Jens.Kammann@dlr.de
2005-Dec-14 07:50 UTC
[Asterisk-Users] Video calls (MS Messenger, Tandberg)
Hi, According to http://www.voip-info.org/wiki-Asterisk+video it should be possible to place video calls using asterisk. So far I managed to get both Microsoft Messenger and a video conference system from Tandberg to register with asterisk. Voice calls between both stations work perfectly (using ulaw codec). Video calls fail with asterisk putting the tandberg system "on hold" (playing Music-on-hold). Despite both clients claim "H261/H263" codecs, SDP negotiation results:> Capabilities: us - 0xc020e(GSM|ULAW|ALAW|SPEEX|H261|H263), peer -audio=0xe(GSM|ULAW|ALAW)/video 0x0(EMPTY), combined - 0xe(GSM|ULAW|ALAW)>Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined -0x1(G723) So no common video codec was negotiated (thus connections is voice-only) Any ideas? Do I need to active the H261/H263 codecs somewhere? I tried forcing theses codecs in sip.conf, but no luck either. regards, Jens SIP/SDP Debug for Sip read: INVITE sip:52800 SIP/2.0 Via: SIP/2.0/UDP 129.247.XXX.XXX:5060;branch=z9hG4bK2375534000-17264122 Max-Forwards: 70 From: 59999<sip:59999@129.247.XXX.XXX>;epid=82052805FAC6AK;tag=plcm_2375520000 -17264121 To: <sip:52800> Call-ID: 2375519000-17264119 CSeq: 2 INVITE Session-Expires: 90 Supported: timer Contact: <sip:129.247.XXX.XXX:5060;transport=udp> Content-Type: application/sdp Proxy-Authorization: Digest username="59999",realm="asterisk",nonce="6f505027",uri="sip:52800",respo nse="d260786708039bed1a05af94a4a69fb3",algorithm=md5 User-Agent: Polycom VSX 7000A Release 8.0.3 - 06Oct2005 13:49 Content-Length: 990 v=0 o=DLR-KN 1353514857 0 IN IP4 129.247.173.207 s=- c=IN IP4 129.247.XXX.XXX b=AS:384 t=0 0 m=audio 49184 RTP/AVP 99 98 97 102 101 103 9 15 0 8 18 a=rtpmap:99 SIREN14/16000 a=fmtp:99 bitrate=48000 a=rtpmap:98 SIREN14/16000 a=fmtp:98 bitrate=32000 a=rtpmap:97 SIREN14/16000 a=fmtp:97 bitrate=24000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:101 G7221/16000 a=fmtp:101 bitrate=24000 a=rtpmap:103 G7221/16000 a=fmtp:103 bitrate=16000 a=rtpmap:9 G722/8000 a=rtpmap:15 G728/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729A/8000 a=fmtp:18 annexb=no a=sendrecv m=video 49186 RTP/AVP 109 34 96 31 b=TIAS:384000 a=rtpmap:109 H264/90000 a=fmtp:109 profile-level-id=42800c; max-mbps=10000; max-fs=1792; max-br=775 a=rtpmap:34 H263/90000 a=rtpmap:96 H263-1998/90000 a=fmtp:96 CIF4=2;CIF=1;QCIF=1;SQCIF=1;F;J;T a=rtp 14 headers, 34 lines Using latest request as basis request Sending to 129.247.XXX.XXX : 5060 (NAT) Found RTP audio format 99 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 102 Found RTP audio format 101 Found RTP audio format 103 Found RTP audio format 9 Found RTP audio format 15 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Peer audio RTP is at port 129.247.XXX.XXX:49184 Found description format SIREN14 Found description format SIREN14 Found description format SIREN14 Found description format G7221 Found description format G7221 Found description format G7221 Found description format G722 Found description format G728 Found description format PCMU Found description format PCMA Found description format G729A Found description format H264 Found description format H263 Found description format H263-1998 Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x50c(ULAW|ALAW|G729A|ILBC)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found user '59999' Looking for 52800 in sip_dlrpbx list_route: hop: <sip:129.247.XXX.XXX:5060;transport=udp> Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 129.247.XXX.XXX:5060;branch=z9hG4bK2375534000-17264122;received=129.247. XXX.XXX;rport=5060 From: 59999<sip:59999@129.247.XXX.XXX>;epid=82052805FAC6AK;tag=plcm_2375520000 -17264121 To: <sip:52800>;tag=as7a4b0ce2 Call-ID: 2375519000-17264119 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:52800@129.247.XXX.XXX> Content-Length: 0