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Re: A proposal for how we would use the SDP that comes out of the MMUSIC interm

From: Peter Thatcher <pthatcher@google.com>
Date: Fri, 9 Oct 2015 08:30:34 -0700
Message-ID: <CAJrXDUGkCg=mJXDYhZ25AkrJkMiLxaf4=ZrHKyVAPoC7obychw@mail.gmail.com>
To: Harald Alvestrand <harald@alvestrand.no>
Cc: "public-webrtc@w3.org" <public-webrtc@w3.org>
On Fri, Oct 9, 2015 at 5:25 AM, Harald Alvestrand <harald@alvestrand.no>
wrote:

> Some comments on proposal; I'm not sure I would advocate adding it yet.
>
>
> Den 09. okt. 2015 04:06, skrev Peter Thatcher:
> > Heading into the MMUSIC interim, I think it would be wise to have some
> > idea of what SDP we would want to use (what subset of what MMUSIC
> > decides on would WebRTC support?).  Having spoken to several people in
> > the MMUSIC WG working on the draft for simulcast and several people in
> > the WebRTC WG, I've come up with the following proposal for what subset
> > we could use, and what the API around it would look like.  I hope this
> > proposal and ensuing discussion help us prepare for the interim.
> >
> >
> > At the f2f, we had several ideas with different qualities which I
> > describe her in JSON form.
>
> I'm hesitant to call these "plans", they're not fleshed out enough, but
> I think I can map your points to some of the things I remember...
>

You are right.  At the f2f, they were called "Option A", "Option B", etc,
not "Plan A", etc. ​


>
> >
> > Plan A: {api: little, sdp:big}
>
> This would be the "just set the number of layers and let SDP do the
> rest" plan?
>

​Yes, that was Option A.​


>
> > Plan B: {api: big,    sdp: none}
>
> This would be the "drop multicast from WebRTC-1.0 API, leave it to
> lower-layer APIs in NV" plan?
>
>
​No, Option B was "call RtpSender.setParameters with multiple encodings
regardless of the SDP".


> > Plan C: {api: none,   sdp:none}
>


> This would be the "just say no" plan?


​Yes, Option C (or maybe it was D) was "not in 1.0".​



> >
> > This proposal (let it be Plan X) tries to combine the best of both and
> > have the qualities {api: moderate, sdp: moderate}.
> >
> >
> > Here is a graphical representation of the qualities:
> >
> > +
> > API Big | B
> > |
> > |
> > |
> > |
> > |
> > |
> > |
> > |
> > | X
> > |
> > |
> > |
> > |
> > |
> > |
> > API Small | A
> > | C
> > +------------------------------------+
> >
> > SDP Small SDP Big
> >
> >
> >
> >
> > Here's how it would work with an example:
> >
> > var video = pc.addTransceiver(track,
> >   {send: {encodings: [{scale: 1.0}, {scale: 2.0}, {scale: 4.0}]}});
> >
> > // This now has a *subset* of the SDP from MMUSIC
> > pc.createOffer().then(signalOffer)
> > var answer = ... ; //  wait for the answer
> >
> > // This accepts a *subset* of the SDP from MMUSIC
> > pc.setRemoteDescription(answer);
> >
> > // The app can later decide to change parameters, such as
> > // stop sending the top layer
> > var params = video.sender.getParameters();
> > params.encodings[0].active = false;
> > video.sender.setParameters(params);
> >
> >
> > The key parts are:
> > 1.  It builds on top of addTransceiver, expanding {send: true} to {send:
> > ...} where the ... can express the desire for simulcast.
>
> Actually this is the API part I like the least. While we did this for
> constraints, the result is not so pretty. I'd prefer to just add a
> dictionary member called "sendLayers".
> ​ ​
> If "send" is modifiable in the

transceiver (as I think it should be), this also allows us to prepare a
> transceiver for simulcast without turning on the spigot at first
> negotiation.
>

​That's a good idea.  Let's change it to:
​
dictionary RTCRtpTransceiverInit {
  // .... the rest as-is
  sequence<RTCRtpEncodingParameters> sendEncodings;
}​


> Syntax seems fine, except that I don't know what it means; it is
> strictly spatial resolution differences, or is it a bandwidth target
> that can be hit by any combination of spatial and temporal shenanigans?
>
>
​At the f2f we were calling this "resolutionScale".  Let's change it to
that to be more clear:

dictionary RTCRtpEncodingParameters {
  // ... the rest as-is
  double resolutionScale;
  unsigned long rsid;  // RTP Source Stream ID
}


Now are example would be:

pc.addTransceiver("video", {send: true, sendEncodings:
   [{resolutionScale: 1.0}, {resolutionScale: 2.0}, {resolutionScale:
4.0}]});



>
> > 2.  It uses a subset of the SDP from MMUSIC in the offer and answer.
> > 3.  It gives JS some control over each layer: .active, .maxBandwidth,
> > .scale.
> > 4.  The additions to the API and to the SDP are simple.
> >
> >
> >
> > Here's the WebIDL:
> >
> > dictionary RTCRtpTransceiverInit {
> >   (boolean or RTCRtpParameters) send;
> >   // .. the rest as-is
> > }
> >
> > dictionary RTCRtpEncodingParameters {
> >   double scale;  // Resolution scale
> >   unsigned long rsid;  // RTP Source Stream ID
> >   // ... the rest as-is
> > }
> >
> > And here's the *subset* of the SDP from MMUSIC we could use in the offer
> > (obviously subject to change based on the results of the interim):
> >
> > m=video ...
> > ...
> > a=rsid send 1
> > a=rsid send 2
> > a=rsid send 3
> > a=simulcast rsids=1,2,3
> >
> >
> > And here's the *subset* of the SDP from MMUSIC we could use in the
> answer:
> > m=video ...
> > ...
> > a=rsid recv 1
> > a=rsid recv 2
> > a=rsid recv 3
> > a=simulcast rsids=1,2,3
> >
> >
> > That's it.  That's all I think we need: a simple addition to
> > addTransceiver plus a simple subset of the SDP from MMUSIC.
> >
> >
> > The last thing I would note is that I propose we *do not* use the
> > entirety of the MMUSIC draft in WebRTC.  In particular, not the PT
> > overloading or the more extensive attributes that don't map well to
> > RTCRtpEncodingParameters (max-width, max-height, max-fps, max-fs,
> max-pps).
> >
> >
> >
> >
> >
>
Received on Friday, 9 October 2015 15:31:42 UTC

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