[Asterisk-video] Asterisk H264

[John Martin]

Hi Matteo,
  Some H.264 endpoints do indeed need the fmtp to be specified before
they will make a video call. Some just assume defaults - though it can
mean you end up with a low resolution, low frame rate call. Some
endpoints are also a bit fussy about which scope the fmtp is declared in
within the SDP.
  Adding the fmtp to the SDP is "under construction" in Asterisk. We are
waiting for 1.4 to get underway before we add support for decoding and
encoding fmtp for video and audio. Most of the work is now done, but it
was deemed too different to be put into 1.4. To get things going you
could try and add the fmtp profile, level and maxbr (for instance) to
the SDP in add_codec_to_sdp() with something like...

	if (codec == AST_FORMAT_H264)
	 ast_build_string(a_buf, a_size, "a=fmtp:%d
profile-level-id=42800C; packetization-mode=0; max-br=384;
max-mbps=10000\r\n", rtp_code);

  I think trunk has all those variables available but I've had to change
various things in my build :-( I had this line of code in my builds for
some time before I got around to doing it more fully.

  The upcoming support that's needed within asterisk is to make all
these variables configurable (from the conf files) and negotiated across
the various call legs of Asterisk.

  I have all of this working, I think, so we should be able to get it
into trunk once 1.4 is out of the way.

John

John Martin
John.Martin at AuPix.com
http://www.AuPix.com


> -----Original Message-----
> From: asterisk-video-bounces at lists.digium.com [mailto:asterisk-video-
> bounces at lists.digium.com] On Behalf Of Matteo Brancaleoni
> Sent: 19 July 2006 14:14
> To: Development discussion of video media support in Asterisk
> Subject: RE: [Asterisk-video] Asterisk H264
> 
> I think that I've spotted where the issue is with asterisk
> in middle of 2 h264 streams.
> H264 requires in SDP the format of the codec. When you make
> a call with H264 there's a a:ftmp that specifies at least
> the profile-level-id of the h264 stream.
> Asterisk should copy that information to the other peer,
> in addition to the rtpmap.
> 
> matteo.
> 
> --
> Matteo Brancaleoni
> R&D Director
> Tel  +39.02.70633354
> Sip  matteo at sip.voismart.it
> Iax2 matteo at iax.voismart.it
> 
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