Re: [rtcweb] Why is required to have local streams before running ICE gathering? (another SDP limitation?)

Justin Uberti <juberti@google.com> Sat, 15 June 2013 23:40 UTC

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From: Justin Uberti <juberti@google.com>
Date: Sat, 15 Jun 2013 16:40:28 -0700
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Subject: Re: [rtcweb] Why is required to have local streams before running ICE gathering? (another SDP limitation?)
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On Thu, Jun 13, 2013 at 7:55 AM, Iñaki Baz Castillo <ibc@aliax.net> wrote:

> Hi, let me expose the following simple case:
>
>
> 1) SIP application in the web.
>
> 2) Alice receives INVITE from Bob with audio and video.
>
> 3) Alice JS UA is ringing, Alice does not answer or reject the call yet.
>
> 4) Alice JS UA does not know whether Alice would answer with audio and/or
> video.
>
> 5) Alice JS UA starts ICE gathering and obtains all its ICE candidates.
>
> 6) Alice presses "Answer with audio and video".
>
> 7) Alice JS UA calls getUserMedia, creates the PC, gets the SDP and
> passes it to Bob JS UA in a SIP 200 response.
>
>
> Well, this is not possible in WebRTC due to step 5. Unfortunately
> before ICE gathering, the local PC must be set with the local SDP, so
> getUserMedia is required to happen before ICE gathering.
>

This should be possible today. You can set the local description with no
MediaStreams but the constraint OfferToReceiveVideo:true to start candidate
gathering for both audio and video.

When you do eventually get the stream, you can call setLocalDescription
again, to indicate whether video is to be used or not.

>
> But we cannot prompt Alice with the getUserMedia pop-up while she has
> not yet decided to answer the incoming call.
>
> The fact is that both the audio and video will use the same UDP
> channel/transport, so having ICE candidates for the audio and the
> video m lines is just unneeded.
>
> So IMHO this is another issue due to the usage of SDP, right? or may
> be I am wrong? If so please let me know a use case in which the
> browser would mantain two different UDP channels/transports for the
> same "call" (a single SDP O/A).
>

This happens in any non-BUNDLE scenario.

>
> This is just for "legacy interoperability", right? this is, for the
> case in which the remote peer is a not-full-WebRTC device and uses
> different UDP channels/transports for each m line in the SDP. If so,
> would not be MUCH easier just to *require* as per WebRTC specs that
> two peers MUST use a single UDP channel/transport for each SDP O/A?


>
>
> --
> Iñaki Baz Castillo
> <ibc@aliax.net>
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> rtcweb@ietf.org
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>