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Subject: [rtcweb] New Non-WG Mailing List: rtcweb -- Real-Time Communication in WEB-browsers (proposed) working group list
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A new IETF non-working group email list has been created.

List address: rtcweb@ietf.org
Archive: http://www.ietf.org/mail-archive/web/rtcweb/
To subscribe: https://www.ietf.org/mailman/listinfo/rtcweb

Description:  This proposed RTCWeb working group is discussing direct
interactive rich communication between two peers' web-browsers, enabling
audio, video, collaboration, games, etc. 

For additional information, please contact the list administrators.


From ietf@meetecho.com  Tue Apr  5 09:36:34 2011
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Subject: [rtcweb] RTCWEB recording available -- update
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Dear all,the full recording (synchronized video, audio, slides and jabber=
 room) of the RTCWEB BOF at IETF80 is available at the following URL:http=
://ietf.conf.meetecho.comYou can either choose to watch it online (two di=
fferent approaches are available) or download and replay it whenever you =
want.In case of problems with the playout, just drop an e-mail to ietf-su=
pport@meetecho.com.For the chair(s): please feel free to put the link to =
the recording in the minutes, if you think this might be useful.Cheers,th=
e Meetecho Team

--_=__=_XaM3_.1302021495.2A.350528.42.28319.52.42.007.1707768203
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=0A<div class=3D"xam_msg_class">=0A<div style=3D"font: normal 13px Arial;=
 color:rgb(0, 0, 0);">Dear all,<br><br>the full recording (synchronized v=
ideo, audio, slides and jabber room) of the RTCWEB BOF at IETF80 is avail=
able at the following URL:<br>http://ietf.conf.meetecho.com<br><br>You ca=
n either choose to watch it online (two different approaches are availabl=
e) or download and replay it whenever you want.<br><br>In case of problem=
s with the playout, just drop an e-mail to ietf-support@meetecho.com.<br>=
<br>For the chair(s): please feel free to put the link to the recording i=
n the minutes, if you think this might be useful.<br><br>Cheers,<br>the M=
eetecho Team<br></div>=0A</div>=0A

--_=__=_XaM3_.1302021495.2A.350528.42.28319.52.42.007.1707768203--


From harald@alvestrand.no  Fri Apr  8 04:54:21 2011
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Date: Fri, 08 Apr 2011 13:56:03 +0200
From: Harald Alvestrand <harald@alvestrand.no>
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There's always a quiet period when changing mailing lists.
Therefore, someone should send out a message that's provocative enough 
to get a few responses, so that everyone's filters get tuned.... here's 
my try.

Controversial things that need more discussions, culled from Friday's notes:

- Non-media data: RTP, UDP-without-RTP, TCP, or "something else"?
   Congestion control is obviously needed - both for this and for media, 
naturally.

- Mandatory to implement codecs: How can we figure out how to get the 
arguments documented, and make the call on whether to have them or not?

- Interim meeting:
When, with whom, and using what technology?

Enough ... that should set a few tongues wagging....


From tim@phonefromhere.com  Fri Apr  8 05:57:10 2011
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On 8 Apr 2011, at 12:56, Harald Alvestrand wrote:

> There's always a quiet period when changing mailing lists.
> Therefore, someone should send out a message that's provocative enough =
to get a few responses, so that everyone's filters get tuned.... here's =
my try.
>=20
> Controversial things that need more discussions, culled from Friday's =
notes:
>=20
> - Non-media data: RTP, UDP-without-RTP, TCP, or "something else"?
>  Congestion control is obviously needed - both for this and for media, =
naturally.
>=20
> - Mandatory to implement codecs: How can we figure out how to get the =
arguments documented, and make the call on whether to have them or not?
>=20
> - Interim meeting:
> When, with whom, and using what technology?
>=20
> Enough ... that should set a few tongues wagging....
>=20
> ________

Ok, I'll bite :-)

On the non-media data, I think it is somewhere between essential and =
_very_ useful.

Our experience of this is running a small java applet softphone that =
implements IAX.
We've been using the TEXT frame type to send data between the pbx and =
the browser.

IAX's TEXT (and in general non-media) frames are small (optionally =
encrypted), sequenced=20
and reliable, but carried over UDP.

Typically it is notification of who is speaking in a conference call, a =
slide change, transcription text
or IVR prompts in text form. So the fact that it is sent directly and =
not polled is essential.
Mostly we send JSON objects, and have the page's javascript interpret =
them as it sees fit.

I'm not saying we should use IAX, but these properties of the TEXT frame =
are very useful=20
in this context, so we could aim to reproduce those properties in any =
solution.

There are mechanisms for passing messages from one browser context to =
another,
see =
http://www.ajaxlines.com/ajax/stuff/article/using_html_postmessage.php - =
we should probably
be adopting that as an interface.

On the codec front, we've found SILK's adaptive nature really valuable =
in situations where a TDM codec
might fail - (e.g. user on the edge of a wifi network with high packet =
loss and variable bandwidth), so
we should aim to include a codec of that kind.

Tim.



From mary.ietf.barnes@gmail.com  Fri Apr  8 06:22:40 2011
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Date: Fri, 8 Apr 2011 08:24:21 -0500
Message-ID: <BANLkTimLf5qfsn1u_Ax9aPCS_kwvF236SQ@mail.gmail.com>
From: Mary Barnes <mary.ietf.barnes@gmail.com>
To: Harald Alvestrand <harald@alvestrand.no>
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--bcaec50165b1d3575b04a068241a
Content-Type: text/plain; charset=ISO-8859-1

One suggestion I have for the group is to please start separate threads on
each of the items as that will make it MUCH easier to track the discussion.

In the spirit of this suggestion, I'll post my comments on the codec issue
separately.

Regards,
Mary.



On Fri, Apr 8, 2011 at 6:56 AM, Harald Alvestrand <harald@alvestrand.no>wrote:

> There's always a quiet period when changing mailing lists.
> Therefore, someone should send out a message that's provocative enough to
> get a few responses, so that everyone's filters get tuned.... here's my try.
>
> Controversial things that need more discussions, culled from Friday's
> notes:
>
> - Non-media data: RTP, UDP-without-RTP, TCP, or "something else"?
>  Congestion control is obviously needed - both for this and for media,
> naturally.
>
> - Mandatory to implement codecs: How can we figure out how to get the
> arguments documented, and make the call on whether to have them or not?
>
> - Interim meeting:
> When, with whom, and using what technology?
>
> Enough ... that should set a few tongues wagging....
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--bcaec50165b1d3575b04a068241a
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

One suggestion I have for the group is to please start separate threads on =
each of the items as that will make it MUCH easier to track the discussion.=
=A0<div><br></div><div>In the spirit of this suggestion, I&#39;ll post my c=
omments on the codec issue separately.=A0</div>
<div><br></div><div>Regards,</div><div>Mary.<br><div><div><br></div><div><b=
r></div><div><br><div class=3D"gmail_quote">On Fri, Apr 8, 2011 at 6:56 AM,=
 Harald Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestran=
d.no">harald@alvestrand.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;">There&#39;s always a quiet period when chan=
ging mailing lists.<br>
Therefore, someone should send out a message that&#39;s provocative enough =
to get a few responses, so that everyone&#39;s filters get tuned.... here&#=
39;s my try.<br>
<br>
Controversial things that need more discussions, culled from Friday&#39;s n=
otes:<br>
<br>
- Non-media data: RTP, UDP-without-RTP, TCP, or &quot;something else&quot;?=
<br>
 =A0Congestion control is obviously needed - both for this and for media, n=
aturally.<br>
<br>
- Mandatory to implement codecs: How can we figure out how to get the argum=
ents documented, and make the call on whether to have them or not?<br>
<br>
- Interim meeting:<br>
When, with whom, and using what technology?<br>
<br>
Enough ... that should set a few tongues wagging....<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>

--bcaec50165b1d3575b04a068241a--

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On Fri, Apr 8, 2011 at 6:56 AM, Harald Alvestrand <harald@alvestrand.no>wrote:


> - Mandatory to implement codecs: How can we figure out how to get the
> arguments documented, and make the call on whether to have them or not?
>

[MB] Given the amount of time spent discussing codecs in the adhoc, I would
suggest that the group focus on the other issues for the time being, so that
some progress can be made initially and then revisit the codec issue once
some of the other decisions are more firm. [/MB]

Regards,
Mary.

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On Fri, Apr 8, 2011 at 6:56 AM, Harald Alvestrand <span dir=3D"ltr">&lt;<a =
href=3D"mailto:harald@alvestrand.no">harald@alvestrand.no</a>&gt;</span> wr=
ote:<br><div class=3D"gmail_quote"><div>=A0</div><blockquote class=3D"gmail=
_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:=
1ex;">

- Mandatory to implement codecs: How can we figure out how to get the argum=
ents documented, and make the call on whether to have them or not?<br></blo=
ckquote><div><br></div><div>[MB] Given the amount of time spent discussing =
codecs in the adhoc, I would suggest that the group focus on the other issu=
es for the time being, so that some progress can be made initially and then=
 revisit the codec issue once some of the other decisions are more firm. [/=
MB]</div>
<div><br></div><div>Regards,</div><div>Mary.=A0</div></div>

--20cf30434702e6e9c204a0682c12--

From lorenzo@meetecho.com  Fri Apr  8 06:27:05 2011
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Hi Harald and all,

a few comments inline.



Il giorno Fri, 08 Apr 2011 13:56:03 +0200
Harald Alvestrand <harald@alvestrand.no> ha scritto:

> There's always a quiet period when changing mailing lists.
> Therefore, someone should send out a message that's provocative
> enough to get a few responses, so that everyone's filters get
> tuned.... here's my try.
> 
> Controversial things that need more discussions, culled from Friday's
> notes:
> 
> - Non-media data: RTP, UDP-without-RTP, TCP, or "something else"?
>    Congestion control is obviously needed - both for this and for
> media, naturally.
> 


I guess that depends on what non-media is.

During the RTCWEB BOF we shared the session using a web-based interface
(some in this list may even have tried it), and it involved both media
(audio and video) and what we may call non-media (the jabber room, the
slides). All this non-media interaction, which was mostly based on
XMPP and a bit of HTTP, was just made available as a JavaScript API for
the webpage to exploit: asynchronous interaction was achieved the usual
way (one or more AJAX channels to send data, a long poll AJAX channel
to get updates) which I guess is the simpliest approach, from a
functional point of view, with respect to this kind of non-media.
Meaning I wouldn't go for a new protocol/solution or force RTP for
that as well, if avoidable.

For media itself, the approach we used so far was using another
JavaScript API to handle the SIP negotiation (masking all the SIP
details), and then relying on a Java applet to handle the RTP frames
and the encoding: I guess that the ability to handle also this last
frame of the picture within the browser would be the most
important outcome of the WG. What may also be interesting would be
debating whether or not an HTTP fallback (or something similar) should
be considered for RTP in constrained scenarios, as I seem to recall
having been mentioned during the BOF as well.


> - Mandatory to implement codecs: How can we figure out how to get the 
> arguments documented, and make the call on whether to have them or
> not?
> 


This is likely to become a hot debate: HTML5 video has thought us
there's no easy way out. As someone already told during the BOF, I'd go
for a very simple (even if not optimal) common ground as mandatory, and
leave advanced codec choices up to developers, rather than saying
"this is how you can negotiate stuff, which stuff is up to you". I have
no strong sentiment towards any specific choice, but I'd go for
something that's not (too?) closed. Forcing AMR and H.264 as mandatory,
for example, would very likely cut a lot of people out of the loop,
which I don't think is what anybody wants.


> - Interim meeting:
> When, with whom, and using what technology?
> 


If a virtual interim meeting is to be setup, you may count on us to
host and support it should you think it could be useful.



> Enough ... that should set a few tongues wagging....
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


L.


-- 
Lorenzo Miniero, COB

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

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>
>
>
> - Interim meeting:
> When, with whom, and using what technology?
>
[MB] I would suggest late May or early June would be a good time.  I had
suggested to the RAI area that folks provide the details for the Interims on
the RAI area wiki as I know there are several groups planning interims and
many of us need to participate in almost all of those:
http://trac.tools.ietf.org/area/rai/trac/wiki

I will note that not all the WG chairs have been adding the information, but
it is really helpful.  I add it for the WGs that I'm subscribed to when I
happen to notice (which is why I suggested the RAI area do this in the first
place because I was having difficulty keeping track of all the meetings -
there were at least 6 interim meetings in the RAI area between IETF-79 and
IETF-80).

As far as technology, I thought Meetecho worked extremely well at IETF-80,
although I'm not sure how well it would work with alot of geographic
separation.  The alternative of course is Webex.

And, as usual, I suggest setting up a doodle poll to find the best date/time
- it's good to include a link to that with a deadline in the wiki, as well.
[/MB]

Regards,
Mary.

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<div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><br>
<br>
- Interim meeting:<br>
When, with whom, and using what technology?<br></blockquote><div>[MB] I wou=
ld suggest late May or early June would be a good time. =A0I had suggested =
to the RAI area that folks provide the details for the Interims on the RAI =
area wiki as I know there are several groups planning interims and many of =
us need to participate in almost all of those: =A0</div>
<div><a href=3D"http://trac.tools.ietf.org/area/rai/trac/wiki">http://trac.=
tools.ietf.org/area/rai/trac/wiki</a></div><div><br></div><div>I will note =
that not all the WG chairs have been adding the information, but it is real=
ly helpful. =A0I add it for the WGs that I&#39;m subscribed to when I happe=
n to notice (which is why I suggested the RAI area do this in the first pla=
ce because I was having difficulty keeping track of all the meetings - ther=
e were at least 6 interim meetings in the RAI area between IETF-79 and IETF=
-80). =A0</div>
<div><br></div><div>As far as technology, I thought Meetecho worked extreme=
ly well at IETF-80, although I&#39;m not sure how well it would work with a=
lot of geographic separation. =A0The alternative of course is Webex.=A0</di=
v>
<div><br></div><div>And, as usual, I suggest setting up a doodle poll to fi=
nd the best date/time - it&#39;s good to include a link to that with a dead=
line in the wiki, as well.=A0</div><div>[/MB]</div><div><br></div><div>Rega=
rds,</div>
<div>Mary.</div></div>

--bcaec53f95ef67f20e04a0686028--

From singer@apple.com  Fri Apr  8 11:17:18 2011
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On Apr 8, 2011, at 4:56 , Harald Alvestrand wrote:

> 
> - Non-media data: RTP, UDP-without-RTP, TCP, or "something else"?
>  Congestion control is obviously needed - both for this and for media, naturally.
> 

My feeling is that we should not forget we're designing something to sit in a browser or browser-like context.  We *have* text chat tools, file upload and download tools, and so on, today, working in browsers.  Yes, they typically go through a server, and going point to point might be an optimization.  But that's the rub; I don't think we should spend a lot of time optimizing tools that are ancillary to the core problem of doing real-time a/v between two users. If someone wants to start a group looking at real-time point-to-point text chat, well, OK, but I think it's a distraction for us, at least initially.

David Singer
Multimedia and Software Standards, Apple Inc.


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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: David Singer <singer@apple.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Tue, 12 Apr 2011 09:19:49 +0200
Thread-Topic: [rtcweb] non-media data
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It is always good to try to limit scope, and I think this is indeed a case =
where we can useful limit scope to RTP-based media (audio, video, real-time=
 text...).

John
=20

> -----Original Message-----
> From: rtcweb-bounces@ietf.org=20
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of David Singer
> Sent: 08 April 2011 19:19
> To: rtcweb@ietf.org
> Subject: [rtcweb] non-media data
>=20
>=20
> On Apr 8, 2011, at 4:56 , Harald Alvestrand wrote:
>=20
> >=20
> > - Non-media data: RTP, UDP-without-RTP, TCP, or "something else"?
> >  Congestion control is obviously needed - both for this and=20
> for media, naturally.
> >=20
>=20
> My feeling is that we should not forget we're designing=20
> something to sit in a browser or browser-like context.  We=20
> *have* text chat tools, file upload and download tools, and=20
> so on, today, working in browsers.  Yes, they typically go=20
> through a server, and going point to point might be an=20
> optimization.  But that's the rub; I don't think we should=20
> spend a lot of time optimizing tools that are ancillary to=20
> the core problem of doing real-time a/v between two users. If=20
> someone wants to start a group looking at real-time=20
> point-to-point text chat, well, OK, but I think it's a=20
> distraction for us, at least initially.
>=20
> David Singer
> Multimedia and Software Standards, Apple Inc.
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =

From tim@phonefromhere.com  Tue Apr 12 01:40:23 2011
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To: "Elwell, John" <john.elwell@siemens-enterprise.com>
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Subject: Re: [rtcweb] non-media data
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Although we should aim to limit scope, it is also a good plan to listen =
to experience in the field.
Our experience is that direct, reliable, sequenced transmission of small =
messages between=20
endpoints hugely adds to the value of a web-embedded RTC component.

Typically these messages are intimately tied to events in the media =
engine (change of speaker,
end of message playback, transcription or recording available). (dynamic =
media meta-data if you like)

They are used to change the visual state of the web page (via =
javascript) so that (for example) the=20
web displayed IVR prompts stay in sync with the audio.

Whilst all of these are achievable via a more circuitous route =
(mediaserver->signalingserver->webserver->long poll-> browser)
the added latency will make the visual changes lag the audio which =
undermines the user experience.

The value in RTC-web isn't the realtime audio/video itself, it's the =
fact that it is an integral part of
the web-page, that integration is aided by non-media data.
Tim.


