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Re: [rtcweb] SDP is not suitable for WebRTC

From: Iñaki Baz Castillo <ibc@aliax.net>
Date: Mon, 29 Jul 2013 19:49:58 +0200
Message-ID: <CALiegfms3r7RSZ=nCH2fSFLnRxc8UjkL4F2d3evFs5XJOWmAbw@mail.gmail.com>
To: Bossiel thioriguel <bossiel@yahoo.fr>
Cc: public-webrtc@w3.org, rtcweb@ietf.org
Thanks a lot for clarifying it. So Christer was wrong ;)

And it makes my scenario even worse. I really hope something will happen
and WebRTC will get rid of SDP...

--
Iñaki Baz Castillo
<ibc@aliax.net>
El 29/07/2013 19:45, "Bossiel thioriguel" <bossiel@yahoo.fr> escribió:

> You said: "2)
>
> SDP seems to allow that the offer and the answer have different number
> of m lines "
>
> No at all:
> RFC 3264:
> For each "m=" line in the offer, there MUST be a corresponding "m="
>
>    line in the answer.  The answer MUST contain exactly the same number
>    of "m=" lines as the offer.  This allows for streams to be matched up
>    based on their order.  This implies that if the offer contained zero
>    "m=" lines, the answer MUST contain zero "m=" lines.
>
> Mamadou.
>
>   ------------------------------
>  *De :* Iñaki Baz Castillo <ibc@aliax.net>
> *À :* "rtcweb@ietf.org" <rtcweb@ietf.org>; "public-webrtc@w3.org" <
> public-webrtc@w3.org>
> *Envoyé le :* Lundi 29 juillet 2013 19h31
> *Objet :* [rtcweb] SDP is not suitable for WebRTC
>
> Hi, I initiated a thread [*] about Plan-Unified and multiple m lines,
> but it was moved to MMUSIC maillist (don't know why since it is about
> WebRTC applications design):
>
> http://www.ietf.org/mail-archive/web/mmusic/current/msg11966.html
>
> Sorry for the cross-posting but at this point I'm a bit lost and do
> not know which is the appropriate group for my concern.
>
>
>
> So my concern is:
>
>
> - Web application with a SIP over WebSocket client running in the web.
>
> - The web user is provided with a conference SIP URI in which there
> are *already* 8 participants (5 of them emitting audio and video and 3
> just emitting audio).
>
> - The user calls, from his webphone, to the given URI to join the
> conference.
>
>
>
> Let's imagine that the JS app knows the number of participant in the
> conference.
> Let's imagine my browser have mic and webcam.
>
>
>
> QUESTION:
>
> How can my browser join the conference without requiring SDP
> renegotiation from the server and, at the same time, being able to
> send audio/video and receive audio/video from others (different tracks
> / m=lines)?
>
>
>
>
> "SOLUTIONS":
>
>
>
> 1)
>
> I tell my browser to generate a SDP offer with:
>
>   - 1 send/receive m=audio line.
>   - 7 recvonly m=audio line.
>   - 1 send/only m=video line.
>   - 4 recvonly m=video line.
>
> (Obviously this is a joke)
>
>
>
> 2)
>
> SDP seems to allow that the offer and the answer have different number
> of m lines (I'm not aware of that but I believe that SDP can do
> "everything"). So my browser generates a SDP offer with 1 m=audio line
> and 1 m=video line, and the server replies with 8 m=audio lines and 4
> m=video lines.
>
> Will my browser understand such a SDP answer with more m lines than
> its generated offer? I assume NOT.
>
>
>
> 3)
>
> My browser generates a SDP offer with 1 m=audio line and 1 m=video
> line and the server too. And later the server sends re-INVITE with all
> the m lines.
>
> Oppss, SDP renegotiation...
>
>
>
>
> SDP is bad for WebRTC. SDP is good for legacy symmetric communications
> in which there is a single-track audio communication and, of course,
> both endpoints emit audio. But SDP is bad for modern RTC protocols in
> which an endpoint can emit tons of tracks to a single endpoint.
>
>
> Do we really want this for WebRTC 1.0 ?
>
>
> --
> Iñaki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
Received on Monday, 29 July 2013 17:50:25 UTC

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