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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