On 12 Apr 2011, at 08:19, Elwell, John wrote:

> It is always good to try to limit scope, and I think this is indeed a =
case where we can useful limit scope to RTP-based media (audio, video, =
real-time text...).
>=20
> John
>=20
>=20
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org=20
>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of David Singer
>> Sent: 08 April 2011 19:19
>> To: rtcweb@ietf.org
>> Subject: [rtcweb] non-media data
>>=20
>>=20
>> On Apr 8, 2011, at 4:56 , Harald Alvestrand wrote:
>>=20
>>>=20
>>> - Non-media data: RTP, UDP-without-RTP, TCP, or "something else"?
>>> Congestion control is obviously needed - both for this and=20
>> for media, naturally.
>>>=20
>>=20
>> My feeling is that we should not forget we're designing=20
>> something to sit in a browser or browser-like context.  We=20
>> *have* text chat tools, file upload and download tools, and=20
>> so on, today, working in browsers.  Yes, they typically go=20
>> through a server, and going point to point might be an=20
>> optimization.  But that's the rub; I don't think we should=20
>> spend a lot of time optimizing tools that are ancillary to=20
>> the core problem of doing real-time a/v between two users. If=20
>> someone wants to start a group looking at real-time=20
>> point-to-point text chat, well, OK, but I think it's a=20
>> distraction for us, at least initially.
>>=20
>> David Singer
>> Multimedia and Software Standards, Apple Inc.
>>=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


From harald@alvestrand.no  Tue Apr 12 05:29:16 2011
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Subject: [rtcweb] Fwd: W3C mailing-list [was Re: Updated W3C draft charter for Web RTC WG]
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This is a multi-part message in MIME format.
--------------080005070907060009030808
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This is to be the official public list of the (still in-formation) W3C 
WG for WebRTC.
Welcome!

                           Harald

-------- Original Message --------
Subject: 	W3C mailing-list [was Re: Updated W3C draft charter for Web 
RTC WG]
Date: 	Tue, 12 Apr 2011 08:48:09 +0200
From: 	Francois Daoust <fd@w3.org>
To: 	Harald Alvestrand <harald@alvestrand.no>



Hi Harald,

The public-webrtc@w3.org mailing-list is up and running.

To subscribe to the mailing-list, one needs to send an email to public-webrtc-request@w3.org with subject "subscribe", or click on the "subscribe to this list" option on the archives page.

The archives are available at:
  http://lists.w3.org/Archives/Public/public-webrtc/

Francois.





On 04/07/2011 11:27 AM, Francois Daoust wrote:
>  On 04/07/2011 10:50 AM, Harald Alvestrand wrote:
>>  Francios,
>>
>>  a) waiting for anything from me?
>
>  No, I'm waiting for replies from reviewers to my own replies [1] before I can get back to W3C management with a disposition of comments to ask for their formal approval of the changes and chairs nomination.
>
>  David Baron's already replied that he's ok with the proposed changes, so I'm waiting for Chaals feedback in particular. Not a big deal if we don't hear back from other reviewers.
>
>
>>  b) are things firm enough that we can create a public-webrtc-whatever@w3.org mailing list and start using it for API discussions?
>
>  That's usually done when the group gets created, but we can certainly do that before, yes. I've just requested the creation of a public-webrtc@w3.org mailing-list. It may take a few days for this request to be processed by the systems team.
>
>  Francois.
>
>  [1] http://www.w3.org/Search/Mail/Member/search?keywords=daoust&hdr-1-name=subject&hdr-1-query=real-time&index-grp=Member__FULL+Public__FULL&index-type=t&type-index=w3c-archive
>
>>
>>  Harald
>>
>>  On 04/04/11 11:11, Francois Daoust wrote:
>>>  Hi Harald,
>>>
>>>  It took some time for lack of time and connection last week, but I've eventually made a few changes to the W3C draft charter for Web RTC. See latest version at:
>>>  http://www.w3.org/2010/12/webrtc-charter.html
>>>
>>>  The diff is available at:
>>>  http://www.w3.org/2007/10/htmldiff?doc1=http%3A%2F%2Fwww.w3.org%2F2010%2F12%2Fwebrtc-charter-initial.html&doc2=http%3A%2F%2Fwww.w3.org%2F2010%2F12%2Fwebrtc-charter.html
>>>
>>>  List of changes:
>>>  - liaison with Web and TV IG (for home networking scenarios) added
>>>  - "any new" added to the mention that audio and video codecs are out of scope to respond to Art Barstow's comment, as that seemed benign enough.
>>>  - milestones section completed. Let me know if you have comments on this initial roadmap. I've merged all functions into one single row to keep things simple.
>>>  - decision policy copied from HTML charter
>>>  - deliverables section completed to put more emphasis on collaboration with IETF without mandating any specific text, using similar wording as that of the latest IETF RTCWeb group's charter.
>>>
>>>  (Note the diff tool loses itself near the end: only the decision policy section has been changed)
>>>
>>>  I stayed silent on the relation with WHATWG, and also did not introduce any change in the charter to address Stefan's detailed comments (e.g. "There must be support for media format negotiation and selection in the solution") to leave that as discussion for the group.
>>>
>>>  Could you review the changes and let me know whether that looks good for you?
>>>  I've read the BoF notes taken by Christer Holmberg. Any other input from the IETF BoF meeting that would trigger additional changes in the charter?
>>>
>>>  If you're fine with the changes, I'll get back to reviewers to get their approval, and to W3C Management with a disposition of comments to have them approve the charter and the WG chairs. Depending on the time it take to get reviewers approval, the decision could be taken this week, next week or the week after that (on Wednesday, each time). The group's creation can be officially announced shortly after that.
>>>
>>>  Do you have a timeline for the creation of the IETF group? Having both groups created at the same time might be good if that's not too much of a pain to do.
>>>
>>>  Dan Burnett proposed himself as co-chair for the Web RTC group. Dan has a good experience of W3C groups and is already chair of the Voice Browser working group and of the HTML Speech Incubator Group. On the minus side, he already has quite a few W3C activities on his plate. I do not know well Dan and Stefan. It doesn't make sense to have 3 chairs for the group. Do you have a strong preference?
>>>
>>>  Who would be the chairs of the IETF working group?
>>>
>>>  Thanks,
>>>  Francois.
>>>
>>
>>


--------------080005070907060009030808
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
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    <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
  </head>
  <body bgcolor="#ffffff" text="#000000">
    This is to be the official public list of the (still in-formation)
    W3C WG for WebRTC.<br>
    Welcome!<br>
    <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
    <br>
    -------- Original Message --------
    <table class="moz-email-headers-table" border="0" cellpadding="0"
      cellspacing="0">
      <tbody>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject: </th>
          <td>W3C mailing-list [was Re: Updated W3C draft charter for
            Web RTC WG]</td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
          <td>Tue, 12 Apr 2011 08:48:09 +0200</td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From: </th>
          <td>Francois Daoust <a class="moz-txt-link-rfc2396E" href="mailto:fd@w3.org">&lt;fd@w3.org&gt;</a></td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To: </th>
          <td>Harald 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>Hi Harald,

The <a class="moz-txt-link-abbreviated" href="mailto:public-webrtc@w3.org">public-webrtc@w3.org</a> mailing-list is up and running.

To subscribe to the mailing-list, one needs to send an email to <a class="moz-txt-link-abbreviated" href="mailto:public-webrtc-request@w3.org">public-webrtc-request@w3.org</a> with subject "subscribe", or click on the "subscribe to this list" option on the archives page.

The archives are available at:
 <a class="moz-txt-link-freetext" href="http://lists.w3.org/Archives/Public/public-webrtc/">http://lists.w3.org/Archives/Public/public-webrtc/</a>

Francois.





On 04/07/2011 11:27 AM, Francois Daoust wrote:
&gt; On 04/07/2011 10:50 AM, Harald Alvestrand wrote:
&gt;&gt; Francios,
&gt;&gt;
&gt;&gt; a) waiting for anything from me?
&gt;
&gt; No, I'm waiting for replies from reviewers to my own replies [1] before I can get back to W3C management with a disposition of comments to ask for their formal approval of the changes and chairs nomination.
&gt;
&gt; David Baron's already replied that he's ok with the proposed changes, so I'm waiting for Chaals feedback in particular. Not a big deal if we don't hear back from other reviewers.
&gt;
&gt;
&gt;&gt; b) are things firm enough that we can create a <a class="moz-txt-link-abbreviated" href="mailto:public-webrtc-whatever@w3.org">public-webrtc-whatever@w3.org</a> mailing list and start using it for API discussions?
&gt;
&gt; That's usually done when the group gets created, but we can certainly do that before, yes. I've just requested the creation of a <a class="moz-txt-link-abbreviated" href="mailto:public-webrtc@w3.org">public-webrtc@w3.org</a> mailing-list. It may take a few days for this request to be processed by the systems team.
&gt;
&gt; Francois.
&gt;
&gt; [1] <a class="moz-txt-link-freetext" href="http://www.w3.org/Search/Mail/Member/search?keywords=daoust&amp;hdr-1-name=subject&amp;hdr-1-query=real-time&amp;index-grp=Member__FULL+Public__FULL&amp;index-type=t&amp;type-index=w3c-archive">http://www.w3.org/Search/Mail/Member/search?keywords=daoust&amp;hdr-1-name=subject&amp;hdr-1-query=real-time&amp;index-grp=Member__FULL+Public__FULL&amp;index-type=t&amp;type-index=w3c-archive</a>
&gt;
&gt;&gt;
&gt;&gt; Harald
&gt;&gt;
&gt;&gt; On 04/04/11 11:11, Francois Daoust wrote:
&gt;&gt;&gt; Hi Harald,
&gt;&gt;&gt;
&gt;&gt;&gt; It took some time for lack of time and connection last week, but I've eventually made a few changes to the W3C draft charter for Web RTC. See latest version at:
&gt;&gt;&gt; <a class="moz-txt-link-freetext" href="http://www.w3.org/2010/12/webrtc-charter.html">http://www.w3.org/2010/12/webrtc-charter.html</a>
&gt;&gt;&gt;
&gt;&gt;&gt; The diff is available at:
&gt;&gt;&gt; <a class="moz-txt-link-freetext" href="http://www.w3.org/2007/10/htmldiff?doc1=http%3A%2F%2Fwww.w3.org%2F2010%2F12%2Fwebrtc-charter-initial.html&amp;doc2=http%3A%2F%2Fwww.w3.org%2F2010%2F12%2Fwebrtc-charter.html">http://www.w3.org/2007/10/htmldiff?doc1=http%3A%2F%2Fwww.w3.org%2F2010%2F12%2Fwebrtc-charter-initial.html&amp;doc2=http%3A%2F%2Fwww.w3.org%2F2010%2F12%2Fwebrtc-charter.html</a>
&gt;&gt;&gt;
&gt;&gt;&gt; List of changes:
&gt;&gt;&gt; - liaison with Web and TV IG (for home networking scenarios) added
&gt;&gt;&gt; - "any new" added to the mention that audio and video codecs are out of scope to respond to Art Barstow's comment, as that seemed benign enough.
&gt;&gt;&gt; - milestones section completed. Let me know if you have comments on this initial roadmap. I've merged all functions into one single row to keep things simple.
&gt;&gt;&gt; - decision policy copied from HTML charter
&gt;&gt;&gt; - deliverables section completed to put more emphasis on collaboration with IETF without mandating any specific text, using similar wording as that of the latest IETF RTCWeb group's charter.
&gt;&gt;&gt;
&gt;&gt;&gt; (Note the diff tool loses itself near the end: only the decision policy section has been changed)
&gt;&gt;&gt;
&gt;&gt;&gt; I stayed silent on the relation with WHATWG, and also did not introduce any change in the charter to address Stefan's detailed comments (e.g. "There must be support for media format negotiation and selection in the solution") to leave that as discussion for the group.
&gt;&gt;&gt;
&gt;&gt;&gt; Could you review the changes and let me know whether that looks good for you?
&gt;&gt;&gt; I've read the BoF notes taken by Christer Holmberg. Any other input from the IETF BoF meeting that would trigger additional changes in the charter?
&gt;&gt;&gt;
&gt;&gt;&gt; If you're fine with the changes, I'll get back to reviewers to get their approval, and to W3C Management with a disposition of comments to have them approve the charter and the WG chairs. Depending on the time it take to get reviewers approval, the decision could be taken this week, next week or the week after that (on Wednesday, each time). The group's creation can be officially announced shortly after that.
&gt;&gt;&gt;
&gt;&gt;&gt; Do you have a timeline for the creation of the IETF group? Having both groups created at the same time might be good if that's not too much of a pain to do.
&gt;&gt;&gt;
&gt;&gt;&gt; Dan Burnett proposed himself as co-chair for the Web RTC group. Dan has a good experience of W3C groups and is already chair of the Voice Browser working group and of the HTML Speech Incubator Group. On the minus side, he already has quite a few W3C activities on his plate. I do not know well Dan and Stefan. It doesn't make sense to have 3 chairs for the group. Do you have a strong preference?
&gt;&gt;&gt;
&gt;&gt;&gt; Who would be the chairs of the IETF working group?
&gt;&gt;&gt;
&gt;&gt;&gt; Thanks,
&gt;&gt;&gt; Francois.
&gt;&gt;&gt;
&gt;&gt;
&gt;&gt;

</pre>
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Harald,

With 3 "RTCWEB" mailing lists now in existence (the original "rtc-web", =
IETF "rtcweb" and now W3C "webrtc"), I get the sense that those of us =
who want to monitor what's going on and participate as we can should =
join ALL three of the lists, correct?  =20

This "rtc-web" list will continue to be for the overall effort, with the =
IETF and W3C lists being used for work within those bodies. =20

Or do you see this "rtc-web" list fading away in preference to the IETF =
and W3C lists?

Just trying to figure how many different lists I need to be on.

Thanks,
Dan


On Apr 12, 2011, at 8:29 AM, Harald Alvestrand wrote:

> This is to be the official public list of the (still in-formation) W3C =
WG for WebRTC.
> Welcome!
>=20
>                           Harald
>=20
> -------- Original Message --------
> Subject:	W3C mailing-list [was Re: Updated W3C draft charter for =
Web RTC WG]
> Date:	Tue, 12 Apr 2011 08:48:09 +0200
> From:	Francois Daoust <fd@w3.org>
> To:	Harald Alvestrand <harald@alvestrand.no>
>=20
> Hi Harald,
>=20
> The public-webrtc@w3.org mailing-list is up and running.
>=20
> To subscribe to the mailing-list, one needs to send an email to =
public-webrtc-request@w3.org with subject "subscribe", or click on the =
"subscribe to this list" option on the archives page.
>=20
> The archives are available at:
>  http://lists.w3.org/Archives/Public/public-webrtc/
>=20
> Francois.

--=20
Dan York, Director of Conversations
Voxeo Corporation   http://www.voxeo.com  dyork@voxeo.com
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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
">Harald,<div><br></div><div>With 3 "RTCWEB" mailing lists now in =
existence (the original "rtc-web", IETF "rtcweb" and now W3C "webrtc"), =
I get the sense that those of us who want to monitor what's going on and =
participate as we can should join ALL three of the lists, correct? =
&nbsp;&nbsp;</div><div><br></div><div>This "rtc-web" list will continue =
to be for the overall effort, with the IETF and W3C lists being used for =
work within those bodies. &nbsp;</div><div><br></div><div>Or do you see =
this "rtc-web" list fading away in preference to the IETF and W3C =
lists?</div><div><br></div><div>Just trying to figure how many different =
lists I need to be =
on.</div><div><br></div><div>Thanks,</div><div>Dan</div><div><br></div><di=
v><br><div><div>On Apr 12, 2011, at 8:29 AM, Harald Alvestrand =
wrote:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite">
<div bgcolor=3D"#ffffff" text=3D"#000000">
    This is to be the official public list of the (still in-formation)
    W3C WG for WebRTC.<br>
    Welcome!<br>
    <br>
    =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; Harald<br>
    <br>
    -------- Original Message --------
    <table class=3D"moz-email-headers-table" border=3D"0" =
cellpadding=3D"0" cellspacing=3D"0">
      <tbody>
        <tr>
          <th align=3D"RIGHT" nowrap=3D"nowrap" =
valign=3D"BASELINE">Subject: </th>
          <td>W3C mailing-list [was Re: Updated W3C draft charter for
            Web RTC WG]</td>
        </tr>
        <tr>
          <th align=3D"RIGHT" nowrap=3D"nowrap" valign=3D"BASELINE">Date: =
</th>
          <td>Tue, 12 Apr 2011 08:48:09 +0200</td>
        </tr>
        <tr>
          <th align=3D"RIGHT" nowrap=3D"nowrap" valign=3D"BASELINE">From: =
</th>
          <td>Francois Daoust <a class=3D"moz-txt-link-rfc2396E" =
href=3D"mailto:fd@w3.org">&lt;fd@w3.org&gt;</a></td>
        </tr>
        <tr>
          <th align=3D"RIGHT" nowrap=3D"nowrap" valign=3D"BASELINE">To: =
</th>
          <td>Harald Alvestrand <a class=3D"moz-txt-link-rfc2396E" =
href=3D"mailto:harald@alvestrand.no">&lt;harald@alvestrand.no&gt;</a></td>=

        </tr>
      </tbody>
    </table>
    <br>
    <br>
    <pre>Hi Harald,

The <a class=3D"moz-txt-link-abbreviated" =
href=3D"mailto:public-webrtc@w3.org">public-webrtc@w3.org</a> =
mailing-list is up and running.

