VS: [rtcweb] SDP is not suitable for WebRTC

Hi,

I was not wrong, but I was maybe unclear. I meant that each entity would offer it’s m- lines as part of separate Offer/Answer transactions. It IS allowed to add m- lines in a subsequent Offer.

Regards,

Christer

Lähettäjä: Iñaki Baz Castillo [mailto:ibc@aliax.net]
Lähetetty: 29. heinäkuuta 2013 20:50
Vastaanottaja: Bossiel thioriguel
Kopio: public-webrtc@w3.org; rtcweb@ietf.org
Aihe: Re: [rtcweb] SDP is not suitable for WebRTC


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<mailto:ibc@aliax.net>>
El 29/07/2013 19:45, "Bossiel thioriguel" <bossiel@yahoo.fr<mailto: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<mailto:ibc@aliax.net>>
À : "rtcweb@ietf.org<mailto:rtcweb@ietf.org>" <rtcweb@ietf.org<mailto:rtcweb@ietf.org>>; "public-webrtc@w3.org<mailto:public-webrtc@w3.org>" <public-webrtc@w3.org<mailto: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<mailto:ibc@aliax.net>>
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb