RTCP typically isn't sent frequently enough to allow for real-time adjustments in bitrate. TFRC provides a nice mechanism for controlling bitrate in real-time, but the work to apply TFRC to RTP has not yet been codified into a standard.<div>
<br></div><div>There was a draft but it has been abandonded (<a href="http://tools.ietf.org/html/draft-ietf-avt-tfrc-profile-10">http://tools.ietf.org/html/draft-ietf-avt-tfrc-profile-10</a>)<br><br><div class="gmail_quote">
On Thu, Jan 20, 2011 at 11:50 PM, Stefan Håkansson LK <span dir="ltr"><<a href="mailto:stefan.lk.hakansson@ericsson.com">stefan.lk.hakansson@ericsson.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div text="#000000" bgcolor="#ffffff">
<div dir="ltr" align="left"><span><font face="Arial" color="#0000ff" size="2">My view: we are discussing a problem already solved! The
common procedure would be to use info in the RTCP reports from the receiving end
to change the transmitted bit rate (if change is required).
</font></span></div><br>
<blockquote style="padding-left:5px;margin-left:5px;border-left:#0000ff 2px solid;margin-right:0px">
<div lang="en-us" dir="ltr" align="left">
<hr>
<font face="Tahoma" size="2"><b>From:</b> Harald Alvestrand
[mailto:<a href="mailto:harald@alvestrand.no" target="_blank">harald@alvestrand.no</a>] <br><b>Sent:</b> den 21 januari 2011
08:46<br><b>To:</b> Henry Sinnreich<br><b>Cc:</b> Stefan Håkansson LK; Stephen
Botzko; Cullen Jennings; <a href="mailto:rtc-web@alvestrand.no" target="_blank">rtc-web@alvestrand.no</a>; DISPATCH
list<br><b>Subject:</b> Rate control and codec adaption (Re: [RTW] [dispatch]
The charter formerly know as RTC-WEB take 3)<br></font><br></div><div><div></div><div class="h5">
<div></div>On 01/21/2011 12:06 AM, Henry Sinnreich wrote:
<blockquote type="cite"><font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size:13pt">></span></font><span style="font-size:13pt"><font color="#0000fe"><font face="Arial">Minor comment:
I think all codecs that have been discussed (except for G.711) are adaptive
in the sense that their bitrate can be adapted.<br></font></font><font face="Calibri, Verdana, Helvetica, Arial"><br>It is not clear to me how to
avoid the codec adaptation mechanism fighting the rate control mechanism,
without some guidance in the standard for developers.<br>Can you
explain?<br></font></span></blockquote>Changing the subject to content of
thread....<br><br>are we reducing to a previously solved problem, or to a
previously unsolved problem?<br>I don't see how this problem actually differs
from the one that people will have when operating RTP under TFRC
(draft-ietf-avt-tfrc-profile-10).<br><br>
<blockquote type="cite"><span style="font-size:13pt"><font face="Calibri,
Verdana, Helvetica, Arial"><br>Thanks,
Henry<br><br><br>On 1/20/11 2:02 PM, "Stefan Håkansson LK" <<a href="http://stefan.lk.hakansson@ericsson.com" target="_blank">stefan.lk.hakansson@ericsson.com</a>>
wrote:<br><br></font></span>
<blockquote><span style="font-size:13pt"><font color="#0000ff"><font face="Arial">Minor comment: I think all codecs that have been discussed
(except for G.711) are adaptive in the sense that their bitrate can be
adapted.<br></font></font><font face="Calibri, Verdana, Helvetica,
Arial"><br></font><font color="#0000ff"><font face="Arial">Br,<br>Stefan<br></font></font><font face="Calibri, Verdana, Helvetica,
Arial"><br></font></span>
<blockquote><span style="font-size:13pt"><font face="Calibri,
Verdana, Helvetica, Arial"><br> <br>
<hr align="center" width="100%" size="3">
</font><font face="Tahoma, Verdana, Helvetica, Arial"><b>From:</b>
Stephen Botzko [<a href="mailto:stephen.botzko@gmail.com" target="_blank">mailto:stephen.botzko@gmail.com</a>]
<br><b>Sent:</b> den 20 januari 2011 16:45<br><b>To:</b> Henry
Sinnreich<br><b>Cc:</b> Stefan Håkansson LK; Cullen Jennings;
DISPATCH list; <a href="http://rtc-web@alvestrand.no" target="_blank">rtc-web@alvestrand.no</a><br><b>Subject:</b> Re:
[dispatch] The charter formerly know as RTC-WEB take
3<br></font><font face="Calibri, Verdana, Helvetica, Arial"><br> <br>>>><br> How
does this fit with adaptive codecs?<br>>>><br>Just
because some codecs can adapt doesn't mean rate
adaptation/congestion control should be left out of the scope. I
think it needs to be
considered.<br><br>>>><br> Hint: codec
selection matters, is actually critical to this
effort.<br>>>><br>Codec selection does matter, but I am
not convinced that mandatory codecs need to be in the RFCs.