To subscribe to the mailing-list, one needs to send an email to <a =
class=3D"moz-txt-link-abbreviated" =
href=3D"mailto:public-webrtc-request@w3.org">public-webrtc-request@w3.org<=
/a> with subject "subscribe", or click on the "subscribe to this list" =
option on the archives page.

The archives are available at:
 <a class=3D"moz-txt-link-freetext" =
href=3D"http://lists.w3.org/Archives/Public/public-webrtc/">http://lists.w=
3.org/Archives/Public/public-webrtc/</a>

Francois.
</pre></div></blockquote></div><br><div>
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auto; -webkit-text-stroke-width: 0px; "><div style=3D"word-wrap: =
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after-white-space; "><span class=3D"Apple-style-span" =
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0px; -webkit-border-vertical-spacing: 0px; color: rgb(0, 0, 0); =
font-family: Helvetica; font-size: 12px; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
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">--&nbsp;</div><div style=3D"margin-top: 0px; margin-right: 0px; =
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Director of Conversations</div><div style=3D"margin-top: 0px; =
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On 04/12/11 14:46, Dan York wrote:
> Harald,
>
> With 3 "RTCWEB" mailing lists now in existence (the original 
> "rtc-web", IETF "rtcweb" and now W3C "webrtc"), I get the sense that 
> those of us who want to monitor what's going on and participate as we 
> can should join ALL three of the lists, correct?
>
> This "rtc-web" list will continue to be for the overall effort, with 
> the IETF and W3C lists being used for work within those bodies.
>
> Or do you see this "rtc-web" list fading away in preference to the 
> IETF and W3C lists?
The ADs in the IETF have stated a preference for using the IETF list 
rather than this private list; it's somewhat clearer that this list's 
mail is an official IETF activity and operates under IETF IPR rules, and 
we don't have to worry about the quality of management at my private server.

Somewhat similar considerations apply to the W3C.

I would also prefer to not have to run this list on my private box 
forever, so I'm in favour of the "rtc-web" list fading out of use as the 
permanent homes for the work get set up.

So I'm hoping to get down to two: rtcweb@ietf.org and 
public-webrtc@w3.org. Some people might have interests that are 
specialized enough to want to be on only one of those.

(We also have a thread about an RTCWeb-related API running on the whatwg 
mailing list - that's a high traffic list, some might not want to sign 
up on that one for that reason.)
>
> Just trying to figure how many different lists I need to be on.
>
> Thanks,
> Dan
>
>
> On Apr 12, 2011, at 8:29 AM, Harald Alvestrand wrote:
>
>> This is to be the official public list of the (still in-formation) 
>> W3C WG for WebRTC.
>> Welcome!
>>
>>                           Harald
>>
>> -------- Original Message --------
>> Subject: 	W3C mailing-list [was Re: Updated W3C draft charter for Web 
>> RTC WG]
>> Date: 	Tue, 12 Apr 2011 08:48:09 +0200
>> From: 	Francois Daoust <fd@w3.org>
>> To: 	Harald Alvestrand <harald@alvestrand.no>
>>
>>
>>
>> Hi Harald,
>>
>> Thepublic-webrtc@w3.org  mailing-list is up and running.
>>
>> To subscribe to the mailing-list, one needs to send an email topublic-webrtc-request@w3.org  with subject "subscribe", or click on the "subscribe to this list" option on the archives page.
>>
>> The archives are available at:
>>   http://lists.w3.org/Archives/Public/public-webrtc/
>>
>> Francois.
>
> -- 
> Dan York, Director of Conversations
> Voxeo Corporation http://www.voxeo.com dyork@voxeo.com 
> <mailto:dyork@voxeo.com>
> Phone: +1-407-455-5859    Skype: danyork
>
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    On 04/12/11 14:46, Dan York wrote:
    <blockquote
      cite="mid:19F5C26C-7EE3-47F8-A5BB-0D537D1B5444@voxeo.com"
      type="cite">Harald,
      <div><br>
      </div>
      <div>With 3 "RTCWEB" mailing lists now in existence (the original
        "rtc-web", IETF "rtcweb" and now W3C "webrtc"), I get the sense
        that those of us who want to monitor what's going on and
        participate as we can should join ALL three of the lists,
        correct? &nbsp;&nbsp;</div>
      <div><br>
      </div>
      <div>This "rtc-web" list will continue to be for the overall
        effort, with the IETF and W3C lists being used for work within
        those bodies. &nbsp;</div>
      <div><br>
      </div>
      <div>Or do you see this "rtc-web" list fading away in preference
        to the IETF and W3C lists?</div>
    </blockquote>
    The ADs in the IETF have stated a preference for using the IETF list
    rather than this private list; it's somewhat clearer that this
    list's mail is an official IETF activity and operates under IETF IPR
    rules, and we don't have to worry about the quality of management at
    my private server.<br>
    <br>
    Somewhat similar considerations apply to the W3C.<br>
    <br>
    I would also prefer to not have to run this list on my private box
    forever, so I'm in favour of the "rtc-web" list fading out of use as
    the permanent homes for the work get set up.<br>
    <br>
    So I'm hoping to get down to two: <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a> and
    <a class="moz-txt-link-abbreviated" href="mailto:public-webrtc@w3.org">public-webrtc@w3.org</a>. Some people might have interests that are
    specialized enough to want to be on only one of those.<br>
    <br>
    (We also have a thread about an RTCWeb-related API running on the
    whatwg mailing list - that's a high traffic list, some might not
    want to sign up on that one for that reason.)<br>
    <blockquote
      cite="mid:19F5C26C-7EE3-47F8-A5BB-0D537D1B5444@voxeo.com"
      type="cite">
      <div><br>
      </div>
      <div>Just trying to figure how many different lists I need to be
        on.</div>
      <div><br>
      </div>
      <div>Thanks,</div>
      <div>Dan</div>
      <div><br>
      </div>
      <div><br>
        <div>
          <div>On Apr 12, 2011, at 8:29 AM, Harald Alvestrand wrote:</div>
          <br class="Apple-interchange-newline">
          <blockquote type="cite">
            <div bgcolor="#ffffff" text="#000000"> This is to be the
              official public list of the (still in-formation) W3C WG
              for WebRTC.<br>
              Welcome!<br>
              <br>
              &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
              <br>
              -------- Original Message --------
              <table class="moz-email-headers-table" border="0"
                cellpadding="0" cellspacing="0">
                <tbody>
                  <tr>
                    <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject:
                    </th>
                    <td>W3C mailing-list [was Re: Updated W3C draft
                      charter for Web RTC WG]</td>
                  </tr>
                  <tr>
                    <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date:
                    </th>
                    <td>Tue, 12 Apr 2011 08:48:09 +0200</td>
                  </tr>
                  <tr>
                    <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From:
                    </th>
                    <td>Francois Daoust <a moz-do-not-send="true"
                        class="moz-txt-link-rfc2396E"
                        href="mailto:fd@w3.org">&lt;fd@w3.org&gt;</a></td>
                  </tr>
                  <tr>
                    <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To:
                    </th>
                    <td>Harald Alvestrand <a moz-do-not-send="true"
                        class="moz-txt-link-rfc2396E"
                        href="mailto:harald@alvestrand.no">&lt;harald@alvestrand.no&gt;</a></td>
                  </tr>
                </tbody>
              </table>
              <br>
              <br>
              <pre>Hi Harald,

The <a moz-do-not-send="true" class="moz-txt-link-abbreviated" href="mailto:public-webrtc@w3.org">public-webrtc@w3.org</a> mailing-list is up and running.

To subscribe to the mailing-list, one needs to send an email to <a moz-do-not-send="true" class="moz-txt-link-abbreviated" href="mailto:public-webrtc-request@w3.org">public-webrtc-request@w3.org</a> with subject "subscribe", or click on the "subscribe to this list" option on the archives page.

The archives are available at:
 <a moz-do-not-send="true" class="moz-txt-link-freetext" href="http://lists.w3.org/Archives/Public/public-webrtc/">http://lists.w3.org/Archives/Public/public-webrtc/</a>

Francois.
</pre>
            </div>
          </blockquote>
        </div>
        <br>
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                                              class="Apple-style-span"
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                                              <div style="margin: 0px;"><span
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                                                  <div style="margin:
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                                                  <div style="margin:
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                                                    Director of
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                                                  <div style="margin:
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From singer@apple.com  Tue Apr 12 11:28:55 2011
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From: David Singer <singer@apple.com>
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On Apr 12, 2011, at 1:40 , Tim Panton wrote:

> Although we should aim to limit scope, it is also a good plan to listen to experience in the field.
> Our experience is that direct, reliable, sequenced transmission of small messages between 
> endpoints hugely adds to the value of a web-embedded RTC component.
> 
> Typically these messages are intimately tied to events in the media engine (change of speaker,
> end of message playback, transcription or recording available). (dynamic media meta-data if you like)
> 
> They are used to change the visual state of the web page (via javascript) so that (for example) the 
> web displayed IVR prompts stay in sync with the audio.
> 
> Whilst all of these are achievable via a more circuitous route (mediaserver->signalingserver->webserver->long poll-> browser)
> the added latency will make the visual changes lag the audio which undermines the user experience.
> 
> The value in RTC-web isn't the realtime audio/video itself, it's the fact that it is an integral part of
> the web-page, that integration is aided by non-media data.
> Tim.
> 

OK, this is fine, but having arbitrary numbers of streams of arbitrary media types is different from designing chat sessions or file transfer as 'part of' the tool.

> 
> On 12 Apr 2011, at 08:19, Elwell, John wrote:
> 
>> It is always good to try to limit scope, and I think this is indeed a case where we can useful limit scope to RTP-based media (audio, video, real-time text...).
>> 
>> John
>> 
>> 
>>> -----Original Message-----
>>> From: rtcweb-bounces@ietf.org 
>>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of David Singer
>>> Sent: 08 April 2011 19:19
>>> To: rtcweb@ietf.org
>>> Subject: [rtcweb] non-media data
>>> 
>>> 
>>> On Apr 8, 2011, at 4:56 , Harald Alvestrand wrote:
>>> 
>>>> 
>>>> - Non-media data: RTP, UDP-without-RTP, TCP, or "something else"?
>>>> Congestion control is obviously needed - both for this and 
>>> for media, naturally.
>>>> 
>>> 
>>> My feeling is that we should not forget we're designing 
>>> something to sit in a browser or browser-like context.  We 
>>> *have* text chat tools, file upload and download tools, and 
>>> so on, today, working in browsers.  Yes, they typically go 
>>> through a server, and going point to point might be an 
>>> optimization.  But that's the rub; I don't think we should 
>>> spend a lot of time optimizing tools that are ancillary to 
>>> the core problem of doing real-time a/v between two users. If 
>>> someone wants to start a group looking at real-time 
>>> point-to-point text chat, well, OK, but I think it's a 
>>> distraction for us, at least initially.
>>> 
>>> David Singer
>>> Multimedia and Software Standards, Apple Inc.
>>> 
>>> _______________________________________________
>>> 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
> 

David Singer
Multimedia and Software Standards, Apple Inc.


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On 4/8/2011 11:19 AM, David Singer wrote:
> My feeling is that we should not forget we're designing something to sit in a browser or browser-like context.  We*have*  text chat tools, file upload and download tools, and so on, today, working in browsers.  Yes, they typically go through a server, and going point to point might be an optimization.  But that's the rub; I don't think we should spend a lot of time optimizing tools that are ancillary to the core problem of doing real-time a/v between two users. If someone wants to start a group looking at real-time point-to-point text chat, well, OK, but I think it's a distraction for us, at least initially.
>

I'd also like to point out that at Mozilla we want to see the browser 
change.  What you're describing is the way that browsers work today, but 
I think we're ready to move beyond that model to something that's 
app-driven, including point to point connections for data as well as 
media.  Latency isn't just an excuse to do point to point, but is an end 
unto itself.

John Elwell points out that limiting scope is important, and I agree.  
But I want to make sure that any APIs or protocols that get designed can 
handle this case as well.

--Chris

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      http-equiv="Content-Type">
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    On 4/8/2011 11:19 AM, David Singer wrote:
    <blockquote
      cite="mid:F4467DDA-816F-4DDD-B66F-8A48F9A0A16C@apple.com"
      type="cite">
      <pre wrap="">My feeling is that we should not forget we're designing something to sit in a browser or browser-like context.  We <b class="moz-txt-star"><span class="moz-txt-tag">*</span>have<span class="moz-txt-tag">*</span></b> text chat tools, file upload and download tools, and so on, today, working in browsers.  Yes, they typically go through a server, and going point to point might be an optimization.  But that's the rub; I don't think we should spend a lot of time optimizing tools that are ancillary to the core problem of doing real-time a/v between two users. If someone wants to start a group looking at real-time point-to-point text chat, well, OK, but I think it's a distraction for us, at least initially.

</pre>
    </blockquote>
    <br>
    I'd also like to point out that at Mozilla we want to see the
    browser change.&nbsp; What you're describing is the way that browsers
    work today, but I think we're ready to move beyond that model to
    something that's app-driven, including point to point connections
    for data as well as media.&nbsp; Latency isn't just an excuse to do point
    to point, but is an end unto itself.<br>
    <br>
    John Elwell points out that limiting scope is important, and I
    agree.&nbsp; But I want to make sure that any APIs or protocols that get
    designed can handle this case as well.<br>
    <br>
    --Chris<br>
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From harald@alvestrand.no  Wed Apr 13 01:43:51 2011
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On 04/08/11 20:19, David Singer wrote:
> On Apr 8, 2011, at 4:56 , Harald Alvestrand wrote:
>
>> - Non-media data: RTP, UDP-without-RTP, TCP, or "something else"?
>>   Congestion control is obviously needed - both for this and for media, naturally.
>>
> My feeling is that we should not forget we're designing something to sit in a browser or browser-like context.  We *have* text chat tools, file upload and download tools, and so on, today, working in browsers.  Yes, they typically go through a server, and going point to point might be an optimization.  But that's the rub; I don't think we should spend a lot of time optimizing tools that are ancillary to the core problem of doing real-time a/v between two users. If someone wants to start a group looking at real-time point-to-point text chat, well, OK, but I think it's a distraction for us, at least initially.
One reason it's somewhat urgent, organizationally, is that there is an 
effort being pursued by some participants in WHATWG to define a protocol 
for data transfer in the RTCWeb context.

http://www.whatwg.org/specs/web-apps/current-work/multipage/dnd.html#the-data-stream 
is the link.

I'd like to make sure we don't end up in a situation where there's a 
thought-through protocol defined by the IETF and a non-thought-through 
protocol that "everyone" implements "just because the spec was there". 
Ideally, I'd like one protocol referenced by both contexts that is well 
designed.

                                   Harald


From stefan.lk.hakansson@ericsson.com  Thu Apr 14 06:37:05 2011
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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: Tim Panton <tim@phonefromhere.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>
Date: Thu, 14 Apr 2011 15:37:01 +0200
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I also think a direct "small message" channel would be very useful: for enh=
ancing conferencing-like services as discussed below as well as for gaming =
(where latency can be extremely important). And once it is available I'm su=
re developers will find other, more innovative, ways to use it!

Stefan

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Tim Panton
Sent: den 12 april 2011 10:40
To: Elwell, John
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] non-media data

Although we should aim to limit scope, it is also a good plan to listen to =
experience in the field.
Our experience is that direct, reliable, sequenced transmission of small me=
ssages between endpoints hugely adds to the value of a web-embedded RTC com=
ponent.

Typically these messages are intimately tied to events in the media engine =
(change of speaker, end of message playback, transcription or recording ava=
ilable). (dynamic media meta-data if you like)

They are used to change the visual state of the web page (via javascript) s=
o that (for example) the web displayed IVR prompts stay in sync with the au=
dio.

Whilst all of these are achievable via a more circuitous route (mediaserver=
->signalingserver->webserver->long poll-> browser) the added latency will m=
ake the visual changes lag the audio which undermines the user experience.

The value in RTC-web isn't the realtime audio/video itself, it's the fact t=
hat it is an integral part of the web-page, that integration is aided by no=
n-media data.
Tim.


On 12 Apr 2011, at 08:19, Elwell, John wrote:

> It is always good to try to limit scope, and I think this is indeed a cas=
e where we can useful limit scope to RTP-based media (audio, video, real-ti=
me text...).
>=20
> John
>=20
>=20
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org
>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of David Singer
>> Sent: 08 April 2011 19:19
>> To: rtcweb@ietf.org
>> Subject: [rtcweb] non-media data
>>=20
>>=20
>> On Apr 8, 2011, at 4:56 , Harald Alvestrand wrote:
>>=20
>>>=20
>>> - Non-media data: RTP, UDP-without-RTP, TCP, or "something else"?
>>> Congestion control is obviously needed - both for this and
>> for media, naturally.
>>>=20
>>=20
>> My feeling is that we should not forget we're designing something to=20
>> sit in a browser or browser-like context.  We
>> *have* text chat tools, file upload and download tools, and so on,=20
>> today, working in browsers.  Yes, they typically go through a server,=20
>> and going point to point might be an optimization.  But that's the=20
>> rub; I don't think we should spend a lot of time optimizing tools=20
>> that are ancillary to the core problem of doing real-time a/v between=20
>> two users. If someone wants to start a group looking at real-time=20
>> point-to-point text chat, well, OK, but I think it's a distraction=20
>> for us, at least initially.
>>=20
>> David Singer
>> Multimedia and Software Standards, Apple Inc.
>>=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

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

From hoene@uni-tuebingen.de  Thu Apr 14 07:00:10 2011
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Hi,

I was wondering whether somebody plan to add a 3D rendering plugin into the
RTC-web browser?
Any plan about this or is it out of scope?

