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From: "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>
To: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-03.txt
Thread-Index: AQHPfkdvMAvn6ASrHUWO6QS4xLiKqg==
Date: Mon, 2 Jun 2014 09:45:49 +0000
Message-ID: <CFB23ABB.8F1BB%rmohanr@cisco.com>
References: <20140519083842.21555.8626.idtracker@ietfa.amsl.com> <CF9FC965.8D5C1%rmohanr@cisco.com> <CFA0E4F6.286D3%eckelcu@cisco.com>
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Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/IlCU8hpgCGqUTFB3Y3tY2HdYrKo
Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-03.txt
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From nobody Mon Jun  2 10:13:57 2014
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From: Harald Alvestrand <harald@alvestrand.no>
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Subject: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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I believe this has been discussed before, but I can't find the 
discussion...... so I'm re-raising it.
(it may have happened here, or on MMUSIC, but I can't find it in either 
place).

I believe this WG has consensus that ICE is used, that ICE-TCP 
candidates are permitted (indeed, required to be supported), and that 
TURN over TCP is permitted; in the latter two cases, RFC 4571 framing is 
used, but SRTP is used, not TLS.

Given the nature of ICE, flows may shift back and forth between TCP 
candidates and UDP candidates over the lifetime of the flow, without any 
new SDP negotiation being invoked.

A bug has been raised about what protocol field to be used in the SDP 
for these flows.

RFC 5764 says:


8.  Session Description for RTP/SAVP over DTLS

    This specification defines new tokens to describe the protocol used
    in SDP media descriptions ("m=" lines and their associated
    parameters).  The new values defined for the proto field are:

    o  When a RTP/SAVP or RTP/SAVPF [RFC5124] stream is transported over
       DTLS with the Datagram Congestion Control Protocol (DCCP), then
       the token SHALL be DCCP/TLS/RTP/SAVP or DCCP/TLS/RTP/SAVPF
       respectively.

    o  When a RTP/SAVP or RTP/SAVPF stream is transported over DTLS with
       UDP, the token SHALL be UDP/TLS/RTP/SAVP or UDP/TLS/RTP/SAVPF
       respectively.

    The "fmt" parameter SHALL be as defined for RTP/SAVP.

    See [RFC5763] for how to use offer/answer with DTLS-SRTP.

    This document does not specify how to protect RTP data transported
    over TCP.  Potential approaches include carrying the RTP over TLS
    over TCP (see [SRTP-NOT-MAND]) or using a mechanism similar to that
    in this document over TCP, either via TLS or DTLS, with DTLS being
    used for consistency between reliable and unreliable transports. In
    the latter case, it would be necessary to profile DTLS so that
    fragmentation and retransmissions no longer occurred.  In either
    case, a new document would be required.


The current implementations both send "RTP/SAVPF" in this field. But 
this usage is not specified by RFC 5764.

Using the specified token, UDP/TLS/RTP/SAVPF, speaks a partial untruth, 
in that sometimes, UDP is not used. But it is a token people have seen 
before.

What says the wisdom of the group? What shall we put in the spec?

Harald





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<html>
  <head>

    <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    I believe this has been discussed before, but I can't find the
    discussion...... so I'm re-raising it.<br>
    (it may have happened here, or on MMUSIC, but I can't find it in
    either place).<br>
    <br>
    I believe this WG has consensus that ICE is used, that ICE-TCP
    candidates are permitted (indeed, required to be supported), and
    that TURN over TCP is permitted; in the latter two cases, RFC 4571
    framing is used, but SRTP is used, not TLS.<br>
    <br>
    Given the nature of ICE, flows may shift back and forth between TCP
    candidates and UDP candidates over the lifetime of the flow, without
    any new SDP negotiation being invoked.<br>
    <br>
    A bug has been raised about what protocol field to be used in the
    SDP for these flows.<br>
    <br>
    RFC 5764 says:<br>
    <br>
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    <br>
    8.&nbsp; Session Description for RTP/SAVP over DTLS<br>
    <br>
    &nbsp;&nbsp; This specification defines new tokens to describe the protocol
    used<br>
    &nbsp;&nbsp; in SDP media descriptions ("m=" lines and their associated<br>
    &nbsp;&nbsp; parameters).&nbsp; The new values defined for the proto field are:<br>
    <br>
    &nbsp;&nbsp; o&nbsp; When a RTP/SAVP or RTP/SAVPF [RFC5124] stream is transported
    over<br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; DTLS with the Datagram Congestion Control Protocol (DCCP),
    then<br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; the token SHALL be DCCP/TLS/RTP/SAVP or DCCP/TLS/RTP/SAVPF<br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; respectively.<br>
    <br>
    &nbsp;&nbsp; o&nbsp; When a RTP/SAVP or RTP/SAVPF stream is transported over DTLS
    with<br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; UDP, the token SHALL be UDP/TLS/RTP/SAVP or UDP/TLS/RTP/SAVPF<br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; respectively.<br>
    <br>
    &nbsp;&nbsp; The "fmt" parameter SHALL be as defined for RTP/SAVP.<br>
    <br>
    &nbsp;&nbsp; See [RFC5763] for how to use offer/answer with DTLS-SRTP.<br>
    <br>
    &nbsp;&nbsp; This document does not specify how to protect RTP data
    transported<br>
    &nbsp;&nbsp; over TCP.&nbsp; Potential approaches include carrying the RTP over TLS<br>
    &nbsp;&nbsp; over TCP (see [SRTP-NOT-MAND]) or using a mechanism similar to
    that<br>
    &nbsp;&nbsp; in this document over TCP, either via TLS or DTLS, with DTLS
    being<br>
    &nbsp;&nbsp; used for consistency between reliable and unreliable transports.&nbsp;
    In<br>
    &nbsp;&nbsp; the latter case, it would be necessary to profile DTLS so that<br>
    &nbsp;&nbsp; fragmentation and retransmissions no longer occurred.&nbsp; In either<br>
    &nbsp;&nbsp; case, a new document would be required.<br>
    <br>
    <br>
    The current implementations both send "RTP/SAVPF" in this field. But
    this usage is not specified by RFC 5764.<br>
    <br>
    Using the specified token, UDP/TLS/RTP/SAVPF, speaks a partial
    untruth, in that sometimes, UDP is not used. But it is a token
    people have seen before.<br>
    <br>
    What says the wisdom of the group? What shall we put in the spec?<br>
    <br>
    Harald<br>
    <br>
    <br>
    <br>
    <br>
  </body>
</html>

--------------090009010206020701040405--


From nobody Mon Jun  2 11:43:33 2014
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From: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-03.txt
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On 2 June 2014 02:45, Ram Mohan R (rmohanr) <rmohanr@cisco.com> wrote:
> <charles said>
> I should have also stated that this would apply only for cases in which
> such use of multiple DSCPs was appropriate, as determined by DART.
> </charles>
>
>
>  We did discuss this earlier and there was some support to use highest
> priority DSCP in case of BUNDLE. Here was the previous thread -
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg11777.html
> I did not see any conclusion though on that topic.
>
>
> I am ok to have STUN packets to have highest DSCP when multiple DSCP
> values are used for a given 5-tuple (like in DART). The only issue I see
> is when there is a congestion then your STUN consent check packets would
> get preference over some of the other packets. I think this is ok as
> dropping a consent message means the sender is likely to stop sending
> media itself,  if it does not see a consent refresh response.
>
> I will add the following text. Does this look ok ?
>
> "It is possible that different Diffserv Codepoints are used by
>    different media over the same transport address as described in
>    [I-D.ietf-tsvwg-rtcweb-qos].  In such cases, it is recommended to use
> the same DSCP for the STUN messages
> as that used for the highest priority media stream(s)."

I know that I haven't raised this point before, and I apologize for
not paying enough attention, but I don't think that we should be
talking about DSCP in this draft.

Use of DSCP with STUN and the additional complications that BUNGLE
adds are best dealt with in those contexts.  This was a simple
document, we should keep the scope constrained as much as possible.


From nobody Mon Jun  2 11:54:09 2014
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Subject: Re: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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On 2 June 2014 10:13, Harald Alvestrand <harald@alvestrand.no> wrote:
> The current implementations both send "RTP/SAVPF" in this field. But this
> usage is not specified by RFC 5764.

Sounds easy to me.  The SDP doesn't matter and can say anything.
"RTP/SAVPF" is as good as any other sequence of octets.

I'm sure that JSEP can include a requirement that is directly
contradictory to an existing RFC.  That happens all the time.  At
least this time it can be explicit.


From nobody Mon Jun  2 12:51:42 2014
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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2014-06-02 19:13 GMT+02:00 Harald Alvestrand <harald@alvestrand.no>:
> The current implementations both send "RTP/SAVPF" in this field. But this
> usage is not specified by RFC 5764.
>
> Using the specified token, UDP/TLS/RTP/SAVPF, speaks a partial untruth, i=
n
> that sometimes, UDP is not used. But it is a token people have seen befor=
e.
>
> What says the wisdom of the group? What shall we put in the spec?

I don't think SDP is ready or enough powerful to allow this case in
which different transports can be used. The "UDP/TLS/RTP/SAVPF" token
is just that: a token. And there is no reason at all to mandate that
the m=3D line defines the transport to use given that, at least in
WebRTC, the transport is determined by the ICE candidates. The
"transport" in the m=3D line is as useless as the "port" field in the m=3D
line when ICE is mandated.

RFC 5764 "DTLS Extension to Establish Keys for SRTP" clearly states
that "there is no chance" to allow two different transports for the
same media:

-----------------------------
8.  Session Description for RTP/SAVP over DTLS

   This specification defines new tokens to describe the protocol used
   in SDP media descriptions ("m=3D" lines and their associated
   parameters).  The new values defined for the proto field are:

   o  When a RTP/SAVP or RTP/SAVPF [RFC5124] stream is transported over
      DTLS with the Datagram Congestion Control Protocol (DCCP), then
      the token SHALL be DCCP/TLS/RTP/SAVP or DCCP/TLS/RTP/SAVPF
      respectively.

   o  When a RTP/SAVP or RTP/SAVPF stream is transported over DTLS with
      UDP, the token SHALL be UDP/TLS/RTP/SAVP or UDP/TLS/RTP/SAVPF
      respectively.
------------------------------


So basically SDP is not valid for WebRTC, but since it has been forced
to be valid we need a workaround to hide this fact. I would propose
that the m=3D line "transport" for WebRTC MUST always be the token
"UDP/TLS/RTP/SAVPF" regardless the nature of transports in the ICE
candidates (so basically it becomes a useless but mandatory field in
the SDP, as many others already). Said that I would also be happy if
the chosen token is "CHICKEN".


Regards.


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


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A1. Section 4, 1st paragraph

I think the full name of the protocol better establishes context. It
should be used as a section's first reference to the protocol. Then
subsequent references may be less explicit. (see A4)

I would change "This protocol..." to "The Data Channel Establishment
Protocol..."

A2. Section 4, first bullet

Should be simplified to read: Reliable or unreliable message
transmission. In the case of unreliable transmissions, the same level
of unreliability is used.

A3. Section 4, second bullet

Should be simplified to read: In-order or out-of- order message
delivery.

A4. Section 4, third paragraph

Change the subsequent reference of "The Data Channel Establishment
Protocol..." to "This protocol..."

A5. Section 4, third paragraph

"...uses a two way handshake to open a data channel by combining two
SCTP streams, one in each direction, with the same SCTP stream
identifier."

For more clarity change to:

"...uses a two way handshake to open a data channel. The handshake
pairs one incoming and one outgoing SCTP stream, each having an
identical stream identifier, into a single bidirectional channel."

A6. Section 4, third paragraph

I think the sentence that begins "Please note" should be separated out
into a new paragraph and labeled as a note as follows:

Note: The opening side can only send user messages before the
DATA_CHANNEL_ACK is received.

A7. Section 4, fourth paragraph

I think we should formally reference the title of Internet- Drafts
and then link to their location, rather than using the link as the
subject of sentences. I would change the sentence that begins "When
using [I-D.ietf-tsvwg-sctp-dtls-encaps], the method..." to the
following:

=E2=80=9CWhen using SCTP over DTLS [I-D.ietf-tsvwg-sctp-dtls-encaps], the m=
ethod=E2=80=A6=E2=80=9D


--=20
Doug Leonard
Developer  |  Jive Communications, Inc.
Jive.com  |  dleonard at getjive . com

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<br><div><br></div><div>A1. Section 4, 1st paragraph=C2=A0</div><div><br></=
div><div>I think the full name of the protocol better establishes context. =
It</div><div>should be used as a section&#39;s first reference to the proto=
col. Then</div>
<div>subsequent references may be less explicit. (see A4)=C2=A0</div><div><=
br></div><div>I would change &quot;This protocol...&quot; to &quot;The Data=
 Channel Establishment</div><div>Protocol...&quot;</div><div><br></div><div=
>
A2. Section 4, first bullet=C2=A0</div><div><br></div><div>Should be simpli=
fied to read: Reliable or unreliable message</div><div>transmission. In the=
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y is used.</div>
<div><br></div><div>A3. Section 4, second bullet=C2=A0</div><div><br></div>=
<div>Should be simplified to read: In-order or out-of- order message</div><=
div>delivery.</div><div><br></div><div>A4. Section 4, third paragraph=C2=A0=
</div>
<div><br></div><div>Change the subsequent reference of &quot;The Data Chann=
el Establishment</div><div>Protocol...&quot; to &quot;This protocol...&quot=
;</div><div><br></div><div>A5. Section 4, third paragraph=C2=A0</div><div><=
br>
</div><div>&quot;...uses a two way handshake to open a data channel by comb=
ining two</div><div>SCTP streams, one in each direction, with the same SCTP=
 stream</div><div>identifier.&quot;</div><div><br></div><div>For more clari=
ty change to:</div>
<div><br></div><div>&quot;...uses a two way handshake to open a data channe=
l. The handshake</div><div>pairs one incoming and one outgoing SCTP stream,=
 each having an</div><div>identical stream identifier, into a single bidire=
ctional channel.&quot;</div>
<div><br></div><div>A6. Section 4, third paragraph=C2=A0</div><div><br></di=
v><div>I think the sentence that begins &quot;Please note&quot; should be s=
eparated out</div><div>into a new paragraph and labeled as a note as follow=
s:</div>
<div><br></div><div>Note: The opening side can only send user messages befo=
re the</div><div>DATA_CHANNEL_ACK is received.</div><div><br></div><div>A7.=
 Section 4, fourth paragraph=C2=A0</div><div><br></div><div>I think we shou=
ld formally reference the title of Internet- Drafts</div>
<div>and then link to their location, rather than using the link as the</di=
v><div>subject of sentences. I would change the sentence that begins &quot;=
When</div><div>using [I-D.ietf-tsvwg-sctp-dtls-encaps], the method...&quot;=
 to the</div>
<div>following:</div><div><br></div><div>=E2=80=9CWhen using SCTP over DTLS=
 [I-D.ietf-tsvwg-sctp-dtls-encaps], the method=E2=80=A6=E2=80=9D</div><div>=
<br></div><div><br></div>-- <br><div dir=3D"ltr"><div>Doug Leonard</div><di=
v>Developer =C2=A0| =C2=A0Jive Communications, Inc.</div>
<div>Jive.com =C2=A0| =C2=A0dleonard at getjive . com</div></div>
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I think there is value in specifying UDP/TLS/RTP/SAVPF, in that it
unambiguously indicates the desire to do DTLS-SRTP. Otherwise, one has to
guess at this from the presence of a=fingerprint, which seems dodgy, given
that endpoints are free to ignore SDP attributes they don't understand.

The fact that UDP and TLS appear in the m= line is kind of unfortunate, but
I think it can be interpreted as "do DTLS-SRTP".


On Mon, Jun 2, 2014 at 11:53 AM, Martin Thomson <martin.thomson@gmail.com>
wrote:

> On 2 June 2014 10:13, Harald Alvestrand <harald@alvestrand.no> wrote:
> > The current implementations both send "RTP/SAVPF" in this field. But this
> > usage is not specified by RFC 5764.
>
> Sounds easy to me.  The SDP doesn't matter and can say anything.
> "RTP/SAVPF" is as good as any other sequence of octets.
>
> I'm sure that JSEP can include a requirement that is directly
> contradictory to an existing RFC.  That happens all the time.  At
> least this time it can be explicit.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">I think there is value in specifying UDP/TLS/RTP/SAVPF, in=
 that it unambiguously indicates the desire to do DTLS-SRTP. Otherwise, one=
 has to guess at this from the presence of a=3Dfingerprint, which seems dod=
gy, given that endpoints are free to ignore SDP attributes they don&#39;t u=
nderstand.<div>

<br></div><div>The fact that UDP and TLS appear in the m=3D line is kind of=
 unfortunate, but I think it can be interpreted as &quot;do DTLS-SRTP&quot;=
.</div></div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">=
On Mon, Jun 2, 2014 at 11:53 AM, Martin Thomson <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:martin.thomson@gmail.com" target=3D"_blank">martin.thomson@gma=
il.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"">On 2 June 2014 10:13, Harald=
 Alvestrand &lt;<a href=3D"mailto:harald@alvestrand.no">harald@alvestrand.n=
o</a>&gt; wrote:<br>


&gt; The current implementations both send &quot;RTP/SAVPF&quot; in this fi=
eld. But this<br>
&gt; usage is not specified by RFC 5764.<br>
<br>
</div>Sounds easy to me. =C2=A0The SDP doesn&#39;t matter and can say anyth=
ing.<br>
&quot;RTP/SAVPF&quot; is as good as any other sequence of octets.<br>
<br>
I&#39;m sure that JSEP can include a requirement that is directly<br>
contradictory to an existing RFC. =C2=A0That happens all the time. =C2=A0At=
<br>
least this time it can be explicit.<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>

--047d7b86c2668e52ae04fae849d5--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Continous consent when no media is sent
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Subject: [rtcweb] Continous consent when no media is sent
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Hi,

Section 4.1 of draft-ietf-rtcweb-stun-consent-freshness-03 says:

"An endpoint that is only receiving application traffic (recvonly)
                does not need to obtain consent which can slightly conserve=
 its
                resources."

I think it would be good to be more generic, and talk about cases where the=
 sender does not send media. That is obviously true in case of "recvonly", =
but it could also be true in other cases.

In addition, I guess the no-need-to-send-consent rule also applies to "inac=
tive" streams.

Regards,

Christer

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<div class=3D"WordSection1">
<p class=3D"MsoNormal">Hi,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Section 4.1 of draft-ietf-rtcweb-stun-consent-freshn=
ess-03 says:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">&#8220;An endpoint that=
 is only receiving application traffic (recvonly)<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; does not need to obtain consent which can=
 slightly conserve its<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; resources.&#8221;<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I think it would be good to be more generic, and tal=
k about cases where the sender does not send media. That is obviously true =
in case of &#8220;recvonly&#8221;, but it could also be true in other cases=
.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">In addition, I guess the no-need-to-send-consent rul=
e also applies to &#8220;inactive&#8221; streams.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Christer<o:p></o:p></p>
</div>
</body>
</html>

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Subject: Re: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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On the other side, keeping "RTP/SAVFP" would have allowed a server 
(SBC/MCU/etc) to send an offer without knowing in advance if the other 
side supports SDES or DTLS, by sending both the crypto and fingerprint 
attributes and let the receiver choose which one to use (as chrome has 
done until recently). I know that it is not important for webrtc, as 
DTLS is mandatory and SDES is a must not, but for supporting other 
legacy clients it may be easier for some of us.

Best regards
Sergio

El 03/06/2014 8:04, Justin Uberti escribió:
> I think there is value in specifying UDP/TLS/RTP/SAVPF, in that it 
> unambiguously indicates the desire to do DTLS-SRTP. Otherwise, one has 
> to guess at this from the presence of a=fingerprint, which seems 
> dodgy, given that endpoints are free to ignore SDP attributes they 
> don't understand.
>
> The fact that UDP and TLS appear in the m= line is kind of 
> unfortunate, but I think it can be interpreted as "do DTLS-SRTP".
>
>
> On Mon, Jun 2, 2014 at 11:53 AM, Martin Thomson 
> <martin.thomson@gmail.com <mailto:martin.thomson@gmail.com>> wrote:
>
>     On 2 June 2014 10:13, Harald Alvestrand <harald@alvestrand.no
>     <mailto:harald@alvestrand.no>> wrote:
>     > The current implementations both send "RTP/SAVPF" in this field.
>     But this
>     > usage is not specified by RFC 5764.
>
>     Sounds easy to me.  The SDP doesn't matter and can say anything.
>     "RTP/SAVPF" is as good as any other sequence of octets.
>
>     I'm sure that JSEP can include a requirement that is directly
>     contradictory to an existing RFC.  That happens all the time.  At
>     least this time it can be explicit.
>
>     _______________________________________________
>     rtcweb mailing list
>     rtcweb@ietf.org <mailto: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 class="moz-cite-prefix">On the other side, keeping "RTP/SAVFP"
      would have allowed a server (SBC/MCU/etc) to send an offer without
      knowing in advance if the other side supports SDES or DTLS, by
      sending both the crypto and fingerprint attributes and let the
      receiver choose which one to use (as chrome has done until
      recently). I know that it is not important for webrtc, as DTLS is
      mandatory and SDES is a must not, but for supporting other legacy
      clients it may be easier for some of us. <br>
      <br>
      Best regards<br>
      Sergio<br>
      <br>
      El 03/06/2014 8:04, Justin Uberti escribi&oacute;:<br>
    </div>
    <blockquote
cite="mid:CAOJ7v-2U0vMQ5xgOkLR1LdG8NPMo=mC8qiUQaVcm4A0X+hq3kQ@mail.gmail.com"
      type="cite">
      <div dir="ltr">I think there is value in specifying
        UDP/TLS/RTP/SAVPF, in that it unambiguously indicates the desire
        to do DTLS-SRTP. Otherwise, one has to guess at this from the
        presence of a=fingerprint, which seems dodgy, given that
        endpoints are free to ignore SDP attributes they don't
        understand.
        <div>
          <br>
        </div>
        <div>The fact that UDP and TLS appear in the m= line is kind of
          unfortunate, but I think it can be interpreted as "do
          DTLS-SRTP".</div>
      </div>
      <div class="gmail_extra"><br>
        <br>
        <div class="gmail_quote">On Mon, Jun 2, 2014 at 11:53 AM, Martin
          Thomson <span dir="ltr">&lt;<a moz-do-not-send="true"
              href="mailto:martin.thomson@gmail.com" target="_blank">martin.thomson@gmail.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 class="">On 2 June 2014 10:13, Harald Alvestrand &lt;<a
                moz-do-not-send="true"
                href="mailto:harald@alvestrand.no">harald@alvestrand.no</a>&gt;
              wrote:<br>
              &gt; The current implementations both send "RTP/SAVPF" in
              this field. But this<br>
              &gt; usage is not specified by RFC 5764.<br>
              <br>
            </div>
            Sounds easy to me. &nbsp;The SDP doesn't matter and can say
            anything.<br>
            "RTP/SAVPF" is as good as any other sequence of octets.<br>
            <br>
            I'm sure that JSEP can include a requirement that is
            directly<br>
            contradictory to an existing RFC. &nbsp;That happens all the
            time. &nbsp;At<br>
            least this time it can be explicit.<br>
            <br>
            _______________________________________________<br>
            rtcweb mailing list<br>
            <a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
            <a moz-do-not-send="true"
              href="https://www.ietf.org/mailman/listinfo/rtcweb"
              target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
          </blockquote>
        </div>
        <br>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
  </body>
</html>

--------------090302010602070801010801--


From nobody Tue Jun  3 02:08:20 2014
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Date: Tue, 3 Jun 2014 11:08:07 +0200
From: Lorenzo Miniero <lorenzo@meetecho.com>
To: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
Message-ID: <20140603110807.0baf1982@lminiero>
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Subject: Re: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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On Tue, 03 Jun 2014 10:32:43 +0200
Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com> wrote:

> On the other side, keeping "RTP/SAVFP" would have allowed a server=20
> (SBC/MCU/etc) to send an offer without knowing in advance if the
> other side supports SDES or DTLS, by sending both the crypto and
> fingerprint attributes and let the receiver choose which one to use
> (as chrome has done until recently). I know that it is not important
> for webrtc, as DTLS is mandatory and SDES is a must not, but for
> supporting other legacy clients it may be easier for some of us.
>=20
> Best regards
> Sergio
>=20


I guess that was the rationale for having RTP/SAVPF in the first
place, as initially both SDES and DTLS-SRTP were allowed. There are
people confused about not having UDP/TLS/RTP/SAVPF now that DTLS-SRTP
is mandatory (Asterisk developers, for instance, although getting it to
work from a signalling point of view is a trivial replace matter in
JavaScript), but I agree it makes sense to keep RTP/SAVPF, if not just
because all browser implementations right now (the legacy of a few
months from now on) only accept that, which would render them useless
having this change.

Lorenzo


=20
> El 03/06/2014 8:04, Justin Uberti escribi=C3=B3:
> > I think there is value in specifying UDP/TLS/RTP/SAVPF, in that it=20
> > unambiguously indicates the desire to do DTLS-SRTP. Otherwise, one
> > has to guess at this from the presence of a=3Dfingerprint, which
> > seems dodgy, given that endpoints are free to ignore SDP attributes
> > they don't understand.
> >
> > The fact that UDP and TLS appear in the m=3D line is kind of=20
> > unfortunate, but I think it can be interpreted as "do DTLS-SRTP".
> >
> >
> > On Mon, Jun 2, 2014 at 11:53 AM, Martin Thomson=20
> > <martin.thomson@gmail.com <mailto:martin.thomson@gmail.com>> wrote:
> >
> >     On 2 June 2014 10:13, Harald Alvestrand <harald@alvestrand.no
> >     <mailto:harald@alvestrand.no>> wrote:
> >     > The current implementations both send "RTP/SAVPF" in this
> >     > field.
> >     But this
> >     > usage is not specified by RFC 5764.
> >
> >     Sounds easy to me.  The SDP doesn't matter and can say anything.
> >     "RTP/SAVPF" is as good as any other sequence of octets.
> >
> >     I'm sure that JSEP can include a requirement that is directly
> >     contradictory to an existing RFC.  That happens all the time.
> > At least this time it can be explicit.
> >
> >     _______________________________________________
> >     rtcweb mailing list
> >     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
> >     https://www.ietf.org/mailman/listinfo/rtcweb
> >
> >
> >
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Subject: Re: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Hi Harald,

>Given the nature of ICE, flows may shift back and forth between TCP candid=
ates and UDP candidates over the lifetime of the flow, without any new SDP =
negotiation being invoked.

Is that really true?

1) When the selected candidates have been chosen, and updated offer is sent=
 where the c/m- lines represent the selected candidates

2) Media is sent only using the selected candidates

3) Keep-alives etc are sent only using the selected candidates (which means=
 that switching to other candidates "on the fly" may not even work)

4) If you want to change the selected candidates, you need to do an ICE res=
tart, new candidate nominations, and new selected (which means a new offer)=
.

Or, did I misunderstand what you said? :)

Regards,

Christer


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 3 Jun 2014 11:35:03 +0200
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Subject: Re: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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2014-06-03 11:16 GMT+02:00 Christer Holmberg <christer.holmberg@ericsson.co=
m>:
> 1) When the selected candidates have been chosen, and updated offer is se=
nt where the c/m- lines represent the selected candidates
>
> 2) Media is sent only using the selected candidates
>
> 3) Keep-alives etc are sent only using the selected candidates (which mea=
ns that switching to other candidates "on the fly" may not even work)
>
> 4) If you want to change the selected candidates, you need to do an ICE r=
estart, new candidate nominations, and new selected (which means a new offe=
r).


Please, may we forget this constrain/requirement in RFC 5245 that is
100% useless?

By following ICE procedures both peers agree on which candidate-pair
to use, and there is no need at all to force a "new SDP offer"
indicating the selected pair since both peers already know that.

And to prove it, let's take a look to WebRTC browsers as none of them
"generates a new SDP offer if the winning ICE candidate-pair does not
match the candidate in the m=3D line".

Best regards.


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


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Subject: Re: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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On 03.06.14, 12:35, Iħaki Baz Castillo wrote:
> 2014-06-03 11:16 GMT+02:00 Christer Holmberg <christer.holmberg@ericsson.com>:
>> 1) When the selected candidates have been chosen, and updated offer is sent where the c/m- lines represent the selected candidates
>>
>> 2) Media is sent only using the selected candidates
>>
>> 3) Keep-alives etc are sent only using the selected candidates (which means that switching to other candidates "on the fly" may not even work)
>>
>> 4) If you want to change the selected candidates, you need to do an ICE restart, new candidate nominations, and new selected (which means a new offer).
>
>
> Please, may we forget this constrain/requirement in RFC 5245 that is
> 100% useless?
>
> By following ICE procedures both peers agree on which candidate-pair
> to use, and there is no need at all to force a "new SDP offer"
> indicating the selected pair since both peers already know that.
>
> And to prove it, let's take a look to WebRTC browsers as none of them
> "generates a new SDP offer if the winning ICE candidate-pair does not
> match the candidate in the m= line".

I believe Christer was talking about the need to send a new offer when 
you do an ICE restart. I don't think was about the part in RFC5245 that 
tells you to send a new offer once ICE is ready and that is only 
necessary for the benefit of signalling intermediaries.

Emil

-- 
https://jitsi.org


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Subject: Re: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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2014-06-03 11:51 GMT+02:00 Emil Ivov <emcho@jitsi.org>:
> I believe Christer was talking about the need to send a new offer when yo=
u
> do an ICE restart.

Once a candidate-pair is selected and used to transmit media, a device
may check (after some time) the same candidate-pairs. And if it finds
a better pair it may select it. There may be no need for a new SDP O/A
in this case.
Of course, in the case of mobility, interface changes, etc, signaling
the ICE candidates from scratch is needed.


> I don't think was about the part in RFC5245 that tells
> you to send a new offer once ICE is ready and that is only necessary for =
the
> benefit of signalling intermediaries.

That is 100% focused on SIP.

Regards.




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


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On 03 Jun 2014 1:14 PM, "I=C3=B1aki Baz Castillo" <ibc@aliax.net> wrote:
>
> 2014-06-03 11:51 GMT+02:00 Emil Ivov <emcho@jitsi.org>:
> > I believe Christer was talking about the need to send a new offer when
you
> > do an ICE restart.
>
> Once a candidate-pair is selected and used to transmit media, a device
> may check (after some time) the same candidate-pairs. And if it finds
> a better pair it may select it. There may be no need for a new SDP O/A
> in this case.

That wouldn't work with current  RFC5245 implementations (and I believe
would cause problems) but if you want to make a proposal, mmusic would be
the right place.

> That is 100% focused on SIP.

Yes, correct

Emil

--sent from my mobile

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<p dir=3D"ltr"><br>
On 03 Jun 2014 1:14 PM, &quot;I=C3=B1aki Baz Castillo&quot; &lt;<a href=3D"=
mailto:ibc@aliax.net">ibc@aliax.net</a>&gt; wrote:<br>
&gt;<br>
&gt; 2014-06-03 11:51 GMT+02:00 Emil Ivov &lt;<a href=3D"mailto:emcho@jitsi=
.org">emcho@jitsi.org</a>&gt;:<br>
&gt; &gt; I believe Christer was talking about the need to send a new offer=
 when you<br>
&gt; &gt; do an ICE restart.<br>
&gt;<br>
&gt; Once a candidate-pair is selected and used to transmit media, a device=
<br>
&gt; may check (after some time) the same candidate-pairs. And if it finds<=
br>
&gt; a better pair it may select it. There may be no need for a new SDP O/A=
<br>
&gt; in this case.</p>
<p dir=3D"ltr">That wouldn&#39;t work with current=C2=A0 RFC5245 implementa=
tions (and I believe would cause problems) but if you want to make a propos=
al, mmusic would be the right place.</p>
<p dir=3D"ltr">&gt; That is 100% focused on SIP.</p>
<p dir=3D"ltr">Yes, correct</p>
<p dir=3D"ltr">Emil</p>
<p dir=3D"ltr">--sent from my mobile<br>
</p>

--047d7bd8fe9439673504faece625--


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On 06/02/2014 08:43 PM, Martin Thomson wrote:
> On 2 June 2014 02:45, Ram Mohan R (rmohanr) <rmohanr@cisco.com> wrote:
>> <charles said>
>> I should have also stated that this would apply only for cases in which
>> such use of multiple DSCPs was appropriate, as determined by DART.
>> </charles>
>>
>>
>>   We did discuss this earlier and there was some support to use highest
>> priority DSCP in case of BUNDLE. Here was the previous thread -
>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg11777.html
>> I did not see any conclusion though on that topic.
>>
>>
>> I am ok to have STUN packets to have highest DSCP when multiple DSCP
>> values are used for a given 5-tuple (like in DART). The only issue I see
>> is when there is a congestion then your STUN consent check packets would
>> get preference over some of the other packets. I think this is ok as
>> dropping a consent message means the sender is likely to stop sending
>> media itself,  if it does not see a consent refresh response.
>>
>> I will add the following text. Does this look ok ?
>>
>> "It is possible that different Diffserv Codepoints are used by
>>     different media over the same transport address as described in
>>     [I-D.ietf-tsvwg-rtcweb-qos].  In such cases, it is recommended to use
>> the same DSCP for the STUN messages
>> as that used for the highest priority media stream(s)."
> I know that I haven't raised this point before, and I apologize for
> not paying enough attention, but I don't think that we should be
> talking about DSCP in this draft.
>
> Use of DSCP with STUN and the additional complications that BUNGLE
> adds are best dealt with in those contexts.  This was a simple
> document, we should keep the scope constrained as much as possible.

I think having BUNDLE depend on consent-freshness would be a Bad Thing.

Probably the chairs should draw a tree of the various drafts and decide 
which draft can usefully say something about the relationship between 
RTP multiplexing (an -rtp- matter), DSCP (a -tsvwg- matter) and use of 
STUN (a -consent-freshness matter).




>
> _______________________________________________
> 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: Emil Ivov <emcho@jitsi.org>, =?utf-8?B?ScOxYWtpIEJheiBDYXN0aWxsbw==?= <ibc@aliax.net>
Thread-Topic: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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From: "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-03.txt
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-03.txt
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Please see inline

-----Original Message-----
From: Harald Alvestrand <harald@alvestrand.no>
Date: Tuesday, 3 June 2014 5:22 pm
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] I-D Action:
draft-ietf-rtcweb-stun-consent-freshness-03.txt

>On 06/02/2014 08:43 PM, Martin Thomson wrote:
>> On 2 June 2014 02:45, Ram Mohan R (rmohanr) <rmohanr@cisco.com> wrote:
>>> <charles said>
>>> I should have also stated that this would apply only for cases in which
>>> such use of multiple DSCPs was appropriate, as determined by DART.
>>> </charles>
>>>
>>>
>>>   We did discuss this earlier and there was some support to use highest
>>> priority DSCP in case of BUNDLE. Here was the previous thread -
>>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg11777.html
>>> I did not see any conclusion though on that topic.
>>>
>>>
>>> I am ok to have STUN packets to have highest DSCP when multiple DSCP
>>> values are used for a given 5-tuple (like in DART). The only issue I
>>>see
>>> is when there is a congestion then your STUN consent check packets
>>>would
>>> get preference over some of the other packets. I think this is ok as
>>> dropping a consent message means the sender is likely to stop sending
>>> media itself,  if it does not see a consent refresh response.
>>>
>>> I will add the following text. Does this look ok ?
>>>
>>> "It is possible that different Diffserv Codepoints are used by
>>>     different media over the same transport address as described in
>>>     [I-D.ietf-tsvwg-rtcweb-qos].  In such cases, it is recommended to
>>>use
>>> the same DSCP for the STUN messages
>>> as that used for the highest priority media stream(s)."
>> I know that I haven't raised this point before, and I apologize for
>> not paying enough attention, but I don't think that we should be
>> talking about DSCP in this draft.
>>
>> Use of DSCP with STUN and the additional complications that BUNGLE
>> adds are best dealt with in those contexts.  This was a simple
>> document, we should keep the scope constrained as much as possible.
>
>I think having BUNDLE depend on consent-freshness would be a Bad Thing.
>
>Probably the chairs should draw a tree of the various drafts and decide
>which draft can usefully say something about the relationship between
>RTP multiplexing (an -rtp- matter), DSCP (a -tsvwg- matter) and use of
>STUN (a -consent-freshness matter).


We added this text in this document initially as we did not have any other
place holder.  All other STUN usage drafts (like ICE RFC) do have text
about DiffServ treatment and so we felt this draft also should talk about
DiffServ treatment for STUN consent freshness.  Consent freshness when
BUNDLE is used has dependency on other documents( BUNDLE draft, DART
draft) which are a work in progress. I agree having the text on consent
freshness for BUNDLE would make the scope of this draft larger and perhaps
slow the progress of this document.  I am ok to move the text related to
BUNDLE out of this draft but then it has to be put some where and linked
with consent freshness.

I will leave it to the Chairs to decide which is the best way forward for
this.=20


Ram

>
>
>
>
>>
>> _______________________________________________
>> 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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Subject: Re: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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On 06/03/2014 01:56 PM, Christer Holmberg wrote:
> Hi,
>
>>>> Once a candidate-pair is selected and used to transmit media, a device
>>>> may check (after some time) the same candidate-pairs. And if it finds
>>>> a better pair it may select it.
> Nobody can of course prevent implementations from checking whether previously nominated (but not selected) candidates still work. However, I am not sure whether entities e.g. must respond to connectivity checks from such entities.
>
>>>> There may be no need for a new SDP O/A
>>>> in this case.
>>>>
>>>> That wouldn't work with current  RFC5245 implementations (and I believe would cause problems) but if you want to make a proposal, mmusic would be the right place.
>>> That is 100% focused on SIP.
>> Yes, correct
> It is focused on environments where intermediaries make media decisions based on the signaling. SIP is of course an obvious example.
>
> But, never the less, as far as I know we have adopted RFC5245 - not a profile of RFC5245 - which means:
>
> 1) ICE restart in case you want to change the selected candidate pair; and
>
> 2) sending a new offer if a new candidate pair is chosen as the selected

Does it?

11.1.3.  Procedures for All Implementations

    ICE has interactions with jitter buffer adaptation mechanisms. An
    RTP stream can begin using one candidate, and switch to another one,
    though this happens rarely with ICE.  The newer candidate may result
    in RTP packets taking a different path through the network -- one
    with different delay characteristics.  As discussed below, agents are
    encouraged to re-adjust jitter buffers when there are changes in
    source or destination address of media packets.  Furthermore, many
    audio codecs use the marker bit to signal the beginning of a
    talkspurt, for the purposes of jitter buffer adaptation.  For such
    codecs, it is RECOMMENDED that the sender set the marker bit
    [RFC3550] when an agent switches transmission of media from one
    candidate pair to another.

11.2.  Receiving Media

    ICE implementations MUST be prepared to receive media on each
    component on any candidates provided for that component in the most
    recent offer/answer exchange (in the case of RTP, this would include
    both RTP and RTCP if candidates were provided for both).

    It is RECOMMENDED that, when an agent receives an RTP packet with a
    new source or destination IP address for a particular media stream,
    that the agent re-adjust its jitter buffers.

I would never want to call RFC 5245 a model of clarity, but it seems to 
me that this language was written in the expectation that a sender might 
choose to switch to a different pair at any time.

I can't find the corresponding paragraph that says when the sender might 
do so, though.

>
> *IF* we decide to change it, then it should be properly documented.
>
> Regards,
>
> Christer
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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2014-06-03 14:16 GMT+02:00 Harald Alvestrand <harald@alvestrand.no>:
> I would never want to call RFC 5245 a model of clarity, but it seems to m=
e
> that this language was written in the expectation that a sender might cho=
ose
> to switch to a different pair at any time.

+1


> I can't find the corresponding paragraph that says when the sender might =
do
> so, though.

IMHO if it's not defined when it can do that then it MUST be able to
do that at any time.


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


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From: "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Continous consent when no media is sent
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--_000_CFB3BBF88F7C2rmohanrciscocom_
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Hi

Agree. I will remove the =93recvonly=94 keyword from that sentence. Then I =
believe it will cover for all cases where a endpoint is only receiving medi=
a and not sending. Is that sufficient ?

Regards,
Ram

From: Christer Holmberg <christer.holmberg@ericsson.com<mailto:christer.hol=
mberg@ericsson.com>>
Date: Tuesday, 3 June 2014 1:25 pm
To: "rtcweb@ietf.org<mailto:rtcweb@ietf.org>" <rtcweb@ietf.org<mailto:rtcwe=
b@ietf.org>>
Subject: [rtcweb] Continous consent when no media is sent

Hi,

Section 4.1 of draft-ietf-rtcweb-stun-consent-freshness-03 says:

=93An endpoint that is only receiving application traffic (recvonly)
                does not need to obtain consent which can slightly conserve=
 its
                resources.=94

I think it would be good to be more generic, and talk about cases where the=
 sender does not send media. That is obviously true in case of =93recvonly=
=94, but it could also be true in other cases.

In addition, I guess the no-need-to-send-consent rule also applies to =93in=
active=94 streams.

Regards,

Christer

--_000_CFB3BBF88F7C2rmohanrciscocom_
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ly: Calibri, sans-serif;">
<div>Hi&nbsp;</div>
<div><br>
</div>
<div>Agree. I will remove the =93recvonly=94 keyword from that sentence. Th=
en I believe it will cover for all cases where a endpoint is only receiving=
 media and not sending. Is that sufficient ?</div>
<div><br>
</div>
<div>Regards,</div>
<div>Ram</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div style=3D"font-family:Calibri; font-size:11pt; text-align:left; color:b=
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<span style=3D"font-weight:bold">From: </span>Christer Holmberg &lt;<a href=
=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson.com</=
a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Tuesday, 3 June 2014 1:25 pm<=
br>
<span style=3D"font-weight:bold">To: </span>&quot;<a href=3D"mailto:rtcweb@=
ietf.org">rtcweb@ietf.org</a>&quot; &lt;<a href=3D"mailto:rtcweb@ietf.org">=
rtcweb@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>[rtcweb] Continous consent=
 when no media is sent<br>
</div>
<div><br>
</div>
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<p class=3D"MsoNormal">Hi,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Section 4.1 of draft-ietf-rtcweb-stun-consent-freshn=
ess-03 says:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">=93An endpoint that is =
only receiving application traffic (recvonly)<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; does not need to obtain consent which can=
 slightly conserve its<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; resources.=94<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I think it would be good to be more generic, and tal=
k about cases where the sender does not send media. That is obviously true =
in case of =93recvonly=94, but it could also be true in other cases.<o:p></=
o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">In addition, I guess the no-need-to-send-consent rul=
e also applies to =93inactive=94 streams.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Christer<o:p></o:p></p>
</div>
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--_000_CFB3BBF88F7C2rmohanrciscocom_--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
Thread-Index: AQHPfoYJ1ITXr2v8bkWsEJ3GsOpDmZtfFtqw///ow4CAAAR+gIAABnuAgAAWfQCAACRUEP//52uAgAAidEA=
Date: Tue, 3 Jun 2014 12:26:34 +0000
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References: <538CB0C7.6050700@alvestrand.no> <7594FB04B1934943A5C02806D1A2204B1D34DE48@ESESSMB209.ericsson.se> <CALiegfmHCMMrZ2v0puHjR=wkuY0+88+mvdHEdOsf=g_EjBMX0w@mail.gmail.com> <538D9A8B.4060002@jitsi.org> <CALiegf=UfhcjT0L0vAg2fmpjSRo_qf9d669Z3yMu3Eg9TpfMsg@mail.gmail.com> <CAPvvaaJzNyPPnDZZ7Pvb231kQ1tc3CJOLdhXyqKXd9h39+Udaw@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D34E2D2@ESESSMB209.ericsson.se> <538DBCB3.3030408@alvestrand.no>
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Subject: Re: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Hi,

>>>>> Once a candidate-pair is selected and used to transmit media, a=20
>>>>> device may check (after some time) the same candidate-pairs. And if=20
>>>>> it finds a better pair it may select it.
>> Nobody can of course prevent implementations from checking whether previ=
ously nominated (but not selected) candidates still work. However, I am not=
 sure whether entities e.g. must respond to connectivity checks from such e=
ntities.
>>
>>>>> There may be no need for a new SDP O/A in this case.
>>>>>
>>>>> That wouldn't work with current  RFC5245 implementations (and I belie=
ve would cause problems) but if you want to make a proposal, mmusic would b=
e the right place.
>>>> That is 100% focused on SIP.
>>> Yes, correct
>> It is focused on environments where intermediaries make media decisions =
based on the signaling. SIP is of course an obvious example.
>>
>> But, never the less, as far as I know we have adopted RFC5245 - not a pr=
ofile of RFC5245 - which means:
>>
>> 1) ICE restart in case you want to change the selected candidate pair;=20
>> and
>>
>> 2) sending a new offer if a new candidate pair is chosen as the=20
>> selected
>
> Does it?
>
> 11.1.3.  Procedures for All Implementations
>
>    ICE has interactions with jitter buffer adaptation mechanisms. An
>    RTP stream can begin using one candidate, and switch to another one,
>    though this happens rarely with ICE.  The newer candidate may result
>    in RTP packets taking a different path through the network -- one
>    with different delay characteristics.  As discussed below, agents are
>    encouraged to re-adjust jitter buffers when there are changes in
>    source or destination address of media packets.  Furthermore, many
>    audio codecs use the marker bit to signal the beginning of a
>    talkspurt, for the purposes of jitter buffer adaptation.  For such
>    codecs, it is RECOMMENDED that the sender set the marker bit
>    [RFC3550] when an agent switches transmission of media from one
>    candidate pair to another.
>
>
>11.2.  Receiving Media
>
>    ICE implementations MUST be prepared to receive media on each
>    component on any candidates provided for that component in the most
>    recent offer/answer exchange (in the case of RTP, this would include
>    both RTP and RTCP if candidates were provided for both).
>
>    It is RECOMMENDED that, when an agent receives an RTP packet with a
>    new source or destination IP address for a particular media stream,
>    that the agent re-adjust its jitter buffers.

I guess this could happen during session establishment if you use aggressiv=
e nomination, where you may indicate a candidate pair as selected, but then=
 switch to another. But, the RFC still says that you send a new offer when =
such switch occur.

> I would never want to call RFC 5245 a model of clarity, but it seems to m=
e that this language was written in the expectation that a sender might cho=
ose to switch to a different pair at any time.
>
> I can't find the corresponding paragraph that says when the sender might =
do so, though.

Perhaps what you are talking about is related to the MICE work?

It's a while since I read the MICE draft, so I don't remember the details. =
There, I think you can use multiple candidate pairs alive, but I don't reme=
mber whether you can switch without sending a new offer.

In any case, if there is something unclear about ICE, I guess we should bri=
ng it to MMUSIC :)

Regards,

Christer


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Continous consent when no media is sent
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Hi,

>Agree. I will remove the "recvonly" keyword from that sentence. Then I bel=
ieve it will cover for all cases where a endpoint is only receiving media a=
nd not sending. Is that sufficient ?

If the stream is inactive, the endpoint won't receive either, so I think yo=
u only need to talk about sending. Whether the endpoint receives media or n=
ot is irrelevant, isn't it?

Regards,

Christer






From: Christer Holmberg <christer.holmberg@ericsson.com>
Date: Tuesday, 3 June 2014 1:25 pm
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: [rtcweb] Continous consent when no media is sent

Hi,
=A0
Section 4.1 of draft-ietf-rtcweb-stun-consent-freshness-03 says:
=A0
       "An endpoint that is only receiving application traffic (recvonly)
=A0=A0 =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 does not need to obtain consent=
 which can slightly conserve its
=A0=A0 =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 resources."
=A0
I think it would be good to be more generic, and talk about cases where the=
 sender does not send media. That is obviously true in case of "recvonly", =
but it could also be true in other cases.
=A0
In addition, I guess the no-need-to-send-consent rule also applies to "inac=
tive" streams.
=A0
Regards,
=A0
Christer


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Subject: Re: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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2014-06-03 14:26 GMT+02:00 Christer Holmberg <christer.holmberg@ericsson.co=
m>:
> But, the RFC still says that you send a new offer when such switch occur.

Yes, for making SIP B2BUAs happy ;)

May we just ignore that? :)


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


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To: =?utf-8?B?ScOxYWtpIEJheiBDYXN0aWxsbw==?= <ibc@aliax.net>, "Harald Alvestrand" <harald@alvestrand.no>
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Subject: Re: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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On 03.06.14, 15:16, Harald Alvestrand wrote:
> I would never want to call RFC 5245 a model of clarity, but it seems to
> me that this language was written in the expectation that a sender might
> choose to switch to a different pair at any time.

I agree with Christer that the intention here was likely to make that 
possible while ICE processing is still ongoing.

Note how section 8.3.1 says:

    Once ICE processing
    has reached the Completed state for all peers for media streams using
    those candidates, the agent SHOULD wait an additional three seconds,
    and then it MAY cease responding to checks or generating triggered
    checks on that candidate.  It MAY free the candidate at that time.
    Freeing of server reflexive candidates is never explicit; it happens
    by lack of a keepalive.  The three-second delay handles cases when
    aggressive nomination is used, and the selected pairs can quickly
    change after ICE has completed.

8.3.2. Lite Implementation Procedures


    A lite implementation MAY free candidates not selected by ICE as soon
    as ICE processing has reached the Completed state for all peers for
    all media streams using those candidates.

Emil

-- 
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Subject: Re: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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On 03.06.14, 15:31, Iħaki Baz Castillo wrote:
> 2014-06-03 14:26 GMT+02:00 Christer Holmberg <christer.holmberg@ericsson.com>:
>> But, the RFC still says that you send a new offer when such switch occur.
>
> Yes, for making SIP B2BUAs happy ;)
>
> May we just ignore that? :)

B2BUAs may have very valid reasons for requiring this. Typically, some 
would add relay candidates as a fall back and they need to know when and 
where to release them.

Emil


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: =?utf-8?B?ScOxYWtpIEJheiBDYXN0aWxsbw==?= <ibc@aliax.net>
Thread-Topic: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Subject: Re: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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From nobody Tue Jun  3 05:40:22 2014
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From: "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Continous consent when no media is sent
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Date: Tue, 3 Jun 2014 12:39:50 +0000
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Subject: Re: [rtcweb] Continous consent when no media is sent
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Agree. I think checking for sending is good enough. If both the endpoints
are not sending media they won=B9t do consent with their peers.
That will take care of all the cases

Regards,
Ram
-----Original Message-----
From: Christer Holmberg <christer.holmberg@ericsson.com>
Date: Tuesday, 3 June 2014 5:59 pm
To: Ram Mohan Ravindranath <rmohanr@cisco.com>, "rtcweb@ietf.org"
<rtcweb@ietf.org>
Subject: RE: [rtcweb] Continous consent when no media is sent

>Hi,
>
>>Agree. I will remove the "recvonly" keyword from that sentence. Then I
>>believe it will cover for all cases where a endpoint is only receiving
>>media and not sending. Is that sufficient ?
>
>If the stream is inactive, the endpoint won't receive either, so I think
>you only need to talk about sending. Whether the endpoint receives media
>or not is irrelevant, isn't it?
>
>Regards,
>
>Christer
>
>
>
>
>
>
>From: Christer Holmberg <christer.holmberg@ericsson.com>
>Date: Tuesday, 3 June 2014 1:25 pm
>To: "rtcweb@ietf.org" <rtcweb@ietf.org>
>Subject: [rtcweb] Continous consent when no media is sent
>
>Hi,
>=20
>Section 4.1 of draft-ietf-rtcweb-stun-consent-freshness-03 says:
>=20
>       "An endpoint that is only receiving application traffic (recvonly)
>                does not need to obtain consent which can slightly
>conserve its
>                resources."
>=20
>I think it would be good to be more generic, and talk about cases where
>the sender does not send media. That is obviously true in case of
>"recvonly", but it could also be true in other cases.
>=20
>In addition, I guess the no-need-to-send-consent rule also applies to
>"inactive" streams.
>=20
>Regards,
>=20
>Christer


From nobody Tue Jun  3 05:42:25 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Continous consent when no media is sent
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Subject: Re: [rtcweb] Continous consent when no media is sent
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And, we of course also need to make sure that the remote endpoint does not =
expect to receive cont.cons. in such cases :)

Regards,

Christer

-----Original Message-----
From: Ram Mohan R (rmohanr) [mailto:rmohanr@cisco.com]=20
Sent: 3. kes=E4kuuta 2014 15:40
To: Christer Holmberg; rtcweb@ietf.org
Subject: Re: [rtcweb] Continous consent when no media is sent

Agree. I think checking for sending is good enough. If both the endpoints a=
re not sending media they won=B9t do consent with their peers.
That will take care of all the cases

Regards,
Ram
-----Original Message-----
From: Christer Holmberg <christer.holmberg@ericsson.com>
Date: Tuesday, 3 June 2014 5:59 pm
To: Ram Mohan Ravindranath <rmohanr@cisco.com>, "rtcweb@ietf.org"
<rtcweb@ietf.org>
Subject: RE: [rtcweb] Continous consent when no media is sent

>Hi,
>
>>Agree. I will remove the "recvonly" keyword from that sentence. Then I=20
>>believe it will cover for all cases where a endpoint is only receiving=20
>>media and not sending. Is that sufficient ?
>
>If the stream is inactive, the endpoint won't receive either, so I=20
>think you only need to talk about sending. Whether the endpoint=20
>receives media or not is irrelevant, isn't it?
>
>Regards,
>
>Christer
>
>
>
>
>
>
>From: Christer Holmberg <christer.holmberg@ericsson.com>
>Date: Tuesday, 3 June 2014 1:25 pm
>To: "rtcweb@ietf.org" <rtcweb@ietf.org>
>Subject: [rtcweb] Continous consent when no media is sent
>
>Hi,
>=20
>Section 4.1 of draft-ietf-rtcweb-stun-consent-freshness-03 says:
>=20
>       "An endpoint that is only receiving application traffic (recvonly)
>                does not need to obtain consent which can slightly=20
>conserve its
>                resources."
>=20
>I think it would be good to be more generic, and talk about cases where=20
>the sender does not send media. That is obviously true in case of=20
>"recvonly", but it could also be true in other cases.
>=20
>In addition, I guess the no-need-to-send-consent rule also applies to=20
>"inactive" streams.
>=20
>Regards,
>=20
>Christer


From nobody Tue Jun  3 05:44:08 2014
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 3 Jun 2014 14:43:37 +0200
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Subject: Re: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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2014-06-03 14:32 GMT+02:00 Christer Holmberg <christer.holmberg@ericsson.co=
m>:
>>> I would never want to call RFC 5245 a model of clarity, but it seems
>>> to me that this language was written in the expectation that a sender
>>> might choose to switch to a different pair at any time.
>>
>> +1
>
> We can ask MMUSIC about that.


