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 <cmrah...@yahoo.com> To: Development discussion of video media support in Asterisk <asterisk-video@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 <li...@nowin.de> To: Development discussion of video media support in Asterisk <asterisk-video@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 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-video mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-video
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-video mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-video