With best regards,
 Christian Hoene



From magnus.westerlund@ericsson.com  Thu Apr 14 07:02:14 2011
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Christian Hoene skrev 2011-04-14 16:00:
> Hi,
> 
> I was wondering whether somebody plan to add a 3D rendering plugin into the
> RTC-web browser?
> Any plan about this or is it out of scope?

Are there any additional requirements on RTCWEB than seeing it as an
media type with some media type specific signalling, most likely as part
of the RTP payload type used to carry the data between the peers?

Cheers

Magnus Westerlund

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

From singer@apple.com  Thu Apr 14 08:35:05 2011
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On Apr 14, 2011, at 7:02 , Magnus Westerlund wrote:

> Christian Hoene skrev 2011-04-14 16:00:
>> Hi,
>> 
>> I was wondering whether somebody plan to add a 3D rendering plugin into the
>> RTC-web browser?
>> Any plan about this or is it out of scope?
> 
> Are there any additional requirements on RTCWEB than seeing it as an
> media type with some media type specific signalling, most likely as part
> of the RTP payload type used to carry the data between the peers?


It's hard to tell from such a brief message, alas.

But perhaps we are in agreement.  I am envisaging markup/DOM that makes the camera and microphone visible, the ability to connect them and the audio/video elements to modules that do real-time communication, javascript APIs control the markup and those modules. I expect that the modules will instantiate discovery, signalling/connectivity, actual real-time streaming, encode/decode, and monitoring.

Within that, I am expecting there to be initial modules defined and probably mandated to do an initial set of protocols, probably the classic RTP/SIP for signalling and flows, and that those modules will be spec'd with 'enough' NAT/firewall handling to work most of the time. Connection to TURN servers should probably be possible. Other modules are possible; explore, refine, develop, prototype, etc.

So, on the subject of 3D: if your browser supports a stereo video media type for rendering in the video element, and for capture by a camera, and your codecs can encode/decode that type, and RTP encapsulate it, and your protocol modules carry it (and it should be transparent)...why not?

Quite a few use-cases do not need arbitrary discovery.  
* I bring up a web page from an organization that offers to connect me to a department at that organization (e.g. customer support), in an integrated context. The 'server cloud' at that corporation does not need to 'discover' a customer support rep's IP address, it can know it.
* I am on a cloud-service based portal page (like a yahoo, google, etc. page), which has frames for email, chat, news, and video-phone. The page indicates that someone in my (cloudbased) address book is online and call-able. It already knows that and where they are, and has put a 'green phone' icon next to them. No discovery is needed.

Indeed, what makes the browser attractive is that it offers an integrating service environment, and doing a classic 'blind call' to an address in my address book, where full discovery is needed ('is fred online', 'what is his IP address?' etc.) is not obviously benefiting much from this. Videophone applications already do this (quite well).


David Singer
Multimedia and Software Standards, Apple Inc.


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I tend to think we should separate "RTP session" (i.e. PeerConnection) from
"network transport"; a RTP session will run over network transports, but I
don't know if we want to add generic transport facility to the RTP session
API. As in, if reliable message delivery is desired, do we want to go down
the road of TCP-over-RTP?

If we expose raw transport as a lower-level API, then RTP sessions and
generic sessions can make use of this transport service.

On Thu, Apr 14, 2011 at 6:37 AM, Stefan H=C3=A5kansson LK <
stefan.lk.hakansson@ericsson.com> wrote:

> I also think a direct "small message" channel would be very useful: for
> enhancing conferencing-like services as discussed below as well as for
> gaming (where latency can be extremely important). And once it is availab=
le
> I'm sure developers will find other, more innovative, ways to use it!
>
> Stefan
>
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
> Of Tim Panton
> Sent: den 12 april 2011 10:40
> To: Elwell, John
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] non-media data
>
> Although we should aim to limit scope, it is also a good plan to listen t=
o
> experience in the field.
> Our experience is that direct, reliable, sequenced transmission of small
> messages between endpoints hugely adds to the value of a web-embedded RTC
> component.
>
> Typically these messages are intimately tied to events in the media engin=
e
> (change of speaker, end of message playback, transcription or recording
> available). (dynamic media meta-data if you like)
>
> They are used to change the visual state of the web page (via javascript)
> so that (for example) the web displayed IVR prompts stay in sync with the
> audio.
>
> Whilst all of these are achievable via a more circuitous route
> (mediaserver->signalingserver->webserver->long poll-> browser) the added
> latency will make the visual changes lag the audio which undermines the u=
ser
> experience.
>
> The value in RTC-web isn't the realtime audio/video itself, it's the fact
> that it is an integral part of the web-page, that integration is aided by
> non-media data.
> Tim.
>
>
> On 12 Apr 2011, at 08:19, Elwell, John wrote:
>
> > It is always good to try to limit scope, and I think this is indeed a
> case where we can useful limit scope to RTP-based media (audio, video,
> real-time text...).
> >
> > John
> >
> >
> >> -----Original Message-----
> >> From: rtcweb-bounces@ietf.org
> >> [mailto:rtcweb-bounces@ietf.org] On Behalf Of David Singer
> >> Sent: 08 April 2011 19:19
> >> To: rtcweb@ietf.org
> >> Subject: [rtcweb] non-media data
> >>
> >>
> >> On Apr 8, 2011, at 4:56 , Harald Alvestrand wrote:
> >>
> >>>
> >>> - Non-media data: RTP, UDP-without-RTP, TCP, or "something else"?
> >>> Congestion control is obviously needed - both for this and
> >> for media, naturally.
> >>>
> >>
> >> My feeling is that we should not forget we're designing something to
> >> sit in a browser or browser-like context.  We
> >> *have* text chat tools, file upload and download tools, and so on,
> >> today, working in browsers.  Yes, they typically go through a server,
> >> and going point to point might be an optimization.  But that's the
> >> rub; I don't think we should spend a lot of time optimizing tools
> >> that are ancillary to the core problem of doing real-time a/v between
> >> two users. If someone wants to start a group looking at real-time
> >> point-to-point text chat, well, OK, but I think it's a distraction
> >> for us, at least initially.
> >>
> >> David Singer
> >> Multimedia and Software Standards, Apple Inc.
> >>
> >> _______________________________________________
> >> 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
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--001636418125e09d6e04a0ef47e7
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

I tend to think we should separate &quot;RTP session&quot; (i.e. PeerConnec=
tion) from &quot;network transport&quot;; a RTP session will run over netwo=
rk transports, but I don&#39;t know if we want to add generic transport fac=
ility to the RTP session API. As in, if reliable message delivery is desire=
d, do we want to go down the road of TCP-over-RTP?<div>

<br></div><div>If we expose raw transport as a lower-level API, then RTP se=
ssions and generic sessions can make use of this transport service.<br><br>=
<div class=3D"gmail_quote">On Thu, Apr 14, 2011 at 6:37 AM, Stefan H=C3=A5k=
ansson LK <span dir=3D"ltr">&lt;<a href=3D"mailto:stefan.lk.hakansson@erics=
son.com">stefan.lk.hakansson@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;">I also think a direct &quot;small message&q=
uot; channel would be very useful: for enhancing conferencing-like services=
 as discussed below as well as for gaming (where latency can be extremely i=
mportant). And once it is available I&#39;m sure developers will find other=
, more innovative, ways to use it!<br>


<font color=3D"#888888"><br>
Stefan<br>
</font><div class=3D"im"><br>
-----Original Message-----<br>
From: <a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a=
> [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.or=
g</a>] On Behalf Of Tim Panton<br>
Sent: den 12 april 2011 10:40<br>
To: Elwell, John<br>
Cc: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
Subject: Re: [rtcweb] non-media data<br>
<br>
Although we should aim to limit scope, it is also a good plan to listen to =
experience in the field.<br>
Our experience is that direct, reliable, sequenced transmission of small me=
ssages between endpoints hugely adds to the value of a web-embedded RTC com=
ponent.<br>
<br>
Typically these messages are intimately tied to events in the media engine =
(change of speaker, end of message playback, transcription or recording ava=
ilable). (dynamic media meta-data if you like)<br>
<br>
They are used to change the visual state of the web page (via javascript) s=
o that (for example) the web displayed IVR prompts stay in sync with the au=
dio.<br>
<br>
Whilst all of these are achievable via a more circuitous route (mediaserver=
-&gt;signalingserver-&gt;webserver-&gt;long poll-&gt; browser) the added la=
tency will make the visual changes lag the audio which undermines the user =
experience.<br>


<br>
The value in RTC-web isn&#39;t the realtime audio/video itself, it&#39;s th=
e fact that it is an integral part of the web-page, that integration is aid=
ed by non-media data.<br>
Tim.<br>
<br>
<br>
</div><div class=3D"im">On 12 Apr 2011, at 08:19, Elwell, John wrote:<br>
<br>
&gt; It is always good to try to limit scope, and I think this is indeed a =
case where we can useful limit scope to RTP-based media (audio, video, real=
-time text...).<br>
&gt;<br>
&gt; John<br>
&gt;<br>
&gt;<br>
&gt;&gt; -----Original Message-----<br>
&gt;&gt; From: <a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ie=
tf.org</a><br>
&gt;&gt; [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@=
ietf.org</a>] On Behalf Of David Singer<br>
&gt;&gt; Sent: 08 April 2011 19:19<br>
&gt;&gt; To: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt;&gt; Subject: [rtcweb] non-media data<br>
&gt;&gt;<br>
&gt;&gt;<br>
</div><div class=3D"im">&gt;&gt; On Apr 8, 2011, at 4:56 , Harald Alvestran=
d wrote:<br>
&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; - Non-media data: RTP, UDP-without-RTP, TCP, or &quot;somethin=
g else&quot;?<br>
&gt;&gt;&gt; Congestion control is obviously needed - both for this and<br>
&gt;&gt; for media, naturally.<br>
&gt;&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; My feeling is that we should not forget we&#39;re designing someth=
ing to<br>
&gt;&gt; sit in a browser or browser-like context. =C2=A0We<br>
&gt;&gt; *have* text chat tools, file upload and download tools, and so on,=
<br>
&gt;&gt; today, working in browsers. =C2=A0Yes, they typically go through a=
 server,<br>
&gt;&gt; and going point to point might be an optimization. =C2=A0But that&=
#39;s the<br>
&gt;&gt; rub; I don&#39;t think we should spend a lot of time optimizing to=
ols<br>
&gt;&gt; that are ancillary to the core problem of doing real-time a/v betw=
een<br>
&gt;&gt; two users. If someone wants to start a group looking at real-time<=
br>
&gt;&gt; point-to-point text chat, well, OK, but I think it&#39;s a distrac=
tion<br>
&gt;&gt; for us, at least initially.<br>
&gt;&gt;<br>
</div><div class=3D"im">&gt;&gt; David Singer<br>
&gt;&gt; Multimedia and Software Standards, Apple Inc.<br>
&gt;&gt;<br>
&gt;&gt; _______________________________________________<br>
</div><div><div></div><div class=3D"h5">&gt;&gt; rtcweb mailing list<br>
&gt;&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D=
"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;&gt;<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<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>
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>

--001636418125e09d6e04a0ef47e7--

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Justin Uberti skrev 2011-04-15 08:37:
> I tend to think we should separate "RTP session" (i.e. PeerConnection)
> from "network transport"; a RTP session will run over network
> transports, but I don't know if we want to add generic transport
> facility to the RTP session API. As in, if reliable message delivery is
> desired, do we want to go down the road of TCP-over-RTP?

I don't like all this talk about data over RTP. Generic data over RTP
isn't the most suitable model for a number of reasons which I will come
back to below. From my perspective we do want data transport that
provide a unreliable datagram based transport. That would in my view
include make available to the web application through the API this. And
let the WEBAPP decided what is put in each datagram. However, we clearly
needs some requirements on that datagram service so that it remains safe
and fair when it comes to transport. To me that means requirements such as:

- Congestion Controlled
- Confidentiality and Integrity Protected
- Providing consent to receive data prior to allow transmission of
substantial amounts of data
- Some type of source authentication to prevent man in the middles

There is a question of how much additional functionality one likes to
offer around an unreliable datagram transport. I think we need to go
through http://www.rfc-editor.org/rfc/rfc5405.txt and see what is
desired to have as part of the browser implementation and what is left
to the web application to implement.

Then there is the question if we desire to provide additional transport
properties, like reliable datagram, or reliable byte stream modes. There
there exist several possible ways of doing that. Separate protocol
stacks, reliability extension on top of an unreliable datagram transport
etc. I also think it is important that we do discuss what requirements
exist for each of the transport properties.

I promised to return to why I think TCP over RTP or more general,
generic data over RTP isn't the most suitable model.
- RTP is a protocol built towards real-time media. Yes the definition of
media can be pretty loose here. But carrying the individual datagrams of
a TCP connection over RTP is plainly stupid, when TCP over UDP will get
you all the desired functionality anyway with less overhead.
- RTP requires RTP payload formats for anything it carries. That is to
ensure that the application level framing ideas are as well captured for
a particular media encoding as possible. That optimization is not
possible for general data.
- If one can't do general optimization one goes with more general
transport solutions, thus why not use a more general data transport for
the general data?
- RTP and RTCP monitoring is also designed with more continous medias in
mind. Here we don't know how much such properties will be true. Also the
congestion controls available might actually have not the most suitable
properties for certain type of data.

> 
> If we expose raw transport as a lower-level API, then RTP sessions and
> generic sessions can make use of this transport service.
> 

I understand your thinking, but there is important to remember that the
NAT/FW traversal solution is likely needing to operate on top of the
bottom most transport protocol, most likely UDP. On top of that we have
thus may ICE's STUN servers multiplexed with one RTP session on the same
5-tuple. Isn't it likely that one will from an API perspective request
an RTP sessison to be established to the peer with a certain set of RTP
payload types to carry a particular media type for a particular purpose.
In the same way I see that a unreliable datagram transport will be
requested and as part of that request indicate what particular data one
wants to run on top, for example using media types like for the UDP and
TCP protocol identifiers in SDP, or another label scheme. But the web
application and perhaps the browser will have an understanding what a
particular flow will be used to.

When it comes to actually doing a data transport of any kind. I do think
this is an individual module that can be negotiated for availability
between two peers. However, I do think there are a number of use cases
that are so interesting that at least unreliable datagram transport
should be aimed at completing at the same time as setup and negotiation,
NAT/FW traversal, and the real-time media modules. But more use cases
and more opinions on this are desirable.

At the same time I would like to hear if anyone has use cases for
reliable data services that goes directly peer to peer, either message
based or byte stream based. So far the arguments I have seen for this
are reducing load on relays, and lower transport delay. Some views from
someone actually needs such transport services as part of the solution
would be great.

Cheers

Magnus Westerlund

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

From tim@phonefromhere.com  Fri Apr 15 01:40:18 2011
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To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [rtcweb] non-media data
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On 15 Apr 2011, at 08:42, Magnus Westerlund wrote:

>=20
> At the same time I would like to hear if anyone has use cases for
> reliable data services that goes directly peer to peer, either message
> based or byte stream based. So far the arguments I have seen for this
> are reducing load on relays, and lower transport delay. Some views =
from
> someone actually needs such transport services as part of the solution
> would be great.

It's true that there are always multiple routes to solve a problem, but=20=

take my example of sending visible IVR prompts to a browserbased =
softphone.

(So if you call your bank from a web-embedded softphone you get a visual =
menu
to go with the spoken IVR, prompts can have more explanatory help text =
and
chosen faster as most people can read faster than an IVR can speak. The =
prompts=20
change as you move through the IVR tree - at some point you may get =
handed off to
a real person)

Ideally these messages are carried over reliable datagrams, in =
reasonably close
sync with the media. You don't want to miss them and you don't want them =
to be delivered
out of sequence. A TCP stream wouldn't be a good fit as these are =
messages not streams.

The case is much more like DTMF, except more generalised and hopefully =
without the
ugliness of the "send the same thing many times in the hope that one =
will get through"
that asymmetric RTP requires when doing DTMF.

This is a near-term transitional example, I'm sure that in the future, =
there will be more=20
extreme examples say involving lip-sync animation of locally drawn =
(SVG?) avatars, again where
the timing is critical.

Sure you could define a new media-type for lip-sync animation, but which =
web coder will do that
if they can throw a JSON packet up the wire ? In this case a generalist =
web-friendly mechanism
is going to win out.

Tim.=

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Since this RFC effectively tells us that we should abandon the "cname is 
user@host" convention, this RFC may be of interest to this WG.

Of special interest is the explicit statement that a CNAME is used to 
identify streams that need to be synchronized between multiple RTP 
sessions; I take it that the inverse is also true - that streams that do 
not need to be synchronized do not need to have the same CNAME.