I strongly don't agree that any pure network-based feature requires
MMUSIC (SDP) to accept it.

Now: imagine a WebRTC device (a mobile) running a audio/video session.
The user walks, looses WiFi and connects to 3G. The WebRTC application
is probably notified about "network changed" and could restart ICE
without the need of signaling a new SDP offer (it may signal it if it
wants). Why do we need to mandate the new SDP O/A?

ICE protocol should have been defined as a separate pure network
protocol (a pure STUN usage) without involving SIP and SDP at all. And
SIP should have its own MMUSIC RFC "ICE usage for SIP and SDP". And we
wouldn't be here debating about the need of sending a fake SDP offer
just because ICE does it work in background.

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


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On 03 Jun 2014, at 14:43, I=F1aki Baz Castillo <ibc@aliax.net> wrote:

> 2014-06-03 14:32 GMT+02:00 Christer Holmberg <christer.holmberg@ericsson.=
com>:
>>>> I would never want to call RFC 5245 a model of clarity, but it seems
>>>> to me that this language was written in the expectation that a sender
>>>> might choose to switch to a different pair at any time.
>>>=20
>>> +1
>>=20
>> We can ask MMUSIC about that.
>=20
>=20
> I strongly don't agree that any pure network-based feature requires
> MMUSIC (SDP) to accept it.
>=20
> Now: imagine a WebRTC device (a mobile) running a audio/video session.
> The user walks, looses WiFi and connects to 3G. The WebRTC application
> is probably notified about "network changed" and could restart ICE
> without the need of signaling a new SDP offer (it may signal it if it
> wants). Why do we need to mandate the new SDP O/A?
>=20
If there are NATs and FWs involved and you change your IP address there is =
a good chance you need connectivity checks from both sides to get through a=
ll the NATs. To do that you need to send your candidates to the remote side=
 so he can start doing the ICE magic.

There is the MICE draft discussing how that can be done. Trickle ICE also h=
ave the potential to better handle those mobility scenarios. But this needs=
 more work to iron out all the corner cases.

> ICE protocol should have been defined as a separate pure network
> protocol (a pure STUN usage) without involving SIP and SDP at all. And
> SIP should have its own MMUSIC RFC "ICE usage for SIP and SDP". And we
> wouldn't be here debating about the need of sending a fake SDP offer
> just because ICE does it work in background.
>=20
That is work in progress within the MMUSIC.

.-.
P=E5l-Erik

> --=20
> I=F1aki Baz Castillo
> <ibc@aliax.net>
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Tue Jun  3 06:34:34 2014
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Usage of draft-ietf-rtcweb-overview
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I was working through this document, and assessing whether it could form a =
conformance assessment for a rtcweb implementation.

I came to the conclusion that it could not as it currently stands.

While it is standards track, it seems to be missing RFC 2119 language compl=
etely.

As such I end up wondering whether some of the referenced drafts are mandat=
ory to implement, optional to implement (or could just be ignored - unlikel=
y).

So what is the philosophy here - Would I be right in considering that the i=
ntent is that all the referenced normative references are mandatory, i.e. a=
ll implementations implement all these documents?

But if that is the case, how do I treat text in the document that reference=
s draft-cbran-rtcweb-codec, which is an informative reference, but which th=
e text treats with equal weight as other normative references?

Regards

Keith=


From nobody Tue Jun  3 07:05:54 2014
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On 06/03/2014 03:34 PM, DRAGE, Keith (Keith) wrote:
> I was working through this document, and assessing whether it could form a conformance assessment for a rtcweb implementation.
>
> I came to the conclusion that it could not as it currently stands.
>
> While it is standards track, it seems to be missing RFC 2119 language completely.
>
> As such I end up wondering whether some of the referenced drafts are mandatory to implement, optional to implement (or could just be ignored - unlikely).
In that case, the language is unclear, and needs fixing.
I'll review the document with this in mind; doing things like changing

    The data transport protocols used by RTCWEB are described in
    [I-D.ietf-rtcweb-transports].

into

    An RTCWEB implementation MUST implement the data transport protocols 
described in
    [I-D.ietf-rtcweb-transports].

is pretty mechanical.
>
> So what is the philosophy here - Would I be right in considering that the intent is that all the referenced normative references are mandatory, i.e. all implementations implement all these documents?

The intent of the normative references is that the referenced RFCs have 
to be read, and the implications understood, in order to correctly 
implement an RTCWEB system.

I believe that's the definition of a normative reference. Many of these 
(RFC 2119 being the prime example) are, by definition, unimplementable.

Some of them are classified wrongly, like -data-protocol, which should 
be normative. (or deleted, since we now have a reference to it in 
-transport).

I'm not sure where the W3C API documents should be placed; for browsers, 
they're normative, but I'm not sure having a normative reference to them 
in this document is the Right Thing. Opinions?

>
> But if that is the case, how do I treat text in the document that references draft-cbran-rtcweb-codec, which is an informative reference, but which the text treats with equal weight as other normative references?

The reference to draft-cbran is due to be deleted, since that draft is 
(I believe) dead. This is a placeholder for a decision on the MTI video 
codec discussion.

The particular sentence referencing this draft is:

    The mandatory to implement codecs, as well as any profiling
    requirements for both mandatory and optional codecs, is described in
    <WORKING GROUP DRAFT "MEDIA PROCESSING"> (candidate draft:
    [I-D.cbran-rtcweb-codec].

When I wrote this, I thought it was pretty clear that an implementation 
MUST implement the mandatory to implement codecs, but that the draft 
specifying them did not exist yet (and the reference to the 
wannabe-decision therefore had to be informative). But if you read it as 
"treat with equal weight", I guess that wasn't written clearly enough?

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


From nobody Tue Jun  3 07:16:47 2014
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: "Pal Martinsen (palmarti)" <palmarti@cisco.com>, =?iso-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>, Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Strong "+1" for Christer's comments on taking it to MMUSIC.
Please see my comments inserted below.

> > Now: imagine a WebRTC device (a mobile) running a audio/video session.
> > The user walks, looses WiFi and connects to 3G. The WebRTC application
> > is probably notified about "network changed" and could restart ICE
> > without the need of signaling a new SDP offer (it may signal it if it
> > wants). Why do we need to mandate the new SDP O/A?

[Raju] Such a new SDP O/A is needed for couple of reasons:
1. QoS: It is true that ICE client can do all inband without additional SDP=
 O/A.
   But there may be middle boxes in between that may apply QoS per the chan=
ged=20
   ICE candidate(s). One such example is, when going from WiFi to LTE the=20
   network may provide better QoS via PCRF/PGW dedicated bearers. It can on=
ly do
   so if it knows about the new media flow. Please note that the new media =
flow may
   go thru completely new network elements who may not be aware of the old=
=20
   flow at all. Yes, one can provide a "costly" solution to solve this prob=
lem
   by using DPI (and/or SDN), but such a solution is not only costly but al=
so complex.
   IMHO, webrtc should facilitate achieving necessary QoS desired for sessi=
ons=20
   in the network. While doing so, I don't see any functionality being lost=
 at=20
   WebRTC, rather it's a gain and greater adaption of WebRTC.

   For the same reason, I also agree with the comment that, a new SDP O/A m=
ust be sent
   if the selected candidate is not the one listed in c/m=3D lines.
   WebRTC should not assume that middle boxes don't exist. Middle boxes do =
exist for
   Very valid reasons (not just SIP/IMS, which is only one example). Especi=
ally with
   the trend towards SDN, the control plane and data/user plane are being s=
eparated
   for good and the control plane needs triggers to reconfigure the network=
 for better
   QoS. Sending new O/A (please do not equate this to SIP, it can be any si=
gnaling
   protocol and even not necessarily SDP format) achieves that.

2. Resource allocation at middle boxes
   In cases like changing transport (not necessarily access network change)=
=20
   from UDP to TCP or vice-versa, it is highly desirable to have a new SDP =
O/A.=20
   Without such offer, it is a huge burden on middle boxes to be prepared=20
   for any kind of mid-session transport (UDP or TCP) changes. Middle boxes=
 can=20
   have complex logic to reserve all possible resources and keep them for a=
ny
   future mid-session changes without new SDP O/A. Again, it's not only ine=
fficient
   but also makes middle boxes logic complicated. Also, as pointed by Emil =
Ivov,
   ICE RFC says you MAY free remaining candidates once ICE is completed. Ev=
en if
   an implementation wants to keep ALL the candidates even after ICE is com=
pleted,
   it can't expect the peer to do the same, especially ice lite implementat=
ions
   typically won't keep the remaining candidates.


BR
Raju


From nobody Tue Jun  3 07:38:53 2014
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 3 Jun 2014 16:38:24 +0200
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Subject: Re: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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2014-06-03 16:16 GMT+02:00 Makaraju, Maridi Raju (Raju)
<Raju.Makaraju@alcatel-lucent.com>:

> [Raju] Such a new SDP O/A is needed for couple of reasons:
> 1. QoS: It is true that ICE client can do all inband without additional S=
DP O/A.
>    But there may be middle boxes in between that may apply QoS per the ch=
anged
>    ICE candidate(s). One such example is, when going from WiFi to LTE the
>    network may provide better QoS via PCRF/PGW dedicated bearers.

If you are talking about "middle boxes" then you mean a fixed and
specific *signaling* protocol, right? even more, you probably mean
SIP.
Given the fact that WebRTC is supposed to be "signaling protocol"
agnostic I don't buy your rationale.


>    For the same reason, I also agree with the comment that, a new SDP O/A=
 must be sent
>    if the selected candidate is not the one listed in c/m=3D lines.

I'm glad that that does not occur now, and no one WebRTC browser cares
about that.


>    WebRTC should not assume that middle boxes don't exist. Middle boxes d=
o exist for
>    Very valid reasons (not just SIP/IMS, which is only one example). Espe=
cially with
>    the trend towards SDN, the control plane and data/user plane are being=
 separated
>    for good and the control plane needs triggers to reconfigure the netwo=
rk for better
>    QoS. Sending new O/A (please do not equate this to SIP, it can be any =
signaling
>    protocol and even not necessarily SDP format) achieves that.

If so it can be signaled *anyhow* (up to the specific application /
vendor), but we are discussing here about the requirement of signaling
a new SDP (as the RFC mandates).


Regards.



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


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From nobody Tue Jun  3 08:20:18 2014
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 3 Jun 2014 17:19:47 +0200
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To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
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Subject: Re: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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2014-06-03 17:14 GMT+02:00 Makaraju, Maridi Raju (Raju)
<Raju.Makaraju@alcatel-lucent.com>:

> Yes, it is up to the specific application/vendor. But, to achieve this un=
derlying PC/JSEP API should provide mechanisms to:

> 1. Inform application when default candidate changes/selected. So, that a=
pplication
>    can do whatever it wants to do with that info.

OK


> 2. Make sure c/m=3D lines are formatting is consistent with the default/s=
elected
>    candidate. That means using of RTP/AVPF independent of actual transpor=
t is
>    not correct.

OK


> Existing PC callback oniceconnectionstatechange() may be used to solve bo=
th of these.
> However, createOffer() called after this callback should have the updated=
 c/m=3D
> lines per the selected candidate(s). Not sure if browsers do either or bo=
th of these today?! Without clear requirements browsers may not implement t=
his trigger. I think, this is what we need from the PC API and browsers so =
that some applications can take advantage of this for better QoS or other r=
easons.

My main concern is about whether a ICE endpoint should allow
"on-the-fly" candidate-pair changes when such a new SDP O/A does not
take place. That is what worries me, and the RFC is not clear about
that.


Regards.




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


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The other thread has been hijacked by an ICE discussion, so reasking here.
Should use of DTLS be indicated by:
- UDP/TLS/RTP/SAVPF on the m=3D line
- RTP/SAVPF, and presence of an a=3Dfingerprint line

Some have expressed a preference for RTP/SAVPF, as being more backwards
compatible. However, others have expressed a preference for
UDP/TLS/RTP/SAVPF, as that is what RFC 5763/4 says. The last time this came
up, in September 2013, the UDP/TLS/RTP/SAVPF proponents were able to
convince me to include this into jsep-06, so if we want to change it back,
I'd like to see a clear consensus.

On Tue, Jun 3, 2014 at 2:08 AM, Lorenzo Miniero <lorenzo@meetecho.com>
wrote:

> On Tue, 03 Jun 2014 10:32:43 +0200
> Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com> wrote:
>
> > On the other side, keeping "RTP/SAVFP" would have allowed a server
> > (SBC/MCU/etc) to send an offer without knowing in advance if the
> > other side supports SDES or DTLS, by sending both the crypto and
> > fingerprint attributes and let the receiver choose which one to use
> > (as chrome has done until recently). I know that it is not important
> > for webrtc, as DTLS is mandatory and SDES is a must not, but for
> > supporting other legacy clients it may be easier for some of us.
> >
> > Best regards
> > Sergio
> >
>
>
> I guess that was the rationale for having RTP/SAVPF in the first
> place, as initially both SDES and DTLS-SRTP were allowed. There are
> people confused about not having UDP/TLS/RTP/SAVPF now that DTLS-SRTP
> is mandatory (Asterisk developers, for instance, although getting it to
> work from a signalling point of view is a trivial replace matter in
> JavaScript), but I agree it makes sense to keep RTP/SAVPF, if not just
> because all browser implementations right now (the legacy of a few
> months from now on) only accept that, which would render them useless
> having this change.
>
> Lorenzo
>
>
>
> > El 03/06/2014 8:04, Justin Uberti escribi=C3=B3:
> > > I think there is value in specifying UDP/TLS/RTP/SAVPF, in that it
> > > unambiguously indicates the desire to do DTLS-SRTP. Otherwise, one
> > > has to guess at this from the presence of a=3Dfingerprint, which
> > > seems dodgy, given that endpoints are free to ignore SDP attributes
> > > they don't understand.
> > >
> > > The fact that UDP and TLS appear in the m=3D line is kind of
> > > unfortunate, but I think it can be interpreted as "do DTLS-SRTP".
> > >
> > >
> > > On Mon, Jun 2, 2014 at 11:53 AM, Martin Thomson
> > > <martin.thomson@gmail.com <mailto:martin.thomson@gmail.com>> wrote:
> > >
> > >     On 2 June 2014 10:13, Harald Alvestrand <harald@alvestrand.no
> > >     <mailto:harald@alvestrand.no>> wrote:
> > >     > The current implementations both send "RTP/SAVPF" in this
> > >     > field.
> > >     But this
> > >     > usage is not specified by RFC 5764.
> > >
> > >     Sounds easy to me.  The SDP doesn't matter and can say anything.
> > >     "RTP/SAVPF" is as good as any other sequence of octets.
> > >
> > >     I'm sure that JSEP can include a requirement that is directly
> > >     contradictory to an existing RFC.  That happens all the time.
> > > At least this time it can be explicit.
> > >
> > >     _______________________________________________
> > >     rtcweb mailing list
> > >     rtcweb@ietf.org <mailto: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
>

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<div dir=3D"ltr">The other thread has been hijacked by an ICE discussion, s=
o reasking here. Should use of DTLS be indicated by:<div>- UDP/TLS/RTP/SAVP=
F on the m=3D line<br><div>- RTP/SAVPF, and presence of an a=3Dfingerprint =
line<br>

<div><div><div class=3D"gmail_extra"><br>Some have expressed a preference f=
or RTP/SAVPF, as being more backwards compatible. However, others have expr=
essed a preference for UDP/TLS/RTP/SAVPF, as that is what RFC 5763/4 says. =
The last time this came up, in September 2013, the=C2=A0UDP/TLS/RTP/SAVPF p=
roponents were able to convince me to include this into jsep-06, so if we w=
ant to change it back, I&#39;d like to see a clear consensus.<br>

<br><div class=3D"gmail_quote">On Tue, Jun 3, 2014 at 2:08 AM, Lorenzo Mini=
ero <span dir=3D"ltr">&lt;<a href=3D"mailto:lorenzo@meetecho.com" target=3D=
"_blank">lorenzo@meetecho.com</a>&gt;</span> wrote:<br><blockquote class=3D=
"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;borde=
r-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">

<div class=3D"">On Tue, 03 Jun 2014 10:32:43 +0200<br>
Sergio Garcia Murillo &lt;<a href=3D"mailto:sergio.garcia.murillo@gmail.com=
">sergio.garcia.murillo@gmail.com</a>&gt; wrote:<br>
<br>
&gt; On the other side, keeping &quot;RTP/SAVFP&quot; would have allowed a =
server<br>
&gt; (SBC/MCU/etc) to send an offer without knowing in advance if the<br>
&gt; other side supports SDES or DTLS, by sending both the crypto and<br>
&gt; fingerprint attributes and let the receiver choose which one to use<br=
>
&gt; (as chrome has done until recently). I know that it is not important<b=
r>
&gt; for webrtc, as DTLS is mandatory and SDES is a must not, but for<br>
&gt; supporting other legacy clients it may be easier for some of us.<br>
&gt;<br>
&gt; Best regards<br>
&gt; Sergio<br>
&gt;<br>
<br>
<br>
</div>I guess that was the rationale for having RTP/SAVPF in the first<br>
place, as initially both SDES and DTLS-SRTP were allowed. There are<br>
people confused about not having UDP/TLS/RTP/SAVPF now that DTLS-SRTP<br>
is mandatory (Asterisk developers, for instance, although getting it to<br>
work from a signalling point of view is a trivial replace matter in<br>
JavaScript), but I agree it makes sense to keep RTP/SAVPF, if not just<br>
because all browser implementations right now (the legacy of a few<br>
months from now on) only accept that, which would render them useless<br>
having this change.<br>
<br>
Lorenzo<br>
<div class=3D""><br>
<br>
<br>
&gt; El 03/06/2014 8:04, Justin Uberti escribi=C3=B3:<br>
&gt; &gt; I think there is value in specifying UDP/TLS/RTP/SAVPF, in that i=
t<br>
&gt; &gt; unambiguously indicates the desire to do DTLS-SRTP. Otherwise, on=
e<br>
&gt; &gt; has to guess at this from the presence of a=3Dfingerprint, which<=
br>
&gt; &gt; seems dodgy, given that endpoints are free to ignore SDP attribut=
es<br>
&gt; &gt; they don&#39;t understand.<br>
&gt; &gt;<br>
&gt; &gt; The fact that UDP and TLS appear in the m=3D line is kind of<br>
&gt; &gt; unfortunate, but I think it can be interpreted as &quot;do DTLS-S=
RTP&quot;.<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; On Mon, Jun 2, 2014 at 11:53 AM, Martin Thomson<br>
</div><div class=3D"">&gt; &gt; &lt;<a href=3D"mailto:martin.thomson@gmail.=
com">martin.thomson@gmail.com</a> &lt;mailto:<a href=3D"mailto:martin.thoms=
on@gmail.com">martin.thomson@gmail.com</a>&gt;&gt; wrote:<br>
&gt; &gt;<br>
&gt; &gt; =C2=A0 =C2=A0 On 2 June 2014 10:13, Harald Alvestrand &lt;<a href=
=3D"mailto:harald@alvestrand.no">harald@alvestrand.no</a><br>
</div><div class=3D"">&gt; &gt; =C2=A0 =C2=A0 &lt;mailto:<a href=3D"mailto:=
harald@alvestrand.no">harald@alvestrand.no</a>&gt;&gt; wrote:<br>
&gt; &gt; =C2=A0 =C2=A0 &gt; The current implementations both send &quot;RT=
P/SAVPF&quot; in this<br>
&gt; &gt; =C2=A0 =C2=A0 &gt; field.<br>
&gt; &gt; =C2=A0 =C2=A0 But this<br>
&gt; &gt; =C2=A0 =C2=A0 &gt; usage is not specified by RFC 5764.<br>
&gt; &gt;<br>
&gt; &gt; =C2=A0 =C2=A0 Sounds easy to me. =C2=A0The SDP doesn&#39;t matter=
 and can say anything.<br>
&gt; &gt; =C2=A0 =C2=A0 &quot;RTP/SAVPF&quot; is as good as any other seque=
nce of octets.<br>
&gt; &gt;<br>
&gt; &gt; =C2=A0 =C2=A0 I&#39;m sure that JSEP can include a requirement th=
at is directly<br>
&gt; &gt; =C2=A0 =C2=A0 contradictory to an existing RFC. =C2=A0That happen=
s all the time.<br>
&gt; &gt; At least this time it can be explicit.<br>
&gt; &gt;<br>
&gt; &gt; =C2=A0 =C2=A0 _______________________________________________<br>
&gt; &gt; =C2=A0 =C2=A0 rtcweb mailing list<br>
</div>&gt; &gt; =C2=A0 =C2=A0 <a href=3D"mailto:rtcweb@ietf.org">rtcweb@iet=
f.org</a> &lt;mailto:<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>=
&gt;<br>
<div class=3D""><div class=3D"h5">&gt; &gt; =C2=A0 =C2=A0 <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; &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;<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></div></div></div></div>

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To: Justin Uberti <juberti@google.com>, Lorenzo Miniero <lorenzo@meetecho.com>
Thread-Topic: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to	UDP?
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to	UDP?
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Thread-Topic: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Subject: Re: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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From nobody Tue Jun  3 09:37:50 2014
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Subject: Re: [rtcweb] Review of draft-ietf-rtcweb-data-protocol-05
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On 02 Jun 2014, at 18:22, Doug Leonard <dleonard@getjive.com> wrote:
Hi Doug,

thanks a lot for your review. See my comments inline.

Best regards
Michael
>=20
> -----BEGIN PGP SIGNED MESSAGE-----=20
> Hash: SHA256=20
>=20
>=20
> A1. Section 4, 1st paragraph=20
>=20
> I think the full name of the protocol better establishes context. It
> should be used as a section's first reference to the protocol. Then
> subsequent references may be less explicit. (see A4)=20
>=20
> I would change "This protocol..." to "The Data Channel Establishment
> Protocol..."
Done.
>=20
> A2. Section 4, first bullet=20
>=20
> Should be simplified to read: Reliable or unreliable message
> transmission. In the case of unreliable transmissions, the same level
> of unreliability is used.
Done with using reliable instead of Reliable.
>=20
>=20
> A3. Section 4, second bullet=20
>=20
> Should be simplified to read: In-order or out-of- order message
> delivery.
Done with using in-order instead of In-order.
>=20
>=20
> A4. Section 4, third paragraph=20
>=20
> Change the subsequent reference of "The Data Channel Establishment
> Protocol..." to "This protocol..."
Done.
>=20
> A5. Section 4, third paragraph=20
>=20
> "...uses a two way handshake to open a data channel by combining two
> SCTP streams, one in each direction, with the same SCTP stream
> identifier."
>=20
> For more clarity change to:
>=20
> "...uses a two way handshake to open a data channel. The handshake
> pairs one incoming and one outgoing SCTP stream, each having an
> identical stream identifier, into a single bidirectional channel."
I would prefer:
The handshake pairs one incoming and one outgoing SCTP stream, both =
having the
same SCTP stream identifier, into a single bidirectional channel.
>=20
> A6. Section 4, third paragraph=20
>=20
> I think the sentence that begins "Please note" should be separated out
> into a new paragraph and labeled as a note as follows:
>=20
> Note: The opening side can only send user messages before the
> DATA_CHANNEL_ACK is received.
Why did you insert an "only"?
If you want the note to be in a separate paragraph, I would move it to
the end of the 4th paragraph.
>=20
> A7. Section 4, fourth paragraph=20
>=20
> I think we should formally reference the title of Internet- Drafts
> and then link to their location, rather than using the link as the
> subject of sentences. I would change the sentence that begins "When
> using [I-D.ietf-tsvwg-sctp-dtls-encaps], the method..." to the
> following:
>=20
> =93When using SCTP over DTLS [I-D.ietf-tsvwg-sctp-dtls-encaps], the =
method=85=94
Done.

Best regards
Michael
>=20
>=20
> --=20
> Doug Leonard
> Developer  |  Jive Communications, Inc.
> Jive.com  |  dleonard at getjive . com
>=20
> -----BEGIN PGP SIGNATURE-----=20
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> Comment: http://openpgpjs.org=20
>=20
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> -----END PGP SIGNATURE-----=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 3 Jun 2014 18:37:43 +0200
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Subject: Re: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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2014-06-03 18:25 GMT+02:00 Makaraju, Maridi Raju (Raju)
<Raju.Makaraju@alcatel-lucent.com>:
> [Raju] Let's define "on-the-fly" as changing candidate pair after ICE is =
"completed", i.e. a final candidate has been selected and nominated (via ST=
UN USE-CANDIDATE-ATTRIBUTE). In such a case ICE RFC 5245 "9.1.2.2. Existing=
 Media Streams with ICE Completed" says:
>
> <RFC 5245 excerpt>
>    9.1.2.2.  Existing Media Streams with ICE Completed
>    ...
>    The default destination for media (i.e., the values of the IP
>    addresses and ports in the m and c lines used for that media stream)
>    MUST be the local candidate from the highest-priority nominated pair
>    in the valid list for each component.  This "fixes" the default
>    destination for media to equal the destination ICE has selected for
>    media.
>    The agent MUST include candidate attributes for candidates matching
>    the default destination for each component of the media stream, and
>    MUST NOT include any other candidates.
> </RFC 5245 excerpt>
>
> So, no new candidates can be added when in "completed" state (without ICE=
 restart).
>
> If ICE is in running state, then new candidates can be added. But for suc=
h
> new offers RFC says c/m=3D line must match with default candidate/destina=
tion.
>
> <RFC 5245 excerpt>
>    9.1.2.1.  Existing Media Streams with ICE Running
>    ...
>    The component ID MUST remain the same.  The agent
>    MAY include additional candidates it did not offer previously, but
>    which it has gathered since the last offer/answer exchange, including
>    peer reflexive candidates.
>    The agent MAY change the default destination for media.  As with
>    initial offers, there MUST be a set of candidate attributes in the
>    offer matching this default destination.
>
> </RFC 5245 excerpt>
>
> To add a new ICE candidate in ICE completed state ICE Restarts must be do=
ne. ICE restart preserves existing media and tries to find best new candida=
te pair.
>
> <RFC 5245 excerpt>
>    9.1.1.1.  ICE Restarts
>
>    An agent MAY restart ICE processing for an existing media stream.  An
>    ICE restart, as the name implies, will cause all previous states of
>    ICE processing to be flushed and checks to start anew.  The only
>    difference between an ICE restart and a brand new media session is
>    that, during the restart, media can continue to be sent to the
>    previously validated pair.
>    An agent MUST restart ICE for a media stream if:
>
>    o  The offer is being generated for the purposes of changing the
>       target of the media stream.  In other words, if an agent wants to
>       generate an updated offer that, had ICE not been in use, would
>       result in a new value for the destination of a media component
>
> </RFC 5245 excerpt>
>
> So, if a user moves to a new access network (e.g. LTE to WiFi) then assum=
ing ICE is already completed, ICE must be restarted, thus a new SDP O/A mus=
t be done.


All the above can be summarized as: "RFC 5245 mandates a new SDP O/A
if the selected candidate-pair does not match the address in the c/m
lines".
Well, I already know that, it is written in the RFC. The point is that
the above is useless and a requirement that should not be added by a
network protocol that defines a STUN usage.



> This is also obvious from that fact that new local candidates for the acc=
ess network must be conveyed to the peer so that ICE connectivity checks st=
art and succeed.

Not so obvious. If the mobile is talking to a peer which has public IP
(and peer is a full or lite ICE implementation) then a new SDP O/A is
useless. At the end, and regardless what the initial SDP Offer said,
the peer will discover the new reflexive candidates from the mobile
"on-the-fly" (as the ICE protocol states).




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


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From nobody Tue Jun  3 10:17:30 2014
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Date: Tue, 3 Jun 2014 10:17:19 -0700
Message-ID: <CABkgnnXwEXVapOaeEd+PFQjEM+MBWqnccz22d21AWiS0FjAi=A@mail.gmail.com>
From: Martin Thomson <martin.thomson@gmail.com>
To: "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-03.txt
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On 3 June 2014 05:12, Ram Mohan R (rmohanr) <rmohanr@cisco.com> wrote:
> I agree having the text on consent
> freshness for BUNDLE would make the scope of this draft larger and perhaps
> slow the progress of this document.


The concern is not that there is consent freshness for BUNDLE, it's
that there is DiffServ marking for BUNDLE.  BUNDLE should determine
that (or maybe some other document).  Whatever this document does
should - at most - say that the marking is the same for consent
refresh as it is for initial connectivity checking.


From nobody Tue Jun  3 10:27:25 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>, "Justin Uberti" <juberti@google.com>, Lorenzo Miniero <lorenzo@meetecho.com>
Thread-Topic: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to	UDP?
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From: Eric Rescorla <ekr@rtfm.com>
Date: Tue, 3 Jun 2014 10:26:36 -0700
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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I don't feel strongly about this either way. The original motivation for th=
e
text in 5764 is that people in MMUSIC (not me particularlly) felt we
should be clear on the profile and that DTLS was a different profile. If
that sentiment has changed I would be happy to use RTP/SAVPF
(which is what FFOX uses now, I believe).

-Ekr



On Tue, Jun 3, 2014 at 8:51 AM, Justin Uberti <juberti@google.com> wrote:

> The other thread has been hijacked by an ICE discussion, so reasking here=
.
> Should use of DTLS be indicated by:
> - UDP/TLS/RTP/SAVPF on the m=3D line
> - RTP/SAVPF, and presence of an a=3Dfingerprint line
>
> Some have expressed a preference for RTP/SAVPF, as being more backwards
> compatible. However, others have expressed a preference for
> UDP/TLS/RTP/SAVPF, as that is what RFC 5763/4 says. The last time this ca=
me
> up, in September 2013, the UDP/TLS/RTP/SAVPF proponents were able to
> convince me to include this into jsep-06, so if we want to change it back=
,
> I'd like to see a clear consensus.
>
> On Tue, Jun 3, 2014 at 2:08 AM, Lorenzo Miniero <lorenzo@meetecho.com>
> wrote:
>
>> On Tue, 03 Jun 2014 10:32:43 +0200
>> Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com> wrote:
>>
>> > On the other side, keeping "RTP/SAVFP" would have allowed a server
>> > (SBC/MCU/etc) to send an offer without knowing in advance if the
>> > other side supports SDES or DTLS, by sending both the crypto and
>> > fingerprint attributes and let the receiver choose which one to use
>> > (as chrome has done until recently). I know that it is not important
>> > for webrtc, as DTLS is mandatory and SDES is a must not, but for
>> > supporting other legacy clients it may be easier for some of us.
>> >
>> > Best regards
>> > Sergio
>> >
>>
>>
>> I guess that was the rationale for having RTP/SAVPF in the first
>> place, as initially both SDES and DTLS-SRTP were allowed. There are
>> people confused about not having UDP/TLS/RTP/SAVPF now that DTLS-SRTP
>> is mandatory (Asterisk developers, for instance, although getting it to
>> work from a signalling point of view is a trivial replace matter in
>> JavaScript), but I agree it makes sense to keep RTP/SAVPF, if not just
>> because all browser implementations right now (the legacy of a few
>> months from now on) only accept that, which would render them useless
>> having this change.
>>
>> Lorenzo
>>
>>
>>
>> > El 03/06/2014 8:04, Justin Uberti escribi=C3=B3:
>> > > I think there is value in specifying UDP/TLS/RTP/SAVPF, in that it
>> > > unambiguously indicates the desire to do DTLS-SRTP. Otherwise, one
>> > > has to guess at this from the presence of a=3Dfingerprint, which
>> > > seems dodgy, given that endpoints are free to ignore SDP attributes
>> > > they don't understand.
>> > >
>> > > The fact that UDP and TLS appear in the m=3D line is kind of
>> > > unfortunate, but I think it can be interpreted as "do DTLS-SRTP".
>> > >
>> > >
>> > > On Mon, Jun 2, 2014 at 11:53 AM, Martin Thomson
>> > > <martin.thomson@gmail.com <mailto:martin.thomson@gmail.com>> wrote:
>> > >
>> > >     On 2 June 2014 10:13, Harald Alvestrand <harald@alvestrand.no
>> > >     <mailto:harald@alvestrand.no>> wrote:
>> > >     > The current implementations both send "RTP/SAVPF" in this
>> > >     > field.
>> > >     But this
>> > >     > usage is not specified by RFC 5764.
>> > >
>> > >     Sounds easy to me.  The SDP doesn't matter and can say anything.
>> > >     "RTP/SAVPF" is as good as any other sequence of octets.
>> > >
>> > >     I'm sure that JSEP can include a requirement that is directly
>> > >     contradictory to an existing RFC.  That happens all the time.
>> > > At least this time it can be explicit.
>> > >
>> > >     _______________________________________________
>> > >     rtcweb mailing list
>> > >     rtcweb@ietf.org <mailto: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
>>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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

<div dir=3D"ltr">I don&#39;t feel strongly about this either way. The origi=
nal motivation for the<div>text in 5764 is that people in MMUSIC (not me pa=
rticularlly) felt we</div><div>should be clear on the profile and that DTLS=
 was a different profile. If</div>

<div>that sentiment has changed I would be happy to use RTP/SAVPF</div><div=
>(which is what FFOX uses now, I believe).</div><div><br></div><div>-Ekr</d=
iv><div><br></div><div><div><br></div><div><div><br></div><div><div class=
=3D"gmail_extra">

<div class=3D"gmail_quote">On Tue, Jun 3, 2014 at 8:51 AM, Justin Uberti <s=
pan 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_quo=
te" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"=
>

<div dir=3D"ltr">The other thread has been hijacked by an ICE discussion, s=
o reasking here. Should use of DTLS be indicated by:<div>- UDP/TLS/RTP/SAVP=
F on the m=3D line<br><div>- RTP/SAVPF, and presence of an a=3Dfingerprint =
line<br>



<div><div><div class=3D"gmail_extra"><br>Some have expressed a preference f=
or RTP/SAVPF, as being more backwards compatible. However, others have expr=
essed a preference for UDP/TLS/RTP/SAVPF, as that is what RFC 5763/4 says. =
The last time this came up, in September 2013, the=C2=A0UDP/TLS/RTP/SAVPF p=
roponents were able to convince me to include this into jsep-06, so if we w=
ant to change it back, I&#39;d like to see a clear consensus.<br>



<br><div class=3D"gmail_quote">On Tue, Jun 3, 2014 at 2:08 AM, Lorenzo Mini=
ero <span dir=3D"ltr">&lt;<a href=3D"mailto:lorenzo@meetecho.com" target=3D=
"_blank">lorenzo@meetecho.com</a>&gt;</span> wrote:<br><blockquote class=3D=
"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;borde=
r-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">



<div>On Tue, 03 Jun 2014 10:32:43 +0200<br>
Sergio Garcia Murillo &lt;<a href=3D"mailto:sergio.garcia.murillo@gmail.com=
" target=3D"_blank">sergio.garcia.murillo@gmail.com</a>&gt; wrote:<br>
<br>
&gt; On the other side, keeping &quot;RTP/SAVFP&quot; would have allowed a =
server<br>
&gt; (SBC/MCU/etc) to send an offer without knowing in advance if the<br>
&gt; other side supports SDES or DTLS, by sending both the crypto and<br>
&gt; fingerprint attributes and let the receiver choose which one to use<br=
>
&gt; (as chrome has done until recently). I know that it is not important<b=
r>
&gt; for webrtc, as DTLS is mandatory and SDES is a must not, but for<br>
&gt; supporting other legacy clients it may be easier for some of us.<br>
&gt;<br>
&gt; Best regards<br>
&gt; Sergio<br>
&gt;<br>
<br>
<br>
</div>I guess that was the rationale for having RTP/SAVPF in the first<br>
place, as initially both SDES and DTLS-SRTP were allowed. There are<br>
people confused about not having UDP/TLS/RTP/SAVPF now that DTLS-SRTP<br>
is mandatory (Asterisk developers, for instance, although getting it to<br>
work from a signalling point of view is a trivial replace matter in<br>
JavaScript), but I agree it makes sense to keep RTP/SAVPF, if not just<br>
because all browser implementations right now (the legacy of a few<br>
months from now on) only accept that, which would render them useless<br>
having this change.<br>
<br>
Lorenzo<br>
<div><br>
<br>
<br>
&gt; El 03/06/2014 8:04, Justin Uberti escribi=C3=B3:<br>
&gt; &gt; I think there is value in specifying UDP/TLS/RTP/SAVPF, in that i=
t<br>
&gt; &gt; unambiguously indicates the desire to do DTLS-SRTP. Otherwise, on=
e<br>
&gt; &gt; has to guess at this from the presence of a=3Dfingerprint, which<=
br>
&gt; &gt; seems dodgy, given that endpoints are free to ignore SDP attribut=
es<br>
&gt; &gt; they don&#39;t understand.<br>
&gt; &gt;<br>
&gt; &gt; The fact that UDP and TLS appear in the m=3D line is kind of<br>
&gt; &gt; unfortunate, but I think it can be interpreted as &quot;do DTLS-S=
RTP&quot;.<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; On Mon, Jun 2, 2014 at 11:53 AM, Martin Thomson<br>
</div><div>&gt; &gt; &lt;<a href=3D"mailto:martin.thomson@gmail.com" target=
=3D"_blank">martin.thomson@gmail.com</a> &lt;mailto:<a href=3D"mailto:marti=
n.thomson@gmail.com" target=3D"_blank">martin.thomson@gmail.com</a>&gt;&gt;=
 wrote:<br>


&gt; &gt;<br>
&gt; &gt; =C2=A0 =C2=A0 On 2 June 2014 10:13, Harald Alvestrand &lt;<a href=
=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestrand.no</a>=
<br>
</div><div>&gt; &gt; =C2=A0 =C2=A0 &lt;mailto:<a href=3D"mailto:harald@alve=
strand.no" target=3D"_blank">harald@alvestrand.no</a>&gt;&gt; wrote:<br>
&gt; &gt; =C2=A0 =C2=A0 &gt; The current implementations both send &quot;RT=
P/SAVPF&quot; in this<br>
&gt; &gt; =C2=A0 =C2=A0 &gt; field.<br>
&gt; &gt; =C2=A0 =C2=A0 But this<br>
&gt; &gt; =C2=A0 =C2=A0 &gt; usage is not specified by RFC 5764.<br>
&gt; &gt;<br>
&gt; &gt; =C2=A0 =C2=A0 Sounds easy to me. =C2=A0The SDP doesn&#39;t matter=
 and can say anything.<br>
&gt; &gt; =C2=A0 =C2=A0 &quot;RTP/SAVPF&quot; is as good as any other seque=
nce of octets.<br>
&gt; &gt;<br>
&gt; &gt; =C2=A0 =C2=A0 I&#39;m sure that JSEP can include a requirement th=
at is directly<br>
&gt; &gt; =C2=A0 =C2=A0 contradictory to an existing RFC. =C2=A0That happen=
s all the time.<br>
&gt; &gt; At least this time it can be explicit.<br>
&gt; &gt;<br>
&gt; &gt; =C2=A0 =C2=A0 _______________________________________________<br>
&gt; &gt; =C2=A0 =C2=A0 rtcweb mailing list<br>
</div>&gt; &gt; =C2=A0 =C2=A0 <a href=3D"mailto:rtcweb@ietf.org" target=3D"=
_blank">rtcweb@ietf.org</a> &lt;mailto:<a href=3D"mailto:rtcweb@ietf.org" t=
arget=3D"_blank">rtcweb@ietf.org</a>&gt;<br>
<div><div>&gt; &gt; =C2=A0 =C2=A0 <a href=3D"https://www.ietf.org/mailman/l=
istinfo/rtcweb" target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtc=
web</a><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" target=3D"_blank">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;<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>
</div></div></blockquote></div><br></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></div></div></div>

--f46d043c7b10b909ef04faf1d20d--


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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: Martin Thomson <martin.thomson@gmail.com>, "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-03.txt
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|-----Original Message-----
|From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Martin Thomson
|Sent: Tuesday, June 03, 2014 10:47 PM
|To: Ram Mohan R (rmohanr)
|Cc: rtcweb@ietf.org
|Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness=
-03.txt
|
|On 3 June 2014 05:12, Ram Mohan R (rmohanr) <rmohanr@cisco.com> wrote:
|> I agree having the text on consent
|> freshness for BUNDLE would make the scope of this draft larger and perha=
ps
|> slow the progress of this document.
|
|
|The concern is not that there is consent freshness for BUNDLE, it's
|that there is DiffServ marking for BUNDLE.  BUNDLE should determine
|that (or maybe some other document).  Whatever this document does
|should - at most - say that the marking is the same for consent
|refresh as it is for initial connectivity checking.

That is close to what the draft currently says:

   It is RECOMMENDED that STUN consent checks use the same Diffserv
   Codepoint markings as the media packets sent on that transport
   address.  This follows the recommendation of ICE connectivity check
   described in section 7.1.2.4 of [RFC5245].

Of course, we can replace media packets with ICE connectivity check, if nee=
ded. In addition, we can add a node that the BUNDLE case is out of scope of=
 this document.

Muthu

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


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From: "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>
To: Martin Thomson <martin.thomson@gmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-03.txt
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-----Original Message-----
From: Martin Thomson <martin.thomson@gmail.com>
Date: Tuesday, 3 June 2014 10:47 pm
To: Ram Mohan Ravindranath <rmohanr@cisco.com>
Cc: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org"
<rtcweb@ietf.org>
Subject: Re: [rtcweb] I-D Action:
draft-ietf-rtcweb-stun-consent-freshness-03.txt

>On 3 June 2014 05:12, Ram Mohan R (rmohanr) <rmohanr@cisco.com> wrote:
>> I agree having the text on consent
>> freshness for BUNDLE would make the scope of this draft larger and
>>perhaps
>> slow the progress of this document.
>
>
>The concern is not that there is consent freshness for BUNDLE, it's
>that there is DiffServ marking for BUNDLE.  BUNDLE should determine
>that (or maybe some other document).  Whatever this document does
>should - at most - say that the marking is the same for consent
>refresh as it is for initial connectivity checking.

Agree. Instead of saying pick up highest DSCP value or lowest DSCP value
from the media stream(s) in the BUNDLE for STUN consent messages, we could
just say the DiffServ marking for STUN consent messages is same that is
for initial ICE checks. I believe DiffServ marking for initial ICE checks
when BUNDLE is used is itself a open item and not defined in any draft yet.

Ram
=20

>


From nobody Tue Jun  3 11:29:46 2014
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Date: Tue, 3 Jun 2014 20:29:32 +0200
From: Lorenzo Miniero <lorenzo@meetecho.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to	UDP?
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On Tue, 3 Jun 2014 17:27:12 +0000
Christer Holmberg <christer.holmberg@ericsson.com> wrote:

> +1 for =E2=80=9CUDP/TLS/RTP/SAVPF=E2=80=9D
>=20
> Just for clarification, what is meant by =E2=80=9Cbackward compatibility=
=E2=80=9D? Usage of non-secure RTP? I thought we are not going to allow tha=
t.
>=20


When I mentioned "legacy", I talked about WebRTC implementations out in the=
 wild today, and most importantly browsers. Current up-to-date implementati=
ons of Firefox and Chrome, which both use DTLS-SRTP, don't understand UDP/T=
LS/RTP/SAVPF and so would fail when faced with that in a negotiation. This =
might sound like a minor point, considering the autoupdate in most browsers=
 that could have them aligned when the time comes, but you'd be surprised a=
bout how many older versions are still around that force applications to ha=
ve to deal with all of them with polyfills and the like. This is especially=
 true if we consider the beta vs. stable versions both Chrome and Firefox a=
re making available.

Anyway, I agree this should not be a motivation for sticking to RTP/SAVPF. =
After all, the spec is not stable yet and more things are going to change a=
nyway. Besides, most existing deployments still rely on SDES being availabl=
e in Chrome, and so they'll be forced to update sooner or later when it dis=
appears for good.

Lorenzo


> Regards,
>=20
> Christer
>=20
>=20
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Makaraju, Mari=
di Raju (Raju)
> Sent: 03 June 2014 19:21
> To: Justin Uberti; Lorenzo Miniero
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or =
not to UDP?
>=20
> +1 for =E2=80=9CUDP/TLS/RTP/SAVPF on the m=3D line=E2=80=9D
>=20
> Please don=E2=80=99t treat this like another hijacking attempt =E2=98=BA =
but, unfortunately the rationale for this is somewhat related to the ICE di=
scussion.
> RFC 2327 clearly says transport for RTP/AVP is UDP. RTP/SAVPF RFC 5124 mo=
stly inherits that (and RTP/AVPF RFC 4585) and indicates transport as usual=
ly UDP (alternative being DCCP).
> So, using RTP/SAVPF for DTLS-SRTP does not seem consistent with these RFC=
s. More importantly DTLS-SRTP use of =E2=80=9CUDP/TLS/RTP/SAVPF =E2=80=9C i=
s clearly defined in RFC 5764.
>=20
> Now the case of, =E2=80=9Cchanging from default UDP to final TCP transpor=
t or vice-versa=E2=80=9D for SRTP has to be handled per ICE RFC rules, whic=
h says c/m=3D lines should be consistent with the default destination. So, =
using =E2=80=9CRTP/SAVPF=E2=80=9D to cover all transports does not sound ri=
ght to me, especially when browser has to update the SDP when final selecte=
d candidate and/or transport changes anyway.
>=20
> http://tools.ietf.org/html/rfc5245#section-8.1.2
>=20
>                     If an agent is controlling, it examines the highest-p=
riority
>          nominated candidate pair for each component of each media
>          stream.  If any of those candidate pairs differ from the
>          default candidate pairs in the most recent offer/answer
>          exchange, the controlling agent MUST generate an updated offer
>          as described in Section 9<http://tools.ietf.org/html/rfc5245#sec=
tion-9>.
>=20
> BR
> Raju
>=20
>=20
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Justin Uberti
> Sent: Tuesday, June 03, 2014 10:51 AM
> To: Lorenzo Miniero
> Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
> Subject: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not =
to UDP?
>=20
> The other thread has been hijacked by an ICE discussion, so reasking here=
. Should use of DTLS be indicated by:
> - UDP/TLS/RTP/SAVPF on the m=3D line
> - RTP/SAVPF, and presence of an a=3Dfingerprint line
>=20
> Some have expressed a preference for RTP/SAVPF, as being more backwards c=
ompatible. However, others have expressed a preference for UDP/TLS/RTP/SAVP=
F, as that is what RFC 5763/4 says. The last time this came up, in Septembe=
r 2013, the UDP/TLS/RTP/SAVPF proponents were able to convince me to includ=
e this into jsep-06, so if we want to change it back, I'd like to see a cle=
ar consensus.
> On Tue, Jun 3, 2014 at 2:08 AM, Lorenzo Miniero <lorenzo@meetecho.com<mai=
lto:lorenzo@meetecho.com>> wrote:
> On Tue, 03 Jun 2014 10:32:43 +0200
> Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com<mailto:sergio.garc=
ia.murillo@gmail.com>> wrote:
>=20
> > On the other side, keeping "RTP/SAVFP" would have allowed a server
> > (SBC/MCU/etc) to send an offer without knowing in advance if the
> > other side supports SDES or DTLS, by sending both the crypto and
> > fingerprint attributes and let the receiver choose which one to use
> > (as chrome has done until recently). I know that it is not important
> > for webrtc, as DTLS is mandatory and SDES is a must not, but for
> > supporting other legacy clients it may be easier for some of us.
> >
> > Best regards
> > Sergio
> >
> I guess that was the rationale for having RTP/SAVPF in the first
> place, as initially both SDES and DTLS-SRTP were allowed. There are
> people confused about not having UDP/TLS/RTP/SAVPF now that DTLS-SRTP
> is mandatory (Asterisk developers, for instance, although getting it to
> work from a signalling point of view is a trivial replace matter in
> JavaScript), but I agree it makes sense to keep RTP/SAVPF, if not just
> because all browser implementations right now (the legacy of a few
> months from now on) only accept that, which would render them useless
> having this change.
>=20
> Lorenzo
>=20
>=20
>=20
> > El 03/06/2014 8:04, Justin Uberti escribi=C3=B3:
> > > I think there is value in specifying UDP/TLS/RTP/SAVPF, in that it
> > > unambiguously indicates the desire to do DTLS-SRTP. Otherwise, one
> > > has to guess at this from the presence of a=3Dfingerprint, which
> > > seems dodgy, given that endpoints are free to ignore SDP attributes
> > > they don't understand.
> > >
> > > The fact that UDP and TLS appear in the m=3D line is kind of
> > > unfortunate, but I think it can be interpreted as "do DTLS-SRTP".
> > >
> > >
> > > On Mon, Jun 2, 2014 at 11:53 AM, Martin Thomson
> > > <martin.thomson@gmail.com<mailto:martin.thomson@gmail.com> <mailto:ma=
rtin.thomson@gmail.com<mailto:martin.thomson@gmail.com>>> wrote:
> > >
> > >     On 2 June 2014 10:13, Harald Alvestrand <harald@alvestrand.no<mai=
lto:harald@alvestrand.no>
> > >     <mailto:harald@alvestrand.no<mailto:harald@alvestrand.no>>> wrote:
> > >     > The current implementations both send "RTP/SAVPF" in this
> > >     > field.
> > >     But this
> > >     > usage is not specified by RFC 5764.
> > >
> > >     Sounds easy to me.  The SDP doesn't matter and can say anything.
> > >     "RTP/SAVPF" is as good as any other sequence of octets.
> > >
> > >     I'm sure that JSEP can include a requirement that is directly
> > >     contradictory to an existing RFC.  That happens all the time.
> > > At least this time it can be explicit.
> > >
> > >     _______________________________________________
> > >     rtcweb mailing list
> > >     rtcweb@ietf.org<mailto:rtcweb@ietf.org> <mailto:rtcweb@ietf.org<m=
ailto: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
> >
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org<mailto:rtcweb@ietf.org>
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


--=20
Lorenzo Miniero, COB

Meetecho s.r.l.
Web Conferencing and Collaboration Tools
http://www.meetecho.com


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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Lorenzo Miniero <lorenzo@meetecho.com>, Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to	UDP?
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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2014-06-03 18:20 GMT+02:00 Makaraju, Maridi Raju (Raju)
<Raju.Makaraju@alcatel-lucent.com>:
> +1 for =E2=80=9CUDP/TLS/RTP/SAVPF on the m=3D line=E2=80=9D
>
> Please don=E2=80=99t treat this like another hijacking attempt J but, unf=
ortunately
> the rationale for this is somewhat related to the ICE discussion.
>
> RFC 2327 clearly says transport for RTP/AVP is UDP. RTP/SAVPF RFC 5124
> mostly inherits that (and RTP/AVPF RFC 4585) and indicates transport as
> usually UDP (alternative being DCCP).
>
> So, using RTP/SAVPF for DTLS-SRTP does not seem consistent with these RFC=
s.
> More importantly DTLS-SRTP use of =E2=80=9CUDP/TLS/RTP/SAVPF =E2=80=9C is=
 clearly defined in
> RFC 5764.
>
> Now the case of, =E2=80=9Cchanging from default UDP to final TCP transpor=
t or
> vice-versa=E2=80=9D for SRTP has to be handled per ICE RFC rules, which s=
ays c/m=3D
> lines should be consistent with the default destination. So, using
> =E2=80=9CRTP/SAVPF=E2=80=9D to cover all transports does not sound right =
to me, especially
> when browser has to update the SDP when final selected candidate and/or
> transport changes anyway.


