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Re: [tsvwg] Last Call: <draft-ietf-tsvwg-rtcweb-qos-15.txt> (DSCP and other packet markings for WebRTC QoS) to Proposed Standard

2016-03-23 05:08:40
Hi,

Below is the first round on a question, that looks like it needs to be addressed, thus I bring it into a public discussion.

Den 2016-03-22 kl. 19:48, skrev Paul E. Jones:
>> The other comment I have is the following:
>>
o Flow Type: The browser provides this input as it knows if the
  flow is audio, interactive video with or without audio,
  non-interactive video with or without audio, or data.

Yes, a browser knows if a MediaStreamTrack's source is a live
source, i.e. camera or microphone or a file. Which would indicate
the difference between interactive or non. However, I don't
understand what the Flow Type description for video contains "with
or without audio" as the flow definitions in RTCWEB transport
document all indicate flows as containing a single media type. Can
you please clarify?
>
This relates to the table that follows. The intent is that if a
WebRTC application is sending audio and video (e.g., a
videoconference call), then the same DSCP values might be used for
both the audio and video flows. On the other hand, if only a video
flow is sent alone (perhaps the audio source is an entirely different
device), then the browser can still use the same packet marking.


So, I started commenting on this because, the "Flow Types" in the table are not really defined. And my interpretation was not that the audio could be given the same markings as the Video. I only interpreted it as being valid for the video flow. Thus, I think the actual "flow type" values in Table 1 needs to be better defined. To my knowledge these types are not defined in any other RTCWeb document.

I think what is needed is a definition list for what is meant. I can see that pointers for example into RFC 4594 may help making these definitions more compact.

One thing that might be a bit tricky is actually the difference between interactive and non-interactive (streaming) usages of WebRTC RTP streams. It becomes a question if the WebRTC endpoint actually accurately can classify these differences. Yes, a live media source, like an camera or microphone can on first order be used for classification as interactive, while a file source is non-interactive. But even the first, can in the application context be non-interactive. An example would be an lecturer application. Relaxing the delay from the lecturer to the audience is likely fine, especially if one have a "raise hand" mechanism and only explicitly invites participants to ask questions. To my knowledge there are no API surface to indicate these preferences on the MediaStream or MediaStreamTrack level.

I think this document have a potential valuable difference for the interactive vs non-interactive, but the rest of the current solution has difficulties to utilize this difference. From my perspective I am fine with retaining the difference, but the definition must be clear so that implementers select the right one. And likely the non-interactive will not be much utilized until additional API knobs are created.

Cheers

Magnus Westerlund

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