variant: normal; font-weight: normal; =
letter-spacing: normal; line-height: normal; orphans: 2; text-align: =
auto; text-indent: 0px; text-transform: none; white-space: normal; =
widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0; "><div class=3D"">Tim Panton - =
Web/VoIP consultant and implementor</div><div class=3D""><a =
href=3D"http://www.westhawk.co.uk" =
class=3D"">www.westhawk.co.uk</a></div><div class=3D""><br =
class=3D""></div></span><br class=3D"Apple-interchange-newline">

</div>
<br class=3D""></body></html>=

--Apple-Mail=_FBA1CDB7-4448-4881-B9CC-454FFBBC86B8--


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On 03/31/2015 01:33 AM, Martin Thomson wrote:
> On 30 March 2015 at 16:27, Justin Uberti <juberti@google.com> wrote:
>> If we are generating a 512 kbps Opus stream, we could conceivably make a
>> different decision than a plain PCMU stream. But as that is a pathological
>> case, ignoring circuit breakers for all audio streams would probably be
>> fine.
> I think that I'd be more inclined to use whatever signal we got
> telling us that 512kbps was needed to also tell us that circuit
> breakers is.

Wouldn't it be simpler to tell that circuit breakers are needed from the
fact that more than 100 kilobits are being sent within 1 second? (or 1
Mbit within 10 seconds?)

I haven't yet seen a high-level API that doesn't offer to count the
outgoing bits.


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From: Martin Thomson <martin.thomson@gmail.com>
To: Tim Panton <tim@phonefromhere.com>
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Cc: "Cullen Jennings \(fluffy\)" <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, Colin Perkins <csp@csperkins.org>
Subject: Re: [rtcweb] [Suspected Junk Mail] Endpoints that don't support RTCP
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On 31 March 2015 at 03:14, Tim Panton <tim@phonefromhere.com> wrote:
> It looks to me as if Firefox nightly has stopped doing consent checks, so
> we have a way to see how well that turns out - badly so far this morning ;-(

Here: https://bugzilla.mozilla.org/

FWIW, Firefox has never done continuous consent.  The initial consent
to send is all it has ever had in that area.


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Subject: Re: [rtcweb] [Suspected Junk Mail] Endpoints that don't support RTCP
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On 31 March 2015 at 05:16, Harald Alvestrand <harald@alvestrand.no> wrote:
> Wouldn't it be simpler to tell that circuit breakers are needed from the
> fact that more than 100 kilobits are being sent within 1 second? (or 1
> Mbit within 10 seconds?)

That works too.

