[Asterisk-video] Lifesize HD X10 endpoints & Video Resolution and framerate configuration question

[CM Rahman]

Gunnar,

I have tried to compile with the code you provided in 1.6.x

Here is what I get..

 [CC] chan_sip.c -> chan_sip.o
chan_sip.c: In function âadd_vcodec_to_sdpâ:
chan_sip.c:10077: warning: missing terminating " character
chan_sip.c:10077: error: missing terminating " character
chan_sip.c:10078: error: âprofileâ undeclared (first use in this function)
chan_sip.c:10078: error: (Each undeclared identifier is reported only once
chan_sip.c:10078: error: for each function it appears in.)
chan_sip.c:10078: error: âlevelâ undeclared (first use in this function)
chan_sip.c:10078: error: âidâ undeclared (first use in this function)
chan_sip.c:10078: error: invalid suffix "f" on integer constant
chan_sip.c:10078: error: expected â)â before â;â token
chan_sip.c:10079: error: stray â\â in program
chan_sip.c:10079: error: stray â\â in program
chan_sip.c:10079: warning: missing terminating " character
chan_sip.c:10079: error: missing terminating " character
chan_sip.c:10080: error: expected â;â before â}â token
make[1]: *** [chan_sip.o] Error 1
make: *** [channels] Error 2





________________________________
From: CM Rahman <cmrahman at yahoo.com>
To: Development discussion of video media support in Asterisk 
<asterisk-video at lists.digium.com>
Sent: Wed, April 13, 2011 10:21:20 PM
Subject: Re: [Asterisk-video] Lifesize HD X10 endpoints & Video Resolution and 
framerate configuration question


Hi Gunnar,

I have access to Lifesize equipments. I can help you with this.

Thanks
CM




________________________________
From: Gunnar Schaller <linux at nowin.de>
To: Development discussion of video media support in Asterisk 
<asterisk-video at lists.digium.com>
Sent: Wed, April 13, 2011 5:20:38 PM
Subject: Re: [Asterisk-video] Lifesize HD X10 endpoints & Video Resolution and 
framerate configuration question

Hello,
Try to modify the Asterisk source. In chan_sip.c search for the
function add_vcodec_to_sdp (around line 10316 in 1.8 branch at the
moment). At the end of the function there is a comment:
  /* Add fmtp code here */

Add this lines after the comment:
  if (codec & 0x200000) {
      ast_str_append(a_buf,0,"a=fmtp:%d
      profile-level-id=42801f;max-mbps=245000;max-fs=8192;
      packetization-mode=0\r\n",rtp_code);
  }

I did not test that. I not even tried to compile with that lines ...
So no guarantee for anything. I just had a look to the source in
chan_sip.c and your SIP trace and tried to build together some lines
of code. It is not a solution for all video problems. It's only a hack
for your situation with the Lifesize system.
I do not have access to such a Lifesize conferencing system at the
moment. But I know someone and maybe I will do a test in a few weeks.

Regards,
Gunnar


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