I don't like the above, but I strongly agree with it. So:

- Initially use "UDP/TLS/RTP/SAVPF" as the RFC states.

- If the selected candidate (after ICE procedures) has a different
transport, then fire an event to tell the user about it. Something
like:
    onicestatechange(event)  where:
    - event.readyState=3D"open"
    - event.selectedTransport =3D "TCP/TLS/RTP/SAVPF"
And in case the user calls getLocalDescription() it should retrieve a
SDP with m/c lines matching the transport and address of the selected
local candidate.

(well, I know there is zero interest on implementing that, but the
spec should say something like this in order to make SDP&ICE RFC's
happy).


Regards.


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


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From: "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>
To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>, =?iso-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>
Thread-Topic: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Date: Wed, 4 Jun 2014 03:28:53 +0000
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Subject: Re: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Makaraju, Mari=
di
> Raju (Raju)
> Sent: Tuesday, June 03, 2014 10:46 PM
> To: I=F1aki Baz Castillo
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] DTLS carrying RTP/SAVPF over ICE: To UDP or not to =
UDP?
>=20
>=20
> > All the above can be summarized as: "RFC 5245 mandates a new SDP O/A
> > if the selected candidate-pair does not match the address in the c/m
> > lines".
> > Well, I already know that, it is written in the RFC. The point is that
> > the above is useless and a requirement that should not be added by a
> > network protocol that defines a STUN usage.
>=20
> [Raju] Correct. I think we are on the same page. As we discussed earlier,=
 what
> applies to WebRTC is for the browser to update the SDP and notify the
> application of these events. Then it is up to the application to handle t=
hem
> anyway it wants.
>=20
> > > This is also obvious from that fact that new local candidates for
> > > the
> > access network must be conveyed to the peer so that ICE connectivity
> > checks start and succeed.
> >
> > Not so obvious. If the mobile is talking to a peer which has public IP
> > (and peer is a full or lite ICE implementation) then a new SDP O/A is
>=20
> [Raju] Implementation MUST not assume peer having a public IP.
>=20
> > useless. At the end, and regardless what the initial SDP Offer said,
> > the peer will discover the new reflexive candidates from the mobile
> > "on-the-fly" (as the ICE protocol states).
>=20
> [Raju] I don't think peer discovering new reflexive candidates can happen=
 after
> ICE is completed. First, without ICE restart the mobile which detected a =
new
> local candidate (WiFi) can't send ICE connectivity checks on it if the IC=
E state is
> completed (this is written in the same RFC).=20

If both endpoints support MICE (tools.ietf.org/html/draft-wing-mmusic-ice-m=
obility-06) then after mobility event, endpoint which detected local IP add=
ress change can trigger ICE connectivity checks without depending on ICE re=
start. If the remote peer does not support MICE but the endpoint needs mobi=
lity then it can get that using relayed candidates (TURN client and server =
will have to support "Mobility using TURN"). With MICE, If endpoint has mul=
tiple interfaces then it can maintain unused candidates from interfaces wit=
h lower-priority in standby mode, so that when the active interface is not =
available then media streams can be migrated to any one of these interfaces=
 without depending on ICE restart.
=20
-Tiru

> So, even if there is a NAT in between
> and peer may treat this new candidate as peer reflexive candidate, local =
side
> can't send STUN. Also, implementations should not assume NAT being presen=
t
> locally.
>=20
>=20
> BR
> Raju
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Wed Jun  4 00:33:31 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: =?utf-8?B?ScOxYWtpIEJheiBDYXN0aWxsbw==?= <ibc@aliax.net>, "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
Thread-Topic: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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count me out on this. I see no value in updating the m=3D line protocol
depending on the active protocol in use (e.g. UDP vs TCP).

I suggest that we go ahead with Martin's suggestion of offering RTP/SAVPF,
but also accept UDP/TLS/RTP/SAVPF for remote endpoints that are following
5764 exactly. (What protocol is used in an answer to a UDP/TLS/RTP/SAVPF
offer remains as an open question.)


On Wed, Jun 4, 2014 at 12:33 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> >> +1 for =E2=80=9CUDP/TLS/RTP/SAVPF on the m=3D line=E2=80=9D
> >>
> >> Please don=E2=80=99t treat this like another hijacking attempt J but,
> >> unfortunately the rationale for this is somewhat related to the ICE
> discussion.
> >>
> >> RFC 2327 clearly says transport for RTP/AVP is UDP. RTP/SAVPF RFC 5124
> >> mostly inherits that (and RTP/AVPF RFC 4585) and indicates transport
> >> as usually UDP (alternative being DCCP).
> >>
> >> So, using RTP/SAVPF for DTLS-SRTP does not seem consistent with these
> RFCs.
> >> More importantly DTLS-SRTP use of =E2=80=9CUDP/TLS/RTP/SAVPF =E2=80=9C=
 is clearly
> >> defined in RFC 5764.
> >>
> >> Now the case of, =E2=80=9Cchanging from default UDP to final TCP trans=
port or
> >> vice-versa=E2=80=9D for SRTP has to be handled per ICE RFC rules, whic=
h says
> >> c/m=3D lines should be consistent with the default destination. So,
> >> using =E2=80=9CRTP/SAVPF=E2=80=9D to cover all transports does not sou=
nd right to me,
> >> especially when browser has to update the SDP when final selected
> >> candidate and/or transport changes anyway.
> >
> > I don't like the above, but I strongly agree with it. So:
> >
> > - Initially use "UDP/TLS/RTP/SAVPF" as the RFC states.
> >
> > - If the selected candidate (after ICE procedures) has a different
> transport, then fire an event to tell the user about it. Something
> > like:
> >    onicestatechange(event)  where:
> >    - event.readyState=3D"open"
> >    - event.selectedTransport =3D "TCP/TLS/RTP/SAVPF"
> > And in case the user calls getLocalDescription() it should retrieve a
> SDP with m/c lines matching the transport and address of the selected loc=
al
> candidate.
>
> Let's keep in mind that there are also other data (e.g. port) than the
> transport that may change as part of the ICE procedure.
>
> Regards,
>
> Christer
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">count me out on this. I see no value in updating the m=3D =
line protocol depending on the active protocol in use (e.g. UDP vs TCP).<di=
v><br></div><div>I suggest that we go ahead with Martin&#39;s suggestion of=
 offering RTP/SAVPF, but also accept UDP/TLS/RTP/SAVPF for remote endpoints=
 that are following 5764 exactly. (What protocol is used in an answer to a =
UDP/TLS/RTP/SAVPF offer remains as an open question.)</div>

</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Wed,=
 Jun 4, 2014 at 12:33 AM, 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:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi,<br>
<div class=3D""><br>
&gt;&gt; +1 for =E2=80=9CUDP/TLS/RTP/SAVPF on the m=3D line=E2=80=9D<br>
&gt;&gt;<br>
&gt;&gt; Please don=E2=80=99t treat this like another hijacking attempt J b=
ut,<br>
&gt;&gt; unfortunately the rationale for this is somewhat related to the IC=
E discussion.<br>
&gt;&gt;<br>
&gt;&gt; RFC 2327 clearly says transport for RTP/AVP is UDP. RTP/SAVPF RFC =
5124<br>
&gt;&gt; mostly inherits that (and RTP/AVPF RFC 4585) and indicates transpo=
rt<br>
&gt;&gt; as usually UDP (alternative being DCCP).<br>
&gt;&gt;<br>
&gt;&gt; So, using RTP/SAVPF for DTLS-SRTP does not seem consistent with th=
ese RFCs.<br>
&gt;&gt; More importantly DTLS-SRTP use of =E2=80=9CUDP/TLS/RTP/SAVPF =E2=
=80=9C is clearly<br>
&gt;&gt; defined in RFC 5764.<br>
&gt;&gt;<br>
&gt;&gt; Now the case of, =E2=80=9Cchanging from default UDP to final TCP t=
ransport or<br>
&gt;&gt; vice-versa=E2=80=9D for SRTP has to be handled per ICE RFC rules, =
which says<br>
&gt;&gt; c/m=3D lines should be consistent with the default destination. So=
,<br>
&gt;&gt; using =E2=80=9CRTP/SAVPF=E2=80=9D to cover all transports does not=
 sound right to me,<br>
&gt;&gt; especially when browser has to update the SDP when final selected<=
br>
&gt;&gt; candidate and/or transport changes anyway.<br>
&gt;<br>
&gt; I don&#39;t like the above, but I strongly agree with it. So:<br>
&gt;<br>
&gt; - Initially use &quot;UDP/TLS/RTP/SAVPF&quot; as the RFC states.<br>
&gt;<br>
&gt; - If the selected candidate (after ICE procedures) has a different tra=
nsport, then fire an event to tell the user about it. Something<br>
&gt; like:<br>
&gt; =C2=A0 =C2=A0onicestatechange(event) =C2=A0where:<br>
&gt; =C2=A0 =C2=A0- event.readyState=3D&quot;open&quot;<br>
&gt; =C2=A0 =C2=A0- event.selectedTransport =3D &quot;TCP/TLS/RTP/SAVPF&quo=
t;<br>
&gt; And in case the user calls getLocalDescription() it should retrieve a =
SDP with m/c lines matching the transport and address of the selected local=
 candidate.<br>
<br>
</div>Let&#39;s keep in mind that there are also other data (e.g. port) tha=
n the transport that may change as part of the ICE procedure.<br>
<br>
Regards,<br>
<br>
Christer<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>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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2014-06-04 23:48 GMT+02:00 Justin Uberti <juberti@google.com>:
> count me out on this. I see no value in updating the m=3D line protocol
> depending on the active protocol in use (e.g. UDP vs TCP).
>
> I suggest that we go ahead with Martin's suggestion of offering RTP/SAVPF=
,
> but also accept UDP/TLS/RTP/SAVPF for remote endpoints that are following
> 5764 exactly. (What protocol is used in an answer to a UDP/TLS/RTP/SAVPF
> offer remains as an open question.)

Question: do we have (or will have) a PC event telling us *which* ICE
candidate is the selected one? If so, those who want to play with the
"SDP rules" in RFC 5245 can be notified about the selected candidate
and can even play more by mangling the SDP to "update the default
media address" (after all, I think we all are used to mangle the SDP
with JavaScript...).

Wouldn't that be enough to satisfy all the needs?


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


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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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On Wed, Jun 4, 2014 at 2:48 PM, Justin Uberti <juberti@google.com> wrote:

> count me out on this. I see no value in updating the m=3D line protocol
> depending on the active protocol in use (e.g. UDP vs TCP).
>
> I suggest that we go ahead with Martin's suggestion of offering RTP/SAVPF=
,
> but also accept UDP/TLS/RTP/SAVPF for remote endpoints that are following
> 5764 exactly. (What protocol is used in an answer to a UDP/TLS/RTP/SAVPF
> offer remains as an open question.)
>

This would be acceptable to me, but we probably should update 5764 to make
RTP/SAVPF legal as well.

-Ekr


>
> On Wed, Jun 4, 2014 at 12:33 AM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
>
>> Hi,
>>
>> >> +1 for =E2=80=9CUDP/TLS/RTP/SAVPF on the m=3D line=E2=80=9D
>> >>
>> >> Please don=E2=80=99t treat this like another hijacking attempt J but,
>> >> unfortunately the rationale for this is somewhat related to the ICE
>> discussion.
>> >>
>> >> RFC 2327 clearly says transport for RTP/AVP is UDP. RTP/SAVPF RFC 512=
4
>> >> mostly inherits that (and RTP/AVPF RFC 4585) and indicates transport
>> >> as usually UDP (alternative being DCCP).
>> >>
>> >> So, using RTP/SAVPF for DTLS-SRTP does not seem consistent with these
>> RFCs.
>> >> More importantly DTLS-SRTP use of =E2=80=9CUDP/TLS/RTP/SAVPF =E2=80=
=9C is clearly
>> >> defined in RFC 5764.
>> >>
>> >> Now the case of, =E2=80=9Cchanging from default UDP to final TCP tran=
sport or
>> >> vice-versa=E2=80=9D for SRTP has to be handled per ICE RFC rules, whi=
ch says
>> >> c/m=3D lines should be consistent with the default destination. So,
>> >> using =E2=80=9CRTP/SAVPF=E2=80=9D to cover all transports does not so=
und right to me,
>> >> especially when browser has to update the SDP when final selected
>> >> candidate and/or transport changes anyway.
>> >
>> > I don't like the above, but I strongly agree with it. So:
>> >
>> > - Initially use "UDP/TLS/RTP/SAVPF" as the RFC states.
>> >
>> > - If the selected candidate (after ICE procedures) has a different
>> transport, then fire an event to tell the user about it. Something
>> > like:
>> >    onicestatechange(event)  where:
>> >    - event.readyState=3D"open"
>> >    - event.selectedTransport =3D "TCP/TLS/RTP/SAVPF"
>> > And in case the user calls getLocalDescription() it should retrieve a
>> SDP with m/c lines matching the transport and address of the selected lo=
cal
>> candidate.
>>
>> Let's keep in mind that there are also other data (e.g. port) than the
>> transport that may change as part of the ICE procedure.
>>
>> Regards,
>>
>> Christer
>>
>> _______________________________________________
>> 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"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, Jun 4, 2014 at 2:48 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:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr">count me out on this. I see=
 no value in updating the m=3D line protocol depending on the active protoc=
ol in use (e.g. UDP vs TCP).<div>

<br></div><div>I suggest that we go ahead with Martin&#39;s suggestion of o=
ffering RTP/SAVPF, but also accept UDP/TLS/RTP/SAVPF for remote endpoints t=
hat are following 5764 exactly. (What protocol is used in an answer to a UD=
P/TLS/RTP/SAVPF offer remains as an open question.)</div>

</div></blockquote><div><br></div><div>This would be acceptable to me, but =
we probably should update 5764 to make</div><div>RTP/SAVPF legal as well.</=
div><div><br></div><div>-Ekr</div><div>=C2=A0</div><blockquote class=3D"gma=
il_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-lef=
t:1ex">

<div class=3D"HOEnZb"><div class=3D"h5"><div class=3D"gmail_extra"><br><div=
 class=3D"gmail_quote">On Wed, Jun 4, 2014 at 12:33 AM, Christer Holmberg <=
span dir=3D"ltr">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" targ=
et=3D"_blank">christer.holmberg@ericsson.com</a>&gt;</span> wrote:<br>



<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi,<br>
<div><br>
&gt;&gt; +1 for =E2=80=9CUDP/TLS/RTP/SAVPF on the m=3D line=E2=80=9D<br>
&gt;&gt;<br>
&gt;&gt; Please don=E2=80=99t treat this like another hijacking attempt J b=
ut,<br>
&gt;&gt; unfortunately the rationale for this is somewhat related to the IC=
E discussion.<br>
&gt;&gt;<br>
&gt;&gt; RFC 2327 clearly says transport for RTP/AVP is UDP. RTP/SAVPF RFC =
5124<br>
&gt;&gt; mostly inherits that (and RTP/AVPF RFC 4585) and indicates transpo=
rt<br>
&gt;&gt; as usually UDP (alternative being DCCP).<br>
&gt;&gt;<br>
&gt;&gt; So, using RTP/SAVPF for DTLS-SRTP does not seem consistent with th=
ese RFCs.<br>
&gt;&gt; More importantly DTLS-SRTP use of =E2=80=9CUDP/TLS/RTP/SAVPF =E2=
=80=9C is clearly<br>
&gt;&gt; defined in RFC 5764.<br>
&gt;&gt;<br>
&gt;&gt; Now the case of, =E2=80=9Cchanging from default UDP to final TCP t=
ransport or<br>
&gt;&gt; vice-versa=E2=80=9D for SRTP has to be handled per ICE RFC rules, =
which says<br>
&gt;&gt; c/m=3D lines should be consistent with the default destination. So=
,<br>
&gt;&gt; using =E2=80=9CRTP/SAVPF=E2=80=9D to cover all transports does not=
 sound right to me,<br>
&gt;&gt; especially when browser has to update the SDP when final selected<=
br>
&gt;&gt; candidate and/or transport changes anyway.<br>
&gt;<br>
&gt; I don&#39;t like the above, but I strongly agree with it. So:<br>
&gt;<br>
&gt; - Initially use &quot;UDP/TLS/RTP/SAVPF&quot; as the RFC states.<br>
&gt;<br>
&gt; - If the selected candidate (after ICE procedures) has a different tra=
nsport, then fire an event to tell the user about it. Something<br>
&gt; like:<br>
&gt; =C2=A0 =C2=A0onicestatechange(event) =C2=A0where:<br>
&gt; =C2=A0 =C2=A0- event.readyState=3D&quot;open&quot;<br>
&gt; =C2=A0 =C2=A0- event.selectedTransport =3D &quot;TCP/TLS/RTP/SAVPF&quo=
t;<br>
&gt; And in case the user calls getLocalDescription() it should retrieve a =
SDP with m/c lines matching the transport and address of the selected local=
 candidate.<br>
<br>
</div>Let&#39;s keep in mind that there are also other data (e.g. port) tha=
n the transport that may change as part of the ICE procedure.<br>
<br>
Regards,<br>
<br>
Christer<br>
<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>
</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>

--14dae9cc9b982e4f3204fb09cbde--


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 5 Jun 2014 00:12:16 +0200
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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2014-06-05 0:02 GMT+02:00 Eric Rescorla <ekr@rtfm.com>:
> On Wed, Jun 4, 2014 at 2:48 PM, Justin Uberti <juberti@google.com> wrote:
>>
>> count me out on this. I see no value in updating the m=3D line protocol
>> depending on the active protocol in use (e.g. UDP vs TCP).
>>
>> I suggest that we go ahead with Martin's suggestion of offering RTP/SAVP=
F,
>> but also accept UDP/TLS/RTP/SAVPF for remote endpoints that are followin=
g
>> 5764 exactly. (What protocol is used in an answer to a UDP/TLS/RTP/SAVPF
>> offer remains as an open question.)
>
>
> This would be acceptable to me, but we probably should update 5764 to mak=
e
> RTP/SAVPF legal as well.

Hi, my previous suggestion includes that the m=3D protocol field gets
"UDP/TLS/RTP/SAVPF" as defined by RFC 5764.



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


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--001a11c25860e112a604fb09f69f
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On Wed, Jun 4, 2014 at 3:12 PM, I=C3=B1aki Baz Castillo <ibc@aliax.net> wro=
te:

> 2014-06-05 0:02 GMT+02:00 Eric Rescorla <ekr@rtfm.com>:
> > On Wed, Jun 4, 2014 at 2:48 PM, Justin Uberti <juberti@google.com>
> wrote:
> >>
> >> count me out on this. I see no value in updating the m=3D line protoco=
l
> >> depending on the active protocol in use (e.g. UDP vs TCP).
> >>
> >> I suggest that we go ahead with Martin's suggestion of offering
> RTP/SAVPF,
> >> but also accept UDP/TLS/RTP/SAVPF for remote endpoints that are
> following
> >> 5764 exactly. (What protocol is used in an answer to a UDP/TLS/RTP/SAV=
PF
> >> offer remains as an open question.)
> >
> >
> > This would be acceptable to me, but we probably should update 5764 to
> make
> > RTP/SAVPF legal as well.
>
> Hi, my previous suggestion includes that the m=3D protocol field gets
> "UDP/TLS/RTP/SAVPF" as defined by RFC 5764.


I prefer Justin's suggestion.

-Ekr


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

--001a11c25860e112a604fb09f69f
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, Jun 4, 2014 at 3:12 PM, I=C3=B1aki Baz Castillo <span dir=
=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.n=
et</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">2014-06-05 0:02 GMT+02:00 Eric Rescorla &lt;=
<a href=3D"mailto:ekr@rtfm.com">ekr@rtfm.com</a>&gt;:<br>
<div class=3D"">&gt; On Wed, Jun 4, 2014 at 2:48 PM, Justin Uberti &lt;<a h=
ref=3D"mailto:juberti@google.com">juberti@google.com</a>&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt; count me out on this. I see no value in updating the m=3D line pro=
tocol<br>
&gt;&gt; depending on the active protocol in use (e.g. UDP vs TCP).<br>
&gt;&gt;<br>
&gt;&gt; I suggest that we go ahead with Martin&#39;s suggestion of offerin=
g RTP/SAVPF,<br>
&gt;&gt; but also accept UDP/TLS/RTP/SAVPF for remote endpoints that are fo=
llowing<br>
&gt;&gt; 5764 exactly. (What protocol is used in an answer to a UDP/TLS/RTP=
/SAVPF<br>
&gt;&gt; offer remains as an open question.)<br>
&gt;<br>
&gt;<br>
&gt; This would be acceptable to me, but we probably should update 5764 to =
make<br>
&gt; RTP/SAVPF legal as well.<br>
<br>
</div>Hi, my previous suggestion includes that the m=3D protocol field gets=
<br>
&quot;UDP/TLS/RTP/SAVPF&quot; as defined by RFC 5764.</blockquote><div><br>=
</div><div>I prefer Justin&#39;s suggestion.</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">

<div class=3D"HOEnZb"><div class=3D"h5">
<br>
<br>
--<br>
I=C3=B1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
</div></div></blockquote></div><br></div></div>

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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Justin Uberti <juberti@google.com>, Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
Thread-Index: AQHPgD7abSYqU+IK/kuiw4z1j/rtWJthf6wQ
Date: Wed, 4 Jun 2014 22:15:41 +0000
Message-ID: <E1FE4C082A89A246A11D7F32A95A17828E3E45C1@US70UWXCHMBA02.zam.alcatel-lucent.com>
References: <CAOJ7v-2FTAKGd2ZUNt9PZBW9pFu7c9v7Gx8Z8vFOQo4K3dyr5Q@mail.gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E3DFD6E@US70UWXCHMBA02.zam.alcatel-lucent.com> <CALiegf=jG6jCrAgSqfW9_qCuxr_CEOx3d1s6H3fkQfhzhRWeFQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D34FCBA@ESESSMB209.ericsson.se> <CAOJ7v-0p0nSUnwBX-eRFDr+v03r9uHsCVM4xqCL8wTTLx1+tXA@mail.gmail.com>
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Date: Wed, 4 Jun 2014 16:29:16 -0700
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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On Wed, Jun 4, 2014 at 3:15 PM, Makaraju, Maridi Raju (Raju) <
Raju.Makaraju@alcatel-lucent.com> wrote:

>
>
> count me out on this. I see no value in updating the m= line protocol
> depending on the active protocol in use (e.g. UDP vs TCP).
>
> *[Raju] I see there is a clear value in doing this:*
>
> *1.       *ICE RFC states c/m= lines should be updated to default
> destination for initial offer and sub-sequent offer (selected candidate).
> So, if createOffer() is called after ICE completed then browser has to
> update SDP to reflect this anyway. So, why not browser update SDP and fire
> an event about final candidate been selected? It can fire this event always
> or only when it is different from default candidate that was used on c/m=
> lines. Existing oniceconnectionstatechange () can be used as a trigger and
> application can inspect SDP to learn about the selected candidate.
>
> *2.       *Some applications may want to know the selected candidate and
> transport for various reasons like:
>
> -          Updating middle boxes for better QoS (not just for selected
> transport, but for selected ip, port). I believe this is the main reason
> why ICE RFC says (1) above.
>
> -          Learning about the actual transport that is used, which may
> give clues about the underlying network (restrictive firewalls? wrong
> provisioning on home routers to not allow some type of UDP out?).
>
>              Basically, we may come up with various other reasons, but
> IMHO the main point is, ICE candidate selection
>
>              is an important event and application knowing the selected
> candidate will help do some interesting
>
>              value added things.
>
>
> You could still do that. The question is whether the m= line needs to
explicitly include the protocol used by the active candidate pair.

>
>
> I suggest that we go ahead with Martin's suggestion of offering RTP/SAVPF,
> but also accept UDP/TLS/RTP/SAVPF for remote endpoints that are following
> 5764 exactly. (What protocol is used in an answer to a UDP/TLS/RTP/SAVPF
> offer remains as an open question.)
>
> *[Raju] I think answer should have UDP/TLS/RTP/SAVPF else it will result
> in SDP O/A violation.*
>

Can you point me to the text that requires this? I looked in 3264 before,
but didn't see it.

>

--047d7bacbedc695aa504fb0b00cf
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, Jun 4, 2014 at 3:15 PM, Makaraju, Maridi Raju (Raju) <span =
dir=3D"ltr">&lt;<a href=3D"mailto:Raju.Makaraju@alcatel-lucent.com" target=
=3D"_blank">Raju.Makaraju@alcatel-lucent.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">





<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div><div class=3D"">
<p class=3D"MsoNormal">count me out on this. I see no value in updating the=
 m=3D line protocol depending on the active protocol in use (e.g. UDP vs TC=
P).<u></u><u></u></p>
</div><p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-fam=
ily:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">[Raju] I see =
there is a clear value in doing this:<u></u><u></u></span></i></b></p>
<p><u></u><b><i><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&q=
uot;,&quot;sans-serif&quot;;color:#1f497d"><span>1.<span style=3D"font:7.0p=
t &quot;Times New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span></i></b><u></u><span style=3D"font-size:11.0pt;font-fa=
mily:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">ICE RFC stat=
es c/m=3D lines should be updated to default destination for initial offer =
and sub-sequent offer (selected candidate). So, if createOffer()
 is called after ICE completed then browser has to update SDP to reflect th=
is anyway. So, why not browser update SDP and fire an event about final can=
didate been selected? It can fire this event always or only when it is diff=
erent from default candidate that
 was used on c/m=3D lines. Existing oniceconnectionstatechange () can be us=
ed as a trigger and application can inspect SDP to learn about the selected=
 candidate.<u></u><u></u></span></p>
<p><u></u><b><i><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&q=
uot;,&quot;sans-serif&quot;;color:#1f497d"><span>2.<span style=3D"font:7.0p=
t &quot;Times New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span></i></b><u></u><span style=3D"font-size:11.0pt;font-fa=
mily:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">Some applica=
tions may want to know the selected candidate and transport for various rea=
sons like:<u></u><u></u></span></p>


<p style=3D"margin-left:.75in">
<u></u><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quo=
t;sans-serif&quot;;color:#1f497d"><span>-<span style=3D"font:7.0pt &quot;Ti=
mes New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:11.0pt;font-family:&qu=
ot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">Updating middle boxe=
s for better QoS (not just for selected transport, but for selected ip, por=
t). I believe this is the main reason why ICE RFC says
 (1) above.<u></u><u></u></span></p>
<p style=3D"margin-left:.75in">
<u></u><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quo=
t;sans-serif&quot;;color:#1f497d"><span>-<span style=3D"font:7.0pt &quot;Ti=
mes New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:11.0pt;font-family:&qu=
ot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">Learning about the a=
ctual transport that is used, which may give clues about the underlying net=
work (restrictive firewalls? wrong provisioning on
 home routers to not allow some type of UDP out?).<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0 =C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0Basically, we may come up =
with various other reasons, but IMHO the main point is, ICE candidate selec=
tion
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0is an important event=
 and application knowing the selected candidate will help do some interesti=
ng
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0value added things.<u=
></u><u></u></span></p>
<p style=3D"margin-left:.75in"><br></p></div></div></div></div></blockquote=
><div>You could still do that. The question is whether the m=3D line needs =
to explicitly include the protocol used by the active candidate pair.=C2=A0=
</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div lang=3D"EN-US" link=3D"blue" vlink=3D"p=
urple"><div><div style=3D"border:none;border-left:solid blue 1.5pt;padding:=
0in 0in 0in 4.0pt">

<div><p style=3D"margin-left:.75in"></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div><div class=3D"">
<p class=3D"MsoNormal">I suggest that we go ahead with Martin&#39;s suggest=
ion of offering RTP/SAVPF, but also accept UDP/TLS/RTP/SAVPF for remote end=
points that are following 5764 exactly. (What protocol is used in an answer=
 to a UDP/TLS/RTP/SAVPF offer remains
 as an open question.)<u></u><u></u></p>
</div><p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-fam=
ily:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">[Raju] I thin=
k answer should have UDP/TLS/RTP/SAVPF else it will result in SDP O/A viola=
tion.</span></i></b></p>

</div></div></div></div></div></blockquote><div><br></div><div>Can you poin=
t me to the text that requires this? I looked in 3264 before, but didn&#39;=
t see it.=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 =
0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div><div style=3D"borde=
r:none;border-left:solid blue 1.5pt;padding:0in 0in 0in 4.0pt"><div>
</div>
</div>
</div>
</div>

</blockquote></div><br></div></div>

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Date: Wed, 4 Jun 2014 18:03:06 -0600
From: Eduardo Gueiros <egueiros@jive.com>
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--=C2=A0
Eduardo Gueiros
Sent with Airmail

On June 4, 2014 at 6:01:46 PM, rtcweb=40ietfa.amsl.com (rtcweb=40ietfa.am=
sl.com) wrote:

U4-zbg9p=46jOs
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From nobody Wed Jun  4 17:47:46 2014
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Usage of draft-ietf-rtcweb-overview
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That works for me.

At least getting RFC 2119 language on the ones we know must implement withi=
n the scope of RTCWEB helps.

In terms of implementability of such RFC 2119 language, I think that is an =
issue for the referenced documents, not the overview document. All such ter=
minology really says in that the implementation MUST conform to the MUSTs, =
the SHOULDs and the MAYs in the referenced documents.

I agree how to handle the API documents is a difficult question, and as suc=
h I do not have an answer at the moment. I suggest you leave those as is un=
til we get further input.

If "data formats" is ultimately meant to contain the codec requirements, wh=
y is the audio codec reference elsewhere in the document?

regards


Keith=20

> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of=20
> Harald Alvestrand
> Sent: 03 June 2014 15:05
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] Usage of draft-ietf-rtcweb-overview
>=20
> On 06/03/2014 03:34 PM, DRAGE, Keith (Keith) wrote:
> > I was working through this document, and assessing whether=20
> it could form a conformance assessment for a rtcweb implementation.
> >
> > I came to the conclusion that it could not as it currently stands.
> >
> > While it is standards track, it seems to be missing RFC=20
> 2119 language completely.
> >
> > As such I end up wondering whether some of the referenced=20
> drafts are mandatory to implement, optional to implement (or=20
> could just be ignored - unlikely).
> In that case, the language is unclear, and needs fixing.
> I'll review the document with this in mind; doing things like changing
>=20
>     The data transport protocols used by RTCWEB are described in
>     [I-D.ietf-rtcweb-transports].
>=20
> into
>=20
>     An RTCWEB implementation MUST implement the data=20
> transport protocols described in
>     [I-D.ietf-rtcweb-transports].
>=20
> is pretty mechanical.
> >
> > So what is the philosophy here - Would I be right in=20
> considering that the intent is that all the referenced=20
> normative references are mandatory, i.e. all implementations=20
> implement all these documents?
>=20
> The intent of the normative references is that the referenced=20
> RFCs have to be read, and the implications understood, in=20
> order to correctly implement an RTCWEB system.
>=20
> I believe that's the definition of a normative reference.=20
> Many of these (RFC 2119 being the prime example) are, by=20
> definition, unimplementable.
>=20
> Some of them are classified wrongly, like -data-protocol,=20
> which should be normative. (or deleted, since we now have a=20
> reference to it in -transport).
>=20
> I'm not sure where the W3C API documents should be placed;=20
> for browsers, they're normative, but I'm not sure having a=20
> normative reference to them in this document is the Right=20
> Thing. Opinions?
>=20
> >
> > But if that is the case, how do I treat text in the=20
> document that references draft-cbran-rtcweb-codec, which is=20
> an informative reference, but which the text treats with=20
> equal weight as other normative references?
>=20
> The reference to draft-cbran is due to be deleted, since that=20
> draft is=20
> (I believe) dead. This is a placeholder for a decision on the=20
> MTI video=20
> codec discussion.
>=20
> The particular sentence referencing this draft is:
>=20
>     The mandatory to implement codecs, as well as any profiling
>     requirements for both mandatory and optional codecs, is=20
> described in
>     <WORKING GROUP DRAFT "MEDIA PROCESSING"> (candidate draft:
>     [I-D.cbran-rtcweb-codec].
>=20
> When I wrote this, I thought it was pretty clear that an=20
> implementation=20
> MUST implement the mandatory to implement codecs, but that the draft=20
> specifying them did not exist yet (and the reference to the=20
> wannabe-decision therefore had to be informative). But if you=20
> read it as=20
> "treat with equal weight", I guess that wasn't written clearly enough?
>=20
> > Regards
> >
> > Keith
> > _______________________________________________
> > 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
> =


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Subject: [rtcweb] review [draft-ietf-rtcweb-transports]
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Some thoughts/suggestions on [draft-ietf-rtcweb-transports-04] that
perhaps could help making the document more clear.

- --
1. Paragraph 1, 2, and 3.

Introduction seems to give fragmented information about this document
[draft-ietf-rtcweb-transports] and the general RTCWEB effort. At times
I donât know
to which the introduction is referring to. Consider giving that
information in distinct ways. Suggestion:

- - The IETF RTCWEB effort is aimed at specifying a protocol suite that is
useful for real time multimedia exchange between clients. This protocol
suite is designed for WebRTC and intents to satisfy the security
considerations described in the WebRTC security documents [document link].
- - The IEFT RTCWEB effort is part of the WebRTC cooperation between the
IETF and W3C. The overall effort is described in WebRTC Overview
document, [document link].

1. Paragraph 2, 2nd sentence

âThis document focuses on the dataâĤâ - To me itâs not clear which
document youâre referring to; the one it was just cited, or this
document [draft-ietf-rtcweb-transports].


3.2. 2nd paragraph
âWhen TURN is used, and the TURN server has IPv4 or IPv6 connectivity
   to the peer or its TURN server, candidates of the appropriate types
   MUST be supported.â

This could be simplified to âWhen TURN is used, if the server or the
peer has IPv4 or IPv6 connectivity, candidates of the appropriate types
MUST be supported.â

3.4, 1st Paragraph

Avoid contraction in formal writing, change âitâsâ to âit isâ

3.4, 2nd Paragraph

âthis allows interworking with bothâĤâ A clearer sentence would read:  âa
full implementation allows interworking with bothâĤâ

3.4 3rd Paragraph

missing comma after âNATsâ

3.4 5th Paragraph

missing commas ââĤTURN [comma] using TCP between the client and server
[comma] MUST be supported.â

The sentence could also be rewritten, for clarification, as: âTURN MUST
be supported when using TCP between the client and the server in order
to deal with firewalls that block all UDP traffic. TURN MUST also be
supported when using TLS over TCP between the client and the server.â

3.4, 6th and 7th Paragraphs should be combined.

3.4 7th Paragraph

- - If the intention of the paragraph to show why TURN TCP candidates  do
not provide significant benefit it needs to indicate so: ââĤnot seen as
providing significant benefit since/because:â
- - suggest change âSecondlyâ to âSecondâ
- - suggest change âthat use case is anyway supportedâ to âthis use case
is not supportedâ
- - suggest change âThirdlyâ to âThirdâ

4. 1st Paragraph

It seems to me the word âthatâ in the sentence ââĤprioritization model is
that the application tellsâĤâ should be changed to âwhatâ.

4.1 3rd Paragraph

It seems to me this sentence, â (same as above + DSCP code point)â,
could be clearer  as  â(5-tuple plus DSCP code point)â

- --
Eduardo Gueiros
Developer | Jive Communications, Inc.
Jive.com | egueiros at jive.com

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To: Justin Uberti <juberti@google.com>
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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+1 on Justin's suggestion.


On Wed, Jun 4, 2014 at 4:29 PM, Justin Uberti <juberti@google.com> wrote:

>
>
>
> On Wed, Jun 4, 2014 at 3:15 PM, Makaraju, Maridi Raju (Raju) <
> Raju.Makaraju@alcatel-lucent.com> wrote:
>
>>
>>
>> count me out on this. I see no value in updating the m= line protocol
>> depending on the active protocol in use (e.g. UDP vs TCP).
>>
>> *[Raju] I see there is a clear value in doing this:*
>>
>> *1.       *ICE RFC states c/m= lines should be updated to default
>> destination for initial offer and sub-sequent offer (selected candidate).
>> So, if createOffer() is called after ICE completed then browser has to
>> update SDP to reflect this anyway. So, why not browser update SDP and fire
>> an event about final candidate been selected? It can fire this event always
>> or only when it is different from default candidate that was used on c/m=
>> lines. Existing oniceconnectionstatechange () can be used as a trigger and
>> application can inspect SDP to learn about the selected candidate.
>>
>> *2.       *Some applications may want to know the selected candidate and
>> transport for various reasons like:
>>
>> -          Updating middle boxes for better QoS (not just for selected
>> transport, but for selected ip, port). I believe this is the main reason
>> why ICE RFC says (1) above.
>>
>> -          Learning about the actual transport that is used, which may
>> give clues about the underlying network (restrictive firewalls? wrong
>> provisioning on home routers to not allow some type of UDP out?).
>>
>>              Basically, we may come up with various other reasons, but
>> IMHO the main point is, ICE candidate selection
>>
>>              is an important event and application knowing the selected
>> candidate will help do some interesting
>>
>>              value added things.
>>
>>
>> You could still do that. The question is whether the m= line needs to
> explicitly include the protocol used by the active candidate pair.
>
>>
>>
>> I suggest that we go ahead with Martin's suggestion of offering
>> RTP/SAVPF, but also accept UDP/TLS/RTP/SAVPF for remote endpoints that are
>> following 5764 exactly. (What protocol is used in an answer to a
>> UDP/TLS/RTP/SAVPF offer remains as an open question.)
>>
>> *[Raju] I think answer should have UDP/TLS/RTP/SAVPF else it will result
>> in SDP O/A violation.*
>>
>
> Can you point me to the text that requires this? I looked in 3264 before,
> but didn't see it.
>
>>
>
> _______________________________________________
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> rtcweb@ietf.org
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<div dir=3D"ltr">+1 on Justin&#39;s suggestion.</div><div class=3D"gmail_ex=
tra"><br><br><div class=3D"gmail_quote">On Wed, Jun 4, 2014 at 4:29 PM, Jus=
tin Uberti <span dir=3D"ltr">&lt;<a href=3D"mailto:juberti@google.com" targ=
et=3D"_blank">juberti@google.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr"><br><div class=3D"gmail_ext=
ra"><br><br><div class=3D"gmail_quote"><div class=3D"">On Wed, Jun 4, 2014 =
at 3:15 PM, Makaraju, Maridi Raju (Raju) <span dir=3D"ltr">&lt;<a href=3D"m=
ailto:Raju.Makaraju@alcatel-lucent.com" target=3D"_blank">Raju.Makaraju@alc=
atel-lucent.com</a>&gt;</span> wrote:<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">





<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div><div>
<p class=3D"MsoNormal">count me out on this. I see no value in updating the=
 m=3D line protocol depending on the active protocol in use (e.g. UDP vs TC=
P).<u></u><u></u></p>
</div><p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-fam=
ily:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">[Raju] I see =
there is a clear value in doing this:<u></u><u></u></span></i></b></p>
<p><u></u><b><i><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&q=
uot;,&quot;sans-serif&quot;;color:#1f497d"><span>1.<span style=3D"font:7.0p=
t &quot;Times New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span></i></b><u></u><span style=3D"font-size:11.0pt;font-fa=
mily:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">ICE RFC stat=
es c/m=3D lines should be updated to default destination for initial offer =
and sub-sequent offer (selected candidate). So, if createOffer()
 is called after ICE completed then browser has to update SDP to reflect th=
is anyway. So, why not browser update SDP and fire an event about final can=
didate been selected? It can fire this event always or only when it is diff=
erent from default candidate that
 was used on c/m=3D lines. Existing oniceconnectionstatechange () can be us=
ed as a trigger and application can inspect SDP to learn about the selected=
 candidate.<u></u><u></u></span></p>
<p><u></u><b><i><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&q=
uot;,&quot;sans-serif&quot;;color:#1f497d"><span>2.<span style=3D"font:7.0p=
t &quot;Times New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span></i></b><u></u><span style=3D"font-size:11.0pt;font-fa=
mily:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">Some applica=
tions may want to know the selected candidate and transport for various rea=
sons like:<u></u><u></u></span></p>



<p style=3D"margin-left:.75in">
<u></u><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quo=
t;sans-serif&quot;;color:#1f497d"><span>-<span style=3D"font:7.0pt &quot;Ti=
mes New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:11.0pt;font-family:&qu=
ot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">Updating middle boxe=
s for better QoS (not just for selected transport, but for selected ip, por=
t). I believe this is the main reason why ICE RFC says
 (1) above.<u></u><u></u></span></p>
<p style=3D"margin-left:.75in">
<u></u><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quo=
t;sans-serif&quot;;color:#1f497d"><span>-<span style=3D"font:7.0pt &quot;Ti=
mes New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:11.0pt;font-family:&qu=
ot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">Learning about the a=
ctual transport that is used, which may give clues about the underlying net=
work (restrictive firewalls? wrong provisioning on
 home routers to not allow some type of UDP out?).<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0 =C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0Basically, we may come up =
with various other reasons, but IMHO the main point is, ICE candidate selec=
tion
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0is an important event=
 and application knowing the selected candidate will help do some interesti=
ng
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0value added things.<u=
></u><u></u></span></p>
<p style=3D"margin-left:.75in"><br></p></div></div></div></div></blockquote=
></div><div>You could still do that. The question is whether the m=3D line =
needs to explicitly include the protocol used by the active candidate pair.=
=C2=A0</div>
<div class=3D"">

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div lang=3D"EN-US" link=3D"blue" vlink=3D"p=
urple"><div><div style=3D"border:none;border-left:solid blue 1.5pt;padding:=
0in 0in 0in 4.0pt">


<div><p style=3D"margin-left:.75in"></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div><div>
<p class=3D"MsoNormal">I suggest that we go ahead with Martin&#39;s suggest=
ion of offering RTP/SAVPF, but also accept UDP/TLS/RTP/SAVPF for remote end=
points that are following 5764 exactly. (What protocol is used in an answer=
 to a UDP/TLS/RTP/SAVPF offer remains
 as an open question.)<u></u><u></u></p>
</div><p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-fam=
ily:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">[Raju] I thin=
k answer should have UDP/TLS/RTP/SAVPF else it will result in SDP O/A viola=
tion.</span></i></b></p>


</div></div></div></div></div></blockquote><div><br></div></div><div>Can yo=
u point me to the text that requires this? I looked in 3264 before, but did=
n&#39;t see it.=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">


<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div><div style=3D"borde=
r:none;border-left:solid blue 1.5pt;padding:0in 0in 0in 4.0pt"><div>
</div>
</div>
</div>
</div>

</blockquote></div><br></div></div>
<br>_______________________________________________<br>
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<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>

--001a11c1b548ddeceb04fb0f31f0--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Date: Thu, 5 Jun 2014 07:06:09 +0000
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Eric Rescorla <ekr@rtfm.com>, Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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From nobody Thu Jun  5 01:37:00 2014
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Date: Thu, 05 Jun 2014 10:36:44 +0200
From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Usage of draft-ietf-rtcweb-overview
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On 06/05/2014 02:47 AM, DRAGE, Keith (Keith) wrote:
> That works for me.
>
> At least getting RFC 2119 language on the ones we know must implement within the scope of RTCWEB helps.
>
> In terms of implementability of such RFC 2119 language, I think that is an issue for the referenced documents, not the overview document. All such terminology really says in that the implementation MUST conform to the MUSTs, the SHOULDs and the MAYs in the referenced documents.
>
> I agree how to handle the API documents is a difficult question, and as such I do not have an answer at the moment. I suggest you leave those as is until we get further input.
>
> If "data formats" is ultimately meant to contain the codec requirements, why is the audio codec reference elsewhere in the document?

Because I messed up when inserting it :-)
Will fix.

>
> regards
>
>
> Keith
>
>> -----Original Message-----
>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of
>> Harald Alvestrand
>> Sent: 03 June 2014 15:05
>> To: rtcweb@ietf.org
>> Subject: Re: [rtcweb] Usage of draft-ietf-rtcweb-overview
>>
>> On 06/03/2014 03:34 PM, DRAGE, Keith (Keith) wrote:
>>> I was working through this document, and assessing whether
>> it could form a conformance assessment for a rtcweb implementation.
>>> I came to the conclusion that it could not as it currently stands.
>>>
>>> While it is standards track, it seems to be missing RFC
>> 2119 language completely.
>>> As such I end up wondering whether some of the referenced
>> drafts are mandatory to implement, optional to implement (or
>> could just be ignored - unlikely).
>> In that case, the language is unclear, and needs fixing.
>> I'll review the document with this in mind; doing things like changing
>>
>>      The data transport protocols used by RTCWEB are described in
>>      [I-D.ietf-rtcweb-transports].
>>
>> into
>>
>>      An RTCWEB implementation MUST implement the data
>> transport protocols described in
>>      [I-D.ietf-rtcweb-transports].
>>
>> is pretty mechanical.
>>> So what is the philosophy here - Would I be right in
>> considering that the intent is that all the referenced
>> normative references are mandatory, i.e. all implementations
>> implement all these documents?
>>
>> The intent of the normative references is that the referenced
>> RFCs have to be read, and the implications understood, in
>> order to correctly implement an RTCWEB system.
>>
>> I believe that's the definition of a normative reference.
>> Many of these (RFC 2119 being the prime example) are, by
>> definition, unimplementable.
>>
>> Some of them are classified wrongly, like -data-protocol,
>> which should be normative. (or deleted, since we now have a
>> reference to it in -transport).
>>
>> I'm not sure where the W3C API documents should be placed;
>> for browsers, they're normative, but I'm not sure having a
>> normative reference to them in this document is the Right
>> Thing. Opinions?
>>
>>> But if that is the case, how do I treat text in the
>> document that references draft-cbran-rtcweb-codec, which is
>> an informative reference, but which the text treats with
>> equal weight as other normative references?
>>
>> The reference to draft-cbran is due to be deleted, since that
>> draft is
>> (I believe) dead. This is a placeholder for a decision on the
>> MTI video
>> codec discussion.
>>
>> The particular sentence referencing this draft is:
>>
>>      The mandatory to implement codecs, as well as any profiling
>>      requirements for both mandatory and optional codecs, is
>> described in
>>      <WORKING GROUP DRAFT "MEDIA PROCESSING"> (candidate draft:
>>      [I-D.cbran-rtcweb-codec].
>>
>> When I wrote this, I thought it was pretty clear that an
>> implementation
>> MUST implement the mandatory to implement codecs, but that the draft
>> specifying them did not exist yet (and the reference to the
>> wannabe-decision therefore had to be informative). But if you
>> read it as
>> "treat with equal weight", I guess that wasn't written clearly enough?
>>
>>> Regards
>>>
>>> Keith
>>> _______________________________________________
>>> 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: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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From nobody Thu Jun  5 05:16:04 2014
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Eric Rescorla <ekr@rtfm.com>, Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Date: Thu, 5 Jun 2014 12:15:48 +0000
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 5 Jun 2014 15:12:08 +0200
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To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Hi Makaraju,

I would really appreciate if you don't write your text in red color.
Comments inline:



2014-06-05 14:12 GMT+02:00 Makaraju, Maridi Raju (Raju)
<Raju.Makaraju@alcatel-lucent.com>:
> count me out on this. I see no value in updating the m=3D line protocol
> depending on the active protocol in use (e.g. UDP vs TCP).


> [Raju1] You meant =E2=80=9Cyou could still do that=E2=80=9D using existin=
g procedures?
> Currently I don=E2=80=99t think Chrome updates c/m=3D lines ip/port to ma=
tch with
> final selected candidate.

Of course not, and that is why "you could still do that" (in case we
get some PC event firing when the selected ICE candidate is paired, so
*you* can inspect it at JS level and mangle your SDP).


> Even if Chrome updates c/m=3D lines ip/port to match with selected candid=
ate
> before firing oniceconnectionstatechange(), that may only be partially
> useful as knowing L4 transport protocol (TCP vs. UDP) is also important.

The candidate provides transport info.



> One critical property of offered SDP is transport protocol. This determin=
es
> the type of bearer transport (and # of ports) allocated and the applicati=
on
> profile to be used on top of that. So, an extreme example of having UDP i=
n
> offer and TCP in answer won=E2=80=99t work for sure. Or even TCP/MSRP in =
offer and
> TCP/TLS/MSRP in answer won=E2=80=99t work. So, IMHO I think a general but=
 not so
> clearly written rule with SDP is transports have to match.

Let's make things clear:

The m=3D and c=3D lines in the SDP was originally designed to indicate the
address, transport and protocol of a media stream within the SDP. The
addition of ICE modifies it because:

- ICE lines already contain transport and address information.

- There maybe multiple ICE candidates with different transports and address=
es.

- The only remaining information to be provided is the "protocol",
this is: "RTC", "application", etc.

Said that, the c/m lines are still useful in case peer-A supports ICE
but peer-B does not, so they can perform "legacy" procedures based on
c and m line. THIS IS NOT the case in WebRTC where ICE is a *MUST*.


So honestly I don't think that the W3C should design an API (or make
it more complex) that makes happy old and legacy SDP procedures and
requirements.

In WebRTC, DTLS-SRTP is mandatory (and probably TLS-SRTP will also be
valid soon) so by looking at the ICE candidate transports the receiver
already know if DTLS or TLS is being used (per candidate). We DO NOT
need to signal "DTLS" or "TLS" anywhere else in the SDP, really.

Said that, and after re-thinking all of it a bit, I feel really
comfortable by using "RTP/SAVFP" as that is the *only* information
missing in order to know the nature of the stream: the "protocol". The
transports(s) and the address(es) are already indicated by the
connection parameters: the ICE candidates.

And yes, this "breaks RFC 5764 and 5245". But what it breaks is
useless and legacy and makes no sense in the context of WebRTC. So no
problem.



--=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>
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2014-05-19 13:18 GMT+02:00 Harald Alvestrand <harald@alvestrand.no>:
> The onicecandidate(NULL) always seemed inelegant to me; it wasn't supplyi=
ng
> a candidate, it was indicating something else.
>
> I'd be happy with replacing it with an iceGatheringState change callback =
(if
> we can ensure that this will do the Right Thing in all cases...)

+1




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


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El 05/06/2014 15:18, Iħaki Baz Castillo escribi³:
> 2014-05-19 13:18 GMT+02:00 Harald Alvestrand <harald@alvestrand.no>:
>> The onicecandidate(NULL) always seemed inelegant to me; it wasn't supplying
>> a candidate, it was indicating something else.
>>
>> I'd be happy with replacing it with an iceGatheringState change callback (if
>> we can ensure that this will do the Right Thing in all cases...)
> +1
+1 to onicegatheringstatechange

BR
Sergio


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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [rtcweb] When are ICE candidates added to the SDP
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2014-06-05 15:18 GMT+02:00 I=C3=B1aki Baz Castillo <ibc@aliax.net>:
> 2014-05-19 13:18 GMT+02:00 Harald Alvestrand <harald@alvestrand.no>:
>> The onicecandidate(NULL) always seemed inelegant to me; it wasn't supply=
ing
>> a candidate, it was indicating something else.
>>
>> I'd be happy with replacing it with an iceGatheringState change callback=
 (if
>> we can ensure that this will do the Right Thing in all cases...)
>
> +1

And the opposite is a bug in the spec because if the iceGatheringState
does not fire when indeed the "ICE gathering" status has changed to
"ended", what is it supposed to do?


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


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Date: Thu, 5 Jun 2014 16:06:37 +0200
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To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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2014-06-05 15:48 GMT+02:00 Makaraju, Maridi Raju (Raju)
<Raju.Makaraju@alcatel-lucent.com>:
>> Of course not, and that is why "you could still do that" (in case we
>> get some PC event firing when the selected ICE candidate is paired, so
>> *you* can inspect it at JS level and mangle your SDP).
> [Raju] As I mentioned earlier, how about when TCP and UDP port #s are sam=
e? Take a look at the following example. In this case how do you know the e=
xact Layer 4 transport used?
>
> m=3Daudio 58612 RTP/SAVPF 0 100
> a=3Dcandidate:3275585311 1 udp 2122260223 135.252.136.43 58612 typ host g=
eneration 0
> a=3Dcandidate:2378075119 1 tcp 1518280447 135.252.136.43 58612 typ host g=
eneration 0

Assuming PeerConnection fires a new event
"oniceselectedcandidate(event)" and 'event' contains the selected
local ICECandidate, you get your transport from it (tcp or udp).


>>And yes, this "breaks RFC 5764 and 5245". But what it breaks is
>>useless and legacy and makes no sense in the context of WebRTC. So no
>>problem.
> [Raju] Why intentionally break it and add exceptions?!!

In some draft-rtcweb-xxxxxx:

"This document updates RFC 5764 and 5245 by stating that the m=3D line
does not need to indicate the transport but just the protocol (so
RTP/SAVPF) and blablabla".



> If we go in that direction and set wrong precedence to break RFC rules, t=
hen why not do any or all of the following to have a parallel webrtc univer=
se?

Where must I sign?



> - Just use "RTP" instead of "RTP/SAVPF" and assume DTLS-SRTP always.

As said before in this thread (or in other), the "RTP/SAVPF" is just a
token. The only we need it to provide is the kind of protocol of the
stream ("RTP", "application"...).