I believe market forces are sufficient - SIP itself is one
proof point.<br><br>Stephen Botzko<br><br><br> <br>On Thu,
Jan 20, 2011 at 10:37 AM, Henry Sinnreich <<a href="http://henry.sinnreich@gmail.com" target="_blank">henry.sinnreich@gmail.com</a>>
wrote:<br> <br></font></span>
<blockquote><span style="font-size:13pt"><font face="Calibri, Verdana, Helvetica, Arial">Hi
Stefan,<br> <br><br>> 2. The second one is about rate
adaptation/congestion control. It is not<br>> mentioned at
all. I don't know if it is needed, perhaps it is enough
that<br>> RFC3550 (that is already pointed at) has a section
about it, but I wanted to<br>> highlight it.<br><br>How does
this fit with adaptive codecs?<br>Hint: codec selection matters, is
actually critical to this effort.<br><br>Thanks,
Henry<br><br><br>On 1/20/11 3:52 AM, "Stefan Håkansson LK"
<<a href="http://stefan.lk.hakansson@ericsson.com" target="_blank">stefan.lk.hakansson@ericsson.com</a>><br>wrote:<br> <br> <br> <br><br>>
Hi Cullen,<br>><br>> two comments:<br>><br>> 1. As
requirements on the API are explicitly described, I thinke that
there<br>> should be a comment that the API must support
media format negotiation.<br>> Proposal: "The API must enable
media format negotiation and application<br>> influence over
media format selection".<br>><br>> 2. The second one is
about rate adaptation/congestion control. It is not<br>>
mentioned at all. I don't know if it is needed, perhaps it is
enough that<br>> RFC3550 (that is already pointed at) has a
section about it, but I wanted to<br>> highlight
it.<br>><br>> Br,<br>> Stefan<br>><br>>>
-----Original Message-----<br>>> From: <a href="http://dispatch-bounces@ietf.org" target="_blank">dispatch-bounces@ietf.org</a><br>>>
[<a href="mailto:dispatch-bounces@ietf.org" target="_blank">mailto:dispatch-bounces@ietf.org</a>] On
Behalf Of Cullen Jennings<br>>> Sent: den 18 januari 2011
05:59<br>>> To: DISPATCH list<br>>> Cc: <a href="http://rtc-web@alvestrand.no" target="_blank">rtc-web@alvestrand.no</a><br>>>
Subject: [dispatch] The charter formerly know as RTC-WEB take
3<br>>><br>>><br>>> In my dispatch co-chair
role, I tried to take all the<br>>> comments I had seen on
the list about this charter and see if<br>>> I could
address them in a new version of the charter. I<br>>>
probably messed up in some places. There were some<br>>>
conversation that did not seem to be converging so I did
not<br>>> make any changes for theses. Have a read and if
you think<br>>> something needs to be changed, propose
text changes along<br>>> with the reasons why and we will
keep the evolving this charter.<br>>><br>>>
Thanks<br>>> Cullen<br>>><br>>>
--------------------------------------------------------------<br>>>
--------------------<br>>><br>>> Version:
3<br>>><br>>> Possible
Names:<br>>><br>>> RTCWEB<br>>>
WEBRTC<br>>> STORM: Standardized Transport Oriented for
Realtime Media<br>>> BURN: Browsers Using Realtime
Media<br>>> WAVE: Web And Voice/Video
Enablement<br>>> WAVVE: Web And Voice Video
Enablement<br>>> REALTIME<br>>> WEBCOMM<br>>>
WREALTIME<br>>> WEBTIME<br>>> WEBFLOWS<br>>>
BRAVO Browser Realtime Audio and VideO<br>>> COBWEB
COmmuication Between WEBclients<br>>>
WHEELTIME<br>>><br>>><br>>><br>>>
Body:<br>>><br>>> Many implementations have been
made that use a Web browser to<br>>> support direct,
interactive communications, including voice,<br>>> video,
collaboration, and gaming. In these
implementations,<br>>> the web server acts as the signaling path
between these<br>>> applications, using locally
significant identifiers to set up<br>>> the association.
Up till now, such applications have<br>>> typically
required the installation of plugins or<br>>> non-standard
browser extensions. There is a desire to<br>>>
standardize this functionality, so that this type of<br>>>
application can be run in any compatible browser and
allow<br>>> for high-quality real-time communications
experiences within<br>>> the
browser.<br>>><br>>> Traditionally, the W3C has
defined API and markup languages<br>>> such as HTML that work
in conjunction with with the IETF over<br>>> the wire
protocols such as HTTP to allow web browsers to<br>>>
display media that does not have real time
interactive<br>>> constraints with another
human.<br>>><br>>> The W3C and IETF plan to
collaborate together in their<br>>> traditional way to
meet the evolving needs of browsers.<br>>> Specifically
the IETF will provide a set of on the wire<br>>>
protocols, including RTP, to meet the needs on
interactive<br>>> communications, and the W3C will define
the API and markup to<br>>> allow web application
developers to control the on the wire<br>>> protocols.
This will allow application developers to write<br>>>
applications that run in a browser and facilitate
interactive<br>>> communications between users for voice
and video<br>>> communications, collaboration, and
gaming.<br>>><br>>> This working group will select
and define a minimal set of<br>>> protocols that will
enable browsers to:<br>>><br>>> * have interactive
real time voice and video pairwise
be<br></font></span></blockquote></blockquote></blockquote><pre><fieldset></fieldset>
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