                Harald

-------- Original Message --------
Subject: 	RFC 6222 on Guidelines for Choosing RTP Control Protocol 
(RTCP) Canonical Names (CNAMEs)
Date: 	Thu, 14 Apr 2011 11:26:24 -0700 (PDT)
From: 	rfc-editor@rfc-editor.org
To: 	ietf-announce@ietf.org, rfc-dist@rfc-editor.org
CC: 	avt@ietf.org, rfc-editor@rfc-editor.org



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


         RFC 6222

         Title:      Guidelines for Choosing RTP Control
                     Protocol (RTCP) Canonical Names (CNAMEs)
         Author:     A. Begen, C. Perkins,
                     D. Wing
         Status:     Standards Track
         Stream:     IETF
         Date:       April 2011
         Mailbox:    abegen@cisco.com,
                     csp@csperkins.org,
                     dwing@cisco.com
         Pages:      9
         Characters: 20393
         Updates:    RFC3550

         I-D Tag:    draft-ietf-avt-rtp-cnames-05.txt

         URL:        http://www.rfc-editor.org/rfc/rfc6222.txt

The RTP Control Protocol (RTCP) Canonical Name (CNAME) is a
persistent transport-level identifier for an RTP endpoint.  While the
Synchronization Source (SSRC) identifier of an RTP endpoint may
change if a collision is detected or when the RTP application is
restarted, its RTCP CNAME is meant to stay unchanged, so that RTP
endpoints can be uniquely identified and associated with their RTP
media streams.  For proper functionality, RTCP CNAMEs should be
unique within the participants of an RTP session.  However, the
existing guidelines for choosing the RTCP CNAME provided in the RTP
standard are insufficient to achieve this uniqueness.  This memo
updates those guidelines to allow endpoints to choose unique RTCP
CNAMEs.  [STANDARDS-TRACK]

This document is a product of the Audio/Video Transport Working Group of the IETF.

This is now a Proposed Standard Protocol.

STANDARDS TRACK: This document specifies an Internet standards track
protocol for the Internet community,and requests discussion and suggestions
for improvements.  Please refer to the current edition of the Internet
Official Protocol Standards (STD 1) for the standardization state and
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    Since this RFC effectively tells us that we should abandon the
    "cname is user@host" convention, this RFC may be of interest to this
    WG.<br>
    <br>
    Of special interest is the explicit statement that a CNAME is used
    to identify streams that need to be synchronized between multiple
    RTP sessions; I take it that the inverse is also true - that streams
    that do not need to be synchronized do not need to have the same
    CNAME.<br>
    <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
    <br>
    -------- Original Message --------
    <table class="moz-email-headers-table" border="0" cellpadding="0"
      cellspacing="0">
      <tbody>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject: </th>
          <td>RFC 6222 on Guidelines for Choosing RTP Control Protocol
            (RTCP) Canonical Names (CNAMEs)</td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
          <td>Thu, 14 Apr 2011 11:26:24 -0700 (PDT)</td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From: </th>
          <td><a class="moz-txt-link-abbreviated" href="mailto:rfc-editor@rfc-editor.org">rfc-editor@rfc-editor.org</a></td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To: </th>
          <td><a class="moz-txt-link-abbreviated" href="mailto:ietf-announce@ietf.org">ietf-announce@ietf.org</a>, <a class="moz-txt-link-abbreviated" href="mailto:rfc-dist@rfc-editor.org">rfc-dist@rfc-editor.org</a></td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">CC: </th>
          <td><a class="moz-txt-link-abbreviated" href="mailto:avt@ietf.org">avt@ietf.org</a>, <a class="moz-txt-link-abbreviated" href="mailto:rfc-editor@rfc-editor.org">rfc-editor@rfc-editor.org</a></td>
        </tr>
      </tbody>
    </table>
    <br>
    <br>
    <pre>A new Request for Comments is now available in online RFC libraries.

        
        RFC 6222

        Title:      Guidelines for Choosing RTP Control 
                    Protocol (RTCP) Canonical Names (CNAMEs) 
        Author:     A. Begen, C. Perkins,
                    D. Wing
        Status:     Standards Track
        Stream:     IETF
        Date:       April 2011
        Mailbox:    <a class="moz-txt-link-abbreviated" href="mailto:abegen@cisco.com">abegen@cisco.com</a>, 
                    <a class="moz-txt-link-abbreviated" href="mailto:csp@csperkins.org">csp@csperkins.org</a>, 
                    <a class="moz-txt-link-abbreviated" href="mailto:dwing@cisco.com">dwing@cisco.com</a>
        Pages:      9
        Characters: 20393
        Updates:    RFC3550

        I-D Tag:    draft-ietf-avt-rtp-cnames-05.txt

        URL:        <a class="moz-txt-link-freetext" href="http://www.rfc-editor.org/rfc/rfc6222.txt">http://www.rfc-editor.org/rfc/rfc6222.txt</a>

The RTP Control Protocol (RTCP) Canonical Name (CNAME) is a
persistent transport-level identifier for an RTP endpoint.  While the
Synchronization Source (SSRC) identifier of an RTP endpoint may
change if a collision is detected or when the RTP application is
restarted, its RTCP CNAME is meant to stay unchanged, so that RTP
endpoints can be uniquely identified and associated with their RTP
media streams.  For proper functionality, RTCP CNAMEs should be
unique within the participants of an RTP session.  However, the
existing guidelines for choosing the RTCP CNAME provided in the RTP
standard are insufficient to achieve this uniqueness.  This memo
updates those guidelines to allow endpoints to choose unique RTCP
CNAMEs.  [STANDARDS-TRACK]

This document is a product of the Audio/Video Transport Working Group of the IETF.

This is now a Proposed Standard Protocol.

STANDARDS TRACK: This document specifies an Internet standards track
protocol for the Internet community,and requests discussion and suggestions
for improvements.  Please refer to the current edition of the Internet
Official Protocol Standards (STD 1) for the standardization state and
status of this protocol.  Distribution of this memo is unlimited.

This announcement is sent to the IETF-Announce and rfc-dist lists.
To subscribe or unsubscribe, see
  <a class="moz-txt-link-freetext" href="http://www.ietf.org/mailman/listinfo/ietf-announce">http://www.ietf.org/mailman/listinfo/ietf-announce</a>
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For searching the RFC series, see <a class="moz-txt-link-freetext" href="http://www.rfc-editor.org/rfcsearch.html">http://www.rfc-editor.org/rfcsearch.html</a>.
For downloading RFCs, see <a class="moz-txt-link-freetext" href="http://www.rfc-editor.org/rfc.html">http://www.rfc-editor.org/rfc.html</a>.

Requests for special distribution should be addressed to either the
author of the RFC in question, or to <a class="moz-txt-link-abbreviated" href="mailto:rfc-editor@rfc-editor.org">rfc-editor@rfc-editor.org</a>.  Unless
specifically noted otherwise on the RFC itself, all RFCs are for
unlimited distribution.


The RFC Editor Team
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</pre>
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From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] non-media data
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On 04/15/11 10:40, Tim Panton wrote:
> On 15 Apr 2011, at 08:42, Magnus Westerlund wrote:
>
>> At the same time I would like to hear if anyone has use cases for
>> reliable data services that goes directly peer to peer, either message
>> based or byte stream based. So far the arguments I have seen for this
>> are reducing load on relays, and lower transport delay. Some views from
>> someone actually needs such transport services as part of the solution
>> would be great.
> It's true that there are always multiple routes to solve a problem, but
> take my example of sending visible IVR prompts to a browserbased softphone.
>
> (So if you call your bank from a web-embedded softphone you get a visual menu
> to go with the spoken IVR, prompts can have more explanatory help text and
> chosen faster as most people can read faster than an IVR can speak. The prompts
> change as you move through the IVR tree - at some point you may get handed off to
> a real person)
>
> Ideally these messages are carried over reliable datagrams, in reasonably close
> sync with the media. You don't want to miss them and you don't want them to be delivered
> out of sequence. A TCP stream wouldn't be a good fit as these are messages not streams.
Actually, this use case seem to me like an excellent fit with HTML-based 
menus with menu content carried over XDR. On a good day, the time 
required to replace menu content is ~100 msec; this is far faster than 
most humans can react.

It does point out (to me) a need to insert syncpoints into media streams 
so that the HTML action can be synchronized with the spoken text; the 
syncpoints may be represented as specific timestamps that are part of 
HTML content with values pointing into the media streams, or they might 
be synchronized packets carried on a data channel that arrives in sync 
with the audio.
> The case is much more like DTMF, except more generalised and hopefully without the
> ugliness of the "send the same thing many times in the hope that one will get through"
> that asymmetric RTP requires when doing DTMF.
>
> This is a near-term transitional example, I'm sure that in the future, there will be more
> extreme examples say involving lip-sync animation of locally drawn (SVG?) avatars, again where
> the timing is critical.
>
> Sure you could define a new media-type for lip-sync animation, but which web coder will do that
> if they can throw a JSON packet up the wire ? In this case a generalist web-friendly mechanism
> is going to win out.
For a working example of stuff synchronized with media, check out WebVTT 
(subtitling).

http://www.whatwg.org/specs/web-apps/current-work/webvtt.html


From harald@alvestrand.no  Fri Apr 15 05:55:38 2011
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
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When I wrote draft-alvestrand-dispatch-rtcweb-01, I was trying to 
describe a datagram service with assured "permission to send", which 
could be used to underpin either RTP or a raw datagram service.

It's received relatively little feedback; we might want to pursue it or not.

On 04/15/11 08:37, Justin Uberti wrote:
> I tend to think we should separate "RTP session" (i.e. PeerConnection) 
> from "network transport"; a RTP session will run over network 
> transports, but I don't know if we want to add generic transport 
> facility to the RTP session API. As in, if reliable message delivery 
> is desired, do we want to go down the road of TCP-over-RTP?
>
> If we expose raw transport as a lower-level API, then RTP sessions and 
> generic sessions can make use of this transport service.
>
> On Thu, Apr 14, 2011 at 6:37 AM, Stefan HÃ¥kansson LK 
> <stefan.lk.hakansson@ericsson.com 
> <mailto:stefan.lk.hakansson@ericsson.com>> wrote:
>
>     I also think a direct "small message" channel would be very
>     useful: for enhancing conferencing-like services as discussed
>     below as well as for gaming (where latency can be extremely
>     important). And once it is available I'm sure developers will find
>     other, more innovative, ways to use it!
>
>     Stefan
>
>     -----Original Message-----
>     From: rtcweb-bounces@ietf.org <mailto:rtcweb-bounces@ietf.org>
>     [mailto:rtcweb-bounces@ietf.org <mailto:rtcweb-bounces@ietf.org>]
>     On Behalf Of Tim Panton
>     Sent: den 12 april 2011 10:40
>     To: Elwell, John
>     Cc: rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     Subject: Re: [rtcweb] non-media data
>
>     Although we should aim to limit scope, it is also a good plan to
>     listen to experience in the field.
>     Our experience is that direct, reliable, sequenced transmission of
>     small messages between endpoints hugely adds to the value of a
>     web-embedded RTC component.
>
>     Typically these messages are intimately tied to events in the
>     media engine (change of speaker, end of message playback,
>     transcription or recording available). (dynamic media meta-data if
>     you like)
>
>     They are used to change the visual state of the web page (via
>     javascript) so that (for example) the web displayed IVR prompts
>     stay in sync with the audio.
>
>     Whilst all of these are achievable via a more circuitous route
>     (mediaserver->signalingserver->webserver->long poll-> browser) the
>     added latency will make the visual changes lag the audio which
>     undermines the user experience.
>
>     The value in RTC-web isn't the realtime audio/video itself, it's
>     the fact that it is an integral part of the web-page, that
>     integration is aided by non-media data.
>     Tim.
>
>
>     On 12 Apr 2011, at 08:19, Elwell, John wrote:
>
>     > It is always good to try to limit scope, and I think this is
>     indeed a case where we can useful limit scope to RTP-based media
>     (audio, video, real-time text...).
>     >
>     > John
>     >
>     >
>     >> -----Original Message-----
>     >> From: rtcweb-bounces@ietf.org <mailto:rtcweb-bounces@ietf.org>
>     >> [mailto:rtcweb-bounces@ietf.org
>     <mailto:rtcweb-bounces@ietf.org>] On Behalf Of David Singer
>     >> Sent: 08 April 2011 19:19
>     >> To: rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     >> Subject: [rtcweb] non-media data
>     >>
>     >>
>     >> On Apr 8, 2011, at 4:56 , Harald Alvestrand wrote:
>     >>
>     >>>
>     >>> - Non-media data: RTP, UDP-without-RTP, TCP, or "something else"?
>     >>> Congestion control is obviously needed - both for this and
>     >> for media, naturally.
>     >>>
>     >>
>     >> My feeling is that we should not forget we're designing
>     something to
>     >> sit in a browser or browser-like context.  We
>     >> *have* text chat tools, file upload and download tools, and so on,
>     >> today, working in browsers.  Yes, they typically go through a
>     server,
>     >> and going point to point might be an optimization.  But that's the
>     >> rub; I don't think we should spend a lot of time optimizing tools
>     >> that are ancillary to the core problem of doing real-time a/v
>     between
>     >> two users. If someone wants to start a group looking at real-time
>     >> point-to-point text chat, well, OK, but I think it's a distraction
>     >> for us, at least initially.
>     >>
>     >> David Singer
>     >> Multimedia and Software Standards, Apple Inc.
>     >>
>     >> _______________________________________________
>     >> rtcweb mailing list
>     >> rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     >> https://www.ietf.org/mailman/listinfo/rtcweb
>     >>
>     > _______________________________________________
>     > rtcweb mailing list
>     > rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     > https://www.ietf.org/mailman/listinfo/rtcweb
>
>     _______________________________________________
>     rtcweb mailing list
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     https://www.ietf.org/mailman/listinfo/rtcweb
>     _______________________________________________
>     rtcweb mailing list
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


--------------060107050103020406080307
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
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    When I wrote draft-alvestrand-dispatch-rtcweb-01, I was trying to
    describe a datagram service with assured "permission to send", which
    could be used to underpin either RTP or a raw datagram service.<br>
    <br>
    It's received relatively little feedback; we might want to pursue it
    or not.<br>
    <br>
    On 04/15/11 08:37, Justin Uberti wrote:
    <blockquote
      cite="mid:BANLkTi==JtrJ_fmtYm7=UKSpp+56WdH-Hw@mail.gmail.com"
      type="cite">I tend to think we should separate "RTP session" (i.e.
      PeerConnection) from "network transport"; a RTP session will run
      over network transports, but I don't know if we want to add
      generic transport facility to the RTP session API. As in, if
      reliable message delivery is desired, do we want to go down the
      road of TCP-over-RTP?
      <div>
        <br>
      </div>
      <div>If we expose raw transport as a lower-level API, then RTP
        sessions and generic sessions can make use of this transport
        service.<br>
        <br>
        <div class="gmail_quote">On Thu, Apr 14, 2011 at 6:37 AM, Stefan
          HÃ¥kansson LK <span dir="ltr">&lt;<a moz-do-not-send="true"
              href="mailto:stefan.lk.hakansson@ericsson.com">stefan.lk.hakansson@ericsson.com</a>&gt;</span>
          wrote:<br>
          <blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
            0.8ex; border-left: 1px solid rgb(204, 204, 204);
            padding-left: 1ex;">I also think a direct "small message"
            channel would be very useful: for enhancing
            conferencing-like services as discussed below as well as for
            gaming (where latency can be extremely important). And once
            it is available I'm sure developers will find other, more
            innovative, ways to use it!<br>
            <font color="#888888"><br>
              Stefan<br>
            </font>
            <div class="im"><br>
              -----Original Message-----<br>
              From: <a moz-do-not-send="true"
                href="mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>
              [mailto:<a moz-do-not-send="true"
                href="mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>]
              On Behalf Of Tim Panton<br>
              Sent: den 12 april 2011 10:40<br>
              To: Elwell, John<br>
              Cc: <a moz-do-not-send="true"
                href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
              Subject: Re: [rtcweb] non-media data<br>
              <br>
              Although we should aim to limit scope, it is also a good
              plan to listen to experience in the field.<br>
              Our experience is that direct, reliable, sequenced
              transmission of small messages between endpoints hugely
              adds to the value of a web-embedded RTC component.<br>
              <br>
              Typically these messages are intimately tied to events in
              the media engine (change of speaker, end of message
              playback, transcription or recording available). (dynamic
              media meta-data if you like)<br>
              <br>
              They are used to change the visual state of the web page
              (via javascript) so that (for example) the web displayed
              IVR prompts stay in sync with the audio.<br>
              <br>
              Whilst all of these are achievable via a more circuitous
              route
              (mediaserver-&gt;signalingserver-&gt;webserver-&gt;long
              poll-&gt; browser) the added latency will make the visual
              changes lag the audio which undermines the user
              experience.<br>
              <br>
              The value in RTC-web isn't the realtime audio/video
              itself, it's the fact that it is an integral part of the
              web-page, that integration is aided by non-media data.<br>
              Tim.<br>
              <br>
              <br>
            </div>
            <div class="im">On 12 Apr 2011, at 08:19, Elwell, John
              wrote:<br>
              <br>
              &gt; It is always good to try to limit scope, and I think
              this is indeed a case where we can useful limit scope to
              RTP-based media (audio, video, real-time text...).<br>
              &gt;<br>
              &gt; John<br>
              &gt;<br>
              &gt;<br>
              &gt;&gt; -----Original Message-----<br>
              &gt;&gt; From: <a moz-do-not-send="true"
                href="mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a><br>
              &gt;&gt; [mailto:<a moz-do-not-send="true"
                href="mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>]
              On Behalf Of David Singer<br>
              &gt;&gt; Sent: 08 April 2011 19:19<br>
              &gt;&gt; To: <a moz-do-not-send="true"
                href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
              &gt;&gt; Subject: [rtcweb] non-media data<br>
              &gt;&gt;<br>
              &gt;&gt;<br>
            </div>
            <div class="im">&gt;&gt; On Apr 8, 2011, at 4:56 , Harald
              Alvestrand wrote:<br>
              &gt;&gt;<br>
              &gt;&gt;&gt;<br>
              &gt;&gt;&gt; - Non-media data: RTP, UDP-without-RTP, TCP,
              or "something else"?<br>
              &gt;&gt;&gt; Congestion control is obviously needed - both
              for this and<br>
              &gt;&gt; for media, naturally.<br>
              &gt;&gt;&gt;<br>
              &gt;&gt;<br>
              &gt;&gt; My feeling is that we should not forget we're
              designing something to<br>
              &gt;&gt; sit in a browser or browser-like context. Â We<br>
              &gt;&gt; *have* text chat tools, file upload and download
              tools, and so on,<br>
              &gt;&gt; today, working in browsers. Â Yes, they typically
              go through a server,<br>
              &gt;&gt; and going point to point might be an
              optimization. Â But that's the<br>
              &gt;&gt; rub; I don't think we should spend a lot of time
              optimizing tools<br>
              &gt;&gt; that are ancillary to the core problem of doing
              real-time a/v between<br>
              &gt;&gt; two users. If someone wants to start a group
              looking at real-time<br>
              &gt;&gt; point-to-point text chat, well, OK, but I think
              it's a distraction<br>
              &gt;&gt; for us, at least initially.<br>
              &gt;&gt;<br>
            </div>
            <div class="im">&gt;&gt; David Singer<br>
              &gt;&gt; Multimedia and Software Standards, Apple Inc.<br>
              &gt;&gt;<br>
              &gt;&gt; _______________________________________________<br>
            </div>
            <div>
              <div class="h5">&gt;&gt; rtcweb mailing list<br>
                &gt;&gt; <a moz-do-not-send="true"
                  href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                &gt;&gt; <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>
                &gt;&gt;<br>
                &gt; _______________________________________________<br>
                &gt; rtcweb mailing list<br>
                &gt; <a moz-do-not-send="true"
                  href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                &gt; <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>
                <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>
                _______________________________________________<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>
              </div>
            </div>
          </blockquote>
        </div>
        <br>
      </div>
      <pre wrap="">
<fieldset class="mimeAttachmentHeader"></fieldset>
_______________________________________________
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>

--------------060107050103020406080307--

From harald@alvestrand.no  Fri Apr 15 06:12:47 2011
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Subject: [rtcweb] data-over-RTP vs data-over-datagram (Re: non-media data)
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On 04/15/11 09:42, Magnus Westerlund wrote:
> I don't like all this talk about data over RTP. Generic data over RTP
> isn't the most suitable model for a number of reasons which I will come
> back to below. From my perspective we do want data transport that
> provide a unreliable datagram based transport. That would in my view
> include make available to the web application through the API this. And
> let the WEBAPP decided what is put in each datagram. However, we clearly
> needs some requirements on that datagram service so that it remains safe
> and fair when it comes to transport. To me that means requirements such as:
>
> - Congestion Controlled
> - Confidentiality and Integrity Protected
> - Providing consent to receive data prior to allow transmission of
> substantial amounts of data
> - Some type of source authentication to prevent man in the middles
This is a good list to keep in mind. I'll return to it.
> There is a question of how much additional functionality one likes to
> offer around an unreliable datagram transport. I think we need to go
> through http://www.rfc-editor.org/rfc/rfc5405.txt and see what is
> desired to have as part of the browser implementation and what is left
> to the web application to implement.
> Then there is the question if we desire to provide additional transport
> properties, like reliable datagram, or reliable byte stream modes. There
> there exist several possible ways of doing that. Separate protocol
> stacks, reliability extension on top of an unreliable datagram transport
> etc. I also think it is important that we do discuss what requirements
> exist for each of the transport properties.
>
> I promised to return to why I think TCP over RTP or more general,
> generic data over RTP isn't the most suitable model.
> - RTP is a protocol built towards real-time media. Yes the definition of
> media can be pretty loose here. But carrying the individual datagrams of
> a TCP connection over RTP is plainly stupid, when TCP over UDP will get
> you all the desired functionality anyway with less overhead.
RTP is, in the extreme view, a ~12-byte header and a  set of handling 
rules. We should look at what it is, not so much what it was originally 
thought to be designed for (a component in a framework for multicast 
distribution of unidirectional media channels).
> - RTP requires RTP payload formats for anything it carries. That is to
> ensure that the application level framing ideas are as well captured for
> a particular media encoding as possible. That optimization is not
> possible for general data.
I think "general data" is a red herring. We can define a few different 
data-carrying "modes" that have different properties; if some of these 
are useful, I think it makes sense to define them.
> - If one can't do general optimization one goes with more general
> transport solutions, thus why not use a more general data transport for
> the general data?
Pushing stuff towards a "more general data transport" requires one of 
two things:
a) having an existing more general data transport that fulfils our 
requirements
b) defining a new more general data transport that fulfils our requirements
The second seems tough. The first ..... what's your preferred candidate?
> - RTP and RTCP monitoring is also designed with more continous medias in
> mind. Here we don't know how much such properties will be true. Also the
> congestion controls available might actually have not the most suitable
> properties for certain type of data.
It's not clear to me that there *are* congestion controls available 
(other than media-specific ones).
Which ones are you thinking of?

Now, the flipside: What does RTP *have* that we will otherwise need to 
define?
 From your earlier list:

- Congestion Controlled: eh... maybe not.
- Confidentiality and Integrity Protected - done, we've got SRTP.
- Providing consent to receive data prior to allow transmission of
substantial amounts of data - done, we have ICE, and need it for media too.
- Some type of source authentication to prevent man in the middles - done, we have SRTP

So apart from congestion control, it seems that RTP's 12-byte header actually gives us a lot.


>> If we expose raw transport as a lower-level API, then RTP sessions and
>> generic sessions can make use of this transport service.
>>
> I understand your thinking, but there is important to remember that the
> NAT/FW traversal solution is likely needing to operate on top of the
> bottom most transport protocol, most likely UDP. On top of that we have
> thus may ICE's STUN servers multiplexed with one RTP session on the same
> 5-tuple. Isn't it likely that one will from an API perspective request
> an RTP sessison to be established to the peer with a certain set of RTP
> payload types to carry a particular media type for a particular purpose.
If that's the API, we've got an RTP-level API (the RTP machinery is 
below the API). I believe Justin was talking about a lower-level API, 
where the RTP-level machinery would be above it.

Dropping the rest of the message ... this note is already long enough.


From magnus.westerlund@ericsson.com  Fri Apr 15 07:28:43 2011
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Subject: Re: [rtcweb] data-over-RTP vs data-over-datagram (Re: non-media data)
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Haralad and other interested

Snipped out more parts as I added quite a lot around possible
alternatives for you to think about. See below

Harald Alvestrand skrev 2011-04-15 15:12:
>> I promised to return to why I think TCP over RTP or more general,
>> generic data over RTP isn't the most suitable model.
>> - RTP is a protocol built towards real-time media. Yes the definition of
>> media can be pretty loose here. But carrying the individual datagrams of
>> a TCP connection over RTP is plainly stupid, when TCP over UDP will get
>> you all the desired functionality anyway with less overhead.

> RTP is, in the extreme view, a ~12-byte header and a  set of handling 
> rules. We should look at what it is, not so much what it was originally 
> thought to be designed for (a component in a framework for multicast 
> distribution of unidirectional media channels).

Well, the handling rules include RTCP and an intention to handle a
number of different topologies with multi-party handling. There are also
additional signalling issues, like how to configure the RTCP bandwidth
value, which features to use and all these things. Things you may not
need for your data transport. If you need them, you anyway should define
an RTP payload format for your data format.

And to be frank, the transport and RAI ADs has been very hesitant to
allow any RTP payload format that is defined for just being a bit pipe
over RTP. The two reasons for this has been Application level framing
principle and the lack of congestion control for RTP.

>> - RTP requires RTP payload formats for anything it carries. That is to
>> ensure that the application level framing ideas are as well captured for
>> a particular media encoding as possible. That optimization is not
>> possible for general data.

> I think "general data" is a red herring. We can define a few different 
> data-carrying "modes" that have different properties; if some of these 
> are useful, I think it makes sense to define them.

I agree that different transport protocol provides different properties.
And that you should chose the one most appropriate. However, one usually
has to compromise on some aspects, either by needing to do specific
solution for ones data or accept some other potential limitation. On a
high level comparisons between our transport protocols I think can be
summarized like this:

TCP: unicast only, congestion controlled reliable byte stream

SCTP: unicast, multihoming, congestion controlled, reliable and
unreliable, message oriented.

DCCP: unicast, unreliable, congestion controlled sequence numbered datagram

RTP/UDP: unicast and multicast, unreliable, RTP payload of single MTU
size, medium time scale transport feedback.

UDP: unicast and multicast, unreliable, datagram

>> - If one can't do general optimization one goes with more general
>> transport solutions, thus why not use a more general data transport for
>> the general data?

> Pushing stuff towards a "more general data transport" requires one of 
> two things:
> a) having an existing more general data transport that fulfils our 
> requirements
> b) defining a new more general data transport that fulfils our requirements
> The second seems tough. The first ..... what's your preferred candidate?

Actually the one I think is easiest from a specification point of view
and that does provide a reasonable service for unreliable datagrams are:

DTLS/DCCP/UDP

That would be the following references:
https://datatracker.ietf.org/doc/rfc4340/ (Base Spec)
https://datatracker.ietf.org/doc/rfc4341/ (TCP-Like CC)
https://datatracker.ietf.org/doc/rfc4342/ (TFRC CC)
https://datatracker.ietf.org/doc/rfc5238/ (DTLS for DCCP)
https://datatracker.ietf.org/doc/draft-ietf-dccp-udpencap/ (DCCP over UDP)

My main concern is the maturity of available implementations of DCCP.
Otherwise it provides congestion control and can actually choose which
congestion control algorithms we want supported, TCP-like or TFRC. It
also have some additional features. One is that we get a second port
space providing additional multiplexing points.

Please note that this is just a friendly suggestion. At least check it
out, and possible post some comments why it would be unsuitable.

>> - RTP and RTCP monitoring is also designed with more continous medias in
>> mind. Here we don't know how much such properties will be true. Also the
>> congestion controls available might actually have not the most suitable
>> properties for certain type of data.

> It's not clear to me that there *are* congestion controls available 
> (other than media-specific ones).
> Which ones are you thinking of?

The one that is full defined is to run RTP over DCCP.
https://datatracker.ietf.org/doc/rfc5762/

I guess the biggest issue is the lack of experience with this and
especially how you interface the DCCP stack to work well between media
encoding, RTP stack and the DCCP congestion control.

The alternative, not standardized yet but closest available for RTP over
UDP is TFRC for RTP.
https://datatracker.ietf.org/doc/draft-gharai-avtcore-rtp-tfrc/

That was in reasonable shape prior to its abandonment due to lack of
funding from the ones driving it.


> Now, the flipside: What does RTP *have* that we will otherwise need
> to define? From your earlier list:
> 
> - Congestion Controlled: eh... maybe not. 
> - Confidentiality and  Integrity Protected - done, we've got SRTP.
> - Providing consent to receive data prior to allow transmission of
>   substantial amounts of data - done, we have ICE,
>   and need it for media too.
> - Some type of source authentication to prevent man in the middles
>    - done, we have SRTP
> 
> So apart from congestion control, it seems that RTP's 12-byte header
> actually gives us a lot.

Well, the security parts can be provided by ICE and DTLS over UDP also.
we still are short the congestion control for such a thing plus all
these things that the UDP guidelines (RFC5405) discuss. Using RTP over
UDP does drag all the RTP baggage into something that may not needed it,
but it will provide some parts of what you need. If a particular data
format needs RTP functionality, it should define its own RTP payload
format anyway.

Cheers

Magnus Westerlund

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From magnus.westerlund@ericsson.com  Mon Apr 18 02:57:24 2011
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Harald and others,

This is just one of many things we recommend in our draft on how RTP
should be used in the RTCWEB context.
https://datatracker.ietf.org/doc/draft-perkins-rtcweb-rtp-usage/

We do like to get feedback on our document.

Harald Alvestrand skrev 2011-04-15 14:00:
> Since this RFC effectively tells us that we should abandon the "cname is
> user@host" convention, this RFC may be of interest to this WG.
> 
> Of special interest is the explicit statement that a CNAME is used to
> identify streams that need to be synchronized between multiple RTP
> sessions; I take it that the inverse is also true - that streams that do
> not need to be synchronized do not need to have the same CNAME.
> 

Yes, it is mostly true. And if ones intention is to provide sources that
you don't want to force synchronization with, then using different
CNAMEs are necessary. However, there are a bit of overloading on the
CNAME field that makes one have to be a bit careful when using different
CNAMEs from the same end-point. And I would like to add that
synchronization of SSRCs with the same CNAME is not only between RTP
sessions, but also within the RTP session.

Cheers

Magnus Westerlund

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Subject: Re: [rtcweb] Fwd: RFC 6222 on Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names (CNAMEs)
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On Apr 18, 2011, at 2:57 , Magnus Westerlund wrote:

> Harald and others,
> 
> This is just one of many things we recommend in our draft on how RTP
> should be used in the RTCWEB context.
> https://datatracker.ietf.org/doc/draft-perkins-rtcweb-rtp-usage/
> 
> We do like to get feedback on our document.
> 
> Harald Alvestrand skrev 2011-04-15 14:00:
>> Since this RFC effectively tells us that we should abandon the "cname is
>> user@host" convention, this RFC may be of interest to this WG.
>> 
>> Of special interest is the explicit statement that a CNAME is used to
>> identify streams that need to be synchronized between multiple RTP
>> sessions; I take it that the inverse is also true - that streams that do
>> not need to be synchronized do not need to have the same CNAME.
>> 
> 
> Yes, it is mostly true. And if ones intention is to provide sources that
> you don't want to force synchronization with, then using different
> CNAMEs are necessary. However, there are a bit of overloading on the
> CNAME field that makes one have to be a bit careful when using different
> CNAMEs from the same end-point. And I would like to add that
> synchronization of SSRCs with the same CNAME is not only between RTP
> sessions, but also within the RTP session.
> 
> 

If I am doing a point-to-point call, and I have the connectivity established (i.e. IP addresses and the like), why should I trouble at all with CNAME?  And, indeed, are there privacy concerns with it (well, there are with using <userid>@<host>, I rather suspect).  Can the CNAME problem be handled when aggregation happens?

David Singer
Multimedia and Software Standards, Apple Inc.


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Subject: Re: [rtcweb] Fwd: RFC 6222 on Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names (CNAMEs)
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David Singer skrev 2011-04-18 20:10:
> 
> If I am doing a point-to-point call, and I have the connectivity
> established (i.e. IP addresses and the like), why should I trouble at
> all with CNAME?  And, indeed, are there privacy concerns with it
> (well, there are with using <userid>@<host>, I rather suspect).  Can
> the CNAME problem be handled when aggregation happens?

Because you don't know when a peer is a single user, or a conference
bridge. In the later case of you will need the cname. Also for point to
point you might need it if you have multiple sources that should or
should not be synchronized. The point of the new RFC is that the cname
can be set session specific, by drawing a random number that is being
used. Thus you can avoid the downsides of CNAME from privacy point of
view, and the generation aspects.

cheers

Magnus Westerlund

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

From wwwrun@ietfc.amsl.com  Tue Apr 19 09:48:04 2011
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Subject: [rtcweb] WG Review: Real-Time Communication in WEB-browsers (rtcweb)
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A new IETF working group has been proposed in the Real-Time Applications
and Infrastructure Area.  The IESG has not made any determination as yet.
The following draft charter was submitted, and is provided for
informational purposes only. Please send your comments to the IESG mailing
list (iesg@ietf.org) by Tuesday, April 26, 2011.               
                             

Real-Time Communication in WEB-browsers (rtcweb)
------------------------------------------------
Current Status: Proposed Working Group
Last Updated: 2011-04-14

Chairs:
 TBD

RAI Area Directors:
 Gonzalo Camarillo
 Robert Sparks

RAI Area Advisor:
 Gonzalo Camarillo

Mailing Lists:
 Address:       rtcweb@ietf.org
 To Subscribe:  https://www.ietf.org/mailman/listinfo/rtcweb
 Archive:       http://www.ietf.org/mail-archive/web/rtcweb/

Description of Working Group:
 
There are a number of proprietary implementations that provide direct
interactive rich communication using audio, video, collaboration,
games, etc. between two peers' web-browsers. These are not
interoperable, as they require non-standard extensions or plugins to
work.  There is a desire to standardize the basis for such
communication so that interoperable communication can be established
between any compatible browsers. The goal is to enable innovation on
top of a set of basic components.  One core component is to enable
real-time media like audio and video, a second is to enable data
transfer directly between clients.
 
This work will be done in collaboration with the W3C.  The IETF WG
will produce architecture and requirements for selection and profiling
of the on the wire protocols. The architecture needs to be coordinated
with W3C.  The IETF WG work will identify state information and events
that need to be exposed in the APIs as input to W3C. The W3C will be
responsible for defining APIs to ensure that application developers
can control the components. We will reach out to the developer
community for consultation and early feedback on implementation.
 