> - For data channel, just use "SCTP" instead of "SCTP/DTLS" and assume "DT=
LS" always.

OK. But don't assume DTLS, as TLS may also take place.


> We don't want to do that because webrtc respects and takes advantage of e=
xisting IETF RFCs.

And that's why we are here talking about useless SDP legacy procedures
and requirements, which really, it is NOT an "advantage".



> IHMO, cherry picking is not a good idea... especially here where we don't=
 have a strong case for it. I don't think FF to Chrome interop is a strong =
reason and I already suggested a way to have backward compatibility. IMHO, =
FF and Chrome should have used "UDP/TLS/RTP/SAVPF" for DTLS-SRTP from day 1=
; then we wouldn=E2=80=99t be having this discussion.

Then the discussion about "updating the m line" would take place (in
case the selected ICE candidate uses TLS instead of DTLS).

The standards are wrong since ICE was added to the SDP. The m line no
longer is able to indicate a "transport", since that is now indicated
and chosen via ICE procedures, and there is zero value in updating a
currently useless field in the SDP legacy syntax.



> I don't agree with the comment that "it's legacy"; it may not be prefect =
but since webrtc decided to adapt IETF STUN/ICE/SDP RFCs we should follow t=
hem with as few exceptions (zero should be the goal) as possible.

Unfortunately these kinds of "requirements" were not properly reported
when SDP was selected. Otherwise maybe we wouldn't be now debating
about this.


Regards.





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


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From: Eric Rescorla <ekr@rtfm.com>
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On Thu, Jun 5, 2014 at 7:06 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> wro=
te:

> 2014-06-05 15:48 GMT+02:00 Makaraju, Maridi Raju (Raju)
> <Raju.Makaraju@alcatel-lucent.com>:
> >> Of course not, and that is why "you could still do that" (in case we
> >> get some PC event firing when the selected ICE candidate is paired, so
> >> *you* can inspect it at JS level and mangle your SDP).
> > [Raju] As I mentioned earlier, how about when TCP and UDP port #s are
> same? Take a look at the following example. In this case how do you know
> the exact Layer 4 transport used?
> >
> > m=3Daudio 58612 RTP/SAVPF 0 100
> > a=3Dcandidate:3275585311 1 udp 2122260223 135.252.136.43 58612 typ host
> generation 0
> > a=3Dcandidate:2378075119 1 tcp 1518280447 135.252.136.43 58612 typ host
> generation 0
>
> Assuming PeerConnection fires a new event
> "oniceselectedcandidate(event)" and 'event' contains the selected
> local ICECandidate, you get your transport from it (tcp or udp).
>
>
> >>And yes, this "breaks RFC 5764 and 5245". But what it breaks is
> >>useless and legacy and makes no sense in the context of WebRTC. So no
> >>problem.
> > [Raju] Why intentionally break it and add exceptions?!!
>
> In some draft-rtcweb-xxxxxx:
>
> "This document updates RFC 5764 and 5245 by stating that the m=3D line
> does not need to indicate the transport but just the protocol (so
> RTP/SAVPF) and blablabla".
>
>
>
> > If we go in that direction and set wrong precedence to break RFC rules,
> then why not do any or all of the following to have a parallel webrtc
> universe?
>
> Where must I sign?
>
>
>
> > - Just use "RTP" instead of "RTP/SAVPF" and assume DTLS-SRTP always.
>
> As said before in this thread (or in other), the "RTP/SAVPF" is just a
> token. The only we need it to provide is the kind of protocol of the
> stream ("RTP", "application"...).
>
>
> > - For data channel, just use "SCTP" instead of "SCTP/DTLS" and assume
> "DTLS" always.
>
> OK. But don't assume DTLS, as TLS may also take place.


That's not correct. You use DTLS regardless of whether UDP or TCP
is used.

-Ekr

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Thu, Jun 5, 2014 at 7:06 AM, I=C3=B1aki Baz Castillo <span dir=
=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.n=
et</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">2014-06-05 15:48 GMT+02:00 Makaraju, Maridi =
Raju (Raju)<br>
&lt;<a href=3D"mailto:Raju.Makaraju@alcatel-lucent.com">Raju.Makaraju@alcat=
el-lucent.com</a>&gt;:<br>
<div class=3D"">&gt;&gt; Of course not, and that is why &quot;you could sti=
ll do that&quot; (in case we<br>
&gt;&gt; get some PC event firing when the selected ICE candidate is paired=
, so<br>
&gt;&gt; *you* can inspect it at JS level and mangle your SDP).<br>
&gt; [Raju] As I mentioned earlier, how about when TCP and UDP port #s are =
same? Take a look at the following example. In this case how do you know th=
e exact Layer 4 transport used?<br>
&gt;<br>
&gt; m=3Daudio 58612 RTP/SAVPF 0 100<br>
&gt; a=3Dcandidate:3275585311 1 udp 2122260223 135.252.136.43 58612 typ hos=
t generation 0<br>
&gt; a=3Dcandidate:2378075119 1 tcp 1518280447 135.252.136.43 58612 typ hos=
t generation 0<br>
<br>
</div>Assuming PeerConnection fires a new event<br>
&quot;oniceselectedcandidate(event)&quot; and &#39;event&#39; contains the =
selected<br>
local ICECandidate, you get your transport from it (tcp or udp).<br>
<div class=3D""><br>
<br>
&gt;&gt;And yes, this &quot;breaks RFC 5764 and 5245&quot;. But what it bre=
aks is<br>
&gt;&gt;useless and legacy and makes no sense in the context of WebRTC. So =
no<br>
&gt;&gt;problem.<br>
&gt; [Raju] Why intentionally break it and add exceptions?!!<br>
<br>
</div>In some draft-rtcweb-xxxxxx:<br>
<br>
&quot;This document updates RFC 5764 and 5245 by stating that the m=3D line=
<br>
does not need to indicate the transport but just the protocol (so<br>
RTP/SAVPF) and blablabla&quot;.<br>
<div class=3D""><br>
<br>
<br>
&gt; If we go in that direction and set wrong precedence to break RFC rules=
, then why not do any or all of the following to have a parallel webrtc uni=
verse?<br>
<br>
</div>Where must I sign?<br>
<div class=3D""><br>
<br>
<br>
&gt; - Just use &quot;RTP&quot; instead of &quot;RTP/SAVPF&quot; and assume=
 DTLS-SRTP always.<br>
<br>
</div>As said before in this thread (or in other), the &quot;RTP/SAVPF&quot=
; is just a<br>
token. The only we need it to provide is the kind of protocol of the<br>
stream (&quot;RTP&quot;, &quot;application&quot;...).<br>
<div class=3D""><br>
<br>
&gt; - For data channel, just use &quot;SCTP&quot; instead of &quot;SCTP/DT=
LS&quot; and assume &quot;DTLS&quot; always.<br>
<br>
</div>OK. But don&#39;t assume DTLS, as TLS may also take place.</blockquot=
e><div><br></div><div>That&#39;s not correct. You use DTLS regardless of wh=
ether UDP or TCP</div><div>is used.</div><div>=C2=A0</div><div>-Ekr</div><d=
iv>

<br></div></div></div></div>

--047d7ba979c25cc2b604fb19fa57--


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From nobody Thu Jun  5 12:00:43 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>, "Eric Rescorla" <ekr@rtfm.com>, Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Eric Rescorla <ekr@rtfm.com>, Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>, "Eric Rescorla" <ekr@rtfm.com>, Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Eric Rescorla <ekr@rtfm.com>, Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Eric Rescorla <ekr@rtfm.com>, =?utf-8?B?ScOxYWtpIEJheiBDYXN0aWxsbw==?= <ibc@aliax.net>, Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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From nobody Sat Jun  7 10:15:10 2014
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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On 06/07/2014 03:44 PM, Makaraju, Maridi Raju (Raju) wrote:
>>> - For data channel, just use "SCTP" instead of "SCTP/DTLS" and assume "DTLS" always.
>>> OK. But don't assume DTLS, as TLS may also take place.
>> That's not correct. You use DTLS regardless of whether UDP or TCP
>> is used.
> [Raju] I agree with this. Not to nitpick, but I don't think draft-ietf-rtcweb-transports-04 makes this explicit, which says (3.4):
>    "If TCP connections are used, RTP framing according to [RFC4571] MUST
>    be used, both for the RTP packets and for the DTLS packets used to
>    carry data channels."
>
> TCP ICE RFC 6544 talks about doing RTP over TLS or DTLS. The above text simply says "RTP", should it clarify further to make it explicit like "DTLS-RTP"? Further clarification like "If bundling is used then both data channels and DTLS-SRTP may share same DTLS, else separate DTLS associations are used" may be useful.

Clarification will be done. Proposed language appreciated.

We've all agreed to DTLS for everything since very far back.

>
> BR
> Raju
>  
>


-- 
Surveillance is pervasive. Go Dark.


From nobody Sun Jun  8 21:26:10 2014
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Hi,

The referenced draft is a start for discussion on the implicat ions of =
using different DiffServ treatments for packets in the same IP flow =
(that is, share the same IP 5-tuple.) This will likely be of interest to =
RTCWEB, MMUSIC, AVTCORE, CLUE and possibly other groups that contemplate =
the bundling of multiple media streams on a single IP flow.=20

Please take discussion to the DART working group list (copied.)

Thanks!

Ben.


Begin forwarded message:

> From: I-D-Announce [mailto:i-d-announce-bounces@ietf.org] On Behalf Of =
internet-drafts@ietf.org
> Sent: Friday, June 06, 2014 8:49 PM
> To: i-d-announce@ietf.org
> Subject: I-D Action: draft-york-dart-dscp-rtp-00.txt
>=20
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
>=20
>=20
>        Title           : Differentiated Services (DiffServ) and =
Real-time Communication
>        Authors         : Dan York
>                          David Black
>                          Cullen Jennings
>                          Paul Jones
> 	Filename        : draft-york-dart-dscp-rtp-00.txt
> 	Pages           : 13
> 	Date            : 2014-06-06
>=20
> Abstract:
>   This document describes the interaction between Differentiated
>   Services (DiffServ) network quality of service (QoS) functionality
>   and real-time network communication, including communication based =
on
>   the Real-time Transport Protocol (RTP).  DiffServ is based on =
network
>   nodes applying different forwarding treatments to packets whose IP
>   headers are marked with different DiffServ Code Points (DSCPs).  As =
a
>   result, use of different DSCPs within a single traffic flow may =
cause
>   transport protocol interactions (e.g., due to reordering).  In
>   addition, DSCP markings may be changed or removed between the =
traffic
>   source and destination.  This document covers the implications of
>   these DiffServ aspects for real-time network communication, =
including
>   RTCWEB.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-york-dart-dscp-rtp/
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-york-dart-dscp-rtp-00
>=20


From nobody Sun Jun  8 21:27:59 2014
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Subject: [rtcweb] Fwd: [Dart] I-D Action: draft-york-dart-dscp-rtp-00.txt
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[Resending with the correct address for avtcore. Apologies for the =
repeat]

Hi,

The referenced draft is a start for discussion on the implicat ions of =
using different DiffServ treatments for packets in the same IP flow =
(that is, share the same IP 5-tuple.) This will likely be of interest to =
RTCWEB, MMUSIC, AVTCORE, CLUE and possibly other groups that contemplate =
the bundling of multiple media streams on a single IP flow.=20

Please take discussion to the DART working group list (copied.)

Thanks!

Ben.


Begin forwarded message:

> From: I-D-Announce [mailto:i-d-announce-bounces@ietf.org] On Behalf Of =
internet-drafts@ietf.org
> Sent: Friday, June 06, 2014 8:49 PM
> To: i-d-announce@ietf.org
> Subject: I-D Action: draft-york-dart-dscp-rtp-00.txt
>=20
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
>=20
>=20
>       Title           : Differentiated Services (DiffServ) and =
Real-time Communication
>       Authors         : Dan York
>                         David Black
>                         Cullen Jennings
>                         Paul Jones
> 	Filename        : draft-york-dart-dscp-rtp-00.txt
> 	Pages           : 13
> 	Date            : 2014-06-06
>=20
> Abstract:
>  This document describes the interaction between Differentiated
>  Services (DiffServ) network quality of service (QoS) functionality
>  and real-time network communication, including communication based on
>  the Real-time Transport Protocol (RTP).  DiffServ is based on network
>  nodes applying different forwarding treatments to packets whose IP
>  headers are marked with different DiffServ Code Points (DSCPs).  As a
>  result, use of different DSCPs within a single traffic flow may cause
>  transport protocol interactions (e.g., due to reordering).  In
>  addition, DSCP markings may be changed or removed between the traffic
>  source and destination.  This document covers the implications of
>  these DiffServ aspects for real-time network communication, including
>  RTCWEB.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-york-dart-dscp-rtp/
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-york-dart-dscp-rtp-00
>=20


From nobody Mon Jun  9 07:41:46 2014
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Before this thread went off into varioius strange corners, I think we 
had consensus on the following:

- It's a token. It's not a specification.
- An ANSWER has the same transport token as the OFFER, if acceptable.
   So UDP/TLS/RTP/SAVPF in the offer gets a reply of UDP/TLS/RTP/SAVPF 
in the answer.
   Ditto for RTP/SAVPF if used.
- An updating OFFER has the same transport token as previously agreed.

so the only question we have is:

What should the initial OFFER sent by an RTCWEB-conformant browser contain?

There's a VERY weak compatibility argument with legacy devices for the 
longer string.
There's a VERY weak compatibility argument with deployed browsers for 
the shorter string.
 From the discussion, I don't see much basis for making choice; when I 
flipped a coin, it came up tails, but I'd forgotten to decide which was 
which.

So should we go with UDP/TLS/RTP/SAVPF, and call it a day?



From nobody Mon Jun  9 07:58:07 2014
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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> So should we go with UDP/TLS/RTP/SAVPF, and call it a day?
+1


From nobody Mon Jun  9 08:16:31 2014
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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On 09/06/14 16:41, Harald Alvestrand wrote:
> Before this thread went off into varioius strange corners, I think we
> had consensus on the following:
>=20
> - It's a token. It's not a specification.
> - An ANSWER has the same transport token as the OFFER, if acceptable.
>   So UDP/TLS/RTP/SAVPF in the offer gets a reply of UDP/TLS/RTP/SAVPF i=
n
> the answer.
>   Ditto for RTP/SAVPF if used.
> - An updating OFFER has the same transport token as previously agreed.
>=20
> so the only question we have is:
>=20
> What should the initial OFFER sent by an RTCWEB-conformant browser cont=
ain?
>=20
> There's a VERY weak compatibility argument with legacy devices for the
> longer string.
> There's a VERY weak compatibility argument with deployed browsers for
> the shorter string.

And there's a VERY weak compatibility argument with deployed non-browser
endpoints currently interworking with deployed browsers for the shorter
string.

> From the discussion, I don't see much basis for making choice; when I
> flipped a coin, it came up tails, but I'd forgotten to decide which was=

> which.
>=20
> So should we go with UDP/TLS/RTP/SAVPF, and call it a day?

If there's no compelling reason for changing, can we save ourselves and
other people out there the hassle of fixing their parsers (easy) and
redeploying whatever works today (time-consuming and boring)?

It ain't broken, don't fix it!

Enrico



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From nobody Mon Jun  9 08:51:04 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Date: Mon, 9 Jun 2014 15:50:51 +0000
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Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/QRRiV4t4hoAGmZ7tDikhmBhSNlI
Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Hi,

>So should we go with UDP/TLS/RTP/SAVPF, and call it a day?

+1.

Regards,

Christer


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


From nobody Mon Jun  9 10:14:27 2014
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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On Mon, Jun 9, 2014 at 7:41 AM, Harald Alvestrand <harald@alvestrand.no>
wrote:

> Before this thread went off into varioius strange corners, I think we had
> consensus on the following:
>
> - It's a token. It's not a specification.
> - An ANSWER has the same transport token as the OFFER, if acceptable.
>   So UDP/TLS/RTP/SAVPF in the offer gets a reply of UDP/TLS/RTP/SAVPF in
> the answer.
>   Ditto for RTP/SAVPF if used.
> - An updating OFFER has the same transport token as previously agreed.
>
> so the only question we have is:
>
> What should the initial OFFER sent by an RTCWEB-conformant browser contain?
>
> There's a VERY weak compatibility argument with legacy devices for the
> longer string.
> There's a VERY weak compatibility argument with deployed browsers for the
> shorter string.
> From the discussion, I don't see much basis for making choice; when I
> flipped a coin, it came up tails, but I'd forgotten to decide which was
> which.
>
> So should we go with UDP/TLS/RTP/SAVPF, and call it a day?
>
>
> I am opposed to this, for two reasons:
1) we aren't doing UDP/TLS/SCTP for data channel, so we're inconsistent
2) we're not going to do TCP/TLS/RTP/SAVPF (right?), so it's possibly
incorrect

RTP/SAVPF is similar in terms of how it describes the protocol to the
DTLS/SCTP format that data channel uses, and, as others have pointed out,
is also what existing browsers/apps are using.

So I would prefer to keep RTP/SAVPF as what WebRTC endpoints generate.

--089e013d0502add70504fb6a5776
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Mon, Jun 9, 2014 at 7:41 AM, Harald Alvestrand <span dir=3D"ltr"=
>&lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alves=
trand.no</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Before this thread went off into varioius st=
range corners, I think we had consensus on the following:<br>
<br>
- It&#39;s a token. It&#39;s not a specification.<br>
- An ANSWER has the same transport token as the OFFER, if acceptable.<br>
=C2=A0 So UDP/TLS/RTP/SAVPF in the offer gets a reply of UDP/TLS/RTP/SAVPF =
in the answer.<br>
=C2=A0 Ditto for RTP/SAVPF if used.<br>
- An updating OFFER has the same transport token as previously agreed.<br>
<br>
so the only question we have is:<br>
<br>
What should the initial OFFER sent by an RTCWEB-conformant browser contain?=
<br>
<br>
There&#39;s a VERY weak compatibility argument with legacy devices for the =
longer string.<br>
There&#39;s a VERY weak compatibility argument with deployed browsers for t=
he shorter string.<br>
>From the discussion, I don&#39;t see much basis for making choice; when I f=
lipped a coin, it came up tails, but I&#39;d forgotten to decide which was =
which.<br>
<br>
So should we go with UDP/TLS/RTP/SAVPF, and call it a day?<div class=3D"HOE=
nZb"><div class=3D"h5"><br>
<br></div></div></blockquote><div>I am opposed to this, for two reasons:</d=
iv><div>1) we aren&#39;t doing UDP/TLS/SCTP for data channel, so we&#39;re =
inconsistent</div><div>2) we&#39;re not going to do TCP/TLS/RTP/SAVPF (righ=
t?), so it&#39;s possibly incorrect</div>

<div><br></div><div>RTP/SAVPF is similar in terms of how it describes the p=
rotocol to the DTLS/SCTP format that data channel uses, and, as others have=
 pointed out, is also what existing browsers/apps are using.</div><div>

<br></div><div>So I would prefer to keep RTP/SAVPF as what WebRTC endpoints=
 generate.</div><div>=C2=A0</div></div></div></div>

--089e013d0502add70504fb6a5776--


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From: Roman Shpount <roman@telurix.com>
To: Justin Uberti <juberti@google.com>
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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--f46d04426c2cd636ba04fb6a7e5b
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On Mon, Jun 9, 2014 at 1:14 PM, Justin Uberti <juberti@google.com> wrote:

>
> So I would prefer to keep RTP/SAVPF as what WebRTC endpoints generate.
>
>

I would prefer to keep RTP/SAVPF as well - fewer things for me to fix or
worry about on the interop during the browser version transition.
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">=
On Mon, Jun 9, 2014 at 1:14 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: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 dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"=
gmail_quote">
<div><br></div><div>So I would prefer to keep RTP/SAVPF as what WebRTC endp=
oints generate.</div><div>=C2=A0</div></div></div></div>
</blockquote></div><br></div><div class=3D"gmail_extra">I would prefer to k=
eep RTP/SAVPF as well - fewer things for me to fix or worry about on the in=
terop during the browser version transition.</div><div class=3D"gmail_extra=
">
<div>_____________<br>Roman Shpount</div><div><br></div></div></div>

--f46d04426c2cd636ba04fb6a7e5b--


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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Justin Uberti <juberti@google.com>, Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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On 06/09/2014 07:47 PM, Makaraju, Maridi Raju (Raju) wrote:
>> I am opposed to this, for two reasons:
>> 1) we aren't doing UDP/TLS/SCTP for data channel, so we're inconsistent
> [Raju] That's because there are many application level protocols on top of SCTP data channels. Where as in case of DTLS-SRTP, there is only one protocol.
>
>> 2) we're not going to do TCP/TLS/RTP/SAVPF (right?), so it's possibly incorrect
> [Raju] IMO, for TCP we should use something like "TCP/TLS/RTP/SAVPF" or "TCP/DTLS/RTP/SAVPF". WebRTC may have to register this as RFC 6544 mentions use of DTLS over TCP but did not do registration.

IMO, we should not.

Repeat:
At the time of SDP negotiation, we have no idea whether a TCP or UDP
candidate will be chosen. The transport token merely serves to verify
that we agree on the transport token.

Renegotiating the transport token because of choices made in ICE is
wasteful and useless.

>
>
>> RTP/SAVPF is similar in terms of how it describes the protocol to the DTLS/SCTP >format that data channel uses, and, as others have pointed out, is also what existing >browsers/apps are using.
> [Raju] IMHO, RTP/SAVPF is meant for a specific SRTP-SDES use of transport over UDP. So, it is not same as DTLS-SRTP over UDP. Existing browser apps are using it because both FF and Chrome started with it. If I understand correctly, one of the (main) reasons for Chrome/FF did that way is for easy negotiation between SRTP-SDES and DTLS-SRTP (or was it a bug? as initially accepted by Chrome). IMHO using the same reason now to keep status quo does not sound like a convincing reason to me.
>  
> BR
> Raju
>


-- 
Surveillance is pervasive. Go Dark.


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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Harald Alvestrand <harald@alvestrand.no>, Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Date: Mon, 9 Jun 2014 18:50:06 +0000
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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> >> 2) we're not going to do TCP/TLS/RTP/SAVPF (right?), so it's possibly
> incorrect
> > [Raju] IMO, for TCP we should use something like "TCP/TLS/RTP/SAVPF" or
> "TCP/DTLS/RTP/SAVPF". WebRTC may have to register this as RFC 6544 mentio=
ns
> use of DTLS over TCP but did not do registration.
>=20
> IMO, we should not.
>=20
> Repeat:
> At the time of SDP negotiation, we have no idea whether a TCP or UDP
> candidate will be chosen. The transport token merely serves to verify
> that we agree on the transport token.
[Raju] Repeat from me as well: Per ICE RFC, the c/m=3D lines should reflect=
 the default candidate. If default candidate changes later after ICE comple=
tion then c/m=3D lines should reflect this. Please see notes below as it do=
es not mean new end to end SDP O/A for WebRTC use cases.

>=20
> Renegotiating the transport token because of choices made in ICE is
> wasteful and useless.
[Raju] Sorry, repeat again: I am not suggesting an end to end renegotiation=
. Rather, after ICE completes browser updates the local SDP c/m=3D line(s) =
to reflect the current selected candidate. It's up to the application to ac=
t upon transport change to send a new SDP offer or not. For some QoS needs =
sending new SDP offer, with updated c/m=3D lines, is the best choice and it=
 adds value; WebRTC framework should allow it.
In earlier posts, I mentioned that such update not only serves the purpose =
of RFC compliancy but also allows applications to learn final selected cand=
idate.

If we feel WebRTC just needs a token and not worry about compliancy to RFCs=
 then why not simply register "DTLS-SRTP/SAVPF" (for example) and provide c=
lear semantics and tie it to a=3Dcandidate lines?! This way, RTP/SAVPF use =
won't conflict with standard RFC definition of it (meant for SDES-SRTP).

Even if we go in this direction, the PeerConnection API should still allow =
applications to learn about the selected candidate for various reasons I li=
sted in my earlier post(s).

BR
Raju


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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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El 09/06/2014 20:25, Harald Alvestrand escribió:
> On 06/09/2014 07:47 PM, Makaraju, Maridi Raju (Raju) wrote:
>>> 2) we're not going to do TCP/TLS/RTP/SAVPF (right?), so it's possibly incorrect
>> [Raju] IMO, for TCP we should use something like "TCP/TLS/RTP/SAVPF" or "TCP/DTLS/RTP/SAVPF". WebRTC may have to register this as RFC 6544 mentions use of DTLS over TCP but did not do registration.
> IMO, we should not.
>
> Repeat:
> At the time of SDP negotiation, we have no idea whether a TCP or UDP
> candidate will be chosen. The transport token merely serves to verify
> that we agree on the transport token.
>
> Renegotiating the transport token because of choices made in ICE is
> wasteful and useless.

+1

Best regards
Sergio


From nobody Mon Jun  9 19:18:51 2014
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From: Justin Uberti <juberti@google.com>
Date: Mon, 9 Jun 2014 19:18:26 -0700
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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On Mon, Jun 9, 2014 at 10:47 AM, Makaraju, Maridi Raju (Raju) <
Raju.Makaraju@alcatel-lucent.com> wrote:

> >I am opposed to this, for two reasons:
> >1) we aren't doing UDP/TLS/SCTP for data channel, so we're inconsistent
> [Raju] That's because there are many application level protocols on top of
> SCTP data channels. Where as in case of DTLS-SRTP, there is only one
> protocol.
>

I don't buy this. If we don't care what the underlying transport is for the
data channel, we shouldn't care what it is for media.

However, if we decide that the transport token should reflect the in-use
transport protocol, I don't think that adds any additional signaling - the
app can already decide whether or not it wants to send an updated offer
with the correct default candidate in the m/c lines.


>
> >2) we're not going to do TCP/TLS/RTP/SAVPF (right?), so it's possibly
> incorrect
> [Raju] IMO, for TCP we should use something like "TCP/TLS/RTP/SAVPF" or
> "TCP/DTLS/RTP/SAVPF". WebRTC may have to register this as RFC 6544 mentions
> use of DTLS over TCP but did not do registration.
>
>
> >RTP/SAVPF is similar in terms of how it describes the protocol to the
> DTLS/SCTP >format that data channel uses, and, as others have pointed out,
> is also what existing >browsers/apps are using.
> [Raju] IMHO, RTP/SAVPF is meant for a specific SRTP-SDES use of transport
> over UDP. So, it is not same as DTLS-SRTP over UDP. Existing browser apps
> are using it because both FF and Chrome started with it. If I understand
> correctly, one of the (main) reasons for Chrome/FF did that way is for easy
> negotiation between SRTP-SDES and DTLS-SRTP (or was it a bug? as initially
> accepted by Chrome). IMHO using the same reason now to keep status quo does
> not sound like a convincing reason to me.
>
> BR
> Raju
>
>

--001a11336baa9d256504fb71f297
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Mon, Jun 9, 2014 at 10:47 AM, Makaraju, Maridi Raju (Raju) <span=
 dir=3D"ltr">&lt;<a href=3D"mailto:Raju.Makaraju@alcatel-lucent.com" target=
=3D"_blank">Raju.Makaraju@alcatel-lucent.com</a>&gt;</span> wrote:<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div>&gt;I am opposed to this, for two reaso=
ns:<br>
&gt;1) we aren&#39;t doing UDP/TLS/SCTP for data channel, so we&#39;re inco=
nsistent<br>
</div>[Raju] That&#39;s because there are many application level protocols =
on top of SCTP data channels. Where as in case of DTLS-SRTP, there is only =
one protocol.<br></blockquote><div><br></div><div>I don&#39;t buy this. If =
we don&#39;t care what the underlying transport is for the data channel, we=
 shouldn&#39;t care what it is for media.</div>

<div><br></div>
<div>However, if we decide that the transport token should reflect the in-u=
se transport protocol, I don&#39;t think that adds any additional signaling=
 - the app can already decide whether or not it wants to send an updated of=
fer with the correct default candidate in the m/c lines.</div>

<div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex">
<div><br>
&gt;2) we&#39;re not going to do TCP/TLS/RTP/SAVPF (right?), so it&#39;s po=
ssibly incorrect<br>
</div>[Raju] IMO, for TCP we should use something like &quot;TCP/TLS/RTP/SA=
VPF&quot; or &quot;TCP/DTLS/RTP/SAVPF&quot;. WebRTC may have to register th=
is as RFC 6544 mentions use of DTLS over TCP but did not do registration.<b=
r>



<div><br>
<br>
&gt;RTP/SAVPF is similar in terms of how it describes the protocol to the D=
TLS/SCTP &gt;format that data channel uses, and, as others have pointed out=
, is also what existing &gt;browsers/apps are using.<br>
</div>[Raju] IMHO, RTP/SAVPF is meant for a specific SRTP-SDES use of trans=
port over UDP. So, it is not same as DTLS-SRTP over UDP. Existing browser a=
pps are using it because both FF and Chrome started with it. If I understan=
d correctly, one of the (main) reasons for Chrome/FF did that way is for ea=
sy negotiation between SRTP-SDES and DTLS-SRTP (or was it a bug? as initial=
ly accepted by Chrome). IMHO using the same reason now to keep status quo d=
oes not sound like a convincing reason to me.<br>



=C2=A0<br>
BR<br>
<span><font color=3D"#888888">Raju<br>
<br>
</font></span></blockquote></div><br></div></div>

--001a11336baa9d256504fb71f297--


From nobody Mon Jun  9 23:04:26 2014
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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 Data Channel Establishment Protocol
        Authors         : Randell Jesup
                          Salvatore Loreto
                          Michael Tuexen
	Filename        : draft-ietf-rtcweb-data-protocol-06.txt
	Pages           : 12
	Date            : 2014-06-09

Abstract:
   The Real-Time Communication in WEB-browsers working group is charged
   to provide protocol support for direct interactive rich communication
   using audio, video, and data between two peers' web-browsers.  This
   document specifies a simple protocol for establishing symmetric data
   channels between the peers.  It uses a two way handshake and allows
   sending of user data without waiting for the handshake to complete.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-06

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-data-protocol-06


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From nobody Mon Jun  9 23:05:42 2014
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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 Data Channels
        Authors         : Randell Jesup
                          Salvatore Loreto
                          Michael Tuexen
	Filename        : draft-ietf-rtcweb-data-channel-10.txt
	Pages           : 15
	Date            : 2014-06-09

Abstract:
   The Real-Time Communication in WEB-browsers working group is charged
   to provide protocol support for direct interactive rich communication
   using audio, video, and data between two peers' web-browsers.  This
   document specifies the non-SRTP media data transport aspects of the
   WebRTC framework.  It provides an architectural overview of how the
   Stream Control Transmission Protocol (SCTP) is used in the WebRTC
   context as a generic transport service allowing WEB-browsers to
   exchange generic data from peer to peer.


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

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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>, Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Date: Tue, 10 Jun 2014 13:52:13 +0000
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Hi Harald,

>Repeat:
>At the time of SDP negotiation, we have no idea whether a TCP or UDP candi=
date will be chosen. The transport token merely serves to verify that we ag=
ree on the transport token.
>
>Renegotiating the transport token because of choices made in ICE is wastef=
ul and useless.

I asked this before, and I apologized if I missed your reply: do you assume=
 that the same port would be used for UDP and TCP?

If not, you would still have to "renegotiate" due to the port change - even=
 if the transport token doesn't change.

Regards,

Christer


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 10 Jun 2014 17:35:44 +0200
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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2014-06-10 15:52 GMT+02:00 Christer Holmberg <christer.holmberg@ericsson.co=
m>:
>>Repeat:
>>At the time of SDP negotiation, we have no idea whether a TCP or UDP cand=
idate will be chosen. The transport token merely serves to verify that we a=
gree on the transport token.
>>
>>Renegotiating the transport token because of choices made in ICE is waste=
ful and useless.
>
> I asked this before, and I apologized if I missed your reply: do you assu=
me that the same port would be used for UDP and TCP?
>
> If not, you would still have to "renegotiate" due to the port change - ev=
en if the transport token doesn't change.

There is nothing to re-negotiate at signaling level. What for?


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


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: =?iso-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>
Thread-Topic: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Hi,

>>>Repeat:
>>>At the time of SDP negotiation, we have no idea whether a TCP or UDP can=
didate will be chosen. >>>The transport token merely serves to verify that =
we agree on the transport token.
>>>
>>>Renegotiating the transport token because of choices made in ICE is wast=
eful and useless.
>>
>> I asked this before, and I apologized if I missed your reply: do you ass=
ume that the same port would >> be used for UDP and TCP?
>>
>> If not, you would still have to "renegotiate" due to the port change - e=
ven if the transport token=20
>> doesn't change.
>
> There is nothing to re-negotiate at signaling level. What for?

The m=3D line contains the "transport token" - AND the port number.

Regards,

Christer=


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Dear WG,

This starts a working group last call for our core Data Channel drafts:

draft-ietf-rtcweb-data-protocol-06.txt
draft-ietf-rtcweb-data-channel-10.txt

Please review them in detail, especially for areas which may be of concern
to the broader IETF community.  This WGLC will end on June 27, 2014.

Ted, Sean, Cullen

--e89a8f3bac77d6fd7f04fb7fd53b
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:georgia,=
serif">Dear WG,<br><br><div class=3D"gmail_default" style=3D"font-family:ge=
orgia,serif">This starts a working group last call for our core Data Channe=
l drafts:<br>
<br>draft-ietf-rtcweb-data-protocol-06.txt<br>
draft-ietf-rtcweb-data-channel-10.txt<br><br></div><div class=3D"gmail_defa=
ult" style=3D"font-family:georgia,serif">Please
 review them in detail, especially for areas which may be of concern to=20
the broader IETF community.=C2=A0 This WGLC will end on June 27, 2014.<br>
<br></div>Ted, Sean, Cullen<br></div></div>

--e89a8f3bac77d6fd7f04fb7fd53b--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Ted Hardie <ted.ietf@gmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] WGLC for data channel drafts - sub-protocol usage
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Subject: Re: [rtcweb] WGLC for data channel drafts - sub-protocol usage
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Hi,



A comment on draft-ietf-rtcweb-data-protocol-06:



Section 5.1 says:



"Protocol: Variable Length (sequence of characters)
      The sub-protocol for the channel as a UTF-8 encoded string.  If
      this is an empty string the protocol is unspecified.  If it is a
      non-empty string, it specifies an protocol registered in the
      'WebSocket Subprotocol Name Registry' created in [RFC6455]."





I think "The sub-protocol for the channel as a UTF-8 encoded string." part =
is a little confusing, because there is no such thing as sub-protocol in DC=
EP itself (DCEP only uses the WebSocket sub-protocol registry for registeri=
ng DCEP *protocol* values).



Perhaps we could just remove the sentence, and keep:



"Protocol: Variable Length (sequence of characters)
      If this is an empty string the protocol is unspecified.  If it is a
      non-empty string, it specifies an protocol registered in the
      'WebSocket Subprotocol Name Registry' created in [RFC6455]."



Section 4 gives more information about the usage of the protocol.



(Alternatively, we could use sub-protocol terminology also in DCEP)



Regards,



Christer













________________________________
From: rtcweb [rtcweb-bounces@ietf.org] on behalf of Ted Hardie [ted.ietf@gm=
ail.com]
Sent: Tuesday, 10 June, 2014 9:52 PM
To: rtcweb@ietf.org
Subject: [rtcweb] WGLC for data channel drafts

Dear WG,

This starts a working group last call for our core Data Channel drafts:

draft-ietf-rtcweb-data-protocol-06.txt
draft-ietf-rtcweb-data-channel-10.txt

Please review them in detail, especially for areas which may be of concern =
to the broader IETF community.  This WGLC will end on June 27, 2014.

Ted, Sean, Cullen

--_000_7594FB04B1934943A5C02806D1A2204B1D35B976ESESSMB209erics_
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<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-8859-=
1">
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<body fPStyle=3D"1" ocsi=3D"0">
<div style=3D"direction: ltr;font-family: Tahoma;color: #000000;font-size: =
10pt;">
<p>Hi,</p>
<p>&nbsp;</p>
<p>A comment on draft-ietf-rtcweb-data-protocol-06:</p>
<p>&nbsp;</p>
<p>Section 5.1 says:</p>
<p>&nbsp;</p>
<p>&quot;Protocol: Variable Length (sequence of characters)<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; The sub-protocol for the channel as a UTF-8 =
encoded string.&nbsp; If<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; this is an empty string the protocol is unsp=
ecified.&nbsp; If it is a<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; non-empty string, it specifies an protocol r=
egistered in the<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 'WebSocket Subprotocol Name Registry' create=
d in [RFC6455].&quot;</p>
<p>&nbsp;</p>
<p>&nbsp;</p>
<p>I think &quot;The sub-protocol for the channel as a UTF-8 encoded string=
.&quot; part is a little confusing, because there is no such thing as sub-p=
rotocol in DCEP itself (DCEP only uses the WebSocket sub-protocol registry =
for registering DCEP *protocol* values).</p>
<p>&nbsp;</p>
<p>Perhaps we could just remove the sentence, and keep:</p>
<p>&nbsp;</p>
<p>&quot;Protocol: Variable Length (sequence of characters)<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; If this is an empty string the protocol is u=
nspecified.&nbsp; If it is a<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; non-empty string, it specifies an protocol r=
egistered in the<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 'WebSocket Subprotocol Name Registry' create=
d in [RFC6455].&quot;</p>
<p>&nbsp;</p>
<p>Section 4 gives more information about the usage of the protocol.</p>
<p>&nbsp;</p>
<p>(Alternatively, we could use sub-protocol terminology also in DCEP)</p>
<p>&nbsp;</p>
<p>Regards,</p>
<p>&nbsp;</p>
<p>Christer<br>
</p>
<p>&nbsp;</p>
<p>&nbsp;</p>
<p>&nbsp;</p>
<p>&nbsp;</p>
<p>&nbsp;</p>
<p>&nbsp;</p>
<div style=3D"FONT-SIZE: 16px; FONT-FAMILY: Times New Roman; COLOR: #000000=
">
<hr tabindex=3D"-1">
<div id=3D"divRpF359872" style=3D"DIRECTION: ltr"><font color=3D"#000000" s=
ize=3D"2" face=3D"Tahoma"><b>From:</b> rtcweb [rtcweb-bounces@ietf.org] on =
behalf of Ted Hardie [ted.ietf@gmail.com]<br>
<b>Sent:</b> Tuesday, 10 June, 2014 9:52 PM<br>
<b>To:</b> rtcweb@ietf.org<br>
<b>Subject:</b> [rtcweb] WGLC for data channel drafts<br>
</font><br>
</div>
<div></div>
<div>
<div dir=3D"ltr">
<div class=3D"gmail_default" style=3D"FONT-FAMILY: georgia,serif">Dear WG,<=
br>
<br>
<div class=3D"gmail_default" style=3D"FONT-FAMILY: georgia,serif">This star=
ts a working group last call for our core Data Channel drafts:<br>
<br>
draft-ietf-rtcweb-data-protocol-06.txt<br>
draft-ietf-rtcweb-data-channel-10.txt<br>
<br>
</div>
<div class=3D"gmail_default" style=3D"FONT-FAMILY: georgia,serif">Please re=
view them in detail, especially for areas which may be of concern to the br=
oader IETF community.&nbsp; This WGLC will end on June 27, 2014.<br>
<br>
</div>
Ted, Sean, Cullen<br>
</div>
</div>
</div>
</div>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1D35B976ESESSMB209erics_--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Ted Hardie <ted.ietf@gmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] WGLC for data channel drafts - web terminology
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Hi,



A comment (which has been raised before) on draft-ietf-rtcweb-data-channel-=
10:



The name of the mechanism is "WebRTC Data Channel", and much of the text gi=
ves a picture that the mechanism is only to be used within the WebRTC conte=
xt.



If we want to call it "WebRTC Data Channel", we should make it clear that W=
ebRTC is only one scenario where it can be used, but that it can also be us=
ed elsewhere. For example, we are using it in CLUE.



A few places, where I think we need modifications to reflect it.



Section 3:

------------



Q3_1: s/For general use cases see [I-D.ietf-rtcweb-use-cases-and-requiremen=
ts]./For WebRTC use cases see [I-D.ietf-rtcweb-use-cases-and-requirements].



Q3_2: In sections 3.1 and 3.2, there are a number of WebRTC specific requir=
ements, and "PeerConnection", "JavaScript" etc terminology is used. I think=
 it should be clear that those requirements come from WebRTC.





Section 5:

------------



Q5_1: I suggest to replace:



    "SCTP multihoming will not be used in WebRTC."



...with something like:



    "The WebRTC Data Channel mechanism does not support SCTP multihoming."





Section 6:

------------



Q6_1: In section 6.2, the follwoing sentence:



  "The SCTP association will be set up when the two endpoints of the
   WebRTC PeerConnection agree on opening it, as negotiated by JSEP
   (typically an exchange of SDP) [I-D.ietf-rtcweb-jsep]."



...only applies to WebRTC. But, it needs to be clear that, in other context=
s, other mechanisms can be used.





Q6_2: In section 6.4, I suggest to re-order the paragraphs in the following=
 way:



   "The realization of a bidirectional Data Channel is a pair of one
   incoming stream and one outgoing SCTP stream having the same stream
   SCTP identifier.

   How stream values are selected is protocol and implementation
   dependent.

   Each data channel also has a priority, which is an 2 byte unsigned
   integer value.  These priorities MUST be interpreted as weighted-
   fair-queuing scheduling priorities per the definition of the
   corresponding stream scheduler supporting interleaving in
   [I-D.ietf-tsvwg-sctp-ndata].  For use in WebRTC, the values used
   SHOULD be one of 128 ("below normal"), 256 ("normal"), 512 ("high")
   or 1024 ("extra high").

   The W3C has consensus on defining the application API for WebRTC
   DataChannels to be bidirectional.  They also consider the notions of
   in-sequence, out-of-sequence, reliable and unreliable as properties
   of Channels.  One strong wish is for the application-level API to be
   close to the API for WebSockets, which implies bidirectional streams
   of data and waiting for onopen to fire before sending, a textual
   label used to identify the meaning of the stream, among other things."



...because, the realization is the important part :)



Regards,



Christer











________________________________
From: rtcweb [rtcweb-bounces@ietf.org] on behalf of Ted Hardie [ted.ietf@gm=
ail.com]
Sent: Tuesday, 10 June, 2014 9:52 PM
To: rtcweb@ietf.org
Subject: [rtcweb] WGLC for data channel drafts

Dear WG,

This starts a working group last call for our core Data Channel drafts:

draft-ietf-rtcweb-data-protocol-06.txt
draft-ietf-rtcweb-data-channel-10.txt

Please review them in detail, especially for areas which may be of concern =
to the broader IETF community.  This WGLC will end on June 27, 2014.

Ted, Sean, Cullen

--_000_7594FB04B1934943A5C02806D1A2204B1D35B9DCESESSMB209erics_
Content-Type: text/html; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<html dir=3D"ltr">
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-8859-=
1">
<style id=3D"owaParaStyle">P {
	MARGIN-BOTTOM: 0px; MARGIN-TOP: 0px
}
</style>
</head>
<body fPStyle=3D"1" ocsi=3D"0">
<div style=3D"direction: ltr;font-family: Tahoma;color: #000000;font-size: =
10pt;">
<p>Hi,</p>
<p>&nbsp;</p>
<p>A comment (which has been raised before)&nbsp;on draft-ietf-rtcweb-data-=
channel-10:</p>
<p>&nbsp;</p>
<p>The name of the mechanism is &quot;WebRTC Data Channel&quot;, and much o=
f the text gives a picture that the mechanism is only to be used within the=
 WebRTC context.</p>
<p>&nbsp;</p>
<p>If we want to call it &quot;WebRTC Data Channel&quot;, we should make it=
 clear that WebRTC is only one scenario where it can be used, but that it c=
an also be used elsewhere. For example, we are using it in CLUE.</p>
<p>&nbsp;</p>
<p>A few places, where I think we need modifications to reflect it.</p>
<p>&nbsp;</p>
<p>Section 3:</p>
<p>------------</p>
<p>&nbsp;</p>
<p>Q3_1: s/For general use cases see [I-D.ietf-rtcweb-use-cases-and-require=
ments]./For WebRTC use cases see [I-D.ietf-rtcweb-use-cases-and-requirement=
s].<br>
</p>
<p>&nbsp;</p>
<p>Q3_2: In sections 3.1 and 3.2, there are a number of WebRTC specific req=
uirements, and &quot;PeerConnection&quot;, &quot;JavaScript&quot; etc&nbsp;=
terminology is used. I think it should be clear that those requirements com=
e from WebRTC.</p>
<p>&nbsp;</p>
<p>&nbsp;</p>
<p>Section 5:</p>
<p>------------</p>
<p>&nbsp;</p>
<p>Q5_1: I suggest to replace:</p>
<p>&nbsp;&nbsp;&nbsp;&nbsp;</p>
<p>&nbsp;&nbsp;&nbsp;&nbsp;&quot;SCTP multihoming will not be used in WebRT=
C.&quot;</p>
<p>&nbsp;</p>
<p>...with something like:</p>
<p>&nbsp;</p>
<p>&nbsp;&nbsp;&nbsp;&nbsp;&quot;The WebRTC Data Channel mechanism does not=
 support SCTP multihoming.&quot;</p>
<p>&nbsp;</p>
<p>&nbsp;</p>
<p>Section 6:</p>
<p>------------</p>
<p>&nbsp;</p>
<p>Q6_1: In section 6.2, the follwoing sentence:</p>
<p>&nbsp;</p>
<p>&nbsp; &quot;The SCTP association will be set up when the two endpoints =
of the<br>
&nbsp;&nbsp; WebRTC PeerConnection agree on opening it, as negotiated by JS=
EP<br>
&nbsp;&nbsp; (typically an exchange of SDP) [I-D.ietf-rtcweb-jsep].&quot;</=
p>
<p>&nbsp;</p>
<p>...only applies to WebRTC. But, it needs to be clear that, in other cont=
exts, other mechanisms can be used.</p>
<p>&nbsp;</p>
<p>&nbsp;</p>
<p>Q6_2: In section 6.4, I suggest to re-order the paragraphs in the follow=
ing way:</p>
<p>&nbsp;</p>
<p>&nbsp;&nbsp; &quot;The realization of a bidirectional Data Channel is a =
pair of one<br>
&nbsp;&nbsp; incoming stream and one outgoing SCTP stream having the same s=
tream<br>
&nbsp;&nbsp; SCTP identifier.<br>
<br>
&nbsp;&nbsp; How stream values are selected is protocol and implementation<=
br>
&nbsp;&nbsp; dependent.<br>
<br>
&nbsp;&nbsp; Each data channel also has a priority, which is an 2 byte unsi=
gned<br>
&nbsp;&nbsp; integer value.&nbsp; These priorities MUST be interpreted as w=
eighted-<br>
&nbsp;&nbsp; fair-queuing scheduling priorities per the definition of the<b=
r>
&nbsp;&nbsp; corresponding stream scheduler supporting interleaving in<br>
&nbsp;&nbsp; [I-D.ietf-tsvwg-sctp-ndata].&nbsp; For use in WebRTC, the valu=
es used<br>
&nbsp;&nbsp; SHOULD be one of 128 (&quot;below normal&quot;), 256 (&quot;no=
rmal&quot;), 512 (&quot;high&quot;)<br>
&nbsp;&nbsp; or 1024 (&quot;extra high&quot;).<br>
<br>
&nbsp;&nbsp; The W3C has consensus on defining the application API for WebR=
TC<br>
&nbsp;&nbsp; DataChannels to be bidirectional.&nbsp; They also consider the=
 notions of<br>
&nbsp;&nbsp; in-sequence, out-of-sequence, reliable and unreliable as prope=
rties<br>
&nbsp;&nbsp; of Channels.&nbsp; One strong wish is for the application-leve=
l API to be<br>
&nbsp;&nbsp; close to the API for WebSockets, which implies bidirectional s=
treams<br>
&nbsp;&nbsp; of data and waiting for onopen to fire before sending, a textu=
al<br>
&nbsp;&nbsp; label used to identify the meaning of the stream, among other =
things.&quot;</p>
<p>&nbsp;</p>
<p>...because, the realization is the important part :)<br>
</p>
<p>&nbsp;</p>
<p>Regards,</p>
<p>&nbsp;</p>
<p>Christer</p>
<p>&nbsp;</p>
<p>&nbsp;</p>
<p>&nbsp;</p>
<p>&nbsp;</p>
<p>&nbsp;</p>
<div style=3D"FONT-SIZE: 16px; FONT-FAMILY: Times New Roman; COLOR: #000000=
">
<hr tabindex=3D"-1">
<div id=3D"divRpF222619" style=3D"DIRECTION: ltr"><font color=3D"#000000" s=
ize=3D"2" face=3D"Tahoma"><b>From:</b> rtcweb [rtcweb-bounces@ietf.org] on =
behalf of Ted Hardie [ted.ietf@gmail.com]<br>
<b>Sent:</b> Tuesday, 10 June, 2014 9:52 PM<br>
<b>To:</b> rtcweb@ietf.org<br>
<b>Subject:</b> [rtcweb] WGLC for data channel drafts<br>
</font><br>
</div>
<div></div>
<div>
<div dir=3D"ltr">
<div class=3D"gmail_default" style=3D"FONT-FAMILY: georgia,serif">Dear WG,<=
br>
<br>
<div class=3D"gmail_default" style=3D"FONT-FAMILY: georgia,serif">This star=
ts a working group last call for our core Data Channel drafts:<br>
<br>
draft-ietf-rtcweb-data-protocol-06.txt<br>
draft-ietf-rtcweb-data-channel-10.txt<br>
<br>
</div>
<div class=3D"gmail_default" style=3D"FONT-FAMILY: georgia,serif">Please re=
view them in detail, especially for areas which may be of concern to the br=
oader IETF community.&nbsp; This WGLC will end on June 27, 2014.<br>
<br>
</div>
Ted, Sean, Cullen<br>
</div>
</div>
</div>
</div>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1D35B9DCESESSMB209erics_--


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To: Justin Uberti <juberti@google.com>, "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
Thread-Topic: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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From: Harald Alvestrand <harald@alvestrand.no>
Date: Tue, 10 Jun 2014 23:15:40 +0200
To: Christer Holmberg <christer.holmberg@ericsson.com>, "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>, Justin Uberti <juberti@google.com>
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On 10. juni 2014 15:52:13 CEST, Christer Holmberg <christer.holmberg@ericsson.com> wrote:
>Hi Harald,
>
>>Repeat:
>>At the time of SDP negotiation, we have no idea whether a TCP or UDP
>candidate will be chosen. The transport token merely serves to verify
>that we agree on the transport token.
>>
>>Renegotiating the transport token because of choices made in ICE is
>wasteful and useless.
>
>I asked this before, and I apologized if I missed your reply: do you
>assume that the same port would be used for UDP and TCP?
>
>If not, you would still have to "renegotiate" due to the port change -
>even if the transport token doesn't change.
>
>Regards,
>
>Christer

Even if you wanted the ports to be the same it is unlikely that you could get them consistently. Port change must be accommodated.
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<html><head></head><body><br><br><div class="gmail_quote">On 10. juni 2014 15:52:13 CEST, Christer Holmberg &lt;christer.holmberg@ericsson.com&gt; wrote:<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<pre class="k9mail">Hi Harald,<br /><br /><blockquote class="gmail_quote" style="margin: 0pt 0pt 1ex 0.8ex; border-left: 1px solid #729fcf; padding-left: 1ex;">Repeat:<br />At the time of SDP negotiation, we have no idea whether a TCP or UDP candidate will be chosen. The transport token merely serves to verify that we agree on the transport token.<br /><br />Renegotiating the transport token because of choices made in ICE is wasteful and useless.<br /></blockquote><br />I asked this before, and I apologized if I missed your reply: do you assume that the same port would be used for UDP and TCP?<br /><br />If not, you would still have to "renegotiate" due to the port change - even if the transport token doesn't change.<br /><br />Regards,<br /><br />Christer<br /><br /></pre></blockquote></div><br>
Even if you wanted the ports to be the same it is unlikely that you could get them consistently. Port change must be accommodated.<br>
-- <br>
Sent from my Android device with K-9 Mail. Please excuse my brevity.</body></html>
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 10 Jun 2014 23:40:34 +0200
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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2014-06-10 20:03 GMT+02:00 Christer Holmberg <christer.holmberg@ericsson.co=
m>:
>>> If not, you would still have to "renegotiate" due to the port change - =
even if the transport token
>>> doesn't change.
>>
>> There is nothing to re-negotiate at signaling level. What for?
>
> The m=3D line contains the "transport token" - AND the port number.

So you are just quoting what the RFC says ("the SDP must be updated
once the stream has a winner ICE candidate"). Some points:

- The above does not require AT ALL that a new "SDP renegotiation" is
needed. In fact, SDP O/A was previously negotiated for ICE to work.
IMHO this is clear.

