Hi, I found this same issue with all H.264 devices in the 1.4.x. The problem is that the main branch (correct term?) of Asterisk isn¹t supporting FMTP for the RTP payload in H.264. This is used to declare the H.264 profiles/levels and other options in the call SDP, and without it, the SIP device hears strange voices about H.264, is confused, and does nothing ;-)
Follow this old thread (http://www.mail-archive.com/[email protected]/msg00101.html), it¹s worked well for me using Polycom PVX and other SIP H.264 devices. Regards, Joe Baltes RADVISION, Inc. On 2/10/10 6:05 PM, "Borja SIXTO" <[email protected]> wrote: > Have you try the echo test with each phone ? > > > Le 10/02/2010 18:47, Loïc a écrit : >> > Hi, >> > >> > I'm trying to use Asterisk 1.4.29 to connect two Polycom HDX in High >> > Definition h264 format with SIP. >> > When I use h263 , no problem. However, when I try to use h264 I can not >> > have the video streams. The communication is well established : I have a >> > G711 audio stream but no video. >> > >> > If I use a SIP h264 videophone with a Polycom HDX, the Polycom well >> > receives the video stream but no video on the SIP videophone. >> > >> > Any ideas what is the problem ? >> > >> > sip.conf >> > [general] >> > context=default >> > bindport=5060 >> > bindaddr=0.0.0.0 >> > srvlookup=yes >> > notifymimetype=text/plain >> > videosupport=yes >> > maxcallbitrate=1920 >> > nat=no >> > canreinvite=no >> > >> > [100] >> > type=friend >> > username=100 >> > secret=XXX >> > host=dynamic >> > context=default >> > callerid=XXX<100> >> > disallow=all >> > allow=ulaw >> > allow=alaw >> > allow=h264 >> > qualify=yes >> > >> > >> > [200] >> > type=friend >> > username=200 >> > secret=XXX >> > host=dynamic >> > context=default >> > callerid=XXX<200> >> > disallow=all >> > allow=ulaw >> > allow=alaw >> > allow=h264 >> > qualify=yes >> > >> > >> > Thanks in advance... >> > >> > >> > Loïc >> > >> > > > -- > Borja Sixto, Research& Innovation - http://www.i6net.com > Office: +34 911877477 | Gtalk: [email protected] | Skype: borja.sixto > > > -- > _____________________________________________________________________ > -- 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 >
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