The security and privacy goals and requirements will be developed by
the WG. The security model needs to be coordinated with the W3C.  The
work will also consider where support for extensibility is needed. RTP
functionalities, media formats, security algorithms are example of
things that commonly need extensions, additions or replacement, and
thus some support for negotiation between clients is required.
 
The WG will perform the following work:

1.  Define the communication model in detail, including how session
    management is to occur within the model.

2.  Define a security model that describes the security and privacy
    goals and specifies the security protocol mechanisms necessary
    to achieve those goals.

3.  Define the solution - protocols and API requirements - for
    firewall and NAT traversal.

4.  Define which media functions and extensions shall be supported in
    the client and their usage for real-time media, including media
    adaptation to ensure congestion safe usage.

5.  Define what functionalities in the solution, such as media codecs,
    security algorithms, etc., can be extended and how the
    extensibility mechanisms works.

6.  Define a set of media formats that must or should be supported by
    a client to improve interoperability.

7.  Define how non media data is transported between clients in a
    secure and congestion safe way.

8.  Provide W3C input for the APIs that comes from the communication
    model and the selected components and protocols that are part of
    the solution.

9.  The group will consider options for interworking with legacy VoIP
    equipment.
 
This work will be done primarily by using already defined protocols or
functionalities. If there is identification of missing protocols or
functionalities, such work can be requested to be done in another
working group with a suitable charter or by requests for chartering it
in this WG or another WG. The following topics will be out of scope
for the initial phase of the WG: RTSP, RSVP, NSIS, Location services,
Resource Priority, and IM & Presence specific features.
 
The products of the working group will support security and keying as
required by BCP 61 and be defined for IPv4, IPv6, and dual stack
deployments. The Working Group will consider the possibility of
defining a browser component that implements an existing session
negotiation and management protocol. The working group will follow BCP
79, and adhere to the spirit of BCP 79. The working group cannot
explicitly rule out the possibility of adopting encumbered
technologies; however, the working group will try to avoid encumbered
technologies that require royalties or other encumbrances that would
prevent such technologies from being easy to use in web browsers.
 
Milestones:
 
Aug 2011 Architecture, Security, Privacy and Threat Model sent to W3C
 
Aug 2011 Use cases, Scenarios, and Requirements document (I-D) sent to
         W3C
 
Sep 2011 Architecture and Security, Privacy, and Threat Model
         document(s) to IESG as Informational
 
Sep 2011 Use cases, Scenarios, and Requirements for RTCWeb document
         sent to IESG as Informational
 
Dec 2011 RTCWeb protocol profiles and Media format specification(s) to
         IESG as PS
  
Dec 2011 Information elements and events APIs Input to W3C
 
Apr 2012 API to Protocol mapping document submitted to the IESG as
         Informational (if needed)



From ron.even.tlv@gmail.com  Tue Apr 19 21:39:38 2011
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From: "Roni Even" <ron.even.tlv@gmail.com>
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Subject: Re: [rtcweb] WG Review: Real-Time Communication in WEB-browsers (rtcweb) - Mandatory media formats
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Hi,
I think that during the RTCweb ad-hoc on Friday meeting there was no
consensus if the proposed WG should specify which media format MUST or
Should be used. So I am not sure why the charter has item 6 in the work to
do as well about the deliverables of media format specifications.

Thanks
Roni Even

> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of IESG Secretary
> Sent: Tuesday, April 19, 2011 7:48 PM
> To: IETF Announcement list
> Cc: rtcweb@ietf.org
> Subject: [rtcweb] WG Review: Real-Time Communication in WEB-browsers
> (rtcweb)
> 
> A new IETF working group has been proposed in the Real-Time
> Applications
> and Infrastructure Area.  The IESG has not made any determination as
> yet.
> The following draft charter was submitted, and is provided for
> informational purposes only. Please send your comments to the IESG
> mailing
> list (iesg@ietf.org) by Tuesday, April 26, 2011.
> 
> 
> Real-Time Communication in WEB-browsers (rtcweb)
> ------------------------------------------------
> Current Status: Proposed Working Group
> Last Updated: 2011-04-14
> 
> Chairs:
>  TBD
> 
> RAI Area Directors:
>  Gonzalo Camarillo
>  Robert Sparks
> 
> RAI Area Advisor:
>  Gonzalo Camarillo
> 
> Mailing Lists:
>  Address:       rtcweb@ietf.org
>  To Subscribe:  https://www.ietf.org/mailman/listinfo/rtcweb
>  Archive:       http://www.ietf.org/mail-archive/web/rtcweb/
> 
> Description of Working Group:
> 
> There are a number of proprietary implementations that provide direct
> interactive rich communication using audio, video, collaboration,
> games, etc. between two peers' web-browsers. These are not
> interoperable, as they require non-standard extensions or plugins to
> work.  There is a desire to standardize the basis for such
> communication so that interoperable communication can be established
> between any compatible browsers. The goal is to enable innovation on
> top of a set of basic components.  One core component is to enable
> real-time media like audio and video, a second is to enable data
> transfer directly between clients.
> 
> This work will be done in collaboration with the W3C.  The IETF WG
> will produce architecture and requirements for selection and profiling
> of the on the wire protocols. The architecture needs to be coordinated
> with W3C.  The IETF WG work will identify state information and events
> that need to be exposed in the APIs as input to W3C. The W3C will be
> responsible for defining APIs to ensure that application developers
> can control the components. We will reach out to the developer
> community for consultation and early feedback on implementation.
> 
> The security and privacy goals and requirements will be developed by
> the WG. The security model needs to be coordinated with the W3C.  The
> work will also consider where support for extensibility is needed. RTP
> functionalities, media formats, security algorithms are example of
> things that commonly need extensions, additions or replacement, and
> thus some support for negotiation between clients is required.
> 
> The WG will perform the following work:
> 
> 1.  Define the communication model in detail, including how session
>     management is to occur within the model.
> 
> 2.  Define a security model that describes the security and privacy
>     goals and specifies the security protocol mechanisms necessary
>     to achieve those goals.
> 
> 3.  Define the solution - protocols and API requirements - for
>     firewall and NAT traversal.
> 
> 4.  Define which media functions and extensions shall be supported in
>     the client and their usage for real-time media, including media
>     adaptation to ensure congestion safe usage.
> 
> 5.  Define what functionalities in the solution, such as media codecs,
>     security algorithms, etc., can be extended and how the
>     extensibility mechanisms works.
> 
> 6.  Define a set of media formats that must or should be supported by
>     a client to improve interoperability.
> 
> 7.  Define how non media data is transported between clients in a
>     secure and congestion safe way.
> 
> 8.  Provide W3C input for the APIs that comes from the communication
>     model and the selected components and protocols that are part of
>     the solution.
> 
> 9.  The group will consider options for interworking with legacy VoIP
>     equipment.
> 
> This work will be done primarily by using already defined protocols or
> functionalities. If there is identification of missing protocols or
> functionalities, such work can be requested to be done in another
> working group with a suitable charter or by requests for chartering it
> in this WG or another WG. The following topics will be out of scope
> for the initial phase of the WG: RTSP, RSVP, NSIS, Location services,
> Resource Priority, and IM & Presence specific features.
> 
> The products of the working group will support security and keying as
> required by BCP 61 and be defined for IPv4, IPv6, and dual stack
> deployments. The Working Group will consider the possibility of
> defining a browser component that implements an existing session
> negotiation and management protocol. The working group will follow BCP
> 79, and adhere to the spirit of BCP 79. The working group cannot
> explicitly rule out the possibility of adopting encumbered
> technologies; however, the working group will try to avoid encumbered
> technologies that require royalties or other encumbrances that would
> prevent such technologies from being easy to use in web browsers.
> 
> Milestones:
> 
> Aug 2011 Architecture, Security, Privacy and Threat Model sent to W3C
> 
> Aug 2011 Use cases, Scenarios, and Requirements document (I-D) sent to
>          W3C
> 
> Sep 2011 Architecture and Security, Privacy, and Threat Model
>          document(s) to IESG as Informational
> 
> Sep 2011 Use cases, Scenarios, and Requirements for RTCWeb document
>          sent to IESG as Informational
> 
> Dec 2011 RTCWeb protocol profiles and Media format specification(s) to
>          IESG as PS
> 
> Dec 2011 Information elements and events APIs Input to W3C
> 
> Apr 2012 API to Protocol mapping document submitted to the IESG as
>          Informational (if needed)
> 
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>, "'David Singer'" <singer@apple.com>
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Subject: Re: [rtcweb] Fwd: RFC 6222 on Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names (CNAMEs)
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Hi,
I think that CNAMEs are also useful for third party monitors as =
described in
RFC 3550
Roni Even

> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Magnus Westerlund
> Sent: Tuesday, April 19, 2011 12:05 PM
> To: David Singer
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Fwd: RFC 6222 on Guidelines for Choosing RTP
> Control Protocol (RTCP) Canonical Names (CNAMEs)
>=20
> David Singer skrev 2011-04-18 20:10:
> >
> > If I am doing a point-to-point call, and I have the connectivity
> > established (i.e. IP addresses and the like), why should I trouble =
at
> > all with CNAME?  And, indeed, are there privacy concerns with it
> > (well, there are with using <userid>@<host>, I rather suspect).  Can
> > the CNAME problem be handled when aggregation happens?
>=20
> Because you don't know when a peer is a single user, or a conference
> bridge. In the later case of you will need the cname. Also for point =
to
> point you might need it if you have multiple sources that should or
> should not be synchronized. The point of the new RFC is that the cname
> can be set session specific, by drawing a random number that is being
> used. Thus you can avoid the downsides of CNAME from privacy point of
> view, and the generation aspects.
>=20
> cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
> _______________________________________________
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> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From magnus.westerlund@ericsson.com  Wed Apr 20 02:16:06 2011
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Subject: Re: [rtcweb] WG Review: Real-Time Communication in WEB-browsers	(rtcweb) - Mandatory media formats
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Roni Even skrev 2011-04-20 06:38:
> Hi,
> I think that during the RTCweb ad-hoc on Friday meeting there was no
> consensus if the proposed WG should specify which media format MUST or
> Should be used. So I am not sure why the charter has item 6 in the work to
> do as well about the deliverables of media format specifications.

Roni,

I don't quite understand your comment. Yes, there was discussion and
there is yet no consensus nor proposal on what media formats there
should be selected nor the level of required support in implementations.

But if you re-read bullet 6 from the charter:

>> 6.  Define a set of media formats that must or should be supported by
>>     a client to improve interoperability.

You see that the charter item has quite a lot of wiggle room for the WG
to define what level a produced list applies to the implementations.

Am I misunderstanding your comment?

cheers

Magnus Westerlund

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

From ron.even.tlv@gmail.com  Wed Apr 20 03:47:47 2011
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>
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Magnus,
OK, no problem if this is the meaning. Maybe I have not understood.
Roni

> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Wednesday, April 20, 2011 12:16 PM
> To: Roni Even
> Cc: iesg@ietf.org; rtcweb@ietf.org
> Subject: Re: [rtcweb] WG Review: Real-Time Communication in WEB-
> browsers (rtcweb) - Mandatory media formats
>=20
> Roni Even skrev 2011-04-20 06:38:
> > Hi,
> > I think that during the RTCweb ad-hoc on Friday meeting there was no
> > consensus if the proposed WG should specify which media format MUST
> or
> > Should be used. So I am not sure why the charter has item 6 in the
> work to
> > do as well about the deliverables of media format specifications.
>=20
> Roni,
>=20
> I don't quite understand your comment. Yes, there was discussion and
> there is yet no consensus nor proposal on what media formats there
> should be selected nor the level of required support in
> implementations.
>=20
> But if you re-read bullet 6 from the charter:
>=20
> >> 6.  Define a set of media formats that must or should be supported
> by
> >>     a client to improve interoperability.
>=20
> You see that the charter item has quite a lot of wiggle room for the =
WG
> to define what level a produced list applies to the implementations.
>=20
> Am I misunderstanding your comment?
>=20
> cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------


From harald@alvestrand.no  Wed Apr 27 05:18:59 2011
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Subject: Re: [rtcweb] data-over-RTP vs data-over-datagram (Re: non-media data)
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Returning to this thread after a few weeks.... I'm still being contrary=20
to make sure we have all the issues explicit where they need to be...=20
and I do hope that some of the people who actually want to send data=20
(instead of us types who speculate about what you possibly might want)=20
can speak up!

On 04/15/11 16:28, Magnus Westerlund wrote:
> Haralad and other interested
>
> Snipped out more parts as I added quite a lot around possible
> alternatives for you to think about. See below
>
> Harald Alvestrand skrev 2011-04-15 15:12:
>>> I promised to return to why I think TCP over RTP or more general,
>>> generic data over RTP isn't the most suitable model.
>>> - RTP is a protocol built towards real-time media. Yes the definition=
 of
>>> media can be pretty loose here. But carrying the individual datagrams=
 of
>>> a TCP connection over RTP is plainly stupid, when TCP over UDP will g=
et
>>> you all the desired functionality anyway with less overhead.
>> RTP is, in the extreme view, a ~12-byte header and a  set of handling
>> rules. We should look at what it is, not so much what it was originall=
y
>> thought to be designed for (a component in a framework for multicast
>> distribution of unidirectional media channels).
> Well, the handling rules include RTCP and an intention to handle a
> number of different topologies with multi-party handling. There are als=
o
> additional signalling issues, like how to configure the RTCP bandwidth
> value, which features to use and all these things. Things you may not
> need for your data transport. If you need them, you anyway should defin=
e
> an RTP payload format for your data format.
If these questions have fixed answers in a specific scenario (like=20
point-to-point), they can be handled by specifying a set of reasonable=20
values for that scenario.
> And to be frank, the transport and RAI ADs has been very hesitant to
> allow any RTP payload format that is defined for just being a bit pipe
> over RTP. The two reasons for this has been Application level framing
> principle and the lack of congestion control for RTP.
I do wonder about the ALF principle. In some ways, it seems reasonable=20
under ALF that one should be able to use a framing mechanism that=20
provides some of what we want without necessarily worrying that it=20
provides more than what we want...

The congestion control is definitely an issue. With RTCWEB deployment=20
likely to increase the amount of media streams encountering congestion=20
significantly, I think we have to address this in a way that's=20
applicable to RTP in general - but that's an AVTCORE matter, not an=20
RTCWEB matter, I think.
>>> - RTP requires RTP payload formats for anything it carries. That is t=
o
>>> ensure that the application level framing ideas are as well captured =
for
>>> a particular media encoding as possible. That optimization is not
>>> possible for general data.
>> I think "general data" is a red herring. We can define a few different=

>> data-carrying "modes" that have different properties; if some of these=

>> are useful, I think it makes sense to define them.
> I agree that different transport protocol provides different properties=
=2E
> And that you should chose the one most appropriate. However, one usuall=
y
> has to compromise on some aspects, either by needing to do specific
> solution for ones data or accept some other potential limitation. On a
> high level comparisons between our transport protocols I think can be
> summarized like this:
>
> TCP: unicast only, congestion controlled reliable byte stream
Also: Implemented in kernels, allowed through firewalls,=20
double-NAT-traversal fails frequently.
> SCTP: unicast, multihoming, congestion controlled, reliable and
> unreliable, message oriented.
Also: Not implemented in kernels, firewalls know nothing of it (unless=20
UDP encapsulated). NAT boxes know nothing about it (so UDP encapsulation =

seems a must).
> DCCP: unicast, unreliable, congestion controlled sequence numbered data=
gram
The "also" list here is the same as for SCTP, I think.
> RTP/UDP: unicast and multicast, unreliable, RTP payload of single MTU
> size, medium time scale transport feedback.
>
> UDP: unicast and multicast, unreliable, datagram
Properties of traversal of firewalls and NAT boxes are well known.=20
Kernel implemented.
>>> - If one can't do general optimization one goes with more general
>>> transport solutions, thus why not use a more general data transport f=
or
>>> the general data?
>> Pushing stuff towards a "more general data transport" requires one of
>> two things:
>> a) having an existing more general data transport that fulfils our
>> requirements
>> b) defining a new more general data transport that fulfils our require=
ments
>> The second seems tough. The first ..... what's your preferred candidat=
e?
> Actually the one I think is easiest from a specification point of view
> and that does provide a reasonable service for unreliable datagrams are=
:
>
> DTLS/DCCP/UDP
>
> That would be the following references:
> https://datatracker.ietf.org/doc/rfc4340/ (Base Spec)
> https://datatracker.ietf.org/doc/rfc4341/ (TCP-Like CC)
> https://datatracker.ietf.org/doc/rfc4342/ (TFRC CC)
> https://datatracker.ietf.org/doc/rfc5238/ (DTLS for DCCP)
> https://datatracker.ietf.org/doc/draft-ietf-dccp-udpencap/ (DCCP over U=
DP)
>
> My main concern is the maturity of available implementations of DCCP.
> Otherwise it provides congestion control and can actually choose which
> congestion control algorithms we want supported, TCP-like or TFRC. It
> also have some additional features. One is that we get a second port
> space providing additional multiplexing points.
>
> Please note that this is just a friendly suggestion. At least check it
> out, and possible post some comments why it would be unsuitable.
In a design point spectrum where Ian Hixie's 44-byte datagram header,=20
this proposal and a "data-over-RTP" proposal are the 3 candidates, we=20
should have enough design points that we can start discussing the=20
criteria we evaluate them against (often a good thing).
>>> - RTP and RTCP monitoring is also designed with more continous medias=
 in
>>> mind. Here we don't know how much such properties will be true. Also =
the
>>> congestion controls available might actually have not the most suitab=
le
>>> properties for certain type of data.
>> It's not clear to me that there *are* congestion controls available
>> (other than media-specific ones).
>> Which ones are you thinking of?
> The one that is full defined is to run RTP over DCCP.
> https://datatracker.ietf.org/doc/rfc5762/
>
> I guess the biggest issue is the lack of experience with this and
> especially how you interface the DCCP stack to work well between media
> encoding, RTP stack and the DCCP congestion control.
If we buy DCCP anyway, this is certainly worth investigating.
> The alternative, not standardized yet but closest available for RTP ove=
r
> UDP is TFRC for RTP.
> https://datatracker.ietf.org/doc/draft-gharai-avtcore-rtp-tfrc/
>
> That was in reasonable shape prior to its abandonment due to lack of
> funding from the ones driving it.
Does anyone want to pick it up and see what's missing?
>
>> Now, the flipside: What does RTP *have* that we will otherwise need
>> to define? From your earlier list:
>>
>> - Congestion Controlled: eh... maybe not.
>> - Confidentiality and  Integrity Protected - done, we've got SRTP.
>> - Providing consent to receive data prior to allow transmission of
>>    substantial amounts of data - done, we have ICE,
>>    and need it for media too.
>> - Some type of source authentication to prevent man in the middles
>>     - done, we have SRTP
>>
>> So apart from congestion control, it seems that RTP's 12-byte header
>> actually gives us a lot.
> Well, the security parts can be provided by ICE and DTLS over UDP also.=

> we still are short the congestion control for such a thing plus all
> these things that the UDP guidelines (RFC5405) discuss. Using RTP over
> UDP does drag all the RTP baggage into something that may not needed it=
,
> but it will provide some parts of what you need. If a particular data
> format needs RTP functionality, it should define its own RTP payload
> format anyway.
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> 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
> ----------------------------------------------------------------------
>



From hoene@uni-tuebingen.de  Wed Apr 27 10:48:48 2011
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From: "Christian Hoene" <hoene@uni-tuebingen.de>
To: "'Harald Alvestrand'" <harald@alvestrand.no>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] data-over-RTP vs data-over-datagram (Re: non-media	data)
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> Returning to this thread after a few weeks.... I'm still being =
contrary
> to make sure we have all the issues explicit where they need to be...
> and I do hope that some of the people who actually want to send data
> (instead of us types who speculate about what you possibly might want)
> can speak up!

