Re: [rtcweb] [tsvwg] Diffserv QoS for Video
<Ruediger.Geib@telekom.de> Fri, 13 May 2016 06:49 UTC
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From: Ruediger.Geib@telekom.de
To: david.black@emc.com
Date: Fri, 13 May 2016 08:47:32 +0200
Thread-Topic: [tsvwg] Diffserv QoS for Video
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Cc: rtcweb@ietf.org, tsvwg@ietf.org, brian.e.carpenter@gmail.com
Subject: Re: [rtcweb] [tsvwg] Diffserv QoS for Video
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David, as I mentioned in an earlier mail, all changes where the WG reached agreement are fine with me and others I don't pursue. Regards, Ruediger -----Ursprüngliche Nachricht----- Von: Brian E Carpenter [mailto:brian.e.carpenter@gmail.com] Gesendet: Donnerstag, 12. Mai 2016 22:21 An: Black, David; Geib, Rüdiger; harald@alvestrand.no Cc: rtcweb@ietf.org; tsvwg@ietf.org Betreff: Re: [tsvwg] Diffserv QoS for Video > Also, in addition to the -16 version of the WebRTC QoS draft that was > just uploaded, fwiw, as one of the people who's been commenting, I am happy with the changes that have been made in that version. Regards Brian On 12/05/2016 12:19, Black, David wrote: >> I'm working on transport only and my rtcweb input is >> draft-ietf-tsvwg-rtcweb- qos-15. It states little for audience like >> me, if the subject is audio and video related to a talking person. I >> think the relevant statement are lines in a table. I'd appreciate >> text helping with the interpretation of table. If we can't agree on useful text - leave it as is. > > As draft shepherd, I think the table text is ok as-is; other WebRTC > specs own the definition of what a media flow is, and one of them > would be the appropriate location for text (if appropriate) to point out that audio for video could be sent > on the same or different or different media flow as the video. Among the reasons > for this is that deciding whether to use one or two media flows has > RTP impacts that occur well before DSCP assignment to outbound traffic. > > Also, in addition to the -16 version of the WebRTC QoS draft that was > just uploaded, I've edited that draft's shepherd writeup to reflect > the resolution of the issue around interactive vs. non-interactive > video, including recording the expectation that there will be some > additional text on this topic in the next version of the WebRTC Transports draft. > > Thanks, --David > >> -----Original Message----- >> From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of >> Ruediger.Geib@telekom.de >> Sent: Tuesday, May 10, 2016 4:19 AM >> To: harald@alvestrand.no >> Cc: rtcweb@ietf.org; tsvwg@ietf.org >> Subject: Re: [tsvwg] Diffserv QoS for Video >> >> Hi Harald, >> >> We should avoid to re-open last call. That's not my intent. >> >> I'm working on transport only and my rtcweb input is >> draft-ietf-tsvwg-rtcweb- qos-15. It states little for audience like >> me, if the subject is audio and video related to a talking person. I >> think the relevant statement are lines in a table. I'd appreciate >> text helping with the interpretation of table. If we can't agree on useful text - leave it as is. >> >> A backbone supporting RTP based TV distribution tends to be >> engineered for support of an AF4 like QoS class for low loss. I >> think, telephony codecs, whose RTP streams will be transported by EF, >> are able to deal with higher packet loss, than multicast based IPTV. >> >> To me, an audio or telepresence/videoconference call with QoS require >> admission control. This ensures that they are transported without >> queuing or drops. >> >> If there's congestion however: >> >> An audio flow marked AF41 should face a lower drop rate than a video >> flow marked AF42 in the case of congestion at an network edge node. >> The packets arrive in the same order as they were sent (independent >> of the flow they belong to). >> >> An EF marked audio flow may experience loss events independent from a >> video flow marked AF4x. In the case of queuing, one of the flows will >> be earlier. If both are transported by QoS classes optimized for >> rtp/udp transport, the difference is a few [ms] per congestion point. >> If one of the queues is optimized for general data transport, the delay difference is likely to be a double digit [ms] number. >> The packets of each flow arrive in order as sent, the packets of one >> flow are delayed against those of the other. >> >> Regards, >> >> Ruediger >> >> -----Ursprüngliche Nachricht----- >> Von: Harald Alvestrand [mailto:harald@alvestrand.no] >> Gesendet: Dienstag, 10. Mai 2016 09:36 >> An: Geib, Rüdiger; paulej@packetizer.com >> Cc: rtcweb@ietf.org; tsvwg@ietf.org >> Betreff: Re: [tsvwg] Diffserv QoS for Video >> >> FTR: I don't see such an agreement at all. >> >> On the contrary, my perception is that people want the ability to >> deliver audio with a lower loss probability and lower delay >> probability than video - it's more important to the conversation, and >> there are fewer things the recipient can do to hide the losses. If >> the sender chose to send them on separate flows, they shold have different DSCP markings. >> >> I believe this is what draft-ietf-tsvwg-rtcweb-qos-15 section 5 >> states, and I believe that this is what TSVWG has declared consensus >> on and wrote in the document that passed WG last call and is currently in "waiting for writeup" state. >> >> Changing this determination would, at minimum, require reopening the >> WG Last Call. >> And I'd object. >> >> Harald >> >> >> Den 10. mai 2016 08:56, skrev Ruediger.Geib@telekom.de: >>> Hi Paul, >>> >>> I think we agree, that audio and video frames, if both are part of >>> the same (interactive) media flow should be transported by the same >>> PHB [PJ] or the same queue [RG]. The latter is ensured, if the same >>> PHB is picked for audio and video. To me the text of the draft so >>> far doesn't express that both audio and video are supposed to use an >>> "Interactive Video..." PHB, if both are present. I'd prefer to have >>> text with a non binding standard requirement saying >>> >>> However, if the application wishes to send both interactive >>> video and audio, it is RECOMMENDED to transport audio >>> and video packets by the same per hop behavior. For example, >>> audio and video packets would both be marked as AF42 or >>> AF43. >>> >>> I don't insist on descriptive text proposing to transport audio by >>> an AF4 PHB >> offering a lower drop ratio than that used to transport video. My >> audio/video experts support this and I'm pretty sure, that also Cisco >> representatives mentioned that audio quality ranks above video quality in telepresence sessions. >>> >>> Regards, >>> >>> Ruediger >>> >>> >>> -----Ursprüngliche Nachricht----- >>> Von: Paul E. Jones [mailto:paulej@packetizer.com] >>> Gesendet: Montag, 9. Mai 2016 21:03 >>> An: Geib, Rüdiger >>> Cc: brian.e.carpenter@gmail.com; david.black@emc.com; >>> tsvwg@ietf.org; rtcweb@ietf.org >>> Betreff: Re: AW: [tsvwg] Diffserv QoS for Video >>> >>> Ruediger, >>> >>> Perhaps an example might be helpful. How about I add this text for >>> illustrative >> purposes? >>> >>> To illustrate the use of the above table, let us assume the >>> application assigns a priority of "medium" to audio and video >>> flows. Given that assumption, if the application wishes to send >>> only audio then packets would be marked EF. However, if the >>> application wishes to send both interactive video and audio, >>> then audio and video packets would both be marked as AF42 or >>> AF43. The intent is to ensure that when both audio and video >>> are being sent together that they receive similar per-hop >>> behavior. >>> >>> This doesn't get into the preference for AF42 vs. AF43. If it were >>> me, I'd mark all >> audio as AF42 and only key video frames as AF42. All predictive >> frames would be sent with an AF43 marking. I might even take it a >> step further and classify all audio as "high". However, I've seen a >> tremendous amount of debate on this before, so I'd prefer to not go >> too far in dictating audio markings vs. video. I do think most >> people generally agree about at least ensuring the class is the same, >> otherwise the wildly different PHB introduces skew between A/V packet >> arrival, thus inflating the size of buffers managing the A/V streams. >> However, we do not want to dictate that audio should be treated >> significantly better than audio. For deaf users, for example, the >> audio really isn't important at all. That is perhaps an extreme example, but it nonetheless highlights why we should be cautious about exactly what we normatively mandate. >>> >>> Paul >>> >>> ------ Original Message ------ >>> From: Ruediger.Geib@telekom.de >>> To: paulej@packetizer.com >>> Cc: brian.e.carpenter@gmail.com; david.black@emc.com; >>> tsvwg@ietf.org; rtcweb@ietf.org >>> Sent: 5/9/2016 3:34:25 AM >>> Subject: AW: [tsvwg] Diffserv QoS for Video >>> >>>> Hi Paul, >>>> >>>> I've talked with audio/video experts of Deutsche Telekom and they >>>> too favored what you recommend below: transport audio and video by >>>> the same queue. Your statement below however stops there and the >>>> draft text doesn't clarify the issue: >>>> >>>> If there's interactive video with audio, then they both should be >>>> marked for the same PHB which is: >>>> - EF ? >>>> - AF4? Like AF41 Audio, AF42 Video (AF43 in addition, if P or B >>>> frames are to receive a lower priority / >>>> higher drop precedence PHB)? >>>> >>>> I personally prefer AF4 if audio and video are to be transported in >>>> the same queue. >>>> >>>> I'd also ask for the draft text to be clear about the issue when to >>>> mark audio by the EF PHB. My understanding after reading your >>>> statement below is: Audio marked EF if there's no video flow only. >>>> >>>> ... >>>> BC>Finally, why is audio not also subdivided into interactive and >>>> BC>non-interactive? As far as I can see, both are logically possible. >>>> >>>> [PJ] For WebRTC, audio alone is "interactive" in nature (which is >>>> why it's marked EF). However, if one is sending audio and video >>>> it makes sense to mark them both same way to get the same PHB and >>>> hopefully have them queued in the same buffers along the path. >>>> Sending audio as EF and possibly having a PHB that results in >>>> packets arriving much faster than corresponding video packets >>>> marked as AF42 is not at all helpful for applications that have >>>> to synchronize the audio and video flows. >>>> ... >>>> >>>> Paul >>>> >>>> >>>> Regards, >>>> >>>> Ruediger >>>> >>>
- [rtcweb] Diffserv QoS for Video Black, David
- Re: [rtcweb] [tsvwg] Diffserv QoS for Video Brian E Carpenter
- Re: [rtcweb] [tsvwg] Diffserv QoS for Video Bernard Aboba
- Re: [rtcweb] [tsvwg] Diffserv QoS for Video Brian E Carpenter
- Re: [rtcweb] [tsvwg] Diffserv QoS for Video Bernard Aboba
- Re: [rtcweb] [tsvwg] Diffserv QoS for Video Brian E Carpenter
- Re: [rtcweb] [tsvwg] Diffserv QoS for Video Stefan Håkansson LK
- Re: [rtcweb] [tsvwg] Diffserv QoS for Video Paul E. Jones
- Re: [rtcweb] [tsvwg] Diffserv QoS for Video Ruediger.Geib
- Re: [rtcweb] [tsvwg] Diffserv QoS for Video Paul E. Jones
- Re: [rtcweb] [tsvwg] Diffserv QoS for Video Ruediger.Geib
- Re: [rtcweb] [tsvwg] Diffserv QoS for Video Harald Alvestrand
- Re: [rtcweb] [tsvwg] Diffserv QoS for Video Ruediger.Geib
- Re: [rtcweb] [tsvwg] Diffserv QoS for Video Romascanu, Dan (Dan)
- Re: [rtcweb] [tsvwg] Diffserv QoS for Video Paul E. Jones
- Re: [rtcweb] [tsvwg] Diffserv QoS for Video Black, David
- Re: [rtcweb] [tsvwg] Diffserv QoS for Video Brian E Carpenter
- Re: [rtcweb] [tsvwg] Diffserv QoS for Video Ruediger.Geib
- Re: [rtcweb] [tsvwg] Diffserv QoS for Video Cullen Jennings (fluffy)