- The ICE RFC states that the SDP must be updated and so on. OK. It is
clear (IMHO) that the ICE spec should not mention SDP at all. And we
all agree that "sending a new SDP with the updated m/c line" is just
required for some QoS, B2BUAs or similar systems, and we all know that
this may make sense in certain signaling protocols (SIP and others),
but NOT in WebRTC where there is not a defined/mandated "signaling
protocol" at all (so in case the app wants to indicate the server its
winner ICE candidate it can do that in tons of ways).

- And at the end, IMHO it is clear that browser vendors are not going
to implement the "SDP update" mentioned in the ICE RFC because they
find it not useful (as it clearly is).

Regards.



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


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Date: Wed, 11 Jun 2014 00:29:27 +0200
From: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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El 10/06/2014 23:40, Iħaki Baz Castillo escribi³:
> 2014-06-10 20:03 GMT+02:00 Christer Holmberg <christer.holmberg@ericsson.com>:
>>>> If not, you would still have to "renegotiate" due to the port change - even if the transport token
>>>> doesn't change.
>>> There is nothing to re-negotiate at signaling level. What for?
>> The m= line contains the "transport token" - AND the port number.
> So you are just quoting what the RFC says ("the SDP must be updated
> once the stream has a winner ICE candidate"). Some points:
>
> - The above does not require AT ALL that a new "SDP renegotiation" is
> needed. In fact, SDP O/A was previously negotiated for ICE to work.
> IMHO this is clear.
>
> - The ICE RFC states that the SDP must be updated and so on. OK. It is
> clear (IMHO) that the ICE spec should not mention SDP at all. And we
> all agree that "sending a new SDP with the updated m/c line" is just
> required for some QoS, B2BUAs or similar systems, and we all know that
> this may make sense in certain signaling protocols (SIP and others),
> but NOT in WebRTC where there is not a defined/mandated "signaling
> protocol" at all (so in case the app wants to indicate the server its
> winner ICE candidate it can do that in tons of ways).
>
>

I found this part of the RFC particular insightful:

    This begs the question -- why is the updated offer/answer exchange
    needed at all?  Indeed, in a pure offer/answer environment, it would
    not be.  The offerer and answerer will agree on the candidates to use
    through ICE, and then can begin using them.  As far as the agents
    themselves are concerned, the updated offer/answer provides no new
    information.

Best regards
Sergio




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To: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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From nobody Tue Jun 10 23:40:52 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
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Thread-Topic: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Thread-Topic: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Date: Wed, 11 Jun 2014 11:25:41 +0000
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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El 11/06/2014 13:25, Makaraju, Maridi Raju (Raju) escribi³:
>> Raju, for example, in case of client or MCU mobility, we don't need to exchange media >info again, as it will be exactly the same. In our case we only want to propagate new >ICE candidates to recheck with ICE if there is a different valid pair.  But I think >the solutions commented in the thread are still compatible with your idea. As long as >the first offer/answer messages are still valid you could send them again with the >new candidates to the middle boxes if you need it.
> [Raju]
> Hi Javier, Thanks for explaining your MCU use cases.
> I wanted to assert again that, I am not advocating for an automatic new SDP O/A. We need to separate updating local SDP from generating new SDP offer. The former has to be done by browser and the latter is up to the application and falls into signaling (which webrtc has left to applications).
> At a very high-level, this is how the flow looks like:
> 1. When ICE is completed, browser will update local SDP
>     c/m= lines with selected candidate for:
>     a. Local IP, port
>     b. Matching transport (or to indicate the transport
>        selected. TCP or UDP).
> 2. Browser calls an API (TBD) like onicecandidate(NULL)
>     to indicate end of ICE.

Isn't (2) already defined in current api with 
|oniceconnectionstatechange|with state "connected" ? Maybe it would be 
interesting to also add the selected candidate in the event as I see it 
useful for applications.

As a side note, is anyone using ICE TCP at all? Given that we already 
support TURN over TCP/TLS, is it useful for anyone? As I see it it is 
only useful for servers, but locating a TURN server near it would do the 
same trick.

Best regards
Sergio

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    <div class="moz-cite-prefix">El 11/06/2014 13:25, Makaraju, Maridi
      Raju (Raju) escribi³:<br>
    </div>
    <blockquote
cite="mid:E1FE4C082A89A246A11D7F32A95A17828E407D35@US70UWXCHMBA02.zam.alcatel-lucent.com"
      type="cite">
      <blockquote type="cite">
        <pre wrap="">Raju, for example, in case of client or MCU mobility, we don't need to exchange media &gt;info again, as it will be exactly the same. In our case we only want to propagate new &gt;ICE candidates to recheck with ICE if there is a different valid pair. Â But I think &gt;the solutions commented in the thread are still compatible with your idea. As long as &gt;the first offer/answer messages are still valid you could send them again with the &gt;new candidates to the middle boxes if you need it.
</pre>
      </blockquote>
      <pre wrap="">
[Raju] 
Hi Javier, Thanks for explaining your MCU use cases.
I wanted to assert again that, I am not advocating for an automatic new SDP O/A. We need to separate updating local SDP from generating new SDP offer. The former has to be done by browser and the latter is up to the application and falls into signaling (which webrtc has left to applications).
At a very high-level, this is how the flow looks like:
1. When ICE is completed, browser will update local SDP 
   c/m= lines with selected candidate for:
   a. Local IP, port
   b. Matching transport (or to indicate the transport 
      selected. TCP or UDP). 
2. Browser calls an API (TBD) like onicecandidate(NULL) 
   to indicate end of ICE.
</pre>
    </blockquote>
    <br>
    Isn't (2) already defined in current api with <code style="color:
      rgb(0, 0, 0); font-weight: bold; font-family: monospace;
      font-style: normal; font-variant: normal; letter-spacing: normal;
      line-height: normal; orphans: auto; text-align: start;
      text-indent: 0px; text-transform: none; white-space: normal;
      widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;
      background: rgb(255, 255, 210);">oniceconnectionstatechange</code><span
      style="color: rgb(0, 0, 0); font-family: sans-serif; font-size:
      medium; font-style: normal; font-variant: normal; font-weight:
      normal; letter-spacing: normal; line-height: normal; orphans:
      auto; text-align: start; text-indent: 0px; text-transform: none;
      white-space: normal; widows: auto; word-spacing: 0px;
      -webkit-text-stroke-width: 0px; display: inline !important; float:
      none; background-color: rgb(255, 255, 255);"><span
        class="Apple-converted-space"> with state "connected" ? Maybe it
        would be interesting to also add the selected candidate in the
        event as I see it useful for applications.<br>
        <br>
        As a side note, is anyone using ICE TCP at all? Given that we
        already support TURN over TCP/TLS, is it useful for anyone? As I
        see it it is only useful for servers, but locating a TURN server
        near it would do the same trick.<br>
        <br>
        Best regards<br>
        Sergio<br>
      </span></span>
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>, =?utf-8?B?SmF2aWVyIENlcnZpw7Fv?= <jcague@gmail.com>
Thread-Topic: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Date: Wed, 11 Jun 2014 12:08:18 +0000
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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From nobody Wed Jun 11 05:18:05 2014
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To: Eduardo Gueiros <egueiros@jive.com>, rtcweb@ietf.org
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Subject: Re: [rtcweb] review [draft-ietf-rtcweb-transports]
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On 06/05/2014 02:01 AM, Eduardo Gueiros wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA512
>
> Some thoughts/suggestions on [draft-ietf-rtcweb-transports-04] that
> perhaps could help making it more clear.

Thanks for the feedback!

>
> - --
> 1. Paragraph 1, 2, and 3.
>
> Introduction seems to give fragmented information about this document
> [draft-ietf-rtcweb-transports] and the general RTCWEB effort. At times
> I donât
> know to which the introduction is referring to. Consider giving that
> information in distinct ways. Suggestion:
>
> - - The IETF RTCWEB effort is aimed at specifying a protocol suite that
> is useful for real time multimedia exchange between clients. This
> protocol suite is designed for WebRTC and intents to satisfy the
> security considerations described in the WebRTC security documents
> [document link].
> - - The IEFT RTCWEB effort is part of the WebRTC cooperation between the
> IETF and W3C. The overall effort is described in WebRTC Overview
> document, [document link].

I'm not sure that helps. But we should make it shorter.

I'm removing the whole effort description, because it's irrelevant to 
this document - especially when the RFC is published, and the effort is 
concluded.

>
> 1. Paragraph 2, 2nd sentence
>
> âThis document focuses on the dataâĤâ - To me itâs not clear which
> document youâre referring to; the one it was just cited, or this
> document [draft-ietf-rtcweb-transports].

I think this is clear enough - if it had pointed to the other document, 
it would have said "that document".

>
>
> 3.2. 2nd paragraph
> âWhen TURN is used, and the TURN server has IPv4 or IPv6 connectivity
>     to the peer or its TURN server, candidates of the appropriate types
>     MUST be supported.â
>
> This could be simplified to âWhen TURN is used, if the server or the
> peer has IPv4 or IPv6 connectivity, candidates of the appropriate
> types MUST be supported.â

I don't think the proposed sentence is correct, because it doesn't say 
what the server or the peer has to have connectivity to.

The following cases exist (at least) where IPv6 candidates are required:

Client  --IPv4-> TURN server --IPv6-> Peer
Client --IPv6-> TURN server --IPv6-> Peer

Client --IPv4-> Client TURN server --IPv6-> Peer TURN server -IPv4 -> Peer

However, in the following scenario, IPv6 candidates are NOT required:

Client --IPv6-> TURN server --IPv4->Peer

Because even though the TURN server has wonderful IPv6 connectivity, it 
doesn't have IPv6 connectivity to the peer or the peer's TURN server.

>
> 3.4, 1st Paragraph
>
> Avoid contraction in formal writing, change âitâsâ to âit isâ

45 "it's" in the last 250 RFCs, and 2227 "it is". Non-contractions win.
(There doesn't seem to be any such written rule)

>
> 3.4, 2nd Paragraph
>
> âthis allows interworking with bothâĤâ A clearer sentence would read:
> âa full implementation allows interworking with bothâĤâ

Done.
>
> 3.4 3rd Paragraph
>
> missing comma after âNATsâ
It's not supposed to be there. The sentence parses as

"behind (NATs which perform endpoint-dependent mapping), TURN MUST be 
supported"

There are lots of NATs that don't perform endpoint-dependent mapping.
Changed to "NATs of the type that perform endpoint-dependent mapping".

>
> 3.4 5th Paragraph
>
> missing commas ââĤTURN [comma] using TCP between the client and server
> [comma] MUST be supported.â

Again, this would change the meaning to a claim that TURN always uses 
TCP, which is false.
Rephrased as "the mode of TURN that uses TCP .... MUST be supported".
>
> The sentence could also be rewritten, for clarification, as: âTURN
> MUST be supported when using TCP between the client and the server in
> order to deal with firewalls that block all UDP traffic. TURN MUST
> also be supported when using TLS over TCP between the client and the
> server.â
>
> 3.4, 6th and 7th Paragraphs should be combined.
No, they should not. Paragraph 6 deals with IPv6 candidates, paragraph 7 
deals with TCP candidates.

Perhaps you mean paragraph 7 and 8? That makes more sense, but I still 
want those sentences separated by whitespace, so I'll keep the distinction.
>
> 3.4 7th Paragraph
>
> - - If the intention of the paragraph to show why TURN TCP candidates
> do not provide significant benefit it needs to indicate so: ââĤnot seen
> as providing significant benefit since/because:â
> - - suggest change âSecondlyâ to âSecondâ
> - - suggest change âthat use case is anyway supportedâ to âthis use case
> is not supportedâ
But as long as TURN over TCP is supported, the use case is supported, 
that's the point. Changed "anyway supported" to "supported in a 
different way".
> - - suggest change âThirdlyâ to âThirdâ
Done.

>
> 4. 1st Paragraph
>
> It seems to me the word âthatâ in the sentence ââĤprioritization model
> is that the application tellsâĤâ should be changed to âwhatâ.
I tried "The RTCWEB prioritization model is what the application tells 
the RTCWEB implementation", and couldn't find a meaning for it, so I 
don't think that's right.

The model is that the application tells the implementation something 
about what it wants.

What it tells the implementation ... I don't have a name for.
>
> 4.1 3rd Paragraph
>
> It seems to me this sentence, â (same as above + DSCP code point)â,
> could be clearer  as  â(5-tuple plus DSCP code point)â
Makes sense. Done.
>
> - --
> Eduardo Gueiros
> Developer | Jive Communications, Inc.
> Jive.com | egueiros at jive.com
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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El 11/06/2014 14:08, Makaraju, Maridi Raju (Raju) escribi³:
>
>> Isn't (2) already defined in current api with oniceconnectionstatechange with state >"connected" ? Maybe it would be interesting to also add the selected candidate in the >event as I see it useful for applications.
> [Raju] (2) is done at the time of ICE gathering, which happens before initial offer is sent out. But, we need a trigger when ICE is completed, which happens after initial O/A is completed.

I think that's not correct, from the editor's draft:

|oniceconnectionstatechange|of typeEventHandler, This event handler, of 
event handler event type|iceconnectionstatechange 
<http://www.w3.org/TR/webrtc/#event-iceconnectionstatechange>|,/must/be 
fired by all objects implementing the||RTCPeerConnection| 
<http://www.w3.org/TR/webrtc/#idl-def-RTCPeerConnection>|interface. It 
is called any time theiceConnectionStatechanges.

enum RTCIceConnectionState {
     "new",
     "checking",
     "connected",
     "completed",
     "failed",
     "disconnected",
     "closed"
};

That has nothing to do with

enum RTCIceGatheringState {
     "new",
     "gathering",
     "complete"
};

BTW,  as discussed on other email thread, we should add the 
onicegatheringstatechange and remove the onicecandidate(null)

>
> Trickle ICE (draft-ietf-mmusic-trickle-ice-01) is another powerful use case of leaving SDP O/A to application, but provide APIs (onicecandidate, addicecandidate) for them to achieve their needs. Trickle ICE is possible because browser gives ICE candidates to app as they are discovered. Most apps just need to know only when gathering is completed. But some delay conscious apps can make use of these candidates to perform trickle ICE.
> IMO, asking for end of ICE trigger and updating c/m= lines falls in to the same category that is already allowed by the APIs.
>
>> As a side note, is anyone using ICE TCP at all? Given that we already support TURN >over TCP/TLS, is it useful for anyone? As I see it it is only useful for servers, but >locating a TURN server near it would do the same trick.
> [Raju]
> TURN involves an additional element in the network and bearer path; as a result to reduce TURN costs and bearer delay, using ICE-TCP is advised.
> draft-ietf-rtcweb-transports-04 says:
>     "ICE-TCP candidates [RFC6544] MUST be supported; this may allow
>     applications to communicate to peers with public IP addresses across
>     UDP-blocking firewalls without using a TURN server."
>
You haven't tried to use it, right?

https://code.google.com/p/webrtc/issues/detail?id=2320

TURN is bulletproof, but ICE TCP is so seldom used that it is going to 
be very low tested/cared about on browsers implementations, I am afraid.

Best regards
Sergio


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    <div class="moz-cite-prefix">El 11/06/2014 14:08, Makaraju, Maridi
      Raju (Raju) escribi³:<br>
    </div>
    <blockquote
cite="mid:E1FE4C082A89A246A11D7F32A95A17828E407F64@US70UWXCHMBA02.zam.alcatel-lucent.com"
      type="cite"><br>
      <blockquote type="cite">
        <pre wrap="">Isn't (2) already defined in current api with oniceconnectionstatechange with state &gt;"connected" ? Maybe it would be interesting to also add the selected candidate in the &gt;event as I see it useful for applications.
</pre>
      </blockquote>
      <pre wrap="">
[Raju] (2) is done at the time of ICE gathering, which happens before initial offer is sent out. But, we need a trigger when ICE is completed, which happens after initial O/A is completed.</pre>
    </blockquote>
    <br>
    I think that's not correct, from the editor's draft:<br>
    <br>
    <dt id="widl-RTCPeerConnection-oniceconnectionstatechange"
      style="margin-top: 0px; margin-bottom: 0px; font-weight: normal;
      color: rgb(0, 0, 0); font-family: sans-serif; font-size: medium;
      font-style: normal; font-variant: normal; letter-spacing: normal;
      line-height: normal; orphans: auto; text-align: start;
      text-indent: 0px; text-transform: none; white-space: normal;
      widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;
      background-color: rgb(255, 255, 255);"><code style="color: rgb(0,
        0, 0); font-weight: bold; font-family: monospace; background:
        rgb(255, 255, 210);">oniceconnectionstatechange</code><span
        class="Apple-converted-space">Â </span>of type<span
        class="Apple-converted-space">Â </span><span class="idlAttrType"
        style="color: rgb(0, 90, 156);"><a>EventHandler</a></span>, This
      event handler, of event handler event type<span
        class="Apple-converted-space">Â </span><code style="color:
        rgb(255, 69, 0);"><a
          href="http://www.w3.org/TR/webrtc/#event-iceconnectionstatechange"
          style="color: rgb(102, 0, 153); background: transparent;">iceconnectionstatechange</a></code>,<span
        class="Apple-converted-space">Â </span><em title="MUST"
        class="rfc2119" style="text-transform: lowercase; font-variant:
        small-caps; font-style: normal; color: rgb(153, 0, 0);">must</em><span
        class="Apple-converted-space">Â </span>be fired by all objects
      implementing the<span class="Apple-converted-space">Â </span><code
        style="color: rgb(255, 69, 0);"><a class="idlType"
          href="http://www.w3.org/TR/webrtc/#idl-def-RTCPeerConnection"
          style="color: rgb(102, 0, 153); font-weight: bold;
          text-decoration: none; background: transparent;"><code
            style="color: rgb(255, 69, 0);">RTCPeerConnection</code></a></code><span
        class="Apple-converted-space">Â </span>interface. It is called
      any time the<span class="Apple-converted-space">Â </span><var>iceConnectionState</var><span
        class="Apple-converted-space">Â </span>changes.</dt>
    <br>
    enum RTCIceConnectionState {<br>
    Â Â Â  "new",<br>
    Â Â Â  "checking",<br>
    Â Â Â  "connected",<br>
    Â Â Â  "completed",<br>
    Â Â Â  "failed",<br>
    Â Â Â  "disconnected",<br>
    Â Â Â  "closed"<br>
    };<br>
    <br>
    That has nothing to do with <br>
    <br>
    enum RTCIceGatheringState {<br>
    Â Â Â  "new",<br>
    Â Â Â  "gathering",<br>
    Â Â Â  "complete"<br>
    };<br>
    <br>
    BTW,Â  as discussed on other email thread, we should add the
    onicegatheringstatechange and remove the onicecandidate(null)<br>
    <br>
    <blockquote
cite="mid:E1FE4C082A89A246A11D7F32A95A17828E407F64@US70UWXCHMBA02.zam.alcatel-lucent.com"
      type="cite">
      <pre wrap="">

Trickle ICE (draft-ietf-mmusic-trickle-ice-01) is another powerful use case of leaving SDP O/A to application, but provide APIs (onicecandidate, addicecandidate) for them to achieve their needs. Trickle ICE is possible because browser gives ICE candidates to app as they are discovered. Most apps just need to know only when gathering is completed. But some delay conscious apps can make use of these candidates to perform trickle ICE. 
IMO, asking for end of ICE trigger and updating c/m= lines falls in to the same category that is already allowed by the APIs.

</pre>
      <blockquote type="cite">
        <pre wrap="">As a side note, is anyone using ICE TCP at all? Given that we already support TURN &gt;over TCP/TLS, is it useful for anyone? As I see it it is only useful for servers, but &gt;locating a TURN server near it would do the same trick.
</pre>
      </blockquote>
      <pre wrap="">
[Raju]
TURN involves an additional element in the network and bearer path; as a result to reduce TURN costs and bearer delay, using ICE-TCP is advised.
draft-ietf-rtcweb-transports-04 says:
   "ICE-TCP candidates [RFC6544] MUST be supported; this may allow
   applications to communicate to peers with public IP addresses across
   UDP-blocking firewalls without using a TURN server."

</pre>
    </blockquote>
    You haven't tried to use it, right?<br>
    <br>
    <a class="moz-txt-link-freetext" href="https://code.google.com/p/webrtc/issues/detail?id=2320">https://code.google.com/p/webrtc/issues/detail?id=2320</a><br>
    <br>
    TURN is bulletproof, but ICE TCP is so seldom used that it is going
    to be very low tested/cared about on browsers implementations, I am
    afraid.<br>
    <br>
    Best regards<br>
    Sergio<br>
    <br>
  </body>
</html>

--------------080002080605080206040002--


From nobody Wed Jun 11 07:18:53 2014
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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-transports-05.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           : Transports for RTCWEB
        Author          : Harald Alvestrand
	Filename        : draft-ietf-rtcweb-transports-05.txt
	Pages           : 13
	Date            : 2014-06-11

Abstract:
   This document describes the data transport protocols used by RTCWEB,
   including the protocols used for interaction with intermediate boxes
   such as firewalls, relays and NAT boxes.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-transports-05

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-transports-05


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 Wed Jun 11 07:29:07 2014
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On 10 Jun 2014, at 21:22, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
> =20
> A comment on draft-ietf-rtcweb-data-protocol-06:
> =20
> Section 5.1 says:
> =20
> "Protocol: Variable Length (sequence of characters)
>       The sub-protocol for the channel as a UTF-8 encoded string.  If
>       this is an empty string the protocol is unspecified.  If it is a
>       non-empty string, it specifies an protocol registered in the
>       'WebSocket Subprotocol Name Registry' created in [RFC6455]."
> =20
> =20
> I think "The sub-protocol for the channel as a UTF-8 encoded string." =
part is a little confusing, because there is no such thing as =
sub-protocol in DCEP itself (DCEP only uses the WebSocket sub-protocol =
registry for registering DCEP *protocol* values).
> =20
> Perhaps we could just remove the sentence, and keep:
> =20
> "Protocol: Variable Length (sequence of characters)
>       If this is an empty string the protocol is unspecified.  If it =
is a
>       non-empty string, it specifies an protocol registered in the
>       'WebSocket Subprotocol Name Registry' created in [RFC6455]."
> =20
> Section 4 gives more information about the usage of the protocol.
Do you intend to drop the statement about the UTF-8 encoding?
I think we should keep a statement about the encoding.

Best regards
Michael
> =20
> (Alternatively, we could use sub-protocol terminology also in DCEP)
> =20
> Regards,
> =20
> Christer
> =20
> =20
> =20
> =20
> =20
> =20
> From: rtcweb [rtcweb-bounces@ietf.org] on behalf of Ted Hardie =
[ted.ietf@gmail.com]
> Sent: Tuesday, 10 June, 2014 9:52 PM
> To: rtcweb@ietf.org
> Subject: [rtcweb] WGLC for data channel drafts
>=20
> Dear WG,
>=20
> This starts a working group last call for our core Data Channel =
drafts:
>=20
> draft-ietf-rtcweb-data-protocol-06.txt
> draft-ietf-rtcweb-data-channel-10.txt
>=20
> Please review them in detail, especially for areas which may be of =
concern to the broader IETF community.  This WGLC will end on June 27, =
2014.
>=20
> Ted, Sean, Cullen
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>, =?utf-8?B?SmF2aWVyIENlcnZpw7Fv?= <jcague@gmail.com>
Thread-Topic: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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El 11/06/2014 16:52, Makaraju, Maridi Raju (Raju) escribi³:
>> You haven't tried to use it, right?
>> https://code.google.com/p/webrtc/issues/detail?id=2320
>> TURN is bulletproof, but ICE TCP is so seldom used that it is going to be very low >tested/cared about on browsers implementations, I am afraid.
> [Raju] I don't think it is really about if the implementation is stable or not as implementations do get stable over time. But the real question is what is webrtc implementations required to do? The spec clearly says "MUST".
>
>

My point was if that "MUST" was really needed at all.

Best regards
Sergio


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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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The MUST was a consensus call in London. See the minutes for numbers. I was one of the ones who were surprised at the strength of the consensus.

On 11. juni 2014 18:11:47 CEST, Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com> wrote:
>El 11/06/2014 16:52, Makaraju, Maridi Raju (Raju) escribi³:
>>> You haven't tried to use it, right?
>>> https://code.google.com/p/webrtc/issues/detail?id=2320
>>> TURN is bulletproof, but ICE TCP is so seldom used that it is going
>to be very low >tested/cared about on browsers implementations, I am
>afraid.
>> [Raju] I don't think it is really about if the implementation is
>stable or not as implementations do get stable over time. But the real
>question is what is webrtc implementations required to do? The spec
>clearly says "MUST".
>>
>>
>
>My point was if that "MUST" was really needed at all.
>
>Best regards
>Sergio
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

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<html><head></head><body>The MUST was a consensus call in London. See the minutes for numbers. I was one of the ones who were surprised at the strength of the consensus.<br><br><div class="gmail_quote">On 11. juni 2014 18:11:47 CEST, Sergio Garcia Murillo &lt;sergio.garcia.murillo@gmail.com&gt; wrote:<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<pre class="k9mail">El 11/06/2014 16:52, Makaraju, Maridi Raju (Raju) escribi³:<br /><blockquote class="gmail_quote" style="margin: 0pt 0pt 1ex 0.8ex; border-left: 1px solid #729fcf; padding-left: 1ex;"><blockquote class="gmail_quote" style="margin: 0pt 0pt 1ex 0.8ex; border-left: 1px solid #ad7fa8; padding-left: 1ex;"> You haven't tried to use it, right?<br /> <a href="https://code.google.com/p/webrtc/issues/detail?id=2320">https://code.google.com/p/webrtc/issues/detail?id=2320</a><br /> TURN is bulletproof, but ICE TCP is so seldom used that it is going to be very low &gt;tested/cared about on browsers implementations, I am afraid.<br /></blockquote> [Raju] I don't think it is really about if the implementation is stable or not as implementations do get stable over time. But the real question is what is webrtc implementations required to do? The spec clearly says "MUST".</blockquote><br /><br /><br />My point was if that "MUST" was really needed at all.<br /><br />Best regards<br
/>Sergio<br /><br /><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>
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From nobody Wed Jun 11 10:25:02 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
Thread-Topic: [rtcweb] WGLC for data channel drafts - sub-protocol usage
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Subject: Re: [rtcweb] WGLC for data channel drafts - sub-protocol usage
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Hi Michael,

I agree that we need to keep text about the encoding.

Regards,

Christer

________________________________________
From: Michael Tuexen [Michael.Tuexen@lurchi.franken.de]
Sent: Wednesday, 11 June 2014 5:28 PM
To: Christer Holmberg
Cc: Ted Hardie; rtcweb@ietf.org
Subject: Re: [rtcweb] WGLC for data channel drafts - sub-protocol usage

On 10 Jun 2014, at 21:22, Christer Holmberg <christer.holmberg@ericsson.com=
> wrote:

> Hi,
>
> A comment on draft-ietf-rtcweb-data-protocol-06:
>
> Section 5.1 says:
>
> "Protocol: Variable Length (sequence of characters)
>       The sub-protocol for the channel as a UTF-8 encoded string.  If
>       this is an empty string the protocol is unspecified.  If it is a
>       non-empty string, it specifies an protocol registered in the
>       'WebSocket Subprotocol Name Registry' created in [RFC6455]."
>
>
> I think "The sub-protocol for the channel as a UTF-8 encoded string." par=
t is a little confusing, because there is no such thing as sub-protocol in =
DCEP itself (DCEP only uses the WebSocket sub-protocol registry for registe=
ring DCEP *protocol* values).
>
> Perhaps we could just remove the sentence, and keep:
>
> "Protocol: Variable Length (sequence of characters)
>       If this is an empty string the protocol is unspecified.  If it is a
>       non-empty string, it specifies an protocol registered in the
>       'WebSocket Subprotocol Name Registry' created in [RFC6455]."
>
> Section 4 gives more information about the usage of the protocol.
Do you intend to drop the statement about the UTF-8 encoding?
I think we should keep a statement about the encoding.

Best regards
Michael
>
> (Alternatively, we could use sub-protocol terminology also in DCEP)
>
> Regards,
>
> Christer
>
>
>
>
>
>
> From: rtcweb [rtcweb-bounces@ietf.org] on behalf of Ted Hardie [ted.ietf@=
gmail.com]
> Sent: Tuesday, 10 June, 2014 9:52 PM
> To: rtcweb@ietf.org
> Subject: [rtcweb] WGLC for data channel drafts
>
> Dear WG,
>
> This starts a working group last call for our core Data Channel drafts:
>
> draft-ietf-rtcweb-data-protocol-06.txt
> draft-ietf-rtcweb-data-channel-10.txt
>
> Please review them in detail, especially for areas which may be of concer=
n to the broader IETF community.  This WGLC will end on June 27, 2014.
>
> Ted, Sean, Cullen
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb=


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I am late then.. :)

El 11/06/2014 18:57, Harald Alvestrand escribi³:
> The MUST was a consensus call in London. See the minutes for numbers. 
> I was one of the ones who were surprised at the strength of the consensus.
>
> On 11. juni 2014 18:11:47 CEST, Sergio Garcia Murillo 
> <sergio.garcia.murillo@gmail.com> wrote:
>
>     El 11/06/2014 16:52, Makaraju, Maridi Raju (Raju) escribi³:
>
>             You haven't tried to use it, right?
>             https://code.google.com/p/webrtc/issues/detail?id=2320
>             TURN is bulletproof, but ICE TCP is so seldom used that it
>             is going to be very low >tested/cared about on browsers
>             implementations, I am afraid. 
>
>         [Raju] I don't think it is really about if the implementation
>         is stable or not as implementations do get stable over time.
>         But the real question is what is webrtc implementations
>         required to do? The spec clearly says "MUST".
>
>
>
>
>     My point was if that "MUST" was really needed at all.
>
>     Best regards
>     Sergio
>
>     ------------------------------------------------------------------------
>
>     rtcweb mailing list
>     rtcweb@ietf.org
>     https://www.ietf.org/mailman/listinfo/rtcweb
>
>


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    <div class="moz-cite-prefix">I am late then.. :)<br>
      <br>
      El 11/06/2014 18:57, Harald Alvestrand escribi³:<br>
    </div>
    <blockquote
      cite="mid:6025bc75-db01-45d4-ac4a-50a39740f15a@email.android.com"
      type="cite">The MUST was a consensus call in London. See the
      minutes for numbers. I was one of the ones who were surprised at
      the strength of the consensus.<br>
      <br>
      <div class="gmail_quote">On 11. juni 2014 18:11:47 CEST, Sergio
        Garcia Murillo <a class="moz-txt-link-rfc2396E" href="mailto:sergio.garcia.murillo@gmail.com">&lt;sergio.garcia.murillo@gmail.com&gt;</a> wrote:
        <blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
          0.8ex; border-left: 1px solid rgb(204, 204, 204);
          padding-left: 1ex;">
          <pre class="k9mail">El 11/06/2014 16:52, Makaraju, Maridi Raju (Raju) escribi³:
<blockquote class="gmail_quote" style="margin: 0pt 0pt 1ex 0.8ex; border-left: 1px solid #729fcf; padding-left: 1ex;"><blockquote class="gmail_quote" style="margin: 0pt 0pt 1ex 0.8ex; border-left: 1px solid #ad7fa8; padding-left: 1ex;"> You haven't tried to use it, right?
 <a moz-do-not-send="true" href="https://code.google.com/p/webrtc/issues/detail?id=2320">https://code.google.com/p/webrtc/issues/detail?id=2320</a>
 TURN is bulletproof, but ICE TCP is so seldom used that it is going to be very low &gt;tested/cared about on browsers implementations, I am afraid.
</blockquote> [Raju] I don't think it is really about if the implementation is stable or not as implementations do get stable over time. But the real question is what is webrtc implementations required to do? The spec clearly says "MUST".</blockquote>


My point was if that "MUST" was really needed at all.

Best regards
Sergio

<hr>
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a moz-do-not-send="true" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
        </blockquote>
      </div>
      <br>
    </blockquote>
    <br>
  </body>
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--------------000309030604020703030608--


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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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On 06/11/2014 08:38 PM, Makaraju, Maridi Raju (Raju) wrote:
>> The MUST was a consensus call in London. See the minutes for numbers. I was one of >the ones who were surprised at the strength of the consensus.
> [Raju] I thought one of the implicit (may be explicit in draft-ietf-rtcweb-use-cases-and-requirements-14) requirement for WebRTC is "implementations should work right out of the box without extra boxes (e.g. TURN) in between under UDP restrictive firewalls".
Unfortunately, nobody's found a way to make that true even for all 
common NATs/firewalls that want to permit the communication.

TURN servers are the least obnoxious way to make sure apps can work in 
the presence of symmetric NATs.
We could wish that this wasn't so, but we could also wish that NATs 
would go away and everyone would run IPv6; neither is likely to happen 
this year.

Of course, in the presence of firewalls where the admin does *not* want 
to permit the communication, we neither can nor should make 
communication work.


> Then I see the following requirement in draft-ietf-rtcweb-use-cases-and-requirements-14. It did not mention use of TURN in the text.
>
>     F18     The browser must be able to send streams and
>             data to a peer in the presence of NATs and
>             Firewalls that block UDP traffic.

Yes. Using TURN servers is the way in which it is possible to fulfil 
this requirement.

>
> BR
> Raju
>


From nobody Thu Jun 12 02:58:25 2014
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To: "'Harald Alvestrand'" <harald@alvestrand.no>, "'Makaraju, Maridi Raju \(Raju\)'" <Raju.Makaraju@alcatel-lucent.com>, "'Sergio Garcia Murillo'" <sergio.garcia.murillo@gmail.com>, =?iso-8859-1?Q?'Javier_Cervi=F1o'?= <jcague@gmail.com>
References: <CAOJ7v-2FTAKGd2ZUNt9PZBW9pFu7c9v7Gx8Z8vFOQo4K3dyr5Q@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D35B4E7@ESESSMB209.ericsson.se> <CALiegfncVjR-cQV=coLdmO6OODgbuP-pZ2fxopcWfgGZm+jHyQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D35B859@ESESSMB209.ericsson.se> <CALiegfn7WuyA8qbKWGs4JrNutqF9teXiEN8eqd0UJk5vRrX4TQ@mail.gmail.com> <539786C7.8090806@gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E406A88@US70UWXCHMBA02.zam.alcatel-lucent.com> <CAG=SL7mGHuP0KBuvPGQ8bG0+CxHB7WjemCxZ5WxCXB8XgpOhCQ@mail.gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E407D35@US70UWXCHMBA02.zam.alcatel-lucent.com> <53984062.1090202@gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E407F64@US70UWXCHMBA02.zam.alcatel-lucent.com> <53986129.4050209@gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E40858D@US70UWXCHMBA02.zam.alcatel-lucent.com> <53987FC3.7060408@gmail.com> <6025bc75-db01-45d4-ac4a-50a39740f15a@email.android.com> <E1FE4C082A89A246A11D7F32A95A17828E408FC8@US70UWXCHMBA02.zam.alcatel-lucent.com> <539929C1.3070 205@alvestrand.no>
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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>>     F18     The browser must be able to send streams and
>>             data to a peer in the presence of NATs and
>>             Firewalls that block UDP traffic.

>Yes. Using TURN servers is the way in which it is possible to fulfil =
this
requirement.

Further in draft-ietf-rtcweb-use-cases-and-requirements-14:=20
"the straddling TURN server ... 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."
F20     The browser must support the use of STUN and TURN
           servers that are supplied by entities other than
           the web application (i.e. the network provider).

This is a TURN server with one interface on the Enterprise LAN where the
browser=20
simple shall place the WebRTC media. (The other interface should be =
public.)

The way for the browser to autodiscover such network provided (from the
enterprise and ISP) TURN server is being worked in
draft-patil-tram-turn-serv-disc-01.txt.

This resolves any firewall restriction not allowing WebRTC media (by
paralleling the firewall with the TURN server), and also allows the
enterprise or ISP to point out a better path for the WebRTC media =
(instead
of through the often data crowded firewall default gateway).
(This will also be needed with IPv6.)

/Karl

-----Ursprungligt meddelande-----
Fr=E5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=F6r Harald Alvestrand
Skickat: den 12 juni 2014 06:17
Till: Makaraju, Maridi Raju (Raju); Sergio Garcia Murillo; Javier =
Cervi=F1o
Kopia: rtcweb@ietf.org
=C4mne: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or =
not to
UDP?

On 06/11/2014 08:38 PM, Makaraju, Maridi Raju (Raju) wrote:
>> The MUST was a consensus call in London. See the minutes for numbers. =
I
was one of >the ones who were surprised at the strength of the =
consensus.
> [Raju] I thought one of the implicit (may be explicit in
draft-ietf-rtcweb-use-cases-and-requirements-14) requirement for WebRTC =
is
"implementations should work right out of the box without extra boxes =
(e.g.
TURN) in between under UDP restrictive firewalls".
Unfortunately, nobody's found a way to make that true even for all =
common
NATs/firewalls that want to permit the communication.

TURN servers are the least obnoxious way to make sure apps can work in =
the
presence of symmetric NATs.
We could wish that this wasn't so, but we could also wish that NATs =
would go
away and everyone would run IPv6; neither is likely to happen this year.

Of course, in the presence of firewalls where the admin does *not* want =
to
permit the communication, we neither can nor should make communication =
work.


> Then I see the following requirement in
draft-ietf-rtcweb-use-cases-and-requirements-14. It did not mention use =
of
TURN in the text.
>
>     F18     The browser must be able to send streams and
>             data to a peer in the presence of NATs and
>             Firewalls that block UDP traffic.

Yes. Using TURN servers is the way in which it is possible to fulfil =
this
requirement.

>
> BR
> Raju
>

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


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From: "Pal Martinsen (palmarti)" <palmarti@cisco.com>
To: Karl Stahl <karl.stahl@intertex.se>
Thread-Topic: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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On 12 Jun 2014, at 11:58, Karl Stahl <karl.stahl@intertex.se> wrote:

>>>    F18     The browser must be able to send streams and
>>>            data to a peer in the presence of NATs and
>>>            Firewalls that block UDP traffic.
>=20
>> Yes. Using TURN servers is the way in which it is possible to fulfil thi=
s
> requirement.
>=20
> Further in draft-ietf-rtcweb-use-cases-and-requirements-14:=20
> "the straddling TURN server ... 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."
> F20     The browser must support the use of STUN and TURN
>           servers that are supplied by entities other than
>           the web application (i.e. the network provider).
>=20
> This is a TURN server with one interface on the Enterprise LAN where the
> browser=20
> simple shall place the WebRTC media. (The other interface should be publi=
c.)
>=20
> The way for the browser to autodiscover such network provided (from the
> enterprise and ISP) TURN server is being worked in
> draft-patil-tram-turn-serv-disc-01.txt.
>=20
> This resolves any firewall restriction not allowing WebRTC media (by
> paralleling the firewall with the TURN server), and also allows the
> enterprise or ISP to point out a better path for the WebRTC media (instea=
d
> of through the often data crowded firewall default gateway).
> (This will also be needed with IPv6.)
>=20

I have problems understanding how this solves any issues. I doubt many ente=
rprise network managers would install a TURN server in parallel with their =
FW. Would be fair easier and safer to open up the appropriate ports in the =
firewall towards the TURN server.  They could then allow only UDP to allow =
for DPI and see that it is actually RTP traffic flowing.=20

Having a TURN server supporting UDP/TCP/TLS as communication between the cl=
ient and TURN server would solve almost all connectivity problems. We shoul=
d, as Harald also pointed out, not try to force us throughout the FW if the=
 admin do not want the traffic to go through.=20

.-.
P=E5l-Erik










> /Karl
>=20
> -----Ursprungligt meddelande-----
> Fr=E5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=F6r Harald Alvestrand
> Skickat: den 12 juni 2014 06:17
> Till: Makaraju, Maridi Raju (Raju); Sergio Garcia Murillo; Javier Cervi=
=F1o
> Kopia: rtcweb@ietf.org
> =C4mne: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or n=
ot to
> UDP?
>=20
> On 06/11/2014 08:38 PM, Makaraju, Maridi Raju (Raju) wrote:
>>> The MUST was a consensus call in London. See the minutes for numbers. I
> was one of >the ones who were surprised at the strength of the consensus.
>> [Raju] I thought one of the implicit (may be explicit in
> draft-ietf-rtcweb-use-cases-and-requirements-14) requirement for WebRTC i=
s
> "implementations should work right out of the box without extra boxes (e.=
g.
> TURN) in between under UDP restrictive firewalls".
> Unfortunately, nobody's found a way to make that true even for all common
> NATs/firewalls that want to permit the communication.
>=20
> TURN servers are the least obnoxious way to make sure apps can work in th=
e
> presence of symmetric NATs.
> We could wish that this wasn't so, but we could also wish that NATs would=
 go
> away and everyone would run IPv6; neither is likely to happen this year.
>=20
> Of course, in the presence of firewalls where the admin does *not* want t=
o
> permit the communication, we neither can nor should make communication wo=
rk.
>=20
>=20
>> Then I see the following requirement in
> draft-ietf-rtcweb-use-cases-and-requirements-14. It did not mention use o=
f
> TURN in the text.
>>=20
>>    F18     The browser must be able to send streams and
>>            data to a peer in the presence of NATs and
>>            Firewalls that block UDP traffic.
>=20
> Yes. Using TURN servers is the way in which it is possible to fulfil this
> requirement.
>=20
>>=20
>> BR
>> Raju
>>=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


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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Harald Alvestrand <harald@alvestrand.no>, Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>, =?utf-8?B?SmF2aWVyIENlcnZpw7Fv?= <jcague@gmail.com>
Thread-Topic: [rtcweb] Need for TURN, TCP-ICE (was Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?)
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Date: Thu, 12 Jun 2014 12:12:09 +0000
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References: <CAOJ7v-2FTAKGd2ZUNt9PZBW9pFu7c9v7Gx8Z8vFOQo4K3dyr5Q@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D35B4E7@ESESSMB209.ericsson.se> <CALiegfncVjR-cQV=coLdmO6OODgbuP-pZ2fxopcWfgGZm+jHyQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D35B859@ESESSMB209.ericsson.se> <CALiegfn7WuyA8qbKWGs4JrNutqF9teXiEN8eqd0UJk5vRrX4TQ@mail.gmail.com> <539786C7.8090806@gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E406A88@US70UWXCHMBA02.zam.alcatel-lucent.com> <CAG=SL7mGHuP0KBuvPGQ8bG0+CxHB7WjemCxZ5WxCXB8XgpOhCQ@mail.gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E407D35@US70UWXCHMBA02.zam.alcatel-lucent.com> <53984062.1090202@gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E407F64@US70UWXCHMBA02.zam.alcatel-lucent.com> <53986129.4050209@gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E40858D@US70UWXCHMBA02.zam.alcatel-lucent.com> <53987FC3.7060408@gmail.com> <6025bc75-db01-45d4-ac4a-50a39740f15a@email.android.com> <E1FE4C082A89A246A11D7F32A95A17828E408FC8@US70UWXCHMBA02.zam.alcatel-lucent.com> <057DC1BB-CEDA-4E28-A9EC-41C3FE5673F3@cisco.com>
Date: Thu, 12 Jun 2014 05:37:39 -0700
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From: Ca By <cb.list6@gmail.com>
To: "Pal Martinsen (palmarti)" <palmarti@cisco.com>
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Cc: =?UTF-8?Q?Javier_Cervi=C3=B1o?= <jcague@gmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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On Jun 12, 2014 4:31 AM, "Pal Martinsen (palmarti)" <palmarti@cisco.com>
wrote:
>
>
> On 12 Jun 2014, at 11:58, Karl Stahl <karl.stahl@intertex.se> wrote:
>
> >>>    F18     The browser must be able to send streams and
> >>>            data to a peer in the presence of NATs and
> >>>            Firewalls that block UDP traffic.
> >
> >> Yes. Using TURN servers is the way in which it is possible to fulfil
this
> > requirement.
> >
> > Further in draft-ietf-rtcweb-use-cases-and-requirements-14:
> > "the straddling TURN server ... 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."
> > F20     The browser must support the use of STUN and TURN
> >           servers that are supplied by entities other than
> >           the web application (i.e. the network provider).
> >
> > This is a TURN server with one interface on the Enterprise LAN where th=
e
> > browser
> > simple shall place the WebRTC media. (The other interface should be
public.)
> >
> > The way for the browser to autodiscover such network provided (from the
> > enterprise and ISP) TURN server is being worked in
> > draft-patil-tram-turn-serv-disc-01.txt.
> >
> > This resolves any firewall restriction not allowing WebRTC media (by
> > paralleling the firewall with the TURN server), and also allows the
> > enterprise or ISP to point out a better path for the WebRTC media
(instead
> > of through the often data crowded firewall default gateway).
> > (This will also be needed with IPv6.)
> >
>
> I have problems understanding how this solves any issues. I doubt many
enterprise network managers would install a TURN server in parallel with
their FW. Would be fair easier and safer to open up the appropriate ports
in the firewall towards the TURN server.  They could then allow only UDP to
allow for DPI and see that it is actually RTP traffic flowing.
>

I believe they will install a TURN server in parallel to a firewall, like
many enterprises install SBCs in parallel to firewalls. Enterprise
firewalls are generally not optimized for multimedia traffic.

CB
> Having a TURN server supporting UDP/TCP/TLS as communication between the
client and TURN server would solve almost all connectivity problems. We
should, as Harald also pointed out, not try to force us throughout the FW
if the admin do not want the traffic to go through.
>
> .-.
> P=C3=A5l-Erik
>
>
>
>
>
>
>
>
>
>
> > /Karl
> >
> > -----Ursprungligt meddelande-----
> > Fr=C3=A5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=C3=B6r Harald Alve=
strand
> > Skickat: den 12 juni 2014 06:17
> > Till: Makaraju, Maridi Raju (Raju); Sergio Garcia Murillo; Javier
Cervi=C3=B1o
> > Kopia: rtcweb@ietf.org
> > =C3=84mne: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP=
 or
not to
> > UDP?
> >
> > On 06/11/2014 08:38 PM, Makaraju, Maridi Raju (Raju) wrote:
> >>> The MUST was a consensus call in London. See the minutes for numbers.
I
> > was one of >the ones who were surprised at the strength of the
consensus.
> >> [Raju] I thought one of the implicit (may be explicit in
> > draft-ietf-rtcweb-use-cases-and-requirements-14) requirement for WebRTC
is
> > "implementations should work right out of the box without extra boxes
(e.g.
> > TURN) in between under UDP restrictive firewalls".
> > Unfortunately, nobody's found a way to make that true even for all
common
> > NATs/firewalls that want to permit the communication.
> >
> > TURN servers are the least obnoxious way to make sure apps can work in
the
> > presence of symmetric NATs.
> > We could wish that this wasn't so, but we could also wish that NATs
would go
> > away and everyone would run IPv6; neither is likely to happen this year=
.
> >
> > Of course, in the presence of firewalls where the admin does *not* want
to
> > permit the communication, we neither can nor should make communication
work.
> >
> >
> >> Then I see the following requirement in
> > draft-ietf-rtcweb-use-cases-and-requirements-14. It did not mention use
of
> > TURN in the text.
> >>
> >>    F18     The browser must be able to send streams and
> >>            data to a peer in the presence of NATs and
> >>            Firewalls that block UDP traffic.
> >
> > Yes. Using TURN servers is the way in which it is possible to fulfil
this
> > requirement.
> >
> >>
> >> BR
> >> Raju
> >>
> >
> > _______________________________________________
> > 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

--047d7b86cdc28f071504fba2d3f9
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<p dir=3D"ltr"><br>
On Jun 12, 2014 4:31 AM, &quot;Pal Martinsen (palmarti)&quot; &lt;<a href=
=3D"mailto:palmarti@cisco.com">palmarti@cisco.com</a>&gt; wrote:<br>
&gt;<br>
&gt;<br>
&gt; On 12 Jun 2014, at 11:58, Karl Stahl &lt;<a href=3D"mailto:karl.stahl@=
intertex.se">karl.stahl@intertex.se</a>&gt; wrote:<br>
&gt;<br>
&gt; &gt;&gt;&gt; =C2=A0 =C2=A0F18 =C2=A0 =C2=A0 The browser must be able t=
o send streams and<br>
&gt; &gt;&gt;&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0data to a peer i=
n the presence of NATs and<br>
&gt; &gt;&gt;&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0Firewalls that b=
lock UDP traffic.<br>
&gt; &gt;<br>
&gt; &gt;&gt; Yes. Using TURN servers is the way in which it is possible to=
 fulfil this<br>
&gt; &gt; requirement.<br>
&gt; &gt;<br>
&gt; &gt; Further in draft-ietf-rtcweb-use-cases-and-requirements-14:<br>
&gt; &gt; &quot;the straddling TURN server ... It must be<br>
&gt; &gt; =C2=A0 possible to configure the browsers used in the enterprise =
with<br>
&gt; &gt; =C2=A0 network specific STUN and TURN servers. =C2=A0This should =
be possible to<br>
&gt; &gt; =C2=A0 achieve by auto-configuration methods.&quot;<br>
&gt; &gt; F20 =C2=A0 =C2=A0 The browser must support the use of STUN and TU=
RN<br>
&gt; &gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 servers that are supplied by e=
ntities other than<br>
&gt; &gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 the web application (i.e. the =
network provider).<br>
&gt; &gt;<br>
&gt; &gt; This is a TURN server with one interface on the Enterprise LAN wh=
ere the<br>
&gt; &gt; browser<br>
&gt; &gt; simple shall place the WebRTC media. (The other interface should =
be public.)<br>
&gt; &gt;<br>
&gt; &gt; The way for the browser to autodiscover such network provided (fr=
om the<br>
&gt; &gt; enterprise and ISP) TURN server is being worked in<br>
&gt; &gt; draft-patil-tram-turn-serv-disc-01.txt.<br>
&gt; &gt;<br>
&gt; &gt; This resolves any firewall restriction not allowing WebRTC media =
(by<br>
&gt; &gt; paralleling the firewall with the TURN server), and also allows t=
he<br>
&gt; &gt; enterprise or ISP to point out a better path for the WebRTC media=
 (instead<br>
&gt; &gt; of through the often data crowded firewall default gateway).<br>
&gt; &gt; (This will also be needed with IPv6.)<br>
&gt; &gt;<br>
&gt;<br>
&gt; I have problems understanding how this solves any issues. I doubt many=
 enterprise network managers would install a TURN server in parallel with t=
heir FW. Would be fair easier and safer to open up the appropriate ports in=
 the firewall towards the TURN server. =C2=A0They could then allow only UDP=
 to allow for DPI and see that it is actually RTP traffic flowing.<br>

&gt;<br></p>
<p dir=3D"ltr">I believe they will install a TURN server in parallel to a f=
irewall, like many enterprises install SBCs in parallel to firewalls. Enter=
prise firewalls are generally not optimized for multimedia traffic. </p>

<p dir=3D"ltr">CB<br>
&gt; Having a TURN server supporting UDP/TCP/TLS as communication between t=
he client and TURN server would solve almost all connectivity problems. We =
should, as Harald also pointed out, not try to force us throughout the FW i=
f the admin do not want the traffic to go through.<br>

&gt;<br>
&gt; .-.<br>
&gt; P=C3=A5l-Erik<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; &gt; /Karl<br>
&gt; &gt;<br>
&gt; &gt; -----Ursprungligt meddelande-----<br>
&gt; &gt; Fr=C3=A5n: rtcweb [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.o=
rg">rtcweb-bounces@ietf.org</a>] F=C3=B6r Harald Alvestrand<br>
&gt; &gt; Skickat: den 12 juni 2014 06:17<br>
&gt; &gt; Till: Makaraju, Maridi Raju (Raju); Sergio Garcia Murillo; Javier=
 Cervi=C3=B1o<br>
&gt; &gt; Kopia: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; &gt; =C3=84mne: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: =
To UDP or not to<br>
&gt; &gt; UDP?<br>
&gt; &gt;<br>
&gt; &gt; On 06/11/2014 08:38 PM, Makaraju, Maridi Raju (Raju) wrote:<br>
&gt; &gt;&gt;&gt; The MUST was a consensus call in London. See the minutes =
for numbers. I<br>
&gt; &gt; was one of &gt;the ones who were surprised at the strength of the=
 consensus.<br>
&gt; &gt;&gt; [Raju] I thought one of the implicit (may be explicit in<br>
&gt; &gt; draft-ietf-rtcweb-use-cases-and-requirements-14) requirement for =
WebRTC is<br>
&gt; &gt; &quot;implementations should work right out of the box without ex=
tra boxes (e.g.<br>
&gt; &gt; TURN) in between under UDP restrictive firewalls&quot;.<br>
&gt; &gt; Unfortunately, nobody&#39;s found a way to make that true even fo=
r all common<br>
&gt; &gt; NATs/firewalls that want to permit the communication.<br>
&gt; &gt;<br>
&gt; &gt; TURN servers are the least obnoxious way to make sure apps can wo=
rk in the<br>
&gt; &gt; presence of symmetric NATs.<br>
&gt; &gt; We could wish that this wasn&#39;t so, but we could also wish tha=
t NATs would go<br>
&gt; &gt; away and everyone would run IPv6; neither is likely to happen thi=
s year.<br>
&gt; &gt;<br>
&gt; &gt; Of course, in the presence of firewalls where the admin does *not=
* want to<br>
&gt; &gt; permit the communication, we neither can nor should make communic=
ation work.<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;&gt; Then I see the following requirement in<br>
&gt; &gt; draft-ietf-rtcweb-use-cases-and-requirements-14. It did not menti=
on use of<br>
&gt; &gt; TURN in the text.<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; =C2=A0 =C2=A0F18 =C2=A0 =C2=A0 The browser must be able to se=
nd streams and<br>
&gt; &gt;&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0data to a peer in th=
e presence of NATs and<br>
&gt; &gt;&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0Firewalls that block=
 UDP traffic.<br>
&gt; &gt;<br>
&gt; &gt; Yes. Using TURN servers is the way in which it is possible to ful=
fil this<br>
&gt; &gt; requirement.<br>
&gt; &gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; BR<br>
&gt; &gt;&gt; Raju<br>
&gt; &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">https://=
www.ietf.org/mailman/listinfo/rtcweb</a><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">https://=
www.ietf.org/mailman/listinfo/rtcweb</a><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">https://www.i=
etf.org/mailman/listinfo/rtcweb</a><br>
</p>

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To: "Pal Martinsen (palmarti)" <palmarti@cisco.com>,  Karl Stahl <karl.stahl@intertex.se>
References: <CAOJ7v-2FTAKGd2ZUNt9PZBW9pFu7c9v7Gx8Z8vFOQo4K3dyr5Q@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D35B859@ESESSMB209.ericsson.se> <CALiegfn7WuyA8qbKWGs4JrNutqF9teXiEN8eqd0UJk5vRrX4TQ@mail.gmail.com> <539786C7.8090806@gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E406A88@US70UWXCHMBA02.zam.alcatel-lucent.com> <CAG=SL7mGHuP0KBuvPGQ8bG0+CxHB7WjemCxZ5WxCXB8XgpOhCQ@mail.gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E407D35@US70UWXCHMBA02.zam.alcatel-lucent.com> <53984062.1090202@gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E407F64@US70UWXCHMBA02.zam.alcatel-lucent.com> <53986129.4050209@gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E40858D@US70UWXCHMBA02.zam.alcatel-lucent.com> <53987FC3.7060408@gmail.com> <6025bc75-db01-45d4-ac4a-50a39740f15a@email.android.com> <E1FE4C082A89A246A11D7F32A95A17828E408FC8@US70UWXCHMBA02.zam.alcatel-lucent.com> <539929C1.3070 205@alvestrand.no> <017301cf8624$d4a68060$7df38120$@stahl@intertex.se> <057DC1BB-CEDA-4E28-A9EC-41C3FE5673F3@cisco.com>
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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On 06/12/2014 01:30 PM, Pal Martinsen (palmarti) wrote:
> On 12 Jun 2014, at 11:58, Karl Stahl <karl.stahl@intertex.se> wrote:
>
>>>>     F18     The browser must be able to send streams and
>>>>             data to a peer in the presence of NATs and
>>>>             Firewalls that block UDP traffic.
>>> Yes. Using TURN servers is the way in which it is possible to fulfil this
>> requirement.
>>
>> Further in draft-ietf-rtcweb-use-cases-and-requirements-14:
>> "the straddling TURN server ... 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."
>> F20     The browser must support the use of STUN and TURN
>>            servers that are supplied by entities other than
>>            the web application (i.e. the network provider).
>>
>> This is a TURN server with one interface on the Enterprise LAN where the
>> browser
>> simple shall place the WebRTC media. (The other interface should be public.)
>>
>> The way for the browser to autodiscover such network provided (from the
>> enterprise and ISP) TURN server is being worked in
>> draft-patil-tram-turn-serv-disc-01.txt.
>>
>> This resolves any firewall restriction not allowing WebRTC media (by
>> paralleling the firewall with the TURN server), and also allows the
>> enterprise or ISP to point out a better path for the WebRTC media (instead
>> of through the often data crowded firewall default gateway).
>> (This will also be needed with IPv6.)
>>
> I have problems understanding how this solves any issues. I doubt many enterprise network managers would install a TURN server in parallel with their FW. Would be fair easier and safer to open up the appropriate ports in the firewall towards the TURN server.  They could then allow only UDP to allow for DPI and see that it is actually RTP traffic flowing.