[Christian Hoene] Most RTP payload formats are for audio or video. I =
have not seen that RTP is just for other real time data such as GPS, =
location, orientation, input devices (mouse, kinect, etc...) or gaming =
data. Thus, I do not see that data over RTP is an option that is well =
supported. Real-time-Data-over-RTP seems not to be in the =
standardization traditions of AVT. Of course, this might change at some =
point in future.

Until then, I would suggest to support a plain =
real-time-data-over-datagram option, as we do not have to follow =
standards anyhow to support non-media data over whatsoever.=20

Spoken up,
 Christian





>=20
> On 04/15/11 16:28, Magnus Westerlund wrote:
> > Haralad and other interested
> >
> > Snipped out more parts as I added quite a lot around possible
> > alternatives for you to think about. See below
> >
> > Harald Alvestrand skrev 2011-04-15 15:12:
> >>> I promised to return to why I think TCP over RTP or more general,
> >>> generic data over RTP isn't the most suitable model.
> >>> - RTP is a protocol built towards real-time media. Yes the =
definition of
> >>> media can be pretty loose here. But carrying the individual =
datagrams of
> >>> a TCP connection over RTP is plainly stupid, when TCP over UDP =
will get
> >>> you all the desired functionality anyway with less overhead.
> >> RTP is, in the extreme view, a ~12-byte header and a  set of =
handling
> >> rules. We should look at what it is, not so much what it was =
originally
> >> thought to be designed for (a component in a framework for =
multicast
> >> distribution of unidirectional media channels).
> > Well, the handling rules include RTCP and an intention to handle a
> > number of different topologies with multi-party handling. There are =
also
> > additional signalling issues, like how to configure the RTCP =
bandwidth
> > value, which features to use and all these things. Things you may =
not
> > need for your data transport. If you need them, you anyway should =
define
> > an RTP payload format for your data format.
> If these questions have fixed answers in a specific scenario (like
> point-to-point), they can be handled by specifying a set of reasonable
> values for that scenario.
> > And to be frank, the transport and RAI ADs has been very hesitant to
> > allow any RTP payload format that is defined for just being a bit =
pipe
> > over RTP. The two reasons for this has been Application level =
framing
> > principle and the lack of congestion control for RTP.
> I do wonder about the ALF principle. In some ways, it seems reasonable
> under ALF that one should be able to use a framing mechanism that
> provides some of what we want without necessarily worrying that it
> provides more than what we want...
>=20
> The congestion control is definitely an issue. With RTCWEB deployment
> likely to increase the amount of media streams encountering congestion
> significantly, I think we have to address this in a way that's
> applicable to RTP in general - but that's an AVTCORE matter, not an
> RTCWEB matter, I think.
> >>> - RTP requires RTP payload formats for anything it carries. That =
is to
> >>> ensure that the application level framing ideas are as well =
captured for
> >>> a particular media encoding as possible. That optimization is not
> >>> possible for general data.
> >> I think "general data" is a red herring. We can define a few =
different
> >> data-carrying "modes" that have different properties; if some of =
these
> >> are useful, I think it makes sense to define them.
> > I agree that different transport protocol provides different =
properties.
> > And that you should chose the one most appropriate. However, one
> usually
> > has to compromise on some aspects, either by needing to do specific
> > solution for ones data or accept some other potential limitation. On =
a
> > high level comparisons between our transport protocols I think can =
be
> > summarized like this:
> >
> > TCP: unicast only, congestion controlled reliable byte stream
> Also: Implemented in kernels, allowed through firewalls,
> double-NAT-traversal fails frequently.
> > SCTP: unicast, multihoming, congestion controlled, reliable and
> > unreliable, message oriented.
> Also: Not implemented in kernels, firewalls know nothing of it (unless
> UDP encapsulated). NAT boxes know nothing about it (so UDP =
encapsulation
> seems a must).
> > DCCP: unicast, unreliable, congestion controlled sequence numbered
> datagram
> The "also" list here is the same as for SCTP, I think.
> > RTP/UDP: unicast and multicast, unreliable, RTP payload of single =
MTU
> > size, medium time scale transport feedback.
> >
> > UDP: unicast and multicast, unreliable, datagram
> Properties of traversal of firewalls and NAT boxes are well known.
> Kernel implemented.
> >>> - If one can't do general optimization one goes with more general
> >>> transport solutions, thus why not use a more general data =
transport for
> >>> the general data?
> >> Pushing stuff towards a "more general data transport" requires one =
of
> >> two things:
> >> a) having an existing more general data transport that fulfils our
> >> requirements
> >> b) defining a new more general data transport that fulfils our
> requirements
> >> The second seems tough. The first ..... what's your preferred =
candidate?
> > Actually the one I think is easiest from a specification point of =
view
> > and that does provide a reasonable service for unreliable datagrams =
are:
> >
> > DTLS/DCCP/UDP
> >
> > That would be the following references:
> > https://datatracker.ietf.org/doc/rfc4340/ (Base Spec)
> > https://datatracker.ietf.org/doc/rfc4341/ (TCP-Like CC)
> > https://datatracker.ietf.org/doc/rfc4342/ (TFRC CC)
> > https://datatracker.ietf.org/doc/rfc5238/ (DTLS for DCCP)
> > https://datatracker.ietf.org/doc/draft-ietf-dccp-udpencap/ (DCCP =
over
> UDP)
> >
> > My main concern is the maturity of available implementations of =
DCCP.
> > Otherwise it provides congestion control and can actually choose =
which
> > congestion control algorithms we want supported, TCP-like or TFRC. =
It
> > also have some additional features. One is that we get a second port
> > space providing additional multiplexing points.
> >
> > Please note that this is just a friendly suggestion. At least check =
it
> > out, and possible post some comments why it would be unsuitable.
> In a design point spectrum where Ian Hixie's 44-byte datagram header,
> this proposal and a "data-over-RTP" proposal are the 3 candidates, we
> should have enough design points that we can start discussing the
> criteria we evaluate them against (often a good thing).
> >>> - RTP and RTCP monitoring is also designed with more continous =
medias
> in
> >>> mind. Here we don't know how much such properties will be true. =
Also
> the
> >>> congestion controls available might actually have not the most =
suitable
> >>> properties for certain type of data.
> >> It's not clear to me that there *are* congestion controls available
> >> (other than media-specific ones).
> >> Which ones are you thinking of?
> > The one that is full defined is to run RTP over DCCP.
> > https://datatracker.ietf.org/doc/rfc5762/
> >
> > I guess the biggest issue is the lack of experience with this and
> > especially how you interface the DCCP stack to work well between =
media
> > encoding, RTP stack and the DCCP congestion control.
> If we buy DCCP anyway, this is certainly worth investigating.
> > The alternative, not standardized yet but closest available for RTP =
over
> > UDP is TFRC for RTP.
> > https://datatracker.ietf.org/doc/draft-gharai-avtcore-rtp-tfrc/
> >
> > That was in reasonable shape prior to its abandonment due to lack of
> > funding from the ones driving it.
> Does anyone want to pick it up and see what's missing?
> >
> >> Now, the flipside: What does RTP *have* that we will otherwise need
> >> to define? From your earlier list:
> >>
> >> - Congestion Controlled: eh... maybe not.
> >> - Confidentiality and  Integrity Protected - done, we've got SRTP.
> >> - Providing consent to receive data prior to allow transmission of
> >>    substantial amounts of data - done, we have ICE,
> >>    and need it for media too.
> >> - Some type of source authentication to prevent man in the middles
> >>     - done, we have SRTP
> >>
> >> So apart from congestion control, it seems that RTP's 12-byte =
header
> >> actually gives us a lot.
> > Well, the security parts can be provided by ICE and DTLS over UDP =
also.
> > we still are short the congestion control for such a thing plus all
> > these things that the UDP guidelines (RFC5405) discuss. Using RTP =
over
> > UDP does drag all the RTP baggage into something that may not needed =
it,
> > but it will provide some parts of what you need. If a particular =
data
> > format needs RTP functionality, it should define its own RTP payload
> > format anyway.
> >
> > Cheers
> >
> > Magnus Westerlund
> >
> > =
----------------------------------------------------------------------
> > Multimedia Technologies, Ericsson Research EAB/TVM
> > =
----------------------------------------------------------------------
> > 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
> > =
----------------------------------------------------------------------
> >
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From csp@csperkins.org  Wed Apr 27 11:46:12 2011
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To: Christian Hoene <hoene@uni-tuebingen.de>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] data-over-RTP vs data-over-datagram (Re: non-media	data)
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On 27 Apr 2011, at 18:48, Christian Hoene wrote:
>> Returning to this thread after a few weeks.... I'm still being =
contrary
>> to make sure we have all the issues explicit where they need to be...
>> and I do hope that some of the people who actually want to send data
>> (instead of us types who speculate about what you possibly might =
want)
>> can speak up!
>=20
> [Christian Hoene] Most RTP payload formats are for audio or video. I =
have not seen that RTP is just for other real time data such as GPS, =
location, orientation, input devices (mouse, kinect, etc...) or gaming =
data. Thus, I do not see that data over RTP is an option that is well =
supported. Real-time-Data-over-RTP seems not to be in the =
standardization traditions of AVT. Of course, this might change at some =
point in future.
>=20
> Until then, I would suggest to support a plain =
real-time-data-over-datagram option, as we do not have to follow =
standards anyhow to support non-media data over whatsoever.=20


RTP is designed for real-time data, and is built on the assumption that =
the data is time sensitive, and should be processed subject to time =
constraints at the receiver. There are RTP payload formats for =
non-audio-visual media, for example text conversation [RFC2793 and =
RFC4103] and pointers [RFC2862]. I can envisage RTP being appropriate =
for various types of input device, and for some forms of data being sent =
in games, and for some types of sensor data, since those things are =
timed. It's not a general purpose data transfer protocol though, and =
isn't well suited for non-timed data.

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




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From: Colin Perkins <csp@csperkins.org>
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To: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] data-over-RTP vs data-over-datagram (Re: non-media data)
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On 27 Apr 2011, at 13:18, Harald Alvestrand wrote:
> Returning to this thread after a few weeks.... I'm still being =
contrary to make sure we have all the issues explicit where they need to =
be... and I do hope that some of the people who actually want to send =
data (instead of us types who speculate about what you possibly might =
want) can speak up!
>=20
> On 04/15/11 16:28, Magnus Westerlund wrote:
>> Haralad and other interested
>>=20
>> Snipped out more parts as I added quite a lot around possible
>> alternatives for you to think about. See below
>>=20
>> Harald Alvestrand skrev 2011-04-15 15:12:
>>>> I promised to return to why I think TCP over RTP or more general,
>>>> generic data over RTP isn't the most suitable model.
>>>> - RTP is a protocol built towards real-time media. Yes the =
definition of
>>>> media can be pretty loose here. But carrying the individual =
datagrams of
>>>> a TCP connection over RTP is plainly stupid, when TCP over UDP will =
get
>>>> you all the desired functionality anyway with less overhead.
>>> RTP is, in the extreme view, a ~12-byte header and a  set of =
handling
>>> rules. We should look at what it is, not so much what it was =
originally
>>> thought to be designed for (a component in a framework for multicast
>>> distribution of unidirectional media channels).
>> Well, the handling rules include RTCP and an intention to handle a
>> number of different topologies with multi-party handling. There are =
also
>> additional signalling issues, like how to configure the RTCP =
bandwidth
>> value, which features to use and all these things. Things you may not
>> need for your data transport. If you need them, you anyway should =
define
>> an RTP payload format for your data format.
> If these questions have fixed answers in a specific scenario (like =
point-to-point), they can be handled by specifying a set of reasonable =
values for that scenario.
>> And to be frank, the transport and RAI ADs has been very hesitant to
>> allow any RTP payload format that is defined for just being a bit =
pipe
>> over RTP. The two reasons for this has been Application level framing
>> principle and the lack of congestion control for RTP.
> I do wonder about the ALF principle. In some ways, it seems reasonable =
under ALF that one should be able to use a framing mechanism that =
provides some of what we want without necessarily worrying that it =
provides more than what we want...


A protocol isn't just a framing format though. The semantics of RTP are =
very much focussed on real-time, time sensitive, media delivery. Trying =
to use it outside that space is likely to cause problematic semantic =
mismatch, to the extent that you'd be better duplicating the framing =
(which is the most trivial aspect of RTP) in another protocol that =
doesn't have all the other features of RTP that you don't want.

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




From singer@apple.com  Thu Apr 28 06:44:44 2011
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Subject: Re: [rtcweb] data-over-RTP vs data-over-datagram (Re:	non-media	data)
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On Apr 27, 2011, at 14:46 , Colin Perkins wrote:

> On 27 Apr 2011, at 18:48, Christian Hoene wrote:
>>> Returning to this thread after a few weeks.... I'm still being contrary
>>> to make sure we have all the issues explicit where they need to be...
>>> and I do hope that some of the people who actually want to send data
>>> (instead of us types who speculate about what you possibly might want)
>>> can speak up!
>> 
>> [Christian Hoene] Most RTP payload formats are for audio or video. I have not seen that RTP is just for other real time data such as GPS, location, orientation, input devices (mouse, kinect, etc...) or gaming data. Thus, I do not see that data over RTP is an option that is well supported. Real-time-Data-over-RTP seems not to be in the standardization traditions of AVT. Of course, this might change at some point in future.
>> 
>> Until then, I would suggest to support a plain real-time-data-over-datagram option, as we do not have to follow standards anyhow to support non-media data over whatsoever. 
> 
> 
> RTP is designed for real-time data, and is built on the assumption that the data is time sensitive, and should be processed subject to time constraints at the receiver. There are RTP payload formats for non-audio-visual media, for example text conversation [RFC2793 and RFC4103] and pointers [RFC2862]. I can envisage RTP being appropriate for various types of input device, and for some forms of data being sent in games, and for some types of sensor data, since those things are timed. It's not a general purpose data transfer protocol though, and isn't well suited for non-timed data.
> 

Specifically, I'd say that, given the choice between getting all the data (possibly at some indefinite time in the future) or getting data on time (but possibly incomplete), TCP attempts to do the former, and RTP the latter.  That makes RTP unsuitable if loss is undesirable and/or timeliness is not needed.

David Singer
Multimedia and Software Standards, Apple Inc.


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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Colin Perkins <csp@csperkins.org>, Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] data-over-RTP vs data-over-datagram (Re: non-media	data)
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>=20
> A protocol isn't just a framing format though. The semantics=20
> of RTP are very much focussed on real-time, time sensitive,=20
> media delivery. Trying to use it outside that space is likely=20
> to cause problematic semantic mismatch, to the extent that=20
> you'd be better duplicating the framing (which is the most=20
> trivial aspect of RTP) in another protocol that doesn't have=20
> all the other features of RTP that you don't want.

[JRE] Not to mention all the RTCP stuff comes with RTP and probably isn't r=
elevant to non-real-time data.

John


>=20
> --=20
> Colin Perkins
> http://csperkins.org/
>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =
