[RTW] Rate control and codec adaption (Re: [dispatch] The charter formerly know as RTC-WEB take 3)
Saverio Mascolo
saverio.mascolo at gmail.com
Sun Jan 23 08:58:08 CET 2011
On Sun, Jan 23, 2011 at 6:41 AM, Justin Uberti <juberti at google.com> wrote:
> Google Video Chat uses a TFRC-based algorithm for rate control.
>
what is the source of this information?
-sm
>
> On Sat, Jan 22, 2011 at 6:18 AM, Saverio Mascolo <
> saverio.mascolo at gmail.com> wrote:
>
>>
>>
>> On Fri, Jan 21, 2011 at 8:43 PM, Justin Uberti <juberti at google.com>wrote:
>>
>>> TFRC isn't perfect, but it seems to work pretty well in practice.
>>
>>
>> in practice where????
>>
>> -sm
>>
>> The RTP extension header overhead of 12 bytes per packet is fairly nominal
>>> (1%) at today's video bitrates, as is the cost of the RTCP feedback message.
>>>
>>> I'm not aware of any other standards-track bandwidth estimation
>>> algorithms designed to work with RTP/UDP.
>>>
>>> On Fri, Jan 21, 2011 at 9:46 AM, <tom_harper at logitech.com> wrote:
>>>
>>>> It seems to me neither avpf or tfrc is fully perfect- on the whole tfrc
>>>> seems to be better than avpf in terms of constant measurement of the
>>>> connection-
>>>>
>>>> tfrc seems scary/impractical at low latencies due to the following:
>>>> "The TFRC requirements of receiving feedback once per RTT can at times
>>>> conflict with the AVP RTCP bandwidth constraints, particularly at
>>>> small RTTs of 20 ms or less"
>>>> and the fact that it has to be attached as an extension header to every
>>>> data packet seems like more overhead than is needed, but others opinions may
>>>> differ on this.
>>>>
>>>> We support avpf as defined 5104/4585, but prefer not to use it as in
>>>> some scenarios we have run into the rtcp bandwidth cap- and then you get no
>>>> feedback at all in a timely manner.
>>>>
>>>> Are there any other inband schemes that are up in rfc at this point?
>>>>
>>>> Tom
>>>>
>>>>
>>>>
>>>> [image: Inactive hide details for Stefan H嶡ansson LK ---01/21/2011
>>>> 12:38:33 AM---Isn't it so that with the AVPF profile you can actua]Stefan
>>>> H嶡ansson LK ---01/21/2011 12:38:33 AM---Isn't it so that with the AVPF
>>>> profile you can actually sent RTCP when there is a need (even if a tr
>>>>
>>>> From: Stefan H嶡ansson LK <stefan.lk.hakansson at ericsson.com>
>>>> To: Justin Uberti <juberti at google.com>
>>>> Cc: Cullen Jennings <fluffy at cisco.com>, DISPATCH list <
>>>> dispatch at ietf.org>, Henry Sinnreich <henry.sinnreich at gmail.com>, Harald
>>>> Alvestrand <harald at alvestrand.no>, "rtc-web at alvestrand.no" <
>>>> rtc-web at alvestrand.no>, Stephen Botzko <stephen.botzko at gmail.com>
>>>> Date: 01/21/2011 12:38 AM
>>>>
>>>> Subject: Re: [RTW] Rate control and codec adaption (Re: [dispatch] The
>>>> charter formerly know as RTC-WEB take 3)
>>>> Sent by: rtc-web-bounces at alvestrand.no
>>>> ------------------------------
>>>>
>>>>
>>>>
>>>> Isn't it so that with the AVPF profile you can actually sent RTCP when
>>>> there is a need (even if a transmission is not due)? This way you can
>>>> actually react fast.
>>>>
>>>> ------------------------------
>>>> *From:* Justin Uberti [mailto:juberti at google.com <juberti at google.com>]
>>>> *
>>>> Sent:* den 21 januari 2011 09:13*
>>>> To:* Stefan Håkansson LK*
>>>> Cc:* Harald Alvestrand; Henry Sinnreich; Cullen Jennings;
>>>> rtc-web at alvestrand.no; DISPATCH list; Stephen Botzko*
>>>> Subject:* Re: [RTW] Rate control and codec adaption (Re: [dispatch] The
>>>> charter formerly know as RTC-WEB take 3)
>>>>
>>>> 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*<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 at ericsson.com* <stefan.lk.hakansson at 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 at alvestrand.no*<harald at alvestrand.no>]
>>>> *
>>>> Sent:* den 21 januari 2011 08:46*
>>>> To:* Henry Sinnreich*
>>>> Cc:* Stefan Håkansson LK; Stephen Botzko; Cullen Jennings; *
>>>> rtc-web at alvestrand.no* <rtc-web at 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 at ericsson.com*<http://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 at gmail.com*<stephen.botzko at gmail.com>]
>>>> *
>>>> Sent:* den 20 januari 2011 16:45*
>>>> To:* Henry Sinnreich*
>>>> Cc:* Stefan Håkansson LK; Cullen Jennings;
>>>> DISPATCH list; *rtc-web at alvestrand.no*<http://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 at gmail.com*<http://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 at ericsson.com*<http://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 at ietf.org*<http://dispatch-bounces@ietf.org/>
>>>> >> [*mailto:dispatch-bounces at ietf.org*<dispatch-bounces at ietf.org>]
>>>> On Behalf Of Cullen Jennings
>>>> >> Sent: den 18 januari 2011 05:59
>>>> >> To: DISPATCH list
>>>> >> Cc: *rtc-web at alvestrand.no*<http://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
>>>>
>>>> _______________________________________________
>>>> RTC-Web mailing list
>>>> *RTC-Web at alvestrand.no* <RTC-Web at alvestrand.no>
>>>> *http://www.alvestrand.no/mailman/listinfo/rtc-web*<http://www.alvestrand.no/mailman/listinfo/rtc-web>
>>>>
>>>>
>>>> _______________________________________________
>>>> RTC-Web mailing list*
>>>> **RTC-Web at alvestrand.no* <RTC-Web at alvestrand.no>*
>>>> **http://www.alvestrand.no/mailman/listinfo/rtc-web*<http://www.alvestrand.no/mailman/listinfo/rtc-web>
>>>>
>>>> _______________________________________________
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>>>>
>>>>
>>>> _______________________________________________
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>>>> RTC-Web at alvestrand.no
>>>> http://www.alvestrand.no/mailman/listinfo/rtc-web
>>>>
>>>>
>>>
>>> _______________________________________________
>>> RTC-Web mailing list
>>> RTC-Web at alvestrand.no
>>> http://www.alvestrand.no/mailman/listinfo/rtc-web
>>>
>>>
>>
>>
>> --
>> Prof. Saverio Mascolo
>> Dipartimento di Elettrotecnica ed Elettronica
>> Politecnico di Bari
>> Via Orabona 4, 70125 Bari Italy
>> Tel. +39 080 5963621 <tel:+390805963621>
>> Fax. +39 080 5963410 <tel:+390805963410>
>> email:mascolo at poliba.it <email%3Amascolo at poliba.it>
>>
>> http://c3lab.poliba.it
>>
>>
>> =================================
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>
>
--
Prof. Saverio Mascolo
Dipartimento di Elettrotecnica ed Elettronica
Politecnico di Bari
Via Orabona 4, 70125 Bari Italy
Tel. +39 080 5963621
Fax. +39 080 5963410
email:mascolo at poliba.it <email%3Amascolo at poliba.it>
http://c3lab.poliba.it
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