That is actually a very convenient way to (logically) install a TURN 
server in parallel with your firewall.
They don't have to be on the same physical segments to be in parallel.

The important thing that Cullen (I think it was Cullen) pointed out when 
we discussed this requirement is that the TURN server is designated by 
the local network administrator, not by the entity responsible for the 
Web application that uses WebRTC.

Thus, some way for the browser to find this TURN server is needed.

>
> Having a TURN server supporting UDP/TCP/TLS as communication between the client and TURN server would solve almost all connectivity problems. We should, as Harald also pointed out, not try to force us throughout the FW if the admin do not want the traffic to go through.
>
> .-.
> Pċl-Erik
>
>
>
>
>
>
>
>
>
>
>> /Karl
>>
>> -----Ursprungligt meddelande-----
>> Frċn: rtcweb [mailto:rtcweb-bounces@ietf.org] För Harald Alvestrand
>> Skickat: den 12 juni 2014 06:17
>> Till: Makaraju, Maridi Raju (Raju); Sergio Garcia Murillo; Javier Cerviño
>> Kopia: rtcweb@ietf.org
>> Ämne: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to
>> UDP?
>>
>> On 06/11/2014 08:38 PM, Makaraju, Maridi Raju (Raju) wrote:
>>>> The MUST was a consensus call in London. See the minutes for numbers. I
>> was one of >the ones who were surprised at the strength of the consensus.
>>> [Raju] I thought one of the implicit (may be explicit in
>> draft-ietf-rtcweb-use-cases-and-requirements-14) requirement for WebRTC is
>> "implementations should work right out of the box without extra boxes (e.g.
>> TURN) in between under UDP restrictive firewalls".
>> Unfortunately, nobody's found a way to make that true even for all common
>> NATs/firewalls that want to permit the communication.
>>
>> TURN servers are the least obnoxious way to make sure apps can work in the
>> presence of symmetric NATs.
>> We could wish that this wasn't so, but we could also wish that NATs would go
>> away and everyone would run IPv6; neither is likely to happen this year.
>>
>> Of course, in the presence of firewalls where the admin does *not* want to
>> permit the communication, we neither can nor should make communication work.
>>
>>
>>> Then I see the following requirement in
>> draft-ietf-rtcweb-use-cases-and-requirements-14. It did not mention use of
>> TURN in the text.
>>>     F18     The browser must be able to send streams and
>>>             data to a peer in the presence of NATs and
>>>             Firewalls that block UDP traffic.
>> Yes. Using TURN servers is the way in which it is possible to fulfil this
>> requirement.
>>
>>> BR
>>> Raju
>>>
>> _______________________________________________
>> 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: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 12 Jun 2014 17:18:04 +0200
Message-ID: <CALiegfnektKv9awpcWfnsE+ym=Qgaf2Su=JK1GN1kBsGq+SEdQ@mail.gmail.com>
To: Harald Alvestrand <harald@alvestrand.no>
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Cc: =?UTF-8?Q?Javier_Cervi=C3=B1o?= <jcague@gmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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2014-06-12 16:57 GMT+02:00 Harald Alvestrand <harald@alvestrand.no>:
> The important thing that Cullen (I think it was Cullen) pointed out when =
we
> discussed this requirement is that the TURN server is designated by the
> local network administrator, not by the entity responsible for the Web
> application that uses WebRTC.
>
> Thus, some way for the browser to find this TURN server is needed.

This topic has been discussed many times (and IMHO it deserves a
separate thread). Basically, we need per-browser TURN settings. They
may be global or per-origin (which overrides the global ones).

And then we need an API to get it:

var turnServers =3D navigator.getTurnServers()
// now pass turnServers to PeerConnection as usual.

but... that means that all the web applications must add the above
code into their webs. It may also be "transparent", this is, if the
PeerConnection is not provided with TURN servers it may internally
retrieve them from the browser. But probably that would fail if the
web developer set his own TURN servers and the "corporative firewall"
block them. So we probably also need a "Force TURN servers
[true/false]" within the browser TURN serttins (again global and
per-origin).


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


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From: Ted Hardie <ted.ietf@gmail.com>
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Dear Working Group,

Attached is a draft set of minutes from the meeting.  The secretariat is
currently working on moving the videos of the meeting to the IETF channel
at Youtube; the sources videos are available at
https://drive.google.com/#folders/0B-1skjUwD-8EakhmY2pFdV85dHM if you would
like to download them, but please be aware that they are very large.

Comments and corrections on the minutes to the list please,

Ted, Cullen, Sean

--001a11c1faca8349f204fba598f7
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:georgia,=
serif">Dear Working Group,<br><br>Attached is a draft set of minutes from t=
he meeting.=C2=A0 The secretariat is currently working on moving the videos=
 of the meeting to the IETF channel at Youtube; the sources videos are avai=
lable at <a href=3D"https://drive.google.com/#folders/0B-1skjUwD-8EakhmY2pF=
dV85dHM">https://drive.google.com/#folders/0B-1skjUwD-8EakhmY2pFdV85dHM</a>=
 if you would like to download them, but please be aware that they are very=
 large.<br>
<br></div><div class=3D"gmail_default" style=3D"font-family:georgia,serif">=
Comments and corrections on the minutes to the list please,<br><br>Ted, Cul=
len, Sean<br></div></div>

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--001a11c1faca8349f704fba598f9
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--001a11c1faca8349f704fba598f9--


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References: <CAOJ7v-2FTAKGd2ZUNt9PZBW9pFu7c9v7Gx8Z8vFOQo4K3dyr5Q@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D35B4E7@ESESSMB209.ericsson.se> <CALiegfncVjR-cQV=coLdmO6OODgbuP-pZ2fxopcWfgGZm+jHyQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D35B859@ESESSMB209.ericsson.se> <CALiegfn7WuyA8qbKWGs4JrNutqF9teXiEN8eqd0UJk5vRrX4TQ@mail.gmail.com> <539786C7.8090806@gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E406A88@US70UWXCHMBA02.zam.alcatel-lucent.com> <CAG=SL7mGHuP0KBuvPGQ8bG0+CxHB7WjemCxZ5WxCXB8XgpOhCQ@mail.gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E407D35@US70UWXCHMBA02.zam.alcatel-lucent.com> <53984062.1090202@gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E407F64@US70UWXCHMBA02.zam.alcatel-lucent.com> <53986129.4050209@gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E40858D@US70UWXCHMBA02.zam.alcatel-lucent.com> <53987FC3.7060408@gmail.com> <6025bc75-db01-45d4-ac4a-50a39740f15a@email.android.com> <539929C1.3070205@alvestrand.no> <E1FE4C082A89A246A11D7F32A95A17828E40A80E@US70UWXCHMBA02.zam.alcatel-lucent.com>
Date: Thu, 12 Jun 2014 13:51:04 -0400
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From: Roman Shpount <roman@telurix.com>
To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
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Cc: =?UTF-8?Q?Javier_Cervi=C3=B1o?= <jcague@gmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Need for TURN, TCP-ICE (was Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?)
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--089e014940f075a2d404fba734e2
Content-Type: text/plain; charset=UTF-8

On the related note, is there TLS-ICE? Would not it be simpler to open a
TLS connection to the server with provided fingerprint validation and use
RFC 4571 framing with plain RTP? This will also have a higher chance of
traversing firewalls then plain TCP. Even if DTLS is used over this TLS
connection, it would still make sense to use TLS instead of TCP for better
firewall traversal.

_____________
Roman Shpount


On Thu, Jun 12, 2014 at 8:12 AM, Makaraju, Maridi Raju (Raju) <
Raju.Makaraju@alcatel-lucent.com> wrote:

> Since we are digressing a lot I changed the subject.
>
> > TURN servers are the least obnoxious way to make sure apps can work in
> > the presence of symmetric NATs.
>
> [Raju] Agreed. However, if there is no symmetric NAT(s) in the path webrtc
> can find a connection without the TURN servers. I think, that is highly
> preferred and thus explains the "MUST" for TCP-ICE requirement. No matter
> how much we trust a TURN server, it is still a *middle box* and there will
> be privacy concerns with it!!
>
>
> > > Then I see the following requirement in
> draft-ietf-rtcweb-use-cases-and-
> > requirements-14. It did not mention use of TURN in the text.
> > >
> > >     F18     The browser must be able to send streams and
> > >             data to a peer in the presence of NATs and
> > >             Firewalls that block UDP traffic.
> >
> > Yes. Using TURN servers is the way in which it is possible to fulfil
> > this requirement.
> [Raju] Yes, I think use of STUN servers only if direct connection using
> TCP ICE is not possible (due to symmetric NATs). Due to various issues with
> TURNs or middle boxes (latency, cost of service, increased loss of packets)
> ICE RFC recommends the following priorities for ICE candidates. I don't see
> webrtc has an overriding requirement for this. At the end, it's always
> application/user choice to use or not use TURN or other middle boxes. To
> facilitate this, IMHO browser requirement of MUST for TCP-ICE is justified.
>
> http://tools.ietf.org/html/rfc5245#section-4.1.2.2
>    If these concerns are important, the
>    type preference for relayed candidates SHOULD be lower than host
>    candidates.  The RECOMMENDED values are 126 for host candidates, 100
>    for server reflexive candidates, 110 for peer reflexive candidates,
>    and 0 for relayed candidates.
>
> BR
> Raju
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--089e014940f075a2d404fba734e2
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">On the related note, is there TLS-ICE? Would not it be sim=
pler to open a TLS connection to the server with provided=C2=A0fingerprint =
validation and use RFC 4571 framing with plain RTP? This will also have a h=
igher chance of traversing firewalls then plain TCP. Even if DTLS is used o=
ver this TLS connection, it would still make sense to use TLS instead of TC=
P for better firewall traversal.</div>
<div class=3D"gmail_extra"><br clear=3D"all"><div>_____________<br>Roman Sh=
pount</div>
<br><br><div class=3D"gmail_quote">On Thu, Jun 12, 2014 at 8:12 AM, Makaraj=
u, Maridi Raju (Raju) <span dir=3D"ltr">&lt;<a href=3D"mailto:Raju.Makaraju=
@alcatel-lucent.com" target=3D"_blank">Raju.Makaraju@alcatel-lucent.com</a>=
&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Since we are digressing a lot I changed the =
subject.<br>
<br>
&gt; TURN servers are the least obnoxious way to make sure apps can work in=
<br>
&gt; the presence of symmetric NATs.<br>
<br>
[Raju] Agreed. However, if there is no symmetric NAT(s) in the path webrtc =
can find a connection without the TURN servers. I think, that is highly pre=
ferred and thus explains the &quot;MUST&quot; for TCP-ICE requirement. No m=
atter how much we trust a TURN server, it is still a *middle box* and there=
 will be privacy concerns with it!!<br>

<br>
<br>
&gt; &gt; Then I see the following requirement in draft-ietf-rtcweb-use-cas=
es-and-<br>
&gt; requirements-14. It did not mention use of TURN in the text.<br>
&gt; &gt;<br>
&gt; &gt; =C2=A0 =C2=A0 F18 =C2=A0 =C2=A0 The browser must be able to send =
streams and<br>
&gt; &gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 data to a peer in the p=
resence of NATs and<br>
&gt; &gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Firewalls that block UD=
P traffic.<br>
&gt;<br>
&gt; Yes. Using TURN servers is the way in which it is possible to fulfil<b=
r>
&gt; this requirement.<br>
[Raju] Yes, I think use of STUN servers only if direct connection using TCP=
 ICE is not possible (due to symmetric NATs). Due to various issues with TU=
RNs or middle boxes (latency, cost of service, increased loss of packets) I=
CE RFC recommends the following priorities for ICE candidates. I don&#39;t =
see webrtc has an overriding requirement for this. At the end, it&#39;s alw=
ays application/user choice to use or not use TURN or other middle boxes. T=
o facilitate this, IMHO browser requirement of MUST for TCP-ICE is justifie=
d.<br>

<br>
<a href=3D"http://tools.ietf.org/html/rfc5245#section-4.1.2.2" target=3D"_b=
lank">http://tools.ietf.org/html/rfc5245#section-4.1.2.2</a><br>
=C2=A0 =C2=A0If these concerns are important, the<br>
=C2=A0 =C2=A0type preference for relayed candidates SHOULD be lower than ho=
st<br>
=C2=A0 =C2=A0candidates. =C2=A0The RECOMMENDED values are 126 for host cand=
idates, 100<br>
=C2=A0 =C2=A0for server reflexive candidates, 110 for peer reflexive candid=
ates,<br>
=C2=A0 =C2=A0and 0 for relayed candidates.<br>
<br>
BR<br>
Raju<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>

--089e014940f075a2d404fba734e2--


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References: <CAOJ7v-2FTAKGd2ZUNt9PZBW9pFu7c9v7Gx8Z8vFOQo4K3dyr5Q@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D35B4E7@ESESSMB209.ericsson.se> <CALiegfncVjR-cQV=coLdmO6OODgbuP-pZ2fxopcWfgGZm+jHyQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D35B859@ESESSMB209.ericsson.se> <CALiegfn7WuyA8qbKWGs4JrNutqF9teXiEN8eqd0UJk5vRrX4TQ@mail.gmail.com> <539786C7.8090806@gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E406A88@US70UWXCHMBA02.zam.alcatel-lucent.com> <CAG=SL7mGHuP0KBuvPGQ8bG0+CxHB7WjemCxZ5WxCXB8XgpOhCQ@mail.gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E407D35@US70UWXCHMBA02.zam.alcatel-lucent.com> <53984062.1090202@gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E407F64@US70UWXCHMBA02.zam.alcatel-lucent.com> <53986129.4050209@gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E40858D@US70UWXCHMBA02.zam.alcatel-lucent.com> <53987FC3.7060408@gmail.com> <6025bc75-db01-45d4-ac4a-50a39740f15a@email.android.com> <E1FE4C082A89A246A11D7F32A95A17828E408FC8@US70UWXCHMBA02.zam.alcatel-lucent.com> <057DC1BB-CEDA-4E28-A9EC-41C3FE5673F3@cisco.com> <CAD6AjGTNp9VAj9X9=Y8KdNB_3W5RT+6mZ0eLwG0r3povcTzS=g@mail.gmail.com>
Date: Thu, 12 Jun 2014 14:04:26 -0400
Message-ID: <CAD5OKxtVkuv30=jKhs1tZ8z4R95aG=BZmdxuowt+kXq_CMhHyg@mail.gmail.com>
From: Roman Shpount <roman@telurix.com>
To: Ca By <cb.list6@gmail.com>
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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--089e014940f0377e5504fba76463
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On Thu, Jun 12, 2014 at 8:37 AM, Ca By <cb.list6@gmail.com> wrote:

> I believe they will install a TURN server in parallel to a firewall, like
> many enterprises install SBCs in parallel to firewalls. Enterprise
> firewalls are generally not optimized for multimedia traffic.
>
May I remind that SBCs installed in parallel to firewalls are usually
called SIP ALGs and they are mostly broken (
https://wiki.freeswitch.org/wiki/ALG)?

I am afraid that most of the locally provided TURN servers will be terribly
mis-configured, used when they should not, and generally cause more harm
then good. For local TURN servers to work correctly you would need local
network administrators which are educated in correct WebRTC setup and who
are interested in spending time to make their setup to work correctly. In
most cases this would be highly unlikely. All an administrator would do is
turn on some setting in their firewall labeled "Enable WebRTC TURN" which
would more likely turn on something no one will understand and cause all
types of havoc. Based on our experience with SIP, expecting local network
administrators to provide any working solution for NAT traversal is
extremely optimistic.
_____________
Roman Shpount

--089e014940f0377e5504fba76463
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div>On Thu, Jun 12, 2014 at 8:=
37 AM, Ca By <span dir=3D"ltr">&lt;<a href=3D"mailto:cb.list6@gmail.com" ta=
rget=3D"_blank">cb.list6@gmail.com</a>&gt;</span> wrote:<br></div><div clas=
s=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-left-style:solid;p=
adding-left:1ex"><div class=3D""><div class=3D"h5"><p dir=3D"ltr"><span sty=
le=3D"color:rgb(34,34,34)">I believe they will install a TURN server in par=
allel to a firewall, like many enterprises install SBCs in parallel to fire=
walls. Enterprise firewalls are generally not optimized for multimedia traf=
fic.</span></p>
</div></div></blockquote><div>May I remind that SBCs installed in parallel =
to firewalls are usually called SIP ALGs and they are mostly broken (<a hre=
f=3D"https://wiki.freeswitch.org/wiki/ALG">https://wiki.freeswitch.org/wiki=
/ALG</a>)?</div>
<div><br></div><div>I am afraid that most of the locally provided TURN serv=
ers will be terribly mis-configured, used when they should not, and general=
ly cause more harm then good. For local TURN servers to work correctly you =
would need local network administrators which are educated in correct WebRT=
C setup and who are interested in spending time to make their setup to work=
 correctly. In most cases this would be highly unlikely. All an administrat=
or would do is turn on some setting in their firewall labeled &quot;Enable =
WebRTC TURN&quot; which would more likely turn on something no one will und=
erstand and cause all types of havoc. Based on our experience with SIP, exp=
ecting local network administrators to provide any working solution for NAT=
 traversal is extremely optimistic.</div>
<div>_____________<br>Roman Shpount</div><div>=C2=A0</div></div></div></div=
>

--089e014940f0377e5504fba76463--


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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] Need for TURN, TCP-ICE (was Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?)
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Roman Shpount <roman@telurix.com>, Ca By <cb.list6@gmail.com>
Thread-Topic: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Date: Thu, 12 Jun 2014 17:16:08 -0400
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From: Roman Shpount <roman@telurix.com>
To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
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Subject: Re: [rtcweb] Need for TURN, TCP-ICE (was Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?)
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--047d7bdc0e4ccb89a104fbaa1176
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On Thu, Jun 12, 2014 at 4:07 PM, Makaraju, Maridi Raju (Raju) <
Raju.Makaraju@alcatel-lucent.com> wrote:

> >On the related note, is there TLS-ICE? Would not it be simpler to open a
> TLS connection >to the server with provided fingerprint validation and use
> RFC 4571 framing with plain >RTP? This will also have a higher chance of
> traversing firewalls then plain TCP. Even if >DTLS is used over this TLS
> connection, it would still make sense to use TLS instead of >TCP for better
> firewall traversal.
>
> [Raju] ICE TCP RFC 6544 does cover both TLS and DTLS. Please see page 6
> http://tools.ietf.org/html/rfc6544#section-3
> Yes, you can run plain RTP on top of TLS-ICE, but for webrtc use it was
> already decided to use DTLS-SRTP even for TCP.
>
>
I am not sure this is right since TLS or DTLS are supposed to run on top of
RFC 4571 on top of TCP. What I was asking about was RTP on top of RFC 4571
on top of TLS.
_____________
Roman Shpount

--047d7bdc0e4ccb89a104fbaa1176
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 Thu, Jun 12, 2014 at 4:07 PM, Makaraju, Maridi Raju (Raju) <span dir=3D"=
ltr">&lt;<a href=3D"mailto:Raju.Makaraju@alcatel-lucent.com" target=3D"_bla=
nk">Raju.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;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div>&gt;On the related note, is there TLS-ICE? Would not =
it be simpler to open a TLS connection &gt;to the server with provided fing=
erprint validation and use RFC 4571 framing with plain &gt;RTP? This will a=
lso have a higher chance of traversing firewalls then plain TCP. Even if &g=
t;DTLS is used over this TLS connection, it would still make sense to use T=
LS instead of &gt;TCP for better firewall traversal.<br>


<br>
</div>[Raju] ICE TCP RFC 6544 does cover both TLS and DTLS. Please see page=
 6 <a href=3D"http://tools.ietf.org/html/rfc6544#section-3" target=3D"_blan=
k">http://tools.ietf.org/html/rfc6544#section-3</a><br>
Yes, you can run plain RTP on top of TLS-ICE, but for webrtc use it was alr=
eady decided to use DTLS-SRTP even for TCP.<br>
<br></blockquote><div><br></div><div>I am not sure this is right since TLS =
or DTLS are supposed to run=C2=A0on top of RFC 4571 on top of TCP. What I w=
as asking about was RTP on top of RFC 4571 on top of TLS.</div><div>_______=
______<br>
Roman Shpount</div>
<div>=C2=A0</div></div></div></div>

--047d7bdc0e4ccb89a104fbaa1176--


From nobody Fri Jun 13 05:54:56 2014
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From: "Karl Stahl" <karl.stahl@intertex.se>
To: "'Harald Alvestrand'" <harald@alvestrand.no>, "'Pal Martinsen \(palmarti\)'" <palmarti@cisco.com>
References: <CAOJ7v-2FTAKGd2ZUNt9PZBW9pFu7c9v7Gx8Z8vFOQo4K3dyr5Q@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D35B859@ESESSMB209.ericsson.se> <CALiegfn7WuyA8qbKWGs4JrNutqF9teXiEN8eqd0UJk5vRrX4TQ@mail.gmail.com> <539786C7.8090806@gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E406A88@US70UWXCHMBA02.zam.alcatel-lucent.com> <CAG=SL7mGHuP0KBuvPGQ8bG0+CxHB7WjemCxZ5WxCXB8XgpOhCQ@mail.gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E407D35@US70UWXCHMBA02.zam.alcatel-lucent.com> <53984062.1090202@gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E407F64@US70UWXCHMBA02.zam.alcatel-lucent.com> <53986129.4050209@gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E40858D@US70UWXCHMBA02.zam.alcatel-lucent.com> <53987FC3.7060408@gmail.com> <6025bc75-db01-45d4-ac4a-50a39740f15a@email.android.com> <E1FE4C082A89A246A11D7F32A95A17828E408FC8@US70UWXCHMBA02.zam.alcatel-lucent.com> <539929C1.3070 205@alvestrand.no> <017301cf8624$d4a68060$7df38120$@stahl@intertex.se> <057DC1BB-CEDA-4E28-A9EC-41C3FE5673F3@cisco.com> <5399BFE6.5050 003@alvestrand.no>
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Subject: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?
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Below I am agreeing with Harald that WebRTC browsers need to =
auto-discover
and use network provided TURN servers in favor of application provided =
TURN
servers (that will still remain with its hassles and limitations, for =
some
time of course). -->

/Karl

-----Ursprungligt meddelande-----
Fr=E5n: Harald Alvestrand [mailto:harald@alvestrand.no]=20
Skickat: den 12 juni 2014 16:58
Till: Pal Martinsen (palmarti); Karl Stahl
Kopia: Makaraju, Maridi Raju (Raju); Sergio Garcia Murillo; Javier =
Cervi=F1o;
rtcweb@ietf.org
=C4mne: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or =
not to
UDP?

On 06/12/2014 01:30 PM, Pal Martinsen (palmarti) wrote:
> On 12 Jun 2014, at 11:58, Karl Stahl <karl.stahl@intertex.se> wrote:
>
>>>>     F18     The browser must be able to send streams and
>>>>             data to a peer in the presence of NATs and
>>>>             Firewalls that block UDP traffic.
>>> Yes. Using TURN servers is the way in which it is possible to fulfil =

>>> this
>> requirement.
>>
>> Further in draft-ietf-rtcweb-use-cases-and-requirements-14:
>> "the straddling TURN server ... 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."
>> F20     The browser must support the use of STUN and TURN
>>            servers that are supplied by entities other than
>>            the web application (i.e. the network provider).
>>
>> This is a TURN server with one interface on the Enterprise LAN where=20
>> the browser simple shall place the WebRTC media. (The other interface =

>> should be public.)
>>
>> The way for the browser to autodiscover such network provided (from=20
>> the enterprise and ISP) TURN server is being worked in=20
>> draft-patil-tram-turn-serv-disc-01.txt.
>>
>> This resolves any firewall restriction not allowing WebRTC media (by=20
>> paralleling the firewall with the TURN server), and also allows the=20
>> enterprise or ISP to point out a better path for the WebRTC media=20
>> (instead of through the often data crowded firewall default gateway).
>> (This will also be needed with IPv6.)
>>
> I have problems understanding how this solves any issues. I doubt many
enterprise network managers would install a TURN server in parallel with
their FW. Would be fair easier and safer to open up the appropriate =
ports in
the firewall towards the TURN server.  They could then allow only UDP to
allow for DPI and see that it is actually RTP traffic flowing.

That is actually a very convenient way to (logically) install a TURN =
server
in parallel with your firewall.
They don't have to be on the same physical segments to be in parallel.

The important thing that Cullen (I think it was Cullen) pointed out when =
we
discussed this requirement is that the TURN server is designated by the
local network administrator, not by the entity responsible for the Web
application that uses WebRTC.

[Karl] I fully agree with this - The WebRTC application should be =
off-loaded
of having to provide a TURN-server to allow usage behind (some)
NAT/firewalls (and we should not put that burden on the peer either). =
This
is something we do today to get something going (but still don't succeed
with the most restrictive firewalls), but hopefully don't have to do in =
the
future on "WebRTC-ready" accesses. Such accesses simply put ICE =
initiated
real-time traffic through the TURN server, while data goes as usual =
through
the default gateway (in a future clever firewall/NAT this can be the =
same
default gateway IP-address).

[Karl] Yes, let's hope for network provided turn servers, which could be
boxes in parallel with existing firewalls or built into the firewall. (A
turn server is much cleaner and better standardized to integrate, than
non-working SIP ALGs, where we instead put SBCs in parallel with =
firewalls
just to get SIP voice through (and with quality)).=20

[Karl] Also consider all triple play (surf/telephony/IP-TV) broadband
accesses, that have a quality pipe just to cope with voice. A turn =
server in
such CPE will be required to place demanding WebRTC media on the pipe =
that
can handle WebRTCs real-time traffic. Separate pipes are also often =
deployed
e.g. for SIP trunking enterprise voice (PBXs and UC) paralleling the
firewall. A local turn server could similarly feed WebRTC media into =
such
pipes, then bypassing the ordinary firewall (that may be too restrictive =
to
allow real-time traffic or just is too data crowded to cope with WebRTC
media - NAT/Firewalls are typical congestion points).

Harald> Thus, some way for the browser to find this TURN server is =
needed.

[Karl] Auto discovered, network provided (enterprise or ISP) turn =
servers
are being done in draft-patil-tram-turn-serv-disc-01.txt. I favor the
anycast method suggested. Will WebRTC browsers implement all auto =
discovery
methods suggested there?

[Karl] Network provided turn servers available for the network users, =
need
no authentication - The WebRTC browser can just use them as part of the
Internet access (you get an IP address, a default gateway and a turn =
server
address to simply use). Those turn servers are required/can solve both
traversal and quality problems of many internet accesses.


>
> Having a TURN server supporting UDP/TCP/TLS as communication between =
the
client and TURN server would solve almost all connectivity problems. We
should, as Harald also pointed out, not try to force us throughout the =
FW if
the admin do not want the traffic to go through.
>
> .-.
> P=E5l-Erik
>
>
>
>
>
>
>
>
>
>
>> /Karl
>>
>> -----Ursprungligt meddelande-----
>> Fr=E5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=F6r Harald =
Alvestrand
>> Skickat: den 12 juni 2014 06:17
>> Till: Makaraju, Maridi Raju (Raju); Sergio Garcia Murillo; Javier=20
>> Cervi=F1o
>> Kopia: rtcweb@ietf.org
>> =C4mne: Re: [rtcweb] Retry: DTLS carrying RTP/SAVPF over ICE: To UDP =
or=20
>> not to UDP?
>>
>> On 06/11/2014 08:38 PM, Makaraju, Maridi Raju (Raju) wrote:
>>>> The MUST was a consensus call in London. See the minutes for=20
>>>> numbers. I
>> was one of >the ones who were surprised at the strength of the =
consensus.
>>> [Raju] I thought one of the implicit (may be explicit in
>> draft-ietf-rtcweb-use-cases-and-requirements-14) requirement for=20
>> WebRTC is "implementations should work right out of the box without =
extra
boxes (e.g.
>> TURN) in between under UDP restrictive firewalls".
>> Unfortunately, nobody's found a way to make that true even for all=20
>> common NATs/firewalls that want to permit the communication.
>>
>> TURN servers are the least obnoxious way to make sure apps can work=20
>> in the presence of symmetric NATs.
>> We could wish that this wasn't so, but we could also wish that NATs=20
>> would go away and everyone would run IPv6; neither is likely to =
happen
this year.
>>
>> Of course, in the presence of firewalls where the admin does *not*=20
>> want to permit the communication, we neither can nor should make
communication work.
>>
>>
>>> Then I see the following requirement in
>> draft-ietf-rtcweb-use-cases-and-requirements-14. It did not mention=20
>> use of TURN in the text.
>>>     F18     The browser must be able to send streams and
>>>             data to a peer in the presence of NATs and
>>>             Firewalls that block UDP traffic.
>> Yes. Using TURN servers is the way in which it is possible to fulfil=20
>> this requirement.
>>
>>> BR
>>> Raju
>>>
>> _______________________________________________
>> 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


From nobody Fri Jun 13 05:56:25 2014
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From: "Karl Stahl" <karl.stahl@intertex.se>
To: "'Roman Shpount'" <roman@telurix.com>, "'Makaraju, Maridi Raju \(Raju\)'" <Raju.Makaraju@alcatel-lucent.com>
References: <CAOJ7v-2FTAKGd2ZUNt9PZBW9pFu7c9v7Gx8Z8vFOQo4K3dyr5Q@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D35B4E7@ESESSMB209.ericsson.se> <CALiegfncVjR-cQV=coLdmO6OODgbuP-pZ2fxopcWfgGZm+jHyQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D35B859@ESESSMB209.ericsson.se> <CALiegfn7WuyA8qbKWGs4JrNutqF9teXiEN8eqd0UJk5vRrX4TQ@mail.gmail.com> <539786C7.8090806@gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E406A88@US70UWXCHMBA02.zam.alcatel-lucent.com> <CAG=SL7mGHuP0KBuvPGQ8bG0+CxHB7WjemCxZ5WxCXB8XgpOhCQ@mail.gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E407D35@US70UWXCHMBA02.zam.alcatel-lucent.com> <53984062.1090202@gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E407F64@US70UWXCHMBA02.zam.alcatel-lucent.com> <53986129.4050209@gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E40858D@US70UWXCHMBA02.zam.alcatel-lucent.com> <53987FC3.7060408@gmail.com> <6025bc75-db01-45d4-ac4a-50a39740f15a@email.android.com> <539929C1.3070205@alvestrand.no> <E1FE4C082A89A246A11D7F32A95A17828E40A80E@US70UW XCHMBA02.zam.a lcatel-lucent.com> <CAD5OKxsm=Y=r_9L+07VO+Mu2F8aA7C4zHZfUuKdiSuCnqv8H1Q@mail.gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E40C1AD@US70UWXCHMBA02.zam.alcatel-lucent.com> <CAD5OKxsWXYdw-MKXgZN6FQfEM3O_gtie12af8L0r=JgrmJnmJA@mail.gmail.com>
In-Reply-To: <CAD5OKxsWXYdw-MKXgZN6FQfEM3O_gtie12af8L0r=JgrmJnmJA@mail.gmail.com>
Date: Fri, 13 Jun 2014 14:56:20 +0200
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Cc: =?utf-8?Q?'Javier_Cervi=C3=B1o'?= <jcague@gmail.com>, rtcweb@ietf.org
Subject: Re: [rtcweb] Need for TURN, TCP-ICE (was Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?)
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WebRTC media over TCP/TLS (always open port 80 or 443) must also be =
considered a =E2=80=9Cworkaround=E2=80=9D to get WebRTC connectivity =
through a few more restrictive NAT/firewalls =E2=80=93 It will not cope =
with the most restrictive firewalls having HTTP(S) proxies =
(discussed/worked on at the pnta-list, wasn=E2=80=99t it).

=20

Also, in general, it is bad to put quality demanding WebRTC media over a =
retransmitting TCP connection, especially through the firewall that =
often is a data crowded congestion point.

=20

/Karl

=20

Fr=C3=A5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=C3=B6r Roman =
Shpount
Skickat: den 12 juni 2014 23:16
Till: Makaraju, Maridi Raju (Raju)
Kopia: rtcweb@ietf.org; Javier Cervi=C3=B1o
=C3=84mne: Re: [rtcweb] Need for TURN, TCP-ICE (was Retry: DTLS carrying =
RTP/SAVPF over ICE: To UDP or not to UDP?)

=20

=20

On Thu, Jun 12, 2014 at 4:07 PM, Makaraju, Maridi Raju (Raju) =
<Raju.Makaraju@alcatel-lucent.com> wrote:

>On the related note, is there TLS-ICE? Would not it be simpler to open =
a TLS connection >to the server with provided fingerprint validation and =
use RFC 4571 framing with plain >RTP? This will also have a higher =
chance of traversing firewalls then plain TCP. Even if >DTLS is used =
over this TLS connection, it would still make sense to use TLS instead =
of >TCP for better firewall traversal.

[Raju] ICE TCP RFC 6544 does cover both TLS and DTLS. Please see page 6 =
http://tools.ietf.org/html/rfc6544#section-3
Yes, you can run plain RTP on top of TLS-ICE, but for webrtc use it was =
already decided to use DTLS-SRTP even for TCP.

=20

I am not sure this is right since TLS or DTLS are supposed to run on top =
of RFC 4571 on top of TCP. What I was asking about was RTP on top of RFC =
4571 on top of TLS.

_____________
Roman Shpount

=20


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lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>We=
bRTC media over TCP/TLS (always open port 80 or 443) must also be =
considered a =E2=80=9Cworkaround=E2=80=9D to get WebRTC connectivity =
through a few more restrictive NAT/firewalls =E2=80=93 It will not cope =
with the most restrictive firewalls having HTTP(S) proxies =
(discussed/worked on at the pnta-list, wasn=E2=80=99t =
it).<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Al=
so, in general, it is bad to put quality demanding WebRTC media over a =
retransmitting TCP connection, especially through the firewall that =
often is a data crowded congestion point.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>/K=
arl<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><div style=3D'border:none;border-top:solid =
#B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=3DMsoNormal><b><span =
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span></b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> rtcweb =
[mailto:rtcweb-bounces@ietf.org] <b>F=C3=B6r </b>Roman =
Shpount<br><b>Skickat:</b> den 12 juni 2014 23:16<br><b>Till:</b> =
Makaraju, Maridi Raju (Raju)<br><b>Kopia:</b> rtcweb@ietf.org; Javier =
Cervi=C3=B1o<br><b>=C3=84mne:</b> Re: [rtcweb] Need for TURN, TCP-ICE =
(was Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to =
UDP?)<o:p></o:p></span></p></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><p class=3DMsoNormal>On Thu, =
Jun 12, 2014 at 4:07 PM, Makaraju, Maridi Raju (Raju) &lt;<a =
href=3D"mailto:Raju.Makaraju@alcatel-lucent.com" =
target=3D"_blank">Raju.Makaraju@alcatel-lucent.com</a>&gt; =
wrote:<o:p></o:p></p><div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'>&gt;On the related note, is there =
TLS-ICE? Would not it be simpler to open a TLS connection &gt;to the =
server with provided fingerprint validation and use RFC 4571 framing =
with plain &gt;RTP? This will also have a higher chance of traversing =
firewalls then plain TCP. Even if &gt;DTLS is used over this TLS =
connection, it would still make sense to use TLS instead of &gt;TCP for =
better firewall traversal.<o:p></o:p></p></div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'>[Raju] ICE TCP RFC 6544 does cover both =
TLS and DTLS. Please see page 6 <a =
href=3D"http://tools.ietf.org/html/rfc6544#section-3" =
target=3D"_blank">http://tools.ietf.org/html/rfc6544#section-3</a><br>Yes=
, you can run plain RTP on top of TLS-ICE, but for webrtc use it was =
already decided to use DTLS-SRTP even for TCP.<o:p></o:p></p><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p class=3DMsoNormal>I =
am not sure this is right since TLS or DTLS are supposed to run&nbsp;on =
top of RFC 4571 on top of TCP. What I was asking about was RTP on top of =
RFC 4571 on top of TLS.<o:p></o:p></p></div><div><p =
class=3DMsoNormal>_____________<br>Roman =
Shpount<o:p></o:p></p></div><div><p =
class=3DMsoNormal>&nbsp;<o:p></o:p></p></div></div></div></div></div></bo=
dy></html>
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Karl Stahl <karl.stahl@intertex.se>, "'Roman Shpount'" <roman@telurix.com>
Thread-Topic: [rtcweb] Need for TURN, TCP-ICE (was Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?)
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Subject: Re: [rtcweb] Need for TURN, TCP-ICE (was Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?)
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From: Justin Uberti <juberti@google.com>
Date: Fri, 13 Jun 2014 11:24:49 -0700
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To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
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Subject: Re: [rtcweb] Need for TURN, TCP-ICE (was Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?)
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ICE-TLS doesn't really make sense since you don't have a DNS name for the
remote side, just an IP, which makes certificate management very difficult.

ICE-TCP is a nice optimization for services that don't want to stand up
both a gateway and a TURN server.


On Fri, Jun 13, 2014 at 11:18 AM, Makaraju, Maridi Raju (Raju) <
Raju.Makaraju@alcatel-lucent.com> wrote:

> >WebRTC media over TCP/TLS (always open port 80 or 443) must also be
> considered a >=E2=80=9Cworkaround=E2=80=9D to get WebRTC connectivity thr=
ough a few more
> restrictive NAT/firewalls =E2=80=93 It >will not cope with the most restr=
ictive
> firewalls having HTTP(S) proxies (discussed/worked >on at the pnta-list,
> wasn=E2=80=99t it).
>
> [Raju] To get thru these very restrictive firewalls that allow HTTP only,
> using TCP/TLS with HTTP fallback is an option. But the restrictive
> firewalls may even go great lengths to limit the duration of the connecti=
on
> and/or size of the data (in case of TLS).
> With either HTTP fallback option or otherwise, doing TLS below RFC 4571
> framing has following implications for webrtc use case; a single transpor=
t
> connection need to allow following 3 types of data to be
> multiplexed/bundled:
> 1. Sending STUN req before DTLS is established; unless STUN succeeds ther=
e
> is no point
>    in setting up DTLS/TLS.
> 2. Sending DTLS-SRTP. This requires the RTP headers to be in clear text
> (i.e. not encrypted) while only the payload is encrypted.
> 3. Setting up SCTP on top of DTLS.
>
> If TLS is run below RFC 4571 then (1) and (2) can't be achieved.
>
> BR
> Raju
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">ICE-TLS doesn&#39;t really make sense since you don&#39;t =
have a DNS name for the remote side, just an IP, which makes certificate ma=
nagement very difficult.<div><br></div><div>ICE-TCP is a nice optimization =
for services that don&#39;t want to stand up both a gateway and a TURN serv=
er.</div>

</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Fri,=
 Jun 13, 2014 at 11:18 AM, Makaraju, Maridi Raju (Raju) <span dir=3D"ltr">&=
lt;<a href=3D"mailto:Raju.Makaraju@alcatel-lucent.com" target=3D"_blank">Ra=
ju.Makaraju@alcatel-lucent.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"">&gt;WebRTC media over TCP/TL=
S (always open port 80 or 443) must also be considered a &gt;=E2=80=9Cworka=
round=E2=80=9D to get WebRTC connectivity through a few more restrictive NA=
T/firewalls =E2=80=93 It &gt;will not cope with the most restrictive firewa=
lls having HTTP(S) proxies (discussed/worked &gt;on at the pnta-list, wasn=
=E2=80=99t it).<br>


<br>
</div>[Raju] To get thru these very restrictive firewalls that allow HTTP o=
nly, using TCP/TLS with HTTP fallback is an option. But the restrictive fir=
ewalls may even go great lengths to limit the duration of the connection an=
d/or size of the data (in case of TLS).<br>


With either HTTP fallback option or otherwise, doing TLS below RFC 4571 fra=
ming has following implications for webrtc use case; a single transport con=
nection need to allow following 3 types of data to be multiplexed/bundled:<=
br>


1. Sending STUN req before DTLS is established; unless STUN succeeds there =
is no point<br>
=C2=A0 =C2=A0in setting up DTLS/TLS.<br>
2. Sending DTLS-SRTP. This requires the RTP headers to be in clear text (i.=
e. not encrypted) while only the payload is encrypted.<br>
3. Setting up SCTP on top of DTLS.<br>
<br>
If TLS is run below RFC 4571 then (1) and (2) can&#39;t be achieved.<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
BR<br>
Raju<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>

--001a1136b1f426dd4904fbbbcc20--


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From: Roman Shpount <roman@telurix.com>
To: Justin Uberti <juberti@google.com>
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Subject: Re: [rtcweb] Need for TURN, TCP-ICE (was Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?)
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On Fri, Jun 13, 2014 at 2:24 PM, Justin Uberti <juberti@google.com> wrote:

> ICE-TLS doesn't really make sense since you don't have a DNS name for the
> remote side, just an IP, which makes certificate management very difficult.
>

We have a fingerprint in SDP which can be used to verify the certificates.


> ICE-TCP is a nice optimization for services that don't want to stand up
> both a gateway and a TURN server.
>

Agreed
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On F=
ri, Jun 13, 2014 at 2:24 PM, Justin Uberti <span dir=3D"ltr">&lt;<a href=3D=
"mailto:juberti@google.com" target=3D"_blank">juberti@google.com</a>&gt;</s=
pan> 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 dir=3D"ltr">ICE-TLS doesn&#39;t really make sense sin=
ce you don&#39;t have a DNS name for the remote side, just an IP, which mak=
es certificate management very difficult.</div>
</blockquote><div><br></div><div>We have a fingerprint in SDP which can be =
used to verify the certificates.=C2=A0</div><div><br></div><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;b=
order-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"=
>
<div dir=3D"ltr"><div><br></div><div>ICE-TCP is a nice optimization for ser=
vices that don&#39;t want to stand up both a gateway and a TURN server.</di=
v></div></blockquote><div><br></div><div>Agreed=C2=A0</div><div>___________=
__<br>
Roman Shpount</div><div>=C2=A0</div></div></div></div>

--089e0122f54805956604fbbc2e25--


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From: Justin Uberti <juberti@google.com>
Date: Fri, 13 Jun 2014 16:34:30 -0700
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Subject: Re: [rtcweb] Need for TURN, TCP-ICE (was Retry: DTLS carrying RTP/SAVPF over ICE: To UDP or not to UDP?)
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On Fri, Jun 13, 2014 at 11:52 AM, Roman Shpount <roman@telurix.com> wrote:

> On Fri, Jun 13, 2014 at 2:24 PM, Justin Uberti <juberti@google.com> wrote:
>
>> ICE-TLS doesn't really make sense since you don't have a DNS name for the
>> remote side, just an IP, which makes certificate management very difficult.
>>
>
> We have a fingerprint in SDP which can be used to verify the certificates.
>

Not without crossing layers; also, the fingerprint is in the SDP answer,
which means that ICE-TLS couldn't start until the answer is received.

Generally, I think ICE-TCP to 443 should work fine in most cases, and for
the cases where DPI prevents this, it's not clear the DPI wouldn't also
stop ICE-TLS.

>
>
>> ICE-TCP is a nice optimization for services that don't want to stand up
>> both a gateway and a TURN server.
>>
>
> Agreed
> _____________
> Roman Shpount
>
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Jun 13, 2014 at 11:52 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:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra">=
<div class=3D"gmail_quote"><div class=3D"">On Fri, Jun 13, 2014 at 2:24 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: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 dir=3D"ltr">ICE-TLS doesn&#39;t really make sense sin=
ce you don&#39;t have a DNS name for the remote side, just an IP, which mak=
es certificate management very difficult.</div>


</blockquote><div><br></div></div><div>We have a fingerprint in SDP which c=
an be used to verify the certificates.=C2=A0</div></div></div></div></block=
quote><div><br></div><div>Not without crossing layers; also, the fingerprin=
t is in the SDP answer, which means that ICE-TLS couldn&#39;t start until t=
he answer is received.</div>

<div><br></div><div>Generally, I think ICE-TCP to 443 should work fine in m=
ost cases, and for the cases where DPI prevents this, it&#39;s not clear th=
e DPI wouldn&#39;t also stop ICE-TLS.=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=
 class=3D""><div><br></div><blockquote class=3D"gmail_quote" style=3D"margi=
n: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><br></div><div>ICE-TCP is a nice optimization for ser=
vices that don&#39;t want to stand up both a gateway and a TURN server.</di=
v></div></blockquote><div><br></div></div><div>Agreed=C2=A0</div><div>_____=
________<span class=3D"HOEnZb"><font color=3D"#888888"><br>


Roman Shpount</font></span></div><div>=C2=A0</div></div></div></div>
</blockquote></div><br></div></div>

--001a11c35efeade06c04fbc01f9b--


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From: Justin Uberti <juberti@google.com>
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Sorry for the super long latency on this. We've created issues in the GH
tracker for the points identified below. Specific comments inline.
https://github.com/rtcweb-wg/jsep/issues/51
https://github.com/rtcweb-wg/jsep/issues/52
https://github.com/rtcweb-wg/jsep/issues/53
https://github.com/rtcweb-wg/jsep/issues/54

On Fri, Mar 14, 2014 at 9:48 AM, Justin Uberti <juberti@google.com> wrote:

> Thanks for the detailed review. We will work through these comments over
> the next few days.
>
>
> On Fri, Mar 14, 2014 at 6:05 AM, Magnus Westerlund <
> magnus.westerlund@ericsson.com> wrote:
>
>> Hi,
>>
>> I did review the JSEP -06 prior to the meeting. Now I have had time to
>> write-up my comments.
>>
>> 1) Section 1):
>>
>> [W3C.WD-webrtc-20111027]
>> This is very old version of the API, please update to the currently
>> relevant version.
>>
>
Will do.

>
>> 2) Section 3.4.1.1:
>> The m-line index is a zero-based index, referring to the Nth m-line in
>> the SDP.
>>
>> a) I think the above is a bit confusing. As a zero-based index would
>> mean that the first m=3D section would have index 0. Thus, the referring
>> to the Nth m-line in the SDP is unclear. Is N, the provided index, in
>> which case it would be the N+1th m-line. Please rewrite.
>>
>
Will clarify.


>
>> b) I also wonder if one has to be clearer on what "the SDP" is. I assume
>> it is the one that has been generated by the agent, rather than the
>> latest sent or received?
>>
>
It can be for either the local generated SDP, in the case where the
candidate is gotten from the onicecandidate callback, or the remote SDP, in
the case where the candidates are trickled from the remote side. We'll add
some text to make this clear.

>
>> 3) Section 3.4.1.1:
>> "The MID uses the "media stream identification", as defined in
>>    [RFC5888] , to identify the m-line."
>>
>> a spurious space after ref.
>>
>
OK.

>
>> 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 to gather candidates
>>    for.
>>
>> I find the statement that ICE candidate gathering starts first with
>> setLocalDescription. Doesn't the created offer already contain local
>> candidates, thus one must have already gathered some candidates? To me
>> it appears reasonable to start gathering earlier, rather than later.
>> Thus, as soon as one create the PeerConnection object and have added any
>> MST to it or a data channel, one can start gathering.
>>
>
The initial offer comes out as soon as possible, so it contains no
candidates.

setLocalDescription is the first time the implementation knows exactly how
many m=3D lines it needs to gather for, so gathering can't definitively sta=
rt
until this point. Candidates may be pre-gathered using the candidate pool,
but they aren't released via onicecandidate until setLD.


>
>> 5) Section 3.5.2:
>>    In the parallel forking case where the Javascript application wishes
>>    to simultaneously exchange media with multiple peers, the flow is
>>    slightly more complex, but the Javascript application can follow the
>>    strategy that [RFC3960] describes using UPDATE.  (It is worth noting
>>    that use cases where this is the desired behavior are very unusual.)
>>
>> I mostly react to the last sentence within the paragraph. I think the
>> poster child for semi-parallel forking was the MMORG that like to
>> establish PCs with anyone that is coming close in the virtual world.
>> Thus either needs to have a number of PCs on standby or have just one,
>> but treat that as parallel forking. And if you like to discourage the
>> later, then I think you might need to at least inform about the
>> possibility to have stand-by PCs.
>>
>
Is the MMORG case described in the use-case doc? If so, I agree the
parenthesized text should be changed.


>
>> 6. Section 3.6:
>>
>>    In the event that the local application state is reinitialized,
>>    either due to a user reload of the page, or a decision within the
>>    application to reload itself (perhaps to update to a new version), it
>>    is possible to keep an existing session alive, via a process called
>>    "rehydration".  The explicit goal of rehydration is to carry out this
>>    session resumption with no interaction with the remote side other
>>    than normal call signaling messages.
>>
>> I think this paragraph makes two erronous statements:
>> a) "keep an existing session alive"
>> I think "resume" an existing session would be more correct. Because it
>> would be broken when the reload happens.
>>
>> b) "with no interaction with the remote side"
>> My understanding is that due to DTLS and key handling a DTLS
>> renegotiation will be required with the peer. Isn't also a signalling
>> exchange likely to be required due to change of local DTLS cert as well
>> as ICE candidates as parameters? Any other parameters that will be
>> required to exchanges when rehydrating?
>>
>> This section has been removed, as nobody was working on this and it
wasn't clear exactly what it ought to do.


>  7. Section 3.6:
>> Next, a new offer is
>>    generated by the new PeerConnection; this offer will have new ICE and
>>    possibly new DTLS-SRTP certificate fingerprints (since the old ICE
>>    and SRTP state has been lost).
>>
>> I think this may require quite a bit more details. A forward pointer to
>> where you will specify that?
>>
>
See above

>
>> 8. Section 3.6:
>>    [OPEN ISSUE: EKR proposed an alternative rehydration approach where
>>    the actual internal PeerConnection object in the browser was kept
>>    alive for some time after the web page was killed and provided some
>>    way for a new page to acquire the old PeerConnection object.]
>>
>> Can't we declare this open issue closed? I have seen no details on the
>> alternative proposal. Thus, lets work with the one that is present.
>>
>> See above


>  9. Section 4.1.1:
>> There is no discussion of how the bundle policies relate to the Data
>> Channels? Are they included or do they have a specific set of policies.
>> Please extend this to discuss this.
>>
>
Will do.

>
>> 10. Section 4.1.3:
>>
>> This text doesn't appear to contain a requirement on that one MUST have
>> called setRemote prior to being able to call it.
>>
>
Will fix.

>
>> 11. Section 4.1.4.1:
>>    While this could also be done with an inactive "pranswer", followed
>>    by a sendrecv "answer", the initial "pranswer" leaves the offer-
>>    answer exchange open, which means the caller cannot send an updated
>>    offer during this time.
>>
>> Should one mention that also the callee can't send an offer without
>> doing a "final" answer?
>>
>
The callee could still send more pranswers, so it is not as restricted in
this state as the caller. However, it can't add streams at this point, so
there are some limitations.

If you think this is unclear I could change the text to say "neither side
can send an updated offer during this time"

>
>> 12. Section 4.1.4.1
>>
>> By the time the human has accepted the call and
>>    triggered the new offer, it is likely that the ICE and DTLS
>>    handshaking for all the channels will already be set up.
>>
>> Wouldn't the handshaking have "finished" rather than "set up"?
>>
>
Yes.

>
>> 13. Section 4.1.4.2:
>>    Rollback can only be used to cancel proposed changes; there is no
>>    support for rolling back from a stable state to a previous stable
>>    state.
>>
>> What is meant with "proposed changes"? From my side it is not clear how
>> much you can do after having performed a SetLocal or SetRemote. I think
>> these needs to be treated separately.
>
>
>> For an offerer, they can createOffer, SetLocal, send to peer, receive
>> reject and then undo the SetLocal. That is straight forward, and if they
>> done the SetRemote, they would be consider in a new Stable state and
>> can't roll back anymore?
>>
>> For the answering side, it receive an offer, does SetRemote,
>> CreateAnswer, determines that it doesn't like it, the can undo the
>> SetRemote and send a reject message to offerer.
>>
>> But what about this case? it receive an offer, does SetRemote,
>> CreateAnswer, SetLocal and sends the message, then receive a reject from
>> the peer, because it didn't like the answer? Are the answerer then in a
>> stable state that can't be unrolled, or?
>>
>
If you don't like the answer, I think it's game over. The answerer has
already started acting on the new local/remote description at this point.

IOW, "proposed changes" really only means what has been offered.

>
>> 14. Section 4.1.5:
>>    This API indirectly controls the candidate gathering process.  When a
>>    local description is supplied, and the number of transports currently
>>    in use does not match the number of transports needed by the local
>>    description, the PeerConnection will create transports as needed and
>>    begin gathering candidates for them.
>>
>> Another part of an earlier question regarding what triggers ICE gatherin=
g?
>>
>
See answer above.

>
>> 15. Section 4.1.7:
>>    TODO: Do we need to expose accessors for both the current and
>>    proposed local description?
>>
>> This needs to be resolved, I don't know if I can answer this question.
>>
>
We agreed in the interim that we would expose them.

>
>> 16. Section 4.1.9:
>>
>> Should this section discuss that it might be browser that has overriding
>> controls regarding if other candidates then relay ones may be provided,
>> i.e. privacy mode?
>>
>
 Probably. The use of host candidates is something that might be forbidden
by local policy. I think the right thing to do would be to introduce the
concept of iceTransports in section 3.4 (ICE), and then refer to it here in
updateIce (now called setConfiguration), as well as the ctor section, where
the initial policy can be set.

>
>> 17. Section 4.1.9:
>>    Regardless of the configuration, the gathering process collects all
>>    available candidates, but excluded candidates will not be surfaced in
>>    onicecandidate callback or used for connectivity checks.
>>
>> Doesn't this result in both resource consumption and that the configured
>> TURN server will learn the external IP address and port for the client?
>>
>
I'm not sure what the exact concern is here. Gathering candidates, even
those that aren't desired, allows the withheld candidates to be released as
soon as the app asks for them.

>
>> 18. Section 5.2.1:
>>
>> Structure in relation between RTP media and DATA channel. Maybe you
>> should try to find a structure that makes the difference between the two
>> types of transport more clear, and what applies to the perspective.
>>
>> Agree this could be improved.


>  19. Section 5.2.1:
>>
>>
>>    o  For the current default candidate, a "c=3D" line, as specific in
>>       [RFC5245], Section 4.3., paragraph 6.  If no candidates have been
>>       gathered yet, the default candidate should be set to the 'null'
>>       value defined in [I-D.ietf-mmusic-trickle-ice], Section 5.1.
>>
>> What about host candidates?
>>
>
See above. This text has been clarified in the latest non-public revision.

>
>> 20. Section 5.2.1:
>> The CNAME must be generated
>>       in accordance with draft-rescorla-random-cname-00.  [OPEN ISSUE:
>>       How are CNAMEs specified for MSTs?  Are they randomly generated
>>       for each MediaStream?  If so, can two MediaStreams be synced?]
>>
>> I think you should point to the RTP usage for this particular piece.
>> Because we do have text there both for the CNAME generation and how
>> CNAMES among MSTs and between PeerConnection. Likely to be changed soon
>> based on the mailing list discussion.
>>
>
Agreed.

>
>> 21. Section 5.2.1:
>>    o  [OPEN ISSUE: Handling of a=3Dimageattr]
>>
>> Yes, I think imageattr should be supported to give guidance on suitable
>> resolutions that an endpoint like to receive for a particular MST.
>>
>> Agreed. Harald is working on changes to address this.


>  22. Section 5.2.1:
>>    Once all m=3D sections have been generated, a session-level "a=3Dgrou=
p"
>>    attribute MUST be added as specified in [RFC5888].  This attribute
>>    MUST have semantics "BUNDLE", and MUST include the mid identifiers of
>>    each m=3D section.
>>
>> Isn't this overriding BUNDLE as currently specified, forcing everything
>> into a single bundle group. Thus, preventing the bundle policies to work=
.
>>
>> The BUNDLE policies only affect what is bundle-only. Between two
browsers, a single BUNDLE will always be used.


>  23. Section 5.2.2:
>>
>>    o  If any MediaStreamTracks have been added, and there exist m=3D
>>       sections of the appropriate media type with no associated
>>       MediaStreamTracks
>>
>> Isnt' the last a "MediaStreamTrack" rather than plural ones?
>>
>> Agree this could be clarified.


>  24. 5.2.3.3.  VoiceActivityDetection
>>
>>    If the "VoiceActivityDetection" constraint is specified, with a value
>>    of "true", the offer MUST indicate support for silence suppression by
>>    including comfort noise ("CN") codecs for each supported clock rate,
>>    as specified in [RFC3389], Section 5.1.
>>
>> As currently specified this text requires the usage of a comfort noise
>> codec, currently there is no such on required. I think the "MUST" needs
>> to be rewritten.
>>
>
I think this needs to be specified as a MUST in RTP usage then. Otherwise,
this setting won't work in some browsers, leading to nondeterministic
behavior.

This should also be updated to set usedtx=3D1 for Opus.

>
>> 25. Section 5.3.1:
>>                                                                   Note
>>       that for simplicity, the answerer MAY use different payload types
>>       for codecs than the offerer, as it is not prohibited by
>>       Section 6.1.
>>
>> I react to the word simplicity here. I never consider usage of different
>> PTs per direction anything simple. But, yes it is allowed by RFC 3264.
>> But I kind of protest to encourage the usage.
>>
>>
Clearly this needs more explanation. The basic issue is the fact that using
the same payload types gets really hairy when using RTX and when bundling.
We can add an example of the problem.


 26. Section 5.3.2 etc.
>>
>> When will we see a first draft of the text in the sections:
>> 5.3.2, 5.3.3., 5.4, 5.5, 5.6, 5.7?
>>
>> I really like to see it prior to the Interim meeting.
>>
>
Work in progress, will be in jsep-07 before Toronto.


>
>> 27. Section 6:
>>    Note: This section is still very early and is likely to significantly
>>    change as we get a better understanding of a) the use cases for this
>>    b) the implications at the protocol level c) feedback from
>>    implementors on what they can do.
>>
>> Shouldn't this mention W3C discussion also?
>>
>
Sure.


>
>> 28. Section 7 - Security Considerations:
>>
>> First of all there are no references to the Security Considerations or
>> architecture document.
>>
>> Secondly, I would like to have some security discussion on significant
>> issues with JSEP itself. Resource attacks on the local endpoint for
>> example?
>>
>
We will add a open issue for this.

>
>> 29. section 10
>>
>> There are a couple of draft references that are either published RFCs or
>> at least WG versions. Please update the reference list.
>>
>> OK.


>  30. Section 10.2:
>> Looking at the spec, I think the following references are normative:
>>
>>    [I-D.ietf-mmusic-trickle-ice]
>>               Ivov, E., Rescorla, E., and J. Uberti, "Trickle ICE:
>>               Incremental Provisioning of Candidates for the Interactive
>>               Connectivity Establishment (ICE) Protocol", draft-ietf-
>>               mmusic-trickle-ice-00 (work in progress), March 2013.
>>
>>    [I-D.ietf-rtcweb-data-protocol]
>>               Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel
>>               Protocol", draft-ietf-rtcweb-data-protocol-04 (work in
>>               progress), February 2013.
>>
>>
>> OK.


>  Cheers
>>
>> Magnus Westerlund
>>
>> ----------------------------------------------------------------------
>> Services, Media and Network features, Ericsson Research EAB/TXM
>> ----------------------------------------------------------------------
>> Ericsson AB                 | Phone  +46 10 7148287
>> F=C3=A4r=C3=B6gatan 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
>>
>
>

--001a11339e2ed365af04fbe955f7
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Sorry for the super long latency on this. We&#39;ve create=
d issues in the GH tracker for the points identified below. Specific commen=
ts inline.<div><a href=3D"https://github.com/rtcweb-wg/jsep/issues/51">http=
s://github.com/rtcweb-wg/jsep/issues/51</a></div>

<div><a href=3D"https://github.com/rtcweb-wg/jsep/issues/52">https://github=
.com/rtcweb-wg/jsep/issues/52</a></div><div><a href=3D"https://github.com/r=
tcweb-wg/jsep/issues/53">https://github.com/rtcweb-wg/jsep/issues/53</a></d=
iv>

<div><a href=3D"https://github.com/rtcweb-wg/jsep/issues/54">https://github=
.com/rtcweb-wg/jsep/issues/54</a><br><div class=3D"gmail_extra"><br><div cl=
ass=3D"gmail_quote">On Fri, Mar 14, 2014 at 9:48 AM, Justin Uberti <span di=
r=3D"ltr">&lt;<a href=3D"mailto:juberti@google.com" target=3D"_blank">juber=
ti@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;p=
adding-left:1ex"><div dir=3D"ltr">Thanks for the detailed review. We will w=
ork through these comments over the next few days.</div>

<div>
<div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Fri, =
Mar 14, 2014 at 6:05 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:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">Hi,<br>
<br>
I did review the JSEP -06 prior to the meeting. Now I have had time to<br>
write-up my comments.<br>
<br>
1) Section 1):<br>
<br>
[W3C.WD-webrtc-20111027]<br>
This is very old version of the API, please update to the currently<br>
relevant version.<br></blockquote></div></div></div></div></blockquote><div=
><br></div><div>Will do.=C2=A0</div><blockquote class=3D"gmail_quote" style=
=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(20=
4,204,204);border-left-style:solid;padding-left:1ex">

<div>
<div><div class=3D"gmail_extra"><div class=3D"gmail_quote"><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;b=
order-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"=
>
<br>
2) Section <a href=3D"http://3.4.1.1" target=3D"_blank">3.4.1.1</a>:<br>
The m-line index is a zero-based index, referring to the Nth m-line in<br>
the SDP.<br>
<br>
a) I think the above is a bit confusing. As a zero-based index would<br>
mean that the first m=3D section would have index 0. Thus, the referring<br=
>
to the Nth m-line in the SDP is unclear. Is N, the provided index, in<br>
which case it would be the N+1th m-line. Please rewrite.<br></blockquote></=
div></div></div></div></blockquote><div><br></div><div>Will clarify.</div><=
div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0=
px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-le=
ft-style:solid;padding-left:1ex">



<div><div><div class=3D"gmail_extra"><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-left-style:solid;padding-left=
:1ex">


<br>
b) I also wonder if one has to be clearer on what &quot;the SDP&quot; is. I=
 assume<br>
it is the one that has been generated by the agent, rather than the<br>
latest sent or received?<br></blockquote></div></div></div></div></blockquo=
te><div><br></div><div>It can be for either the local generated SDP, in the=
 case where the candidate is gotten from the onicecandidate callback, or th=
e remote SDP, in the case where the candidates are trickled from the remote=
 side. We&#39;ll add some text to make this clear.=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"><div><div><div class=3D"gmail_extra"><div class=3D"gmail_q=
uote">

<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>
3) Section <a href=3D"http://3.4.1.1" target=3D"_blank">3.4.1.1</a>:<br>
&quot;The MID uses the &quot;media stream identification&quot;, as defined =
in<br>
=C2=A0 =C2=A0[RFC5888] , to identify the m-line.&quot;<br>
<br>
a spurious space after ref.<br></blockquote></div></div></div></div></block=
quote><div><br></div><div>OK.=C2=A0</div><blockquote class=3D"gmail_quote" =
style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:r=
gb(204,204,204);border-left-style:solid;padding-left:1ex">

<div>
<div><div class=3D"gmail_extra"><div class=3D"gmail_quote"><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;b=
order-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"=
>
<br>
4) JSEP applications typically inform the browser to begin ICE gathering<br=
>
=C2=A0 =C2=A0via the information supplied to setLocalDescription, as this i=
s where<br>
=C2=A0 =C2=A0the app specifies the number of media streams to gather candid=
ates<br>
=C2=A0 =C2=A0for.<br>
<br>
I find the statement that ICE candidate gathering starts first with<br>
setLocalDescription. Doesn&#39;t the created offer already contain local<br=
>
candidates, thus one must have already gathered some candidates? To me<br>
it appears reasonable to start gathering earlier, rather than later.<br>
Thus, as soon as one create the PeerConnection object and have added any<br=
>
MST to it or a data channel, one can start gathering.<br></blockquote></div=
></div></div></div></blockquote><div><br></div><div>The initial offer comes=
 out as soon as possible, so it contains no candidates.</div><div><br>



</div><div>setLocalDescription is the first time the implementation knows e=
xactly how many m=3D lines it needs to gather for, so gathering can&#39;t d=
efinitively start until this point. Candidates may be pre-gathered using th=
e candidate pool, but they aren&#39;t released via onicecandidate until set=
LD.</div>



<div>=C2=A0<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);bord=
er-left-style:solid;padding-left:1ex"><div><div><div class=3D"gmail_extra">=
<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-left-style:solid;p=
adding-left:1ex">


<br>
5) Section 3.5.2:<br>
=C2=A0 =C2=A0In the parallel forking case where the Javascript application =
wishes<br>
=C2=A0 =C2=A0to simultaneously exchange media with multiple peers, the flow=
 is<br>
=C2=A0 =C2=A0slightly more complex, but the Javascript application can foll=
ow the<br>
=C2=A0 =C2=A0strategy that [RFC3960] describes using UPDATE. =C2=A0(It is w=
orth noting<br>
=C2=A0 =C2=A0that use cases where this is the desired behavior are very unu=
sual.)<br>
<br>
I mostly react to the last sentence within the paragraph. I think the<br>
poster child for semi-parallel forking was the MMORG that like to<br>
establish PCs with anyone that is coming close in the virtual world.<br>
Thus either needs to have a number of PCs on standby or have just one,<br>
but treat that as parallel forking. And if you like to discourage the<br>
later, then I think you might need to at least inform about the<br>
possibility to have stand-by PCs.<br></blockquote></div></div></div></div><=
/blockquote><div><br></div><div>Is the MMORG case described in the use-case=
 doc? If so, I agree the parenthesized text should be changed.</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"><div><div><div class=3D"gmail_extra"><div clas=
s=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-left-style:solid;p=
adding-left:1ex">


<br>
6. Section 3.6:<br>
<br>
=C2=A0 =C2=A0In the event that the local application state is reinitialized=
,<br>
=C2=A0 =C2=A0either due to a user reload of the page, or a decision within =
the<br>
=C2=A0 =C2=A0application to reload itself (perhaps to update to a new versi=
on), it<br>
=C2=A0 =C2=A0is possible to keep an existing session alive, via a process c=
alled<br>
=C2=A0 =C2=A0&quot;rehydration&quot;. =C2=A0The explicit goal of rehydratio=
n is to carry out this<br>
=C2=A0 =C2=A0session resumption with no interaction with the remote side ot=
her<br>
=C2=A0 =C2=A0than normal call signaling messages.<br>
<br>
I think this paragraph makes two erronous statements:<br>
a) &quot;keep an existing session alive&quot;<br>
I think &quot;resume&quot; an existing session would be more correct. Becau=
se it<br>
would be broken when the reload happens.<br>
<br>
b) &quot;with no interaction with the remote side&quot;<br>
My understanding is that due to DTLS and key handling a DTLS<br>
renegotiation will be required with the peer. Isn&#39;t also a signalling<b=
r>
exchange likely to be required due to change of local DTLS cert as well<br>
as ICE candidates as parameters? Any other parameters that will be<br>
required to exchanges when rehydrating?<br>
<br></blockquote></div></div></div></div></blockquote><div>This section has=
 been removed, as nobody was working on this and it wasn&#39;t clear exactl=
y what it ought to do.</div><div>=C2=A0</div><blockquote class=3D"gmail_quo=
te" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-col=
or:rgb(204,204,204);border-left-style:solid;padding-left:1ex">



<div><div><div class=3D"gmail_extra"><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-left-style:solid;padding-left=
:1ex">


7. Section 3.6:<br>
Next, a new offer is<br>
=C2=A0 =C2=A0generated by the new PeerConnection; this offer will have new =
ICE and<br>
=C2=A0 =C2=A0possibly new DTLS-SRTP certificate fingerprints (since the old=
 ICE<br>
=C2=A0 =C2=A0and SRTP state has been lost).<br>
<br>
I think this may require quite a bit more details. A forward pointer to<br>
where you will specify that?<br></blockquote></div></div></div></div></bloc=
kquote><div><br></div><div>See above=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;padding-left:1ex">



<div><div><div class=3D"gmail_extra"><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-left-style:solid;padding-left=
:1ex">


<br>
8. Section 3.6:<br>
=C2=A0 =C2=A0[OPEN ISSUE: EKR proposed an alternative rehydration approach =
where<br>
=C2=A0 =C2=A0the actual internal PeerConnection object in the browser was k=
ept<br>
=C2=A0 =C2=A0alive for some time after the web page was killed and provided=
 some<br>
=C2=A0 =C2=A0way for a new page to acquire the old PeerConnection object.]<=
br>
<br>
Can&#39;t we declare this open issue closed? I have seen no details on the<=
br>
alternative proposal. Thus, lets work with the one that is present.<br>
<br></blockquote></div></div></div></div></blockquote><div>See above</div><=
div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0=
px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-le=
ft-style:solid;padding-left:1ex">

<div><div>
<div class=3D"gmail_extra"><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-left-style:solid;padding-left:1ex">
9. Section 4.1.1:<br>
There is no discussion of how the bundle policies relate to the Data<br>
Channels? Are they included or do they have a specific set of policies.<br>
Please extend this to discuss this.<br></blockquote></div></div></div></div=
></blockquote><div><br></div><div>Will do.=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;padding-left:1ex">



<div><div><div class=3D"gmail_extra"><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-left-style:solid;padding-left=
:1ex">


<br>
10. Section 4.1.3:<br>
<br>
This text doesn&#39;t appear to contain a requirement on that one MUST have=
<br>
called setRemote prior to being able to call it.<br></blockquote></div></di=
v></div></div></blockquote><div><br></div><div>Will fix.=C2=A0</div><blockq=
uote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-wi=
dth:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-=
left:1ex">



<div><div><div class=3D"gmail_extra"><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-left-style:solid;padding-left=
:1ex">


<br>
11. Section <a href=3D"http://4.1.4.1" target=3D"_blank">4.1.4.1</a>:<br>
=C2=A0 =C2=A0While this could also be done with an inactive &quot;pranswer&=
quot;, followed<br>
=C2=A0 =C2=A0by a sendrecv &quot;answer&quot;, the initial &quot;pranswer&q=
uot; leaves the offer-<br>
=C2=A0 =C2=A0answer exchange open, which means the caller cannot send an up=
dated<br>
=C2=A0 =C2=A0offer during this time.<br>
<br>
Should one mention that also the callee can&#39;t send an offer without<br>
doing a &quot;final&quot; answer?<br></blockquote></div></div></div></div><=
/blockquote><div><br></div><div>The callee could still send more pranswers,=
 so it is not as restricted in this state as the caller. However, it can&#3=
9;t add streams at this point, so there are some limitations.</div>



<div><br></div><div>If you think this is unclear I could change the text to=
 say &quot;neither side can send an updated offer during this time&quot;=C2=
=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8e=
x;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-styl=
e:solid;padding-left:1ex">



<div><div><div class=3D"gmail_extra"><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-left-style:solid;padding-left=
:1ex">


<br>
12. Section 4.1.4.1<br>
<br>
By the time the human has accepted the call and<br>
=C2=A0 =C2=A0triggered the new offer, it is likely that the ICE and DTLS<br=
>
=C2=A0 =C2=A0handshaking for all the channels will already be set up.<br>
<br>
Wouldn&#39;t the handshaking have &quot;finished&quot; rather than &quot;se=
t up&quot;?<br></blockquote></div></div></div></div></blockquote><div><br><=
/div><div>Yes.=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;padding-left:1ex">



<div><div><div class=3D"gmail_extra"><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-left-style:solid;padding-left=
:1ex">


<br>
13. Section <a href=3D"http://4.1.4.2" target=3D"_blank">4.1.4.2</a>:<br>
=C2=A0 =C2=A0Rollback can only be used to cancel proposed changes; there is=
 no<br>
=C2=A0 =C2=A0support for rolling back from a stable state to a previous sta=
ble<br>
=C2=A0 =C2=A0state.<br>
<br>
What is meant with &quot;proposed changes&quot;? From my side it is not cle=
ar how<br>
much you can do after having performed a SetLocal or SetRemote. I think<br>
these needs to be treated separately.<span style=3D"color:rgb(34,34,34)">=
=C2=A0</span></blockquote></div></div></div></div></blockquote><blockquote =
class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1=
px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:=
1ex">



<div><div><div class=3D"gmail_extra"><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-left-style:solid;padding-left=
:1ex">


<br>
For an offerer, they can createOffer, SetLocal, send to peer, receive<br>
reject and then undo the SetLocal. That is straight forward, and if they<br=
>
done the SetRemote, they would be consider in a new Stable state and<br>
can&#39;t roll back anymore?<br>
<br>
For the answering side, it receive an offer, does SetRemote,<br>
CreateAnswer, determines that it doesn&#39;t like it, the can undo the<br>
SetRemote and send a reject message to offerer.<br>
<br>
But what about this case? it receive an offer, does SetRemote,<br>
CreateAnswer, SetLocal and sends the message, then receive a reject from<br=
>
the peer, because it didn&#39;t like the answer? Are the answerer then in a=
<br>
stable state that can&#39;t be unrolled, or?<br></blockquote></div></div></=
div></div></blockquote><div><br></div><div>If you don&#39;t like the answer=
, I think it&#39;s game over. The answerer has already started acting on th=
e new local/remote description at this point.=C2=A0</div>



<div><br></div><div>IOW, &quot;proposed changes&quot; really only means wha=
t has been offered.</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">

<div>
<div><div class=3D"gmail_extra"><div class=3D"gmail_quote"><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;b=
order-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"=
>
<br>
14. Section 4.1.5:<br>
=C2=A0 =C2=A0This API indirectly controls the candidate gathering process. =
=C2=A0When a<br>
=C2=A0 =C2=A0local description is supplied, and the number of transports cu=
rrently<br>
=C2=A0 =C2=A0in use does not match the number of transports needed by the l=
ocal<br>
=C2=A0 =C2=A0description, the PeerConnection will create transports as need=
ed and<br>
=C2=A0 =C2=A0begin gathering candidates for them.<br>
<br>
Another part of an earlier question regarding what triggers ICE gathering?<=
br></blockquote></div></div></div></div></blockquote><div><br></div><div>Se=
e answer above.=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margi=
n: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 class=3D"gmail_extra"><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-left-style:solid;padding-left=
:1ex">


<br>
15. Section 4.1.7:<br>
=C2=A0 =C2=A0TODO: Do we need to expose accessors for both the current and<=
br>
=C2=A0 =C2=A0proposed local description?<br>
<br>
This needs to be resolved, I don&#39;t know if I can answer this question.<=
br></blockquote></div></div></div></div></blockquote><div><br></div><div>We=
 agreed in the interim that we would expose them.=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">



<div><div><div class=3D"gmail_extra"><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-left-style:solid;padding-left=
:1ex">


<br>
16. Section 4.1.9:<br>
<br>
Should this section discuss that it might be browser that has overriding<br=
>
controls regarding if other candidates then relay ones may be provided,<br>
i.e. privacy mode?<br></blockquote></div></div></div></div></blockquote><di=
v><br></div><div>=C2=A0Probably. The use of host candidates is something th=
at might be forbidden by local policy. I think the right thing to do would =
be to introduce the concept of iceTransports in section 3.4 (ICE), and then=
 refer to it here in updateIce (now called setConfiguration), as well as th=
e ctor section, where the initial policy can be set.</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"><div><div><div class=3D"gmail_extra"><div class=3D"gmail_q=
uote">

<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>
17. Section 4.1.9:<br>
=C2=A0 =C2=A0Regardless of the configuration, the gathering process collect=
s all<br>
=C2=A0 =C2=A0available candidates, but excluded candidates will not be surf=
aced in<br>
=C2=A0 =C2=A0onicecandidate callback or used for connectivity checks.<br>
<br>
Doesn&#39;t this result in both resource consumption and that the configure=
d<br>
TURN server will learn the external IP address and port for the client?<br>=
</blockquote></div></div></div></div></blockquote><div><br></div><div>I&#39=
;m not sure what the exact concern is here. Gathering candidates, even thos=
e that aren&#39;t desired, allows the withheld candidates to be released as=
 soon as the app asks for them.</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"><div><div><div class=3D"gmail_extra"><div class=3D"gmail_q=
uote">

<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>
18. Section 5.2.1:<br>
<br>
Structure in relation between RTP media and DATA channel. Maybe you<br>
should try to find a structure that makes the difference between the two<br=
>
types of transport more clear, and what applies to the perspective.<br>
<br></blockquote></div></div></div></div></blockquote><div>Agree this could=
 be improved.</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(20=
4,204,204);border-left-style:solid;padding-left:1ex">

<div><div><div class=3D"gmail_extra">
<div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margi=
n:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204=
);border-left-style:solid;padding-left:1ex">
19. Section 5.2.1:<br>
<br>
<br>
=C2=A0 =C2=A0o =C2=A0For the current default candidate, a &quot;c=3D&quot; =
line, as specific in<br>
=C2=A0 =C2=A0 =C2=A0 [RFC5245], Section 4.3., paragraph 6. =C2=A0If no cand=
idates have been<br>
=C2=A0 =C2=A0 =C2=A0 gathered yet, the default candidate should be set to t=
he &#39;null&#39;<br>
=C2=A0 =C2=A0 =C2=A0 value defined in [I-D.ietf-mmusic-trickle-ice], Sectio=
n 5.1.<br>
<br>
What about host candidates?<br></blockquote></div></div></div></div></block=
quote><div><br></div><div>See above. This text has been clarified in the la=
test non-public revision.=C2=A0</div><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">


<div><div><div class=3D"gmail_extra"><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-left-style:solid;padding-left=
:1ex">


<br>
20. Section 5.2.1:<br>
The CNAME must be generated<br>
=C2=A0 =C2=A0 =C2=A0 in accordance with draft-rescorla-random-cname-00. =C2=
=A0[OPEN ISSUE:<br>
=C2=A0 =C2=A0 =C2=A0 How are CNAMEs specified for MSTs? =C2=A0Are they rand=
omly generated<br>
=C2=A0 =C2=A0 =C2=A0 for each MediaStream? =C2=A0If so, can two MediaStream=
s be synced?]<br>
<br>
I think you should point to the RTP usage for this particular piece.<br>
Because we do have text there both for the CNAME generation and how<br>
CNAMES among MSTs and between PeerConnection. Likely to be changed soon<br>
based on the mailing list discussion.<br></blockquote></div></div></div></d=
iv></blockquote><div><br></div><div>Agreed.=C2=A0</div><blockquote class=3D=
"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;borde=
r-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">


<div><div><div class=3D"gmail_extra"><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-left-style:solid;padding-left=
:1ex">


<br>
21. Section 5.2.1:<br>
=C2=A0 =C2=A0o =C2=A0[OPEN ISSUE: Handling of a=3Dimageattr]<br>
<br>
Yes, I think imageattr should be supported to give guidance on suitable<br>
resolutions that an endpoint like to receive for a particular MST.<br>
<br></blockquote></div></div></div></div></blockquote><div>Agreed. Harald i=
s working on changes to address this.</div><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=
">


<div><div><div class=3D"gmail_extra"><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-left-style:solid;padding-left=
:1ex">


22. Section 5.2.1:<br>
=C2=A0 =C2=A0Once all m=3D sections have been generated, a session-level &q=
uot;a=3Dgroup&quot;<br>
=C2=A0 =C2=A0attribute MUST be added as specified in [RFC5888]. =C2=A0This =
attribute<br>
=C2=A0 =C2=A0MUST have semantics &quot;BUNDLE&quot;, and MUST include the m=
id identifiers of<br>
=C2=A0 =C2=A0each m=3D section.<br>
<br>
Isn&#39;t this overriding BUNDLE as currently specified, forcing everything=
<br>
into a single bundle group. Thus, preventing the bundle policies to work.<b=
r>
<br></blockquote></div></div></div></div></blockquote><div>The BUNDLE polic=
ies only affect what is bundle-only. Between two browsers, a single BUNDLE =
will always be used.</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-style:solid;padding-left:1ex">


<div><div><div class=3D"gmail_extra"><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-left-style:solid;padding-left=
:1ex">


23. Section 5.2.2:<br>
<br>
=C2=A0 =C2=A0o =C2=A0If any MediaStreamTracks have been added, and there ex=
ist m=3D<br>
=C2=A0 =C2=A0 =C2=A0 sections of the appropriate media type with no associa=
ted<br>
=C2=A0 =C2=A0 =C2=A0 MediaStreamTracks<br>
<br>
Isnt&#39; the last a &quot;MediaStreamTrack&quot; rather than plural ones?<=
br>
<br></blockquote></div></div></div></div></blockquote><div>Agree this could=
 be clarified.</div><div>=C2=A0</div><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">

<div><div><div class=3D"gmail_extra">
<div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margi=
n:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204=
);border-left-style:solid;padding-left:1ex">
24. 5.2.3.3. =C2=A0VoiceActivityDetection<br>
<br>
=C2=A0 =C2=A0If the &quot;VoiceActivityDetection&quot; constraint is specif=
ied, with a value<br>
=C2=A0 =C2=A0of &quot;true&quot;, the offer MUST indicate support for silen=
ce suppression by<br>
=C2=A0 =C2=A0including comfort noise (&quot;CN&quot;) codecs for each suppo=
rted clock rate,<br>
=C2=A0 =C2=A0as specified in [RFC3389], Section 5.1.<br>
<br>
As currently specified this text requires the usage of a comfort noise<br>
codec, currently there is no such on required. I think the &quot;MUST&quot;=
 needs<br>
to be rewritten.<br></blockquote></div></div></div></div></blockquote><div>=
<br></div><div>I think this needs to be specified as a MUST in RTP usage th=
en. Otherwise, this setting won&#39;t work in some browsers, leading to non=
deterministic behavior.=C2=A0</div>


<div><br></div><div>This should also be updated to set usedtx=3D1 for Opus.=
</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;b=
order-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:s=
olid;padding-left:1ex">

<div><div><div class=3D"gmail_extra"><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-left-style:solid;p=
adding-left:1ex">
<br>
25. Section 5.3.1:<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 Note<br>
=C2=A0 =C2=A0 =C2=A0 that for simplicity, the answerer MAY use different pa=
yload types<br>
=C2=A0 =C2=A0 =C2=A0 for codecs than the offerer, as it is not prohibited b=
y<br>
=C2=A0 =C2=A0 =C2=A0 Section 6.1.<br>
<br>
I react to the word simplicity here. I never consider usage of different<br=
>
PTs per direction anything simple. But, yes it is allowed by RFC 3264.<br>
But I kind of protest to encourage the usage.<br>
<br></blockquote></div></div></div></div></blockquote><div><br></div><div>C=
learly this needs more explanation. The basic issue is the fact that using =
the same payload types gets really hairy when using RTX and when bundling. =
We can add an example of the problem.</div>

<div><br></div><div><br></div><blockquote class=3D"gmail_quote" style=3D"ma=
rgin: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 class=3D"gmail_extra"><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-left-style:solid;padding-left=
:1ex">


26. Section 5.3.2 etc.<br>
<br>
When will we see a first draft of the text in the sections:<br>
5.3.2, 5.3.3., 5.4, 5.5, 5.6, 5.7?<br>
<br>
I really like to see it prior to the Interim meeting.<br></blockquote></div=
></div></div></div></blockquote><div><br></div><div>Work in progress, will =
be in jsep-07 before Toronto.</div><div>=C2=A0</div><blockquote class=3D"gm=
ail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-l=
eft-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">

<div><div><div class=3D"gmail_extra"><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-left-style:solid;padding-left=
:1ex">


<br>
27. Section 6:<br>
=C2=A0 =C2=A0Note: This section is still very early and is likely to signif=
icantly<br>
=C2=A0 =C2=A0change as we get a better understanding of a) the use cases fo=
r this<br>
=C2=A0 =C2=A0b) the implications at the protocol level c) feedback from<br>
=C2=A0 =C2=A0implementors on what they can do.<br>
<br>
Shouldn&#39;t this mention W3C discussion also?<br></blockquote></div></div=
></div></div></blockquote><div><br></div><div>Sure.</div><div>=C2=A0</div><=
blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-l=
eft-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;pa=
dding-left:1ex">

<div><div><div class=3D"gmail_extra"><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-left-style:solid;padding-left=
:1ex">


<br>
28. Section 7 - Security Considerations:<br>
<br>
First of all there are no references to the Security Considerations or<br>
architecture document.<br>
<br>
Secondly, I would like to have some security discussion on significant<br>
issues with JSEP itself. Resource attacks on the local endpoint for<br>
example?<br></blockquote></div></div></div></div></blockquote><div><br></di=
v><div>We will add a open issue for this.=C2=A0</div><blockquote class=3D"g=
mail_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 class=3D"gmail_extra"><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-left-style:solid;padding-left=
:1ex">


<br>
29. section 10<br>
<br>
There are a couple of draft references that are either published RFCs or<br=
>
at least WG versions. Please update the reference list.<br>
<br></blockquote></div></div></div></div></blockquote><div>OK.</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">

<div><div><div class=3D"gmail_extra"><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-left-style:solid;padding-left=
:1ex">


30. Section 10.2:<br>
Looking at the spec, I think the following references are normative:<br>
<br>
=C2=A0 =C2=A0[I-D.ietf-mmusic-trickle-ice]<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Ivov, E., Rescorla, E., an=
d J. Uberti, &quot;Trickle ICE:<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Incremental Provisioning o=
f Candidates for the Interactive<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Connectivity Establishment=
 (ICE) Protocol&quot;, draft-ietf-<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 mmusic-trickle-ice-00 (wor=
k in progress), March 2013.<br>
<br>
=C2=A0 =C2=A0[I-D.ietf-rtcweb-data-protocol]<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Jesup, R., Loreto, S., and=
 M. Tuexen, &quot;WebRTC Data Channel<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Protocol&quot;, draft-ietf=
-rtcweb-data-protocol-04 (work in<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 progress), February 2013.<=
br>
<br>
<br></blockquote></div></div></div></div></blockquote><div>OK.</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">

<div><div><div class=3D"gmail_extra"><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-left-style:solid;padding-left=
:1ex">


Cheers<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Services, Media and Network features, Ericsson Research EAB/TXM<br>
----------------------------------------------------------------------<br>
Ericsson AB =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | Phone=
 =C2=A0<a href=3D"tel:%2B46%2010%207148287" value=3D"+46107148287" target=
=3D"_blank">+46 10 7148287</a><br>
F=C3=A4r=C3=B6gatan 6 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 | Mobile <a href=3D"tel:%2B46%2073%200949079" value=3D"+46730949079" ta=
rget=3D"_blank">+46 73 0949079</a><br>
SE-164 80 Stockholm, Sweden | mailto: <a href=3D"mailto:magnus.westerlund@e=
ricsson.com" target=3D"_blank">magnus.westerlund@ericsson.com</a><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></div><br></div>
</div></div></blockquote></div><br></div></div></div>

--001a11339e2ed365af04fbe955f7--


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Subject: [rtcweb] draft-ietf-rtcweb-overview-09  Review
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA512

Here is my review of the Overview: Real Time Protocols for
Browser-based Applications draft-ietf-rtcweb-overview-09. Please take
into consideration my suggestions.

Thanks
- -Colton



Section 3 Paragraph 1

I am a native English speaker and I had to look up the definition of
"envisage". Drafts should be written in the plainest language
possible, to make it easier for everyone to understand.

"The model of real-time support for browser-based applications does
   not envisage that the browser will contain all the functions that
   need to be performed in order to have a function such as a telephone
   or a video conferencing unit;"

I suggest replacing "envisage" with one of the following
	- suppose
	- assume



Section 3 Paragraph 5

"Existing protocols (for example SIP or XMPP) could be used between
servers..."

In case the reader is not familiar with SIP or XMPP, the draft should
link to these protocols. Just like it does with other protocols.
Especially since the next paragraphs provide an example of SIP and
XMPP. If the reader does not understand how SIP or XMPP work they
could get confused. Providing links gives them a resource to
learn how those protocols work.


SIP RFC 3261
http://www.ietf.org/rfc/rfc3261.txt

XMPP RFC 3920
http://www.ietf.org/rfc/rfc3920.txt




Spelling error in the Title

Title should say "Browser-based" rather than "Brower-based"








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Thank you for the review!

It is very timely, since I'm looking to revise this now based on other
reviews.
I will take care of the stuff you point out at the same time - thanks
for making such specific suggestions!


On 06/16/2014 06:00 PM, Colton Shields wrote:
> Here is my review of the Overview: Real Time Protocols for
> Browser-based Applications draft-ietf-rtcweb-overview-09. Please take
> into consideration my suggestions.
>
> Thanks
> -Colton
>
>
>
> Section 3 Paragraph 1
>
> I am a native English speaker and I had to look up the definition of
> "envisage". Drafts should be written in the plainest language
> possible, to make it easier for everyone to understand.
>
> "The model of real-time support for browser-based applications does
>    not envisage that the browser will contain all the functions that
>    need to be performed in order to have a function such as a telephone
>    or a video conferencing unit;"
>
> I suggest replacing "envisage" with one of the following
>     - suppose
>     - assume
>
>
>
> Section 3 Paragraph 5
>
> "Existing protocols (for example SIP or XMPP) could be used between
> servers..."
>
> In case the reader is not familiar with SIP or XMPP, the draft should
> link to these protocols. Just like it does with other protocols.
> Especially since the next paragraphs provide an example of SIP and
> XMPP. If the reader does not understand how SIP or XMPP work they
> could get confused. Providing links gives them a resource to
> learn how those protocols work.
>
>
> SIP RFC 3261
> http://www.ietf.org/rfc/rfc3261.txt
>
> XMPP RFC 3920
> http://www.ietf.org/rfc/rfc3920.txt
>
>
>
>
> Spelling error in the Title
>
> Title should say "Browser-based" rather than "Brower-based"
>
>
>
>
>
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

- -- 
Surveillance is pervasive. Go Dark.
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    <br>
    Thank you for the review!<br>
    <br>
    It is very timely, since I'm looking to revise this now based on
    other reviews.<br>
    I will take care of the stuff you point out at the same time -
    thanks for making such specific suggestions!<br>
    <br>
    <br>
    On 06/16/2014 06:00 PM, Colton Shields wrote:<br>
    <span style="white-space: pre;">&gt; Here is my review of the
      Overview: Real Time Protocols for<br>
      &gt; Browser-based Applications draft-ietf-rtcweb-overview-09.
      Please take<br>
      &gt; into consideration my suggestions.<br>
      &gt;<br>
      &gt; Thanks<br>
      &gt; -Colton<br>
      &gt;<br>
      &gt;<br>
      &gt;<br>
      &gt; Section 3 Paragraph 1<br>
      &gt;<br>
      &gt; I am a native English speaker and I had to look up the
      definition of<br>
      &gt; "envisage". Drafts should be written in the plainest language<br>
      &gt; possible, to make it easier for everyone to understand.<br>
      &gt;<br>
      &gt; "The model of real-time support for browser-based
      applications does<br>
      &gt;&nbsp;&nbsp;&nbsp; not envisage that the browser will contain all the
      functions that<br>
      &gt;&nbsp;&nbsp;&nbsp; need to be performed in order to have a function such as a
      telephone<br>
      &gt;&nbsp;&nbsp;&nbsp; or a video conferencing unit;"<br>
      &gt;<br>
      &gt; I suggest replacing "envisage" with one of the following<br>
      &gt;&nbsp;&nbsp;&nbsp;&nbsp; - suppose<br>
      &gt;&nbsp;&nbsp;&nbsp;&nbsp; - assume<br>
      &gt;<br>
      &gt;<br>
      &gt;<br>
      &gt; Section 3 Paragraph 5<br>
      &gt;<br>
      &gt; "Existing protocols (for example SIP or XMPP) could be used
      between<br>
      &gt; servers..."<br>
      &gt;<br>
      &gt; In case the reader is not familiar with SIP or XMPP, the
      draft should<br>
      &gt; link to these protocols. Just like it does with other
      protocols.<br>
      &gt; Especially since the next paragraphs provide an example of
      SIP and<br>
      &gt; XMPP. If the reader does not understand how SIP or XMPP work
      they<br>
      &gt; could get confused. Providing links gives them a resource to<br>
      &gt; learn how those protocols work.<br>
      &gt;<br>
      &gt;<br>
      &gt; SIP RFC 3261<br>
      &gt; <a class="moz-txt-link-freetext" href="http://www.ietf.org/rfc/rfc3261.txt">http://www.ietf.org/rfc/rfc3261.txt</a><br>
      &gt;<br>
      &gt; XMPP RFC 3920<br>
      &gt; <a class="moz-txt-link-freetext" href="http://www.ietf.org/rfc/rfc3920.txt">http://www.ietf.org/rfc/rfc3920.txt</a><br>
      &gt;<br>
      &gt;<br>
      &gt;<br>
      &gt;<br>
      &gt; Spelling error in the Title<br>
      &gt;<br>
      &gt; Title should say "Browser-based" rather than "Brower-based"<br>
      &gt;<br>
      &gt;<br>
      &gt;<br>
      &gt;<br>
      &gt;<br>
      &gt;<br>
      &gt;<br>
      &gt;<br>
      &gt;<br>
      &gt; _______________________________________________<br>
      &gt; rtcweb mailing list<br>
      &gt; <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
      &gt; <a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a></span><br>
    <br>
    - -- <br>
    Surveillance is pervasive. Go Dark.<br>
    -----BEGIN PGP SIGNATURE-----<br>
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    Comment: Using GnuPG with Thunderbird - <a class="moz-txt-link-freetext" href="http://www.enigmail.net/">http://www.enigmail.net/</a><br>
    <br>
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    OC3WdK0YqWayesx4AtMNPtWuT8Z9dgvej/hIKzMEhf8yQlpNA9XsLq20eEIikOr/<br>
    6foa5nqkvx1cvOV/5w/JRRlaUQHfgvBl9GDStBxfpcLRJ6lmtVyD2LdSNLeuGQI=<br>
    =caYy<br>
    -----END PGP SIGNATURE-----<br>
    <br>
  </body>
</html>

--------------060507010207010401000209--


From nobody Tue Jun 17 05:59:28 2014
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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-overview-10.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           : Overview: Real Time Protocols for Browser-based Applications
        Author          : Harald T. Alvestrand
	Filename        : draft-ietf-rtcweb-overview-10.txt
	Pages           : 20
	Date            : 2014-06-17

Abstract:
   This document gives an overview and context of a protocol suite
   intended for use with real-time applications that can be deployed in
   browsers - "real time communication on the Web".

   It intends to serve as a starting and coordination point to make sure
   all the parts that are needed to achieve this goal are findable, and
   that the parts that belong in the Internet protocol suite are fully
   specified and on the right publication track.

   This document is an Applicability Statement - it does not itself
   specify any protocol, but specifies which other specifications RTCWEB
   compliant implementations are supposed to follow.

   This document is a work item of the RTCWEB working group.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-overview-10

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-overview-10


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 Tue Jun 17 06:12:31 2014
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--------------060508060103020300090302
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The most important new words are definitions of terms "WebRTC Browser" 
and "WebRTC Device", and using these as requirement categories - which 
then support a new scattering of MUSTs on the references.

Comments welcome!

Harald


-------- Original Message --------
Subject: 	New Version Notification for draft-ietf-rtcweb-overview-10.txt
Date: 	Tue, 17 Jun 2014 05:59:18 -0700
From: 	internet-drafts@ietf.org
To: 	Harald T. Alvestrand <harald@alvestrand.no>, Harald T. Alvestrand 
<harald@alvestrand.no>



A new version of I-D, draft-ietf-rtcweb-overview-10.txt
has been successfully submitted by Harald T. Alvestrand and posted to the
IETF repository.

Name:		draft-ietf-rtcweb-overview
Revision:	10
Title:		Overview: Real Time Protocols for Browser-based Applications
Document date:	2014-06-17
Group:		rtcweb
Pages:		20
URL:            http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-overview-10.txt
Status:         https://datatracker.ietf.org/doc/draft-ietf-rtcweb-overview/
Htmlized:       http://tools.ietf.org/html/draft-ietf-rtcweb-overview-10
Diff:           http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-overview-10

Abstract:
    This document gives an overview and context of a protocol suite
    intended for use with real-time applications that can be deployed in
    browsers - "real time communication on the Web".

    It intends to serve as a starting and coordination point to make sure
    all the parts that are needed to achieve this goal are findable, and
    that the parts that belong in the Internet protocol suite are fully
    specified and on the right publication track.

    This document is an Applicability Statement - it does not itself
    specify any protocol, but specifies which other specifications RTCWEB
    compliant implementations are supposed to follow.

    This document is a work item of the RTCWEB working group.

                                                                                   


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.

The IETF Secretariat




--------------060508060103020300090302
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Content-Transfer-Encoding: 7bit

<html>
  <head>

    <meta http-equiv="content-type" content="text/html; charset=UTF-8">
  </head>
  <body text="#000000" bgcolor="#FFFFFF">
    The most important new words are definitions of terms "WebRTC
    Browser" and "WebRTC Device", and using these as requirement
    categories - which then support a new scattering of MUSTs on the
    references.<br>
    <br>
    Comments welcome!<br>
    <br>
    Harald<br>
    <div class="moz-forward-container"><br>
      <br>
      -------- Original Message --------
      <table class="moz-email-headers-table" border="0" cellpadding="0"
        cellspacing="0">
        <tbody>
          <tr>
            <th nowrap="nowrap" valign="BASELINE" align="RIGHT">Subject:
            </th>
            <td>New Version Notification for
              draft-ietf-rtcweb-overview-10.txt</td>
          </tr>
          <tr>
            <th nowrap="nowrap" valign="BASELINE" align="RIGHT">Date: </th>
            <td>Tue, 17 Jun 2014 05:59:18 -0700</td>
          </tr>
          <tr>
            <th nowrap="nowrap" valign="BASELINE" align="RIGHT">From: </th>
            <td><a class="moz-txt-link-abbreviated" href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a></td>
          </tr>
          <tr>
            <th nowrap="nowrap" valign="BASELINE" align="RIGHT">To: </th>
            <td>Harald T. Alvestrand <a class="moz-txt-link-rfc2396E" href="mailto:harald@alvestrand.no">&lt;harald@alvestrand.no&gt;</a>,
              Harald T. Alvestrand <a class="moz-txt-link-rfc2396E" href="mailto:harald@alvestrand.no">&lt;harald@alvestrand.no&gt;</a></td>
          </tr>
        </tbody>
      </table>
      <br>
      <br>
      <pre>A new version of I-D, draft-ietf-rtcweb-overview-10.txt
has been successfully submitted by Harald T. Alvestrand and posted to the
IETF repository.

Name:		draft-ietf-rtcweb-overview
Revision:	10
Title:		Overview: Real Time Protocols for Browser-based Applications
Document date:	2014-06-17
Group:		rtcweb
Pages:		20
URL:            <a class="moz-txt-link-freetext" href="http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-overview-10.txt">http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-overview-10.txt</a>
Status:         <a class="moz-txt-link-freetext" href="https://datatracker.ietf.org/doc/draft-ietf-rtcweb-overview/">https://datatracker.ietf.org/doc/draft-ietf-rtcweb-overview/</a>
Htmlized:       <a class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-ietf-rtcweb-overview-10">http://tools.ietf.org/html/draft-ietf-rtcweb-overview-10</a>
Diff:           <a class="moz-txt-link-freetext" href="http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-overview-10">http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-overview-10</a>

Abstract:
   This document gives an overview and context of a protocol suite
   intended for use with real-time applications that can be deployed in
   browsers - "real time communication on the Web".

   It intends to serve as a starting and coordination point to make sure
   all the parts that are needed to achieve this goal are findable, and
   that the parts that belong in the Internet protocol suite are fully
   specified and on the right publication track.

   This document is an Applicability Statement - it does not itself
   specify any protocol, but specifies which other specifications RTCWEB
   compliant implementations are supposed to follow.

   This document is a work item of the RTCWEB working group.

                                                                                  


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.

The IETF Secretariat

</pre>
      <br>
    </div>
    <br>
  </body>
</html>

--------------060508060103020300090302--


From nobody Tue Jun 17 08:54:12 2014
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To: Ted Hardie <ted.ietf@gmail.com>, Cullen Jennings <fluffy@cisco.com>, "Sean Turner" <turners@ieca.com>
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From nobody Tue Jun 17 20:17:31 2014
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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-04.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           : STUN Usage for Consent Freshness
        Authors         : Muthu Arul Mozhi Perumal
                          Dan Wing
                          Ram Mohan Ravindranath
                          Tirumaleswar Reddy
                          Martin Thomson
	Filename        : draft-ietf-rtcweb-stun-consent-freshness-04.txt
	Pages           : 9
	Date            : 2014-06-17

Abstract:
   To prevent sending excessive traffic to an endpoint, periodic consent
   needs to be obtained from that remote endpoint.

   This document describes a consent mechanism using a new STUN usage.
   This same mechanism can also determine connection loss ("liveness")
   with a remote peer.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-freshness-04

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-stun-consent-freshness-04


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 Tue Jun 17 20:33:57 2014
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From: "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-04.txt
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The new revision has:

1) Comments from Christer and Charles incorporated.
2) Incorporated the feedback received in the interim meeting in Section 4
on re-transmissions, revocation of consent
3) Removed the Appendix that had a example as it was no different from
what is mentioned in solution.
4) Updates to section 6 (DiffServ Treatment) based on discussions in the
mailing list.

Please review and provide your feedback.

