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. There was a draft but it has been abandonded ( http://tools.ietf.org/html/draft-ietf-avt-tfrc-profile-10) On Thu, Jan 20, 2011 at 11:50 PM, Stefan Håkansson LK < stefan.lk.hakansson@ericsson.com> wrote:
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).
------------------------------ *From:* Harald Alvestrand [mailto:harald@alvestrand.no] *Sent:* den 21 januari 2011 08:46 *To:* Henry Sinnreich *Cc:* Stefan Håkansson LK; Stephen Botzko; Cullen Jennings; rtc-web@alvestrand.no; DISPATCH list *Subject:* Rate control and codec adaption (Re: [RTW] [dispatch] The charter formerly know as RTC-WEB take 3)
On 01/21/2011 12:06 AM, Henry Sinnreich wrote:
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.
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. Can you explain?
Changing the subject to content of thread....
are we reducing to a previously solved problem, or to a previously unsolved problem? 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).
Thanks, Henry
On 1/20/11 2:02 PM, "Stefan Håkansson LK" < stefan.lk.hakansson@ericsson.com> wrote:
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, Stefan
------------------------------ *From:* Stephen Botzko [mailto:stephen.botzko@gmail.com<stephen.botzko@gmail.com>]
*Sent:* den 20 januari 2011 16:45 *To:* Henry Sinnreich *Cc:* Stefan Håkansson LK; Cullen Jennings; DISPATCH list; rtc-web@alvestrand.no *Subject:* Re: [dispatch] The charter formerly know as RTC-WEB take 3
How does this fit with adaptive codecs?
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.
Hint: codec selection matters, is actually critical to this effort.
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.
Stephen Botzko
On Thu, Jan 20, 2011 at 10:37 AM, Henry Sinnreich < henry.sinnreich@gmail.com> wrote:
Hi Stefan,
2. The second one is about rate adaptation/congestion control. It is not mentioned at all. I don't know if it is needed, perhaps it is enough that RFC3550 (that is already pointed at) has a section about it, but I wanted to highlight it.
How does this fit with adaptive codecs? Hint: codec selection matters, is actually critical to this effort.
Thanks, Henry
On 1/20/11 3:52 AM, "Stefan Håkansson LK" < stefan.lk.hakansson@ericsson.com> wrote:
Hi Cullen,
two comments:
1. As requirements on the API are explicitly described, I thinke that there should be a comment that the API must support media format negotiation. Proposal: "The API must enable media format negotiation and application influence over media format selection".
2. The second one is about rate adaptation/congestion control. It is not mentioned at all. I don't know if it is needed, perhaps it is enough that RFC3550 (that is already pointed at) has a section about it, but I wanted to highlight it.
Br, Stefan
-----Original Message----- From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org <dispatch-bounces@ietf.org>] On Behalf Of Cullen Jennings Sent: den 18 januari 2011 05:59 To: DISPATCH list Cc: rtc-web@alvestrand.no Subject: [dispatch] The charter formerly know as RTC-WEB take 3
In my dispatch co-chair role, I tried to take all the comments I had seen on the list about this charter and see if I could address them in a new version of the charter. I probably messed up in some places. There were some conversation that did not seem to be converging so I did not make any changes for theses. Have a read and if you think something needs to be changed, propose text changes along with the reasons why and we will keep the evolving this charter.
Thanks Cullen
-------------------------------------------------------------- --------------------
Version: 3
Possible Names:
RTCWEB WEBRTC STORM: Standardized Transport Oriented for Realtime Media BURN: Browsers Using Realtime Media WAVE: Web And Voice/Video Enablement WAVVE: Web And Voice Video Enablement REALTIME WEBCOMM WREALTIME WEBTIME WEBFLOWS BRAVO Browser Realtime Audio and VideO COBWEB COmmuication Between WEBclients WHEELTIME
Body:
Many implementations have been made that use a Web browser to support direct, interactive communications, including voice, video, collaboration, and gaming. In these implementations, the web server acts as the signaling path between these applications, using locally significant identifiers to set up the association. Up till now, such applications have typically required the installation of plugins or non-standard browser extensions. There is a desire to standardize this functionality, so that this type of application can be run in any compatible browser and allow for high-quality real-time communications experiences within the browser.
Traditionally, the W3C has defined API and markup languages such as HTML that work in conjunction with with the IETF over the wire protocols such as HTTP to allow web browsers to display media that does not have real time interactive constraints with another human.
The W3C and IETF plan to collaborate together in their traditional way to meet the evolving needs of browsers. Specifically the IETF will provide a set of on the wire protocols, including RTP, to meet the needs on interactive communications, and the W3C will define the API and markup to allow web application developers to control the on the wire protocols. This will allow application developers to write applications that run in a browser and facilitate interactive communications between users for voice and video communications, collaboration, and gaming.
This working group will select and define a minimal set of protocols that will enable browsers to:
* have interactive real time voice and video pairwise be
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