Hi, <br><br>I've a problem configuring my Asterisk. What I try to reach
is to interconnect a Tandberg Visioconference (SIP) world with my
Asterisk (SIP) with 1 constraint I can't change : "every RTP flow needs
to pass THROUGH Asterisk, and are NOT nated"<br>
<br>What I observe :<br> - a call made from a SIP Phone registred in Asterisk to Tandberg works (voice and video bidirectionnal)<br>
- a call made from to Tandberg to the SIP phone doesn't work (voice
bidirectionnal, voice only received by the SIP phone, no incomming
video for Tandberg)<br>
<br>I think the problem may come from codec negotiotation :<br> -
when call is made from the SIP phone, it uses "code" 99 for H.264
codec, as Asterisk. Tdb reply SIP:Ok with the same number for H.264<br> - when call is made from Tbd, it uses "code" 98 for H.264 codec. Asterisk then send the Invite with 99 as codec number <br>
<br><br>I use the version 1.6.2.6 of Asterisk<br><br>Is this kind of
configuration supposed to work ? I know passing video media through
Asterisk may not be optimal, but I really need it, even if I have to
patch Asterisk<br>
<br>Thanks for your help<br><br><br><br>SDP send by Tandberg :<br>--------------------------<br>v=0<br>o=tandberg 1 5 IN IP4 192.168.50.10<br>s=-<br>c=IN IP4 192.168.50.10<br>b=CT:1920<br>t=0 0<br>m=audio 48260 RTP/AVP 100 101 9 8 0 102<br>
b=TIAS:64000<br>a=rtpmap:100 G7221/16000<br>a=fmtp:100 bitrate=32000<br>a=rtpmap:101 G7221/16000<br>a=fmtp:101 bitrate=24000<br>a=rtpmap:9 G722/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:102 telephone-event/8000<br>
a=fmtp:102 0-15<br>a=sendrecv<br>m=video 48262 RTP/AVP 97 98 99 34 31<br>c=IN IP4 192.168.50.10<br>b=TIAS:1920000<br>a=rtpmap:97 H264-RCDO/90000<br>a=fmtp:97 profile-level-id=008016;max-<div>mbps=42000;max-fs=3600;max-smbps=323500<br>
a=rtpmap:98 H264/90000<br>a=fmtp:98 profile-level-id=428016;max-mbps=35000;max-fs=3600;max-smbps=323500<br>a=rtpmap:99 H263-1998/90000<br>a=fmtp:99 custom=1280,800,0;custom=1280,768,0;custom=1280,720,3;custom=1024,768,4;custom=1024,576,2;custom=800,600,3;cif4=2;custo<br>
a=rtpmap:34 H263/90000<br>a=fmtp:34 cif4=2;cif=1;qcif=1;sqcif=1;maxbr=19200<br>a=rtpmap:31 H261/90000<br>a=fmtp:31 cif=1;qcif=1;maxbr=19200<br>a=rtcp-fb:* nack pli<br>a=sendrecv<br>a=content:main<br>a=label:11<br>a=answer:full<br>
m=application 5078 UDP/BFCP *<br>c=IN IP4 192.168.50.10<br>a=floorctrl:c-s<br>a=confid:1<br>a=floorid:2 mstrm:12<br>a=userid:1<br>a=setup:passive<br>a=connection:new<br>m=video 48264 RTP/AVP 99 34 31<br>c=IN IP4 192.168.50.10<br>
b=TIAS:1920000<br>a=rtpmap:99 H263-1998/90000<br>a=fmtp:99 custom=1280,800,0;custom=1280,768,0;custom=1280,720,3;custom=1024,768,4;custom=1024,576,2;custom=800,600,3;cif4=2;custo<br>a=rtpmap:34 H263/90000<br>a=fmtp:34 cif4=2;cif=1;qcif=1;sqcif=1;maxbr=19200<br>
a=rtpmap:31 H261/90000<br>a=fmtp:31 cif=1;qcif=1;maxbr=19200<br>a=rtcp-fb:* nack pli<br>a=sendrecv<br>a=content:slides<br>a=label:12<br><br><br><br>SDP send by Asterisk<br>v=0<br>o=root 1077353049 1077353049 IN IP4 192.168.13.100<br>
s=Asterisk PBX 1.6.2.6<br>c=IN IP4 192.168.13.100<br>b=CT:384<br>t=0 0<br>m=audio 14604 RTP/AVP 0 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>
a=sendrecv<br>m=video 17962 RTP/AVP 99<br>a=rtpmap:99 H264/90000<br>a=sendrecv<br><br><br><br><br><br>Here is my sip.conf<br>--------------------<br>[general]<br>port = 5060<br>bindaddr = 0.0.0.0<br>disallow = all<br>realm = testRealm<br>
<br>allow = ulaw<br>allow = h264<br>videosupport=yes<br><br>canreinvite = no<br>calleridupdate = info<br>usercallerid = no<br>context = default<br><br>[toTandberg]<br>host=192.168.50.53<br>type=friend<br>qualify=yes<br>qualifyreq=1<br>
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