Regards,
Ram

-----Original Message-----
From: "internet-drafts@ietf.org" <internet-drafts@ietf.org>
Date: Wednesday, 18 June 2014 8:47 am
To: "i-d-announce@ietf.org" <i-d-announce@ietf.org>
Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: [rtcweb] I-D Action:
draft-ietf-rtcweb-stun-consent-freshness-04.txt

>
>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           : STUN Usage for Consent Freshness
>        Authors         : Muthu Arul Mozhi Perumal
>                          Dan Wing
>                          Ram Mohan Ravindranath
>                          Tirumaleswar Reddy
>                          Martin Thomson
>	Filename        : draft-ietf-rtcweb-stun-consent-freshness-04.txt
>	Pages           : 9
>	Date            : 2014-06-17
>
>Abstract:
>   To prevent sending excessive traffic to an endpoint, periodic consent
>   needs to be obtained from that remote endpoint.
>
>   This document describes a consent mechanism using a new STUN usage.
>   This same mechanism can also determine connection loss ("liveness")
>   with a remote peer.
>
>
>The IETF datatracker status page for this draft is:
>https://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshness/
>
>There's also a htmlized version available at:
>http://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-freshness-04
>
>A diff from the previous version is available at:
>http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-stun-consent-freshnes=
s-
>04
>
>
>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


From nobody Thu Jun 19 10:44:05 2014
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From: Ted Hardie <ted.ietf@gmail.com>
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The secretariat has now uploaded the recordings from the recent Interim to
the IETF Youtube channel:

https://www.youtube.com/user/ietf

regards,

Ted Hardie

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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:georgia,=
serif">The secretariat has now uploaded the recordings from the recent Inte=
rim to the IETF Youtube channel:<br><br><a href=3D"https://www.youtube.com/=
user/ietf">https://www.youtube.com/user/ietf</a><br>
<br></div><div class=3D"gmail_default" style=3D"font-family:georgia,serif">=
regards,<br><br>Ted Hardie<br></div></div>

--089e01493cba3544aa04fc33ec26--


From nobody Sun Jun 22 22:46:45 2014
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Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/a2I_e9-UwsCvE_CxUce6u6jvMkM
Subject: Re: [rtcweb] WGLC for data channel drafts - sub-protocol usage
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Do you need that statement?
[RFC6455] section 11.5 says:

    Subprotocol Identifier
       The identifier of the subprotocol, as will be used in the
       |Sec-WebSocket-Protocol| header field registered in Section 11.3.4
       of this specification.  The value must conform to the requirements
       given in item 10 of Section 4.1 of this specification -- namely,
       the value must be a token as defined by RFC 2616 [RFC2616].

"Token" is RFC2616 is defined as:
      CHAR           = <any US-ASCII character (octets 0 - 127)>
      token          = 1*<any CHAR except CTLs or separators>

Regards, Christian

On 12/06/2014 3:24 AM, Christer Holmberg wrote:
> Hi Michael,
>
> I agree that we need to keep text about the encoding.
>
> Regards,
>
> Christer
>
> ________________________________________
> From: Michael Tuexen [Michael.Tuexen@lurchi.franken.de]
> Sent: Wednesday, 11 June 2014 5:28 PM
> To: Christer Holmberg
> Cc: Ted Hardie; rtcweb@ietf.org
> Subject: Re: [rtcweb] WGLC for data channel drafts - sub-protocol usage
>
> On 10 Jun 2014, at 21:22, Christer Holmberg <christer.holmberg@ericsson.com> wrote:
>
>> Hi,
>>
>> A comment on draft-ietf-rtcweb-data-protocol-06:
>>
>> Section 5.1 says:
>>
>> "Protocol: Variable Length (sequence of characters)
>>        The sub-protocol for the channel as a UTF-8 encoded string.  If
>>        this is an empty string the protocol is unspecified.  If it is a
>>        non-empty string, it specifies an protocol registered in the
>>        'WebSocket Subprotocol Name Registry' created in [RFC6455]."
>>
>>
>> I think "The sub-protocol for the channel as a UTF-8 encoded string." part is a little confusing, because there is no such thing as sub-protocol in DCEP itself (DCEP only uses the WebSocket sub-protocol registry for registering DCEP *protocol* values).
>>
>> Perhaps we could just remove the sentence, and keep:
>>
>> "Protocol: Variable Length (sequence of characters)
>>        If this is an empty string the protocol is unspecified.  If it is a
>>        non-empty string, it specifies an protocol registered in the
>>        'WebSocket Subprotocol Name Registry' created in [RFC6455]."
>>
>> Section 4 gives more information about the usage of the protocol.
> Do you intend to drop the statement about the UTF-8 encoding?
> I think we should keep a statement about the encoding.
>
> Best regards
> Michael
>> (Alternatively, we could use sub-protocol terminology also in DCEP)
>>
>> Regards,
>>
>> Christer
>>
>>
>>
>>
>>
>>
>> From: rtcweb [rtcweb-bounces@ietf.org] on behalf of Ted Hardie [ted.ietf@gmail.com]
>> Sent: Tuesday, 10 June, 2014 9:52 PM
>> To: rtcweb@ietf.org
>> Subject: [rtcweb] WGLC for data channel drafts
>>
>> Dear WG,
>>
>> This starts a working group last call for our core Data Channel drafts:
>>
>> draft-ietf-rtcweb-data-protocol-06.txt
>> draft-ietf-rtcweb-data-channel-10.txt
>>
>> Please review them in detail, especially for areas which may be of concern to the broader IETF community.  This WGLC will end on June 27, 2014.
>>
>> Ted, Sean, Cullen
>> _______________________________________________
>> 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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Some comments:

draft-ietf-rtcweb-data-channel talks about unreliable use cases. 
Draft-ietf-rtcweb-data-protocol-06 in clause 4 also mentions "reliable 
or unreliable message transmission". However there is no channel type 
for "unreliable" only "DATA_CHANNEL_RELIABLE_xxx" and 
"DATA_CHANNEL_PARTIAL_RELIABLE_xxx". Are the "unreliable" use cases/text 
actually "partial reliable"? The requirements suggest that unreliable 
transport is needed but the solution offered is for reliable and partial 
reliability.

draft-ietf-rtcweb-data-channel-10
---------------------------------
Cl. 6.1 bullet 1: Would "stream reconfiguration" be a better term than 
"stream reset"? RFC6525 allows reset and addition of streams.
Cl. 6.1 bullet 2: "dynamic address reconfiguration" is the name of the 
RFC 5061. Is the part that Data Channel supports only the "Supported 
Extensions" parameter (cl.4.2.7/RFC5061)?

Regards, Christian

On 11/06/2014 4:52 AM, Ted Hardie wrote:
> Dear WG,
>
> This starts a working group last call for our core Data Channel drafts:
>
> draft-ietf-rtcweb-data-protocol-06.txt
> draft-ietf-rtcweb-data-channel-10.txt
>
> Please review them in detail, especially for areas which may be of 
> concern to the broader IETF community.  This WGLC will end on June 27, 
> 2014.
>
> Ted, Sean, Cullen
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Tue Jun 24 08:34:00 2014
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On 06/24/2014 05:28 PM, Hutton, Andrew wrote:
> Section 4 now states âWebRTC devices MUST implement the transport protocols described in [I-D.ietf-rtcweb-transports <http://tools.ietf.org/html/draft-ietf-rtcweb-overview-10#ref-I-D.ietf-rtcweb-transports>]â
>
>  
>
> This might be a comment on âtransports- more than âoverview- but section 3.4 of transports states that âThe implementation MUST be a full ICE implementation, not ICE-Liteâ. This is however not true for âWebRTC devicesâ only for âWebRTC browsersâ since ICE-Lite is sufficient for âWebRTC devicesâ.
>  

Good point. -transports- was last updated before I invented "WebRTC
devices", it uses "devices" and "browsers" interchangeably.

Will make them consistent next round, if WG signs off on the split.


> Regards
> Andy
>
>  
>
>  
>
>  
>
>  
>
> *From:*rtcweb [mailto:rtcweb-bounces@ietf.org] *On Behalf Of *Harald
> Alvestrand
> *Sent:* 17 June 2014 14:12
> *To:* rtcweb@ietf.org
> *Subject:* [rtcweb] Fwd: New Version Notification for
> draft-ietf-rtcweb-overview-10.txt
>
>  
>
> The most important new words are definitions of terms "WebRTC Browser"
> and "WebRTC Device", and using these as requirement categories - which
> then support a new scattering of MUSTs on the references.
>
> Comments welcome!
>
> Harald
>
>
>
> -------- Original Message --------
>
> *Subject: *
>
> 	
>
> New Version Notification for draft-ietf-rtcweb-overview-10.txt
>
> *Date: *
>
> 	
>
> Tue, 17 Jun 2014 05:59:18 -0700
>
> *From: *
>
> 	
>
> internet-drafts@ietf.org <mailto:internet-drafts@ietf.org>
>
> *To: *
>
> 	
>
> Harald T. Alvestrand <harald@alvestrand.no>
> <mailto:harald@alvestrand.no>, Harald T. Alvestrand
> <harald@alvestrand.no> <mailto:harald@alvestrand.no>
>
>  
>
> A new version of I-D, draft-ietf-rtcweb-overview-10.txt
> has been successfully submitted by Harald T. Alvestrand and posted to the
> IETF repository.
>  
> Name:         draft-ietf-rtcweb-overview
> Revision:     10
> Title:        Overview: Real Time Protocols for Browser-based Applications
> Document date: 2014-06-17
> Group:        rtcweb
> Pages:        20
> URL:            http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-overview-10.txt
> Status:         https://datatracker.ietf.org/doc/draft-ietf-rtcweb-overview/
> Htmlized:       http://tools.ietf.org/html/draft-ietf-rtcweb-overview-10
> Diff:           http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-overview-10
>  
> Abstract:
>    This document gives an overview and context of a protocol suite
>    intended for use with real-time applications that can be deployed in
>    browsers - "real time communication on the Web".
>  
>    It intends to serve as a starting and coordination point to make sure
>    all the parts that are needed to achieve this goal are findable, and
>    that the parts that belong in the Internet protocol suite are fully
>    specified and on the right publication track.
>  
>    This document is an Applicability Statement - it does not itself
>    specify any protocol, but specifies which other specifications RTCWEB
>    compliant implementations are supposed to follow.
>  
>    This document is a work item of the RTCWEB working group.
>  
>                                                                                   
>  
>  
> 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.
>  
> The IETF Secretariat
>  
>
>  
>
>  
>


-- 
Surveillance is pervasive. Go Dark.


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    <div class="moz-cite-prefix">On 06/24/2014 05:28 PM, Hutton, Andrew
      wrote:<br>
    </div>
    <blockquote
cite="mid:9F33F40F6F2CD847824537F3C4E37DDF17E08E67@MCHP04MSX.global-ad.net"
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      <div class="WordSection1">
        <pre style="page-break-before:always"><span style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Section 4 now states âWebRTC devices MUST implement the transport protocols described in [<a moz-do-not-send="true" href="http://tools.ietf.org/html/draft-ietf-rtcweb-overview-10#ref-I-D.ietf-rtcweb-transports"><span style="color:#1F497D;text-decoration:none">I-D.ietf-rtcweb-transports</span></a>]â</span><o:p></o:p></pre>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>Â </o:p></span></p>
        <pre style="page-break-before:always"><span style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">This might be a comment on âtransports- more than âoverview- but section 3.4 of transports states that âThe implementation MUST be a full ICE implementation, not ICE-Liteâ. This is however not true for âWebRTC devicesâ only for âWebRTC browsersâ since ICE-Lite is sufficient for âWebRTC devicesâ.<o:p></o:p></span></pre>
        <pre style="page-break-before:always"><span style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>Â </o:p></span></pre>
      </div>
    </blockquote>
    <br>
    Good point. -transports- was last updated before I invented "WebRTC
    devices", it uses "devices" and "browsers" interchangeably.<br>
    <br>
    Will make them consistent next round, if WG signs off on the split.<br>
    <br>
    <br>
    <blockquote
cite="mid:9F33F40F6F2CD847824537F3C4E37DDF17E08E67@MCHP04MSX.global-ad.net"
      type="cite">
      <div class="WordSection1">
        <pre style="page-break-before:always"><span style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards<o:p></o:p></span></pre>
        <pre style="page-break-before:always"><span style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Andy<o:p></o:p></span></pre>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>Â </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>Â </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>Â </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>Â </o:p></span></p>
        <div style="border:none;border-left:solid blue 1.5pt;padding:0cm
          0cm 0cm 4.0pt">
          <div>
            <div style="border:none;border-top:solid #B5C4DF
              1.0pt;padding:3.0pt 0cm 0cm 0cm">
              <p class="MsoNormal"><b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">From:</span></b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">
                  rtcweb [<a class="moz-txt-link-freetext"
                    href="mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a>]
                  <b>On Behalf Of </b>Harald Alvestrand<br>
                  <b>Sent:</b> 17 June 2014 14:12<br>
                  <b>To:</b> <a class="moz-txt-link-abbreviated"
                    href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                  <b>Subject:</b> [rtcweb] Fwd: New Version Notification
                  for draft-ietf-rtcweb-overview-10.txt<o:p></o:p></span></p>
            </div>
          </div>
          <p class="MsoNormal"><o:p>Â </o:p></p>
          <p class="MsoNormal">The most important new words are
            definitions of terms "WebRTC Browser" and "WebRTC Device",
            and using these as requirement categories - which then
            support a new scattering of MUSTs on the references.<br>
            <br>
            Comments welcome!<br>
            <br>
            Harald<o:p></o:p></p>
          <div>
            <p class="MsoNormal"><br>
              <br>
              -------- Original Message -------- <o:p></o:p></p>
            <table class="MsoNormalTable" border="0" cellpadding="0"
              cellspacing="0">
              <tbody>
                <tr>
                  <td style="padding:0cm 0cm 0cm 0cm" nowrap="nowrap"
                    valign="top">
                    <p class="MsoNormal" style="text-align:right"
                      align="right"><b>Subject: <o:p></o:p></b></p>
                  </td>
                  <td style="padding:0cm 0cm 0cm 0cm">
                    <p class="MsoNormal">New Version Notification for
                      draft-ietf-rtcweb-overview-10.txt<o:p></o:p></p>
                  </td>
                </tr>
                <tr>
                  <td style="padding:0cm 0cm 0cm 0cm" nowrap="nowrap"
                    valign="top">
                    <p class="MsoNormal" style="text-align:right"
                      align="right"><b>Date: <o:p></o:p></b></p>
                  </td>
                  <td style="padding:0cm 0cm 0cm 0cm">
                    <p class="MsoNormal">Tue, 17 Jun 2014 05:59:18 -0700<o:p></o:p></p>
                  </td>
                </tr>
                <tr>
                  <td style="padding:0cm 0cm 0cm 0cm" nowrap="nowrap"
                    valign="top">
                    <p class="MsoNormal" style="text-align:right"
                      align="right"><b>From: <o:p></o:p></b></p>
                  </td>
                  <td style="padding:0cm 0cm 0cm 0cm">
                    <p class="MsoNormal"><a moz-do-not-send="true"
                        href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a><o:p></o:p></p>
                  </td>
                </tr>
                <tr>
                  <td style="padding:0cm 0cm 0cm 0cm" nowrap="nowrap"
                    valign="top">
                    <p class="MsoNormal" style="text-align:right"
                      align="right"><b>To: <o:p></o:p></b></p>
                  </td>
                  <td style="padding:0cm 0cm 0cm 0cm">
                    <p class="MsoNormal">Harald T. Alvestrand <a
                        moz-do-not-send="true"
                        href="mailto:harald@alvestrand.no">&lt;harald@alvestrand.no&gt;</a>,
                      Harald T. Alvestrand <a moz-do-not-send="true"
                        href="mailto:harald@alvestrand.no">&lt;harald@alvestrand.no&gt;</a><o:p></o:p></p>
                  </td>
                </tr>
              </tbody>
            </table>
            <p class="MsoNormal" style="margin-bottom:12.0pt"><o:p>Â </o:p></p>
            <pre>A new version of I-D, draft-ietf-rtcweb-overview-10.txt<o:p></o:p></pre>
            <pre>has been successfully submitted by Harald T. Alvestrand and posted to the<o:p></o:p></pre>
            <pre>IETF repository.<o:p></o:p></pre>
            <pre><o:p>Â </o:p></pre>
            <pre>Name:Â Â Â Â Â Â Â Â  draft-ietf-rtcweb-overview<o:p></o:p></pre>
            <pre>Revision:Â Â Â Â  10<o:p></o:p></pre>
            <pre>Title:Â Â Â Â Â Â Â  Overview: Real Time Protocols for Browser-based Applications<o:p></o:p></pre>
            <pre>Document date: 2014-06-17<o:p></o:p></pre>
            <pre>Group:Â Â Â Â Â Â Â  rtcweb<o:p></o:p></pre>
            <pre>Pages:Â Â Â Â Â Â Â  20<o:p></o:p></pre>
            <pre>URL:Â Â Â Â Â Â Â Â Â Â Â  <a moz-do-not-send="true" href="http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-overview-10.txt">http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-overview-10.txt</a><o:p></o:p></pre>
            <pre>Status:Â Â Â Â Â Â Â Â  <a moz-do-not-send="true" href="https://datatracker.ietf.org/doc/draft-ietf-rtcweb-overview/">https://datatracker.ietf.org/doc/draft-ietf-rtcweb-overview/</a><o:p></o:p></pre>
            <pre>Htmlized:Â Â Â Â Â Â  <a moz-do-not-send="true" href="http://tools.ietf.org/html/draft-ietf-rtcweb-overview-10">http://tools.ietf.org/html/draft-ietf-rtcweb-overview-10</a><o:p></o:p></pre>
            <pre>Diff:Â Â Â Â Â Â Â Â Â Â  <a moz-do-not-send="true" href="http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-overview-10">http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-overview-10</a><o:p></o:p></pre>
            <pre><o:p>Â </o:p></pre>
            <pre>Abstract:<o:p></o:p></pre>
            <pre>Â Â  This document gives an overview and context of a protocol suite<o:p></o:p></pre>
            <pre>Â Â  intended for use with real-time applications that can be deployed in<o:p></o:p></pre>
            <pre>Â Â  browsers - "real time communication on the Web".<o:p></o:p></pre>
            <pre><o:p>Â </o:p></pre>
            <pre>Â Â  It intends to serve as a starting and coordination point to make sure<o:p></o:p></pre>
            <pre>Â Â  all the parts that are needed to achieve this goal are findable, and<o:p></o:p></pre>
            <pre>Â Â  that the parts that belong in the Internet protocol suite are fully<o:p></o:p></pre>
            <pre>Â Â  specified and on the right publication track.<o:p></o:p></pre>
            <pre><o:p>Â </o:p></pre>
            <pre>Â Â  This document is an Applicability Statement - it does not itself<o:p></o:p></pre>
            <pre>Â Â  specify any protocol, but specifies which other specifications RTCWEB<o:p></o:p></pre>
            <pre>Â Â  compliant implementations are supposed to follow.<o:p></o:p></pre>
            <pre><o:p>Â </o:p></pre>
            <pre>Â  Â This document is a work item of the RTCWEB working group.<o:p></o:p></pre>
            <pre><o:p>Â </o:p></pre>
            <pre>Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â  <o:p></o:p></pre>
            <pre><o:p>Â </o:p></pre>
            <pre><o:p>Â </o:p></pre>
            <pre>Please note that it may take a couple of minutes from the time of submission<o:p></o:p></pre>
            <pre>until the htmlized version and diff are available at tools.ietf.org.<o:p></o:p></pre>
            <pre><o:p>Â </o:p></pre>
            <pre>The IETF Secretariat<o:p></o:p></pre>
            <pre><o:p>Â </o:p></pre>
            <p class="MsoNormal"><o:p>Â </o:p></p>
          </div>
          <p class="MsoNormal"><o:p>Â </o:p></p>
        </div>
      </div>
    </blockquote>
    <br>
    <br>
    <pre class="moz-signature" cols="72">-- 
Surveillance is pervasive. Go Dark.
</pre>
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Content-Type: text/plain; charset=UTF-8

Howdy,

Please note that this working group last call ends in 3 days, and we have
very little in the way or responses.  Please review the current drafts if
you have not done so.

thanks,

Ted


On Tue, Jun 10, 2014 at 11:52 AM, Ted Hardie <ted.ietf@gmail.com> wrote:

> Dear WG,
>
> This starts a working group last call for our core Data Channel drafts:
>
> draft-ietf-rtcweb-data-protocol-06.txt
> draft-ietf-rtcweb-data-channel-10.txt
>
> Please review them in detail, especially for areas which may be of concern
> to the broader IETF community.  This WGLC will end on June 27, 2014.
>
> Ted, Sean, Cullen
>

--bcaec5186ab02a6fbb04fc98c6bd
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:georgia,=
serif">Howdy,<br><br></div><div class=3D"gmail_default" style=3D"font-famil=
y:georgia,serif">Please note that this working group last call ends in 3 da=
ys, and we have very little in the way or responses.=C2=A0 Please review th=
e current drafts if you have not done so.<br>
<br>thanks,<br><br></div><div class=3D"gmail_default" style=3D"font-family:=
georgia,serif">Ted<br></div></div><div class=3D"gmail_extra"><br><br><div c=
lass=3D"gmail_quote">On Tue, Jun 10, 2014 at 11:52 AM, Ted Hardie <span dir=
=3D"ltr">&lt;<a href=3D"mailto:ted.ietf@gmail.com" target=3D"_blank">ted.ie=
tf@gmail.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_default=
" style=3D"font-family:georgia,serif">Dear WG,<br><br><div class=3D"gmail_d=
efault" style=3D"font-family:georgia,serif">
This starts a working group last call for our core Data Channel drafts:<br>
<br>draft-ietf-rtcweb-data-protocol-06.txt<br>
draft-ietf-rtcweb-data-channel-10.txt<br><br></div><div class=3D"gmail_defa=
ult" style=3D"font-family:georgia,serif">Please
 review them in detail, especially for areas which may be of concern to=20
the broader IETF community.=C2=A0 This WGLC will end on June 27, 2014.<br>
<br></div>Ted, Sean, Cullen<br></div></div>
</blockquote></div><br></div>

--bcaec5186ab02a6fbb04fc98c6bd--


From nobody Fri Jun 27 01:27:16 2014
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From: "Hutton, Andrew" <andrew.hutton@unify.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: draft-hutton-httpbis-connect-protocol-00.txt
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I just submitted this draft for which there is a short agenda slot in Toron=
to httpbis wg meeting.

It was written as a result of our discussion during the recent rtcweb inter=
im meeting in Washington DC.

The draft proposes adding an indication in to HTTP Connect as to what proto=
col is used with the tunnel and specifically for use with WebRTC (I.e. TURN=
/ICE-TCP) so should be of interest to this group.

Discussion should be on the HTTPBIS list.

Regards
Andy


        Title           : HTTP Connect - Tunnel Protocol For WebRTC
        Authors         : Andrew Hutton
                          Justin Uberti
                          Martin Thomson
	Filename        : draft-hutton-httpbis-connect-protocol-00.txt
	Pages           : 7
	Date            : 2014-06-27

Abstract:
   This document describes a mechanism to enable HTTP Clients to provide
   an indication within a HTTP Connect request as to which protocol will
   be used within the tunnel established to the Server identified by the
   target resource.  The tunneled protocol is declared using the Tunnel-
   Protocol HTTP Request header field.  Label usage relating to the use
   of HTTP Connect by WebRTC clients (e.g. turn, webrtc) are described
   in this document.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-hutton-httpbis-connect-protocol/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-hutton-httpbis-connect-protocol-00


Please note that it may take a couple of minutes from the time of submissio=
n
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/

_______________________________________________
I-D-Announce mailing list
I-D-Announce@ietf.org
https://www.ietf.org/mailman/listinfo/i-d-announce
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From: Michael Procter <michael@voip.co.uk>
To: "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>
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Hi Ram,

I have a couple of comments about the word "inverted".  In section
4.1, the draft says:

   receiving a
   matching, authenticated, non-error ICE binding response from the
   inverted remote peer's Transport Address.

I think this means receiving it from the remote peer's Transport
Address, although the presence of 'inverted' made me wonder.  If the
word is unnecessary, please remove it.  If it is needed, please add
more explanation as to what it means.

Further on, the draft says:

   That is, if a valid STUN binding response [...]
   has not been received from the inverted 5-tuple, [...]

I think this means receiving it from the 5-tuple, but again, the
presence of 'inverted' made me wonder.  Is this a naming issue?  I am
happy to send and receive on the same 5-tuple, and I expect my peer to
do the same.  However, my peer's 5-tuple will be inverted with respect
to mine!

Regards,

Michael



On 18 June 2014 04:33, Ram Mohan R (rmohanr) <rmohanr@cisco.com> wrote:
> The new revision has:
>
> 1) Comments from Christer and Charles incorporated.
> 2) Incorporated the feedback received in the interim meeting in Section 4
> on re-transmissions, revocation of consent
> 3) Removed the Appendix that had a example as it was no different from
> what is mentioned in solution.
> 4) Updates to section 6 (DiffServ Treatment) based on discussions in the
> mailing list.
>
> Please review and provide your feedback.
>
> Regards,
> Ram
>
> -----Original Message-----
> From: "internet-drafts@ietf.org" <internet-drafts@ietf.org>
> Date: Wednesday, 18 June 2014 8:47 am
> To: "i-d-announce@ietf.org" <i-d-announce@ietf.org>
> Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
> Subject: [rtcweb] I-D Action:
> draft-ietf-rtcweb-stun-consent-freshness-04.txt
>
>>
>>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           : STUN Usage for Consent Freshness
>>        Authors         : Muthu Arul Mozhi Perumal
>>                          Dan Wing
>>                          Ram Mohan Ravindranath
>>                          Tirumaleswar Reddy
>>                          Martin Thomson
>>       Filename        : draft-ietf-rtcweb-stun-consent-freshness-04.txt
>>       Pages           : 9
>>       Date            : 2014-06-17
>>
>>Abstract:
>>   To prevent sending excessive traffic to an endpoint, periodic consent
>>   needs to be obtained from that remote endpoint.
>>
>>   This document describes a consent mechanism using a new STUN usage.
>>   This same mechanism can also determine connection loss ("liveness")
>>   with a remote peer.
>>
>>
>>The IETF datatracker status page for this draft is:
>>https://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshness/
>>
>>There's also a htmlized version available at:
>>http://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-freshness-04
>>
>>A diff from the previous version is available at:
>>http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-stun-consent-freshness-
>>04
>>
>>
>>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


From nobody Fri Jun 27 06:00:32 2014
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On 06/10/2014 08:52 PM, Ted Hardie wrote:
> Dear WG,
>
> This starts a working group last call for our core Data Channel drafts:
>
> draft-ietf-rtcweb-data-protocol-06.txt
> draft-ietf-rtcweb-data-channel-10.txt
>
> Please review them in detail, especially for areas which may be of 
> concern to the broader IETF community.  This WGLC will end on June 27, 
> 2014.

This is my Last Call review of these two drafts.

First of all: The drafts are ready for IETF Last Call.
Second: There are a number of details that should be fixed. Whether that 
happens before IETF Last Call or not is not of great concern to me.

draft-ietf-rtcweb-data-channel-10
----------------------------------------------
Abstract

Ten years from now, the RFC will exist, but the RTCWEB WG will not. I 
suggest replacing "The Real-Time Communication in Web browsers working 
group is charged to" with "The WebRTC framework specifies".

1. Introduction

It's a little abrupt. Some more words of introduction may be needed 
here. Something like this:

In the WebRTC framework, communication between the parties consists of 
media (for example audio and video) and non-media data. Media is sent 
using SRTP, and is not specified further here. Non-media data is handled 
by...."

I suggest a global replace of "non-SRTP media data" with "non-media 
data" (3 occurences total). "non-SRTP media data" can be parsed as 
"media data that is not carried over SRTP", which is not the intended 
meaning.

4. Requirements

Suggest striking "and that the WebRTC PeerConnection as a whole is fair 
with competing traffic such as TCP". The RMCAT discussions have proved 
how difficult it is just to define that, far less achieve it.
If we want to say something in relation to TCP, I would suggest "and 
that the WebRTC PeerConnection does not cause excessive problems when 
run in parallel with TCP connections".

5. SCTP over DTLS over UDP Considerations

In the paragraph on SCTP multihoming, I suggest using "the DTLS layer" 
instead of "the lower layer". There are multiple lower layers, 
identifying which one is mentioned here improves clarity.

The term "user-land" is used here, while Req. 10 uses "user application 
space". I suggest using "user application space" in both places.

The paragraph "When using DTLS as the lower layer..." repeats the info 
that SCTP multihoming is not used. It may be enough to say this once.

"The partial reliability extension MUST support policies..." - this 
paragraph would be more implementable if specific policies, specified in 
an I-D or RFC, were referenced.

The split between section 5 and section 6 seems odd. In particular, the 
section "This SCTP stack and its upper layer...." down to "exactly once 
and delivered in the order received." seems to belong to section 6 not 
section 5.

6.1 SCTP Protocol Considerations

I think the -ndata messsage interleaving MUST be implemented, according 
to previous consensus.
"SHOULD be used" seems both meaningless and unneccssary.

6.2 Association setup

Nit: "that can negotiated" -> "that can be negotiated"

6.4 Channel definition

The title should probably be "WebRTC Data Channel Definition".

As part of getting the WGs out of the document, the first sentence 
should be "The  WebRTC DataChannels are bidirectional."
"Among other things" is a bad term to have in a document at Last Call 
time. Can we remove it?

The SHOULD for -ndata- interleaving should be MUST implement and MUST 
offer. If it is not successfully negotiated at association setup time 
(is that how it works?), the "SHOULD limit" clause is appropriate.

6.7 Closing a channel

Nit: "available to reuse" -> "available for reuse"
Nit: "before resetting the stream" -> "before the stream is reset"

7 Security considerations

Is there a security worry with an attacker sending over-long messages in 
an attempt to cause OOM situations?

draft-ietf-rtcweb-data-protocol-06
---------------------------------------------
Abstract: Same worry about WG reference.

3. Terminology

Stream is defined in terms of stream, which is unhealthy. A reference to 
a section of a relevant SCTP document would be better.

The term "SCTP stream" occurs at least 18 times in the document (out of 
approx 41 occurences of "stream" overall). We should either use "SCTP 
stream" consistently" or stick to "Stream" consistently.

The term "SCTP stream identifier" (or "stream identifier") should be 
called out in the terminology.

The term "data channel" occurs 36 times. Consider standardizing on 
either Channel or Data Channel, and either capitalizing it or not.

6 Procedures
Nit: "the sender of it can start" -> "the sender of the 
DATA_CHANNEL_OPEN can start"

"If a DATA_CHANNEL_OPEN message is received .... the stream identifier 
corresponds to the role of the peer" - is there a reason to insist that 
the protocol machine enforce the odd/even rule?`(actually the next 
paragraph does not cover the odd/even rule violation case, which is 
probably just an unintentional mistake - I prefer writing this kind of 
thing in "IF <condition> ... ELSE .... - because if one writes it as "IF 
<condition>; IF <opposite condition>", the risk of losing some edge 
cases is high.

An alternative formulation for the second paragraph is "If the 
DATA_CHANNEL_OPEN message doesn't satisfy the conditions above, for 
instance if..."

7 Security Considerations

The "When using..." paragraph sounds a bit off. What about saying

This protocol does not provide privacy, integrity or authentication. It 
needs to be used as part of a protocol suite that contains all these 
things. Such a protocol suite is specified in [-dtls-encaps].

Dependency worries
---------------------------
-protocol- depends normatively on these non-WG documents:

draft-ietf-tsvwg-sctp-ndata ("I-D exists")
draft-ietf-tsvwg-sctp-dtls-encaps ("AD is watching")
draft-ietf-tsvwg-sctp-prpolicies ("I-D exists")
draft-ietf-mmusic-sctp-sdp ("I-D exists")

None of these are in the IESG queue. Are we confident of their speed of 
progress?


This concludes my review. I think the protocol definition is ready, and 
needs no changes.





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    <div class="moz-cite-prefix">On 06/10/2014 08:52 PM, Ted Hardie
      wrote:<br>
    </div>
    <blockquote
cite="mid:CA+9kkMAzdcLhM_g+S6407=_SUBKu7hLhMrYSrqGP5tk3yEsP6w@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_default" style="font-family:georgia,serif">Dear
          WG,<br>
          <br>
          <div class="gmail_default" style="font-family:georgia,serif">This
            starts a working group last call for our core Data Channel
            drafts:<br>
            <br>
            draft-ietf-rtcweb-data-protocol-06.txt<br>
            draft-ietf-rtcweb-data-channel-10.txt<br>
            <br>
          </div>
          <div class="gmail_default" style="font-family:georgia,serif">Please

            review them in detail, especially for areas which may be of
            concern to the broader IETF community.&nbsp; This WGLC will end
            on June 27, 2014.<br>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    This is my Last Call review of these two drafts.<br>
    <br>
    First of all: The drafts are ready for IETF Last Call.<br>
    Second: There are a number of details that should be fixed. Whether
    that happens before IETF Last Call or not is not of great concern to
    me.<br>
    <br>
    draft-ietf-rtcweb-data-channel-10<br>
    ----------------------------------------------<br>
    Abstract<br>
    <br>
    Ten years from now, the RFC will exist, but the RTCWEB WG will not.
    I suggest replacing "The Real-Time Communication in Web browsers
    working group is charged to" with "The WebRTC framework specifies".<br>
    <br>
    1. Introduction<br>
    <br>
    It's a little abrupt. Some more words of introduction may be needed
    here. Something like this:<br>
    <br>
    In the WebRTC framework, communication between the parties consists
    of media (for example audio and video) and non-media data. Media is
    sent using SRTP, and is not specified further here. Non-media data
    is handled by...."<br>
    <br>
    I suggest a global replace of "non-SRTP media data" with "non-media
    data" (3 occurences total). "non-SRTP media data" can be parsed as
    "media data that is not carried over SRTP", which is not the
    intended meaning.<br>
    <br>
    4. Requirements<br>
    <br>
    Suggest striking "and that the WebRTC PeerConnection as a whole is
    fair with competing traffic such as TCP". The RMCAT discussions have
    proved how difficult it is just to define that, far less achieve it.<br>
    If we want to say something in relation to TCP, I would suggest "and
    that the WebRTC PeerConnection does not cause excessive problems
    when run in parallel with TCP connections".<br>
    <br>
    5. SCTP over DTLS over UDP Considerations<br>
    <br>
    In the paragraph on SCTP multihoming, I suggest using "the DTLS
    layer" instead of "the lower layer". There are multiple lower
    layers, identifying which one is mentioned here improves clarity.<br>
    <br>
    The term "user-land" is used here, while Req. 10 uses "user
    application space". I suggest using "user application space" in both
    places.<br>
    <br>
    The paragraph "When using DTLS as the lower layer..." repeats the
    info that SCTP multihoming is not used. It may be enough to say this
    once.<br>
    <br>
    "The partial reliability extension MUST support policies..." - this
    paragraph would be more implementable if specific policies,
    specified in an I-D or RFC, were referenced.<br>
    <br>
    The split between section 5 and section 6 seems odd. In particular,
    the section "This SCTP stack and its upper layer...." down to
    "exactly once and delivered in the order received." seems to belong
    to section 6 not section 5.<br>
    <br>
    6.1 SCTP Protocol Considerations<br>
    <br>
    I think the -ndata messsage interleaving MUST be implemented,
    according to previous consensus.<br>
    "SHOULD be used" seems both meaningless and unneccssary.<br>
    <br>
    6.2 Association setup<br>
    <br>
    Nit: "that can negotiated" -&gt; "that can be negotiated"<br>
    <br>
    6.4 Channel definition<br>
    <br>
    The title should probably be "WebRTC Data Channel Definition".<br>
    <br>
    As part of getting the WGs out of the document, the first sentence
    should be "The&nbsp; WebRTC DataChannels are bidirectional."<br>
    "Among other things" is a bad term to have in a document at Last
    Call time. Can we remove it?<br>
    <br>
    The SHOULD for -ndata- interleaving should be MUST implement and
    MUST offer. If it is not successfully negotiated at association
    setup time (is that how it works?), the "SHOULD limit" clause is
    appropriate.<br>
    <br>
    6.7 Closing a channel<br>
    <br>
    Nit: "available to reuse" -&gt; "available for reuse"<br>
    Nit: "before resetting the stream" -&gt; "before the stream is
    reset"<br>
    <br>
    7 Security considerations<br>
    <br>
    Is there a security worry with an attacker sending over-long
    messages in an attempt to cause OOM situations?<br>
    <br>
    draft-ietf-rtcweb-data-protocol-06<br>
    ---------------------------------------------<br>
    Abstract: Same worry about WG reference.<br>
    <br>
    3. Terminology<br>
    <br>
    Stream is defined in terms of stream, which is unhealthy. A
    reference to a section of a relevant SCTP document would be better.<br>
    <br>
    The term "SCTP stream" occurs at least 18 times in the document (out
    of approx 41 occurences of "stream" overall). We should either use
    "SCTP stream" consistently" or stick to "Stream" consistently.<br>
    <br>
    The term "SCTP stream identifier" (or "stream identifier") should be
    called out in the terminology.<br>
    <br>
    The term "data channel" occurs 36 times. Consider standardizing on
    either Channel or Data Channel, and either capitalizing it or not.<br>
    <br>
    6 Procedures<br>
    Nit: "the sender of it can start" -&gt; "the sender of the
    DATA_CHANNEL_OPEN can start"<br>
    <br>
    "If a DATA_CHANNEL_OPEN message is received .... the stream
    identifier corresponds to the role of the peer" - is there a reason
    to insist that the protocol machine enforce the odd/even
    rule?`(actually the next paragraph does not cover the odd/even rule
    violation case, which is probably just an unintentional mistake - I
    prefer writing this kind of thing in "IF &lt;condition&gt; ... ELSE
    .... - because if one writes it as "IF &lt;condition&gt;; IF
    &lt;opposite condition&gt;", the risk of losing some edge cases is
    high.<br>
    <br>
    An alternative formulation for the second paragraph is "If the
    DATA_CHANNEL_OPEN message doesn't satisfy the conditions above, for
    instance if..."<br>
    <br>
    7 Security Considerations<br>
    <br>
    The "When using..." paragraph sounds a bit off. What about saying<br>
    <br>
    This protocol does not provide privacy, integrity or authentication.
    It needs to be used as part of a protocol suite that contains all
    these things. Such a protocol suite is specified in [-dtls-encaps].<br>
    <br>
    Dependency worries<br>
    ---------------------------<br>
    -protocol- depends normatively on these non-WG documents:<br>
    <br>
    draft-ietf-tsvwg-sctp-ndata ("I-D exists")<br>
    draft-ietf-tsvwg-sctp-dtls-encaps ("AD is watching")<br>
    draft-ietf-tsvwg-sctp-prpolicies ("I-D exists")<br>
    draft-ietf-mmusic-sctp-sdp ("I-D exists")<br>
    <br>
    None of these are in the IESG queue. Are we confident of their speed
    of progress?<br>
    <br>
    <br>
    This concludes my review. I think the protocol definition is ready,
    and needs no changes.<br>
    <br>
    <br>
    <br>
    <br>
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From: "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>
To: Michael Procter <michael@voip.co.uk>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-04.txt
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Date: Sat, 28 Jun 2014 16:52:42 +0000
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Hi Michael,

Thanks for your review.

We used inverted to convey that Consent response (STUN binding response)
from remote peer must be  from inverted 5-tuple (source and dest addr
swapped) of the request sent.
I will see how to re-word it.

Regards,
Ram

-----Original Message-----
From: Michael Procter <michael@voip.co.uk>
Date: Friday, 27 June 2014 3:36 pm
To: Ram Mohan Ravindranath <rmohanr@cisco.com>
Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] I-D Action:
draft-ietf-rtcweb-stun-consent-freshness-04.txt

>Hi Ram,
>
>I have a couple of comments about the word "inverted".  In section
>4.1, the draft says:
>
>   receiving a
>   matching, authenticated, non-error ICE binding response from the
>   inverted remote peer's Transport Address.
>
>I think this means receiving it from the remote peer's Transport
>Address, although the presence of 'inverted' made me wonder.  If the
>word is unnecessary, please remove it.  If it is needed, please add
>more explanation as to what it means.
>
>Further on, the draft says:
>
>   That is, if a valid STUN binding response [...]
>   has not been received from the inverted 5-tuple, [...]
>
>I think this means receiving it from the 5-tuple, but again, the
>presence of 'inverted' made me wonder.  Is this a naming issue?  I am
>happy to send and receive on the same 5-tuple, and I expect my peer to
>do the same.  However, my peer's 5-tuple will be inverted with respect
>to mine!
>
>Regards,
>
>Michael
>
>
>
>On 18 June 2014 04:33, Ram Mohan R (rmohanr) <rmohanr@cisco.com> wrote:
>> The new revision has:
>>
>> 1) Comments from Christer and Charles incorporated.
>> 2) Incorporated the feedback received in the interim meeting in Section
>>4
>> on re-transmissions, revocation of consent
>> 3) Removed the Appendix that had a example as it was no different from
>> what is mentioned in solution.
>> 4) Updates to section 6 (DiffServ Treatment) based on discussions in the
>> mailing list.
>>
>> Please review and provide your feedback.
>>
>> Regards,
>> Ram
>>
>> -----Original Message-----
>> From: "internet-drafts@ietf.org" <internet-drafts@ietf.org>
>> Date: Wednesday, 18 June 2014 8:47 am
>> To: "i-d-announce@ietf.org" <i-d-announce@ietf.org>
>> Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
>> Subject: [rtcweb] I-D Action:
>> draft-ietf-rtcweb-stun-consent-freshness-04.txt
>>
>>>
>>>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           : STUN Usage for Consent Freshness
>>>        Authors         : Muthu Arul Mozhi Perumal
>>>                          Dan Wing
>>>                          Ram Mohan Ravindranath
>>>                          Tirumaleswar Reddy
>>>                          Martin Thomson
>>>       Filename        : draft-ietf-rtcweb-stun-consent-freshness-04.txt
>>>       Pages           : 9
>>>       Date            : 2014-06-17
>>>
>>>Abstract:
>>>   To prevent sending excessive traffic to an endpoint, periodic consent
>>>   needs to be obtained from that remote endpoint.
>>>
>>>   This document describes a consent mechanism using a new STUN usage.
>>>   This same mechanism can also determine connection loss ("liveness")
>>>   with a remote peer.
>>>
>>>
>>>The IETF datatracker status page for this draft is:
>>>https://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshnes
>>>s/
>>>
>>>There's also a htmlized version available at:
>>>http://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-freshness-04
>>>
>>>A diff from the previous version is available at:
>>>http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-stun-consent-freshn=
es
>>>s-
>>>04
>>>
>>>
>>>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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On 06/28/2014 06:52 PM, Ram Mohan R (rmohanr) wrote:
> Hi Michael,
>
> Thanks for your review.
>
> We used inverted to convey that Consent response (STUN binding response)
> from remote peer must be  from inverted 5-tuple (source and dest addr
> swapped) of the request sent.
> I will see how to re-word it.

I think it will be fine if you just put in a definition for the term.
BTW, I'd like it to be an "inverse" 5-tuple, not an "inverted" one.

>
> Regards,
> Ram
>
> -----Original Message-----
> From: Michael Procter <michael@voip.co.uk>
> Date: Friday, 27 June 2014 3:36 pm
> To: Ram Mohan Ravindranath <rmohanr@cisco.com>
> Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
> Subject: Re: [rtcweb] I-D Action:
> draft-ietf-rtcweb-stun-consent-freshness-04.txt
>
>> Hi Ram,
>>
>> I have a couple of comments about the word "inverted".  In section
>> 4.1, the draft says:
>>
>>    receiving a
>>    matching, authenticated, non-error ICE binding response from the
>>    inverted remote peer's Transport Address.
>>
>> I think this means receiving it from the remote peer's Transport
>> Address, although the presence of 'inverted' made me wonder.  If the
>> word is unnecessary, please remove it.  If it is needed, please add
>> more explanation as to what it means.
>>
>> Further on, the draft says:
>>
>>    That is, if a valid STUN binding response [...]
>>    has not been received from the inverted 5-tuple, [...]
>>
>> I think this means receiving it from the 5-tuple, but again, the
>> presence of 'inverted' made me wonder.  Is this a naming issue?  I am
>> happy to send and receive on the same 5-tuple, and I expect my peer to
>> do the same.  However, my peer's 5-tuple will be inverted with respect
>> to mine!
>>
>> Regards,
>>
>> Michael
>>
>>
>>
>> On 18 June 2014 04:33, Ram Mohan R (rmohanr) <rmohanr@cisco.com> wrote:
>>> The new revision has:
>>>
>>> 1) Comments from Christer and Charles incorporated.
>>> 2) Incorporated the feedback received in the interim meeting in Section
>>> 4
>>> on re-transmissions, revocation of consent
>>> 3) Removed the Appendix that had a example as it was no different from
>>> what is mentioned in solution.
>>> 4) Updates to section 6 (DiffServ Treatment) based on discussions in the
>>> mailing list.
>>>
>>> Please review and provide your feedback.
>>>
>>> Regards,
>>> Ram
>>>
>>> -----Original Message-----
>>> From: "internet-drafts@ietf.org" <internet-drafts@ietf.org>
>>> Date: Wednesday, 18 June 2014 8:47 am
>>> To: "i-d-announce@ietf.org" <i-d-announce@ietf.org>
>>> Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
>>> Subject: [rtcweb] I-D Action:
>>> draft-ietf-rtcweb-stun-consent-freshness-04.txt
>>>
>>>> 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           : STUN Usage for Consent Freshness
>>>>         Authors         : Muthu Arul Mozhi Perumal
>>>>                           Dan Wing
>>>>                           Ram Mohan Ravindranath
>>>>                           Tirumaleswar Reddy
>>>>                           Martin Thomson
>>>>        Filename        : draft-ietf-rtcweb-stun-consent-freshness-04.txt
>>>>        Pages           : 9
>>>>        Date            : 2014-06-17
>>>>
>>>> Abstract:
>>>>    To prevent sending excessive traffic to an endpoint, periodic consent
>>>>    needs to be obtained from that remote endpoint.
>>>>
>>>>    This document describes a consent mechanism using a new STUN usage.
>>>>    This same mechanism can also determine connection loss ("liveness")
>>>>    with a remote peer.
>>>>
>>>>
>>>> The IETF datatracker status page for this draft is:
>>>> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshnes
>>>> s/
>>>>
>>>> There's also a htmlized version available at:
>>>> http://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-freshness-04
>>>>
>>>> A diff from the previous version is available at:
>>>> http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-stun-consent-freshnes
>>>> s-
>>>> 04
>>>>
>>>>
>>>> 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
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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References: <20140618031726.3502.90813.idtracker@ietfa.amsl.com> <CFC70488.91833%rmohanr@cisco.com> <CAPms+wQvnYguHTWr72YBqpJr4_4zzi=GFaW0cWgMvC1UB84+-A@mail.gmail.com> <CFD4EF5D.92AC7%rmohanr@cisco.com> <53AFCDA3.5000508@alvestrand.no>
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From: Michael Procter <michael@voip.co.uk>
To: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-04.txt
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On 29 June 2014 09:26, Harald Alvestrand <harald@alvestrand.no> wrote:
> I think it will be fine if you just put in a definition for the term.
> BTW, I'd like it to be an "inverse" 5-tuple, not an "inverted" one.

Adding a definition would be fine for me.  And "inverse" is better,
too.  I had spent some time peering at stun xor-mapped-addresses
before reading this draft, so I might have been overly sensitive to
the word 'inverted'!

Michael


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Dear all,

based on a comment from EKR, I would like to bring up a question:
RTCWeb uses DTLS for deriving the keys for SRTP on the one hand and
for transporting SCTP on the other hand.
The SRTP related documents refer to DTLS version 1.0 as specified in
http://tools.ietf.org/html/rfc4347
The SCTP over DTLS document currently refers to DTLS version 1.2 as =
specified in
http://tools.ietf.org/html/rfc6347

I think we need to use a single version of DTLS for RTCWeb. RFC 4347 is =
obsoleted
by RFC 6347. Wouldn't that imply that it makes sense to use DTLS 1.2 in =
all documents?

Best regards
Michael=


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--f46d04388e95b5564704fd164e6a
Content-Type: text/plain; charset=UTF-8

"Wouldn't that imply that it makes sense to use DTLS 1.2 in all documents?"

[BA] Not necessarily.  As I understand it, existing implementations only
support 1.0 - and if we were to mandate only 1.2 ciphersuites, then this
would create an interoperability problem.


On Mon, Jun 30, 2014 at 4:44 PM, Michael Tuexen <
Michael.Tuexen@lurchi.franken.de> wrote:

> Dear all,
>
> based on a comment from EKR, I would like to bring up a question:
> RTCWeb uses DTLS for deriving the keys for SRTP on the one hand and
> for transporting SCTP on the other hand.
> The SRTP related documents refer to DTLS version 1.0 as specified in
> http://tools.ietf.org/html/rfc4347
> The SCTP over DTLS document currently refers to DTLS version 1.2 as
> specified in
> http://tools.ietf.org/html/rfc6347
>
> I think we need to use a single version of DTLS for RTCWeb. RFC 4347 is
> obsoleted
> by RFC 6347. Wouldn't that imply that it makes sense to use DTLS 1.2 in
> all documents?
>
> Best regards
> Michael
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--f46d04388e95b5564704fd164e6a
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div>&quot;Wouldn&#39;t that imply that it makes sense to =
use DTLS 1.2 in all documents?&quot;<br><br></div>[BA] Not necessarily.=C2=
=A0 As I understand it, existing implementations only support 1.0 - and if =
we were to mandate only 1.2 ciphersuites, then this would create an interop=
erability problem. <br>

</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Mon,=
 Jun 30, 2014 at 4:44 PM, Michael Tuexen <span dir=3D"ltr">&lt;<a href=3D"m=
ailto:Michael.Tuexen@lurchi.franken.de" target=3D"_blank">Michael.Tuexen@lu=
rchi.franken.de</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Dear all,<br>
<br>
based on a comment from EKR, I would like to bring up a question:<br>
RTCWeb uses DTLS for deriving the keys for SRTP on the one hand and<br>
for transporting SCTP on the other hand.<br>
The SRTP related documents refer to DTLS version 1.0 as specified in<br>
<a href=3D"http://tools.ietf.org/html/rfc4347" target=3D"_blank">http://too=
ls.ietf.org/html/rfc4347</a><br>
The SCTP over DTLS document currently refers to DTLS version 1.2 as specifi=
ed in<br>
<a href=3D"http://tools.ietf.org/html/rfc6347" target=3D"_blank">http://too=
ls.ietf.org/html/rfc6347</a><br>
<br>
I think we need to use a single version of DTLS for RTCWeb. RFC 4347 is obs=
oleted<br>
by RFC 6347. Wouldn&#39;t that imply that it makes sense to use DTLS 1.2 in=
 all documents?<br>
<br>
Best regards<br>
Michael<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>

--f46d04388e95b5564704fd164e6a--


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Subject: Re: [rtcweb] DTLS version
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On 01 Jul 2014, at 01:48, Bernard Aboba <bernard.aboba@gmail.com> wrote:

> "Wouldn't that imply that it makes sense to use DTLS 1.2 in all =
documents?"
>=20
> [BA] Not necessarily.  As I understand it, existing implementations =
only support 1.0 - and if we were to mandate only 1.2 ciphersuites, then =
this would create an interoperability problem.=20
There are also several clarifications and corrections to handshake =
issues. This
really helps in interoperability....

Best regards
Michael
>=20
>=20
> On Mon, Jun 30, 2014 at 4:44 PM, Michael Tuexen =
<Michael.Tuexen@lurchi.franken.de> wrote:
> Dear all,
>=20
> based on a comment from EKR, I would like to bring up a question:
> RTCWeb uses DTLS for deriving the keys for SRTP on the one hand and
> for transporting SCTP on the other hand.
> The SRTP related documents refer to DTLS version 1.0 as specified in
> http://tools.ietf.org/html/rfc4347
> The SCTP over DTLS document currently refers to DTLS version 1.2 as =
specified in
> http://tools.ietf.org/html/rfc6347
>=20
> I think we need to use a single version of DTLS for RTCWeb. RFC 4347 =
is obsoleted
> by RFC 6347. Wouldn't that imply that it makes sense to use DTLS 1.2 =
in all documents?
>=20
> Best regards
> Michael
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


From nobody Mon Jun 30 17:21:00 2014
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From: Martin Thomson <martin.thomson@gmail.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] DTLS version
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On 30 June 2014 16:55, Michael Tuexen <Michael.Tuexen@lurchi.franken.de> wrote:
>> "Wouldn't that imply that it makes sense to use DTLS 1.2 in all documents?"
>>
>> [BA] Not necessarily.  As I understand it, existing implementations only support 1.0 - and if we were to mandate only 1.2 ciphersuites, then this would create an interoperability problem.
> There are also several clarifications and corrections to handshake issues. This
> really helps in interoperability....

Bernard is right.  But there are several advantages to an update to
referencing (D)TLS 1.2.  For one, we get better ciphersuites (AES-GCM
and all the AEAD modes).

That said, there are enough implementations out there already that
*requiring* 1.2 might be too aggressive.  I think that I'd prefer that
(so that we can make modern ciphersuite recommendations to fit), but
if someone has a strong need for 1.0, I don't have a problem
suggesting that 1.0 be accepted.

p.s., DTLS 1.2 is coming in Firefox.  As it turns out, NSS supported
it, but it hadn't been turned on.

