Bill H Posted March 12, 2010 Report Share Posted March 12, 2010 I am testing the new Polycom VVX1500d Video Phone. When one VVX1500d calls another VVX1500d and the call is answered there is an immediate video display on each phone showing the caller and called party. This works well when I use the URL Dial feature of the phone. (both phones are on the same physical network and subnet) However, when I place the call from one extension to another through PBXNSIP, the call goes through and a conversation takes place but no video. I have isolated the trouble to some degree. In the initial Invite I see this: [8] 2010/03/12 09:32:40: SIP Rx udp:192.168.11.11:5060: <<<<---- This is the Calling Station INVITE sip:205@192.168.11.123:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.11.11:5060;branch=z9hG4bKb31097eb3B3BA3E6 From: "670 Polycom" <sip:204@192.168.11.123>;tag=1AB7F2BE-264A15FD To: <sip:205@192.168.11.123;user=phone> CSeq: 2 INVITE Call-ID: 2c58c7b2-82fd8e1-c11dd524@192.168.11.11 Contact: <sip:204@192.168.11.11:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomVVX-VVX_1500-UA/3.2.2.0481 Accept-Language: en Supported: 100rel,replaces Allow-Events: talk,hold,conference Authorization: Digest username="204", realm="192.168.11.123", nonce="3cd9ef58f0334be8ef5520db51e29838", uri="sip:205@192.168.11.123:5060;user=phone", response="eeeacfa0845ab0085cdac9d4f79fc1cc", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 688 v=0 o=- 1268404353 1268404353 IN IP4 192.168.11.11 s=Polycom IP Phone c=IN IP4 192.168.11.11 b=AS:448 t=0 0 a=sendrecv m=audio 2242 RTP/AVP 115 102 9 0 8 18 101 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 m=video 2244 RTP/AVP 109 96 34 31 <<<< ---- This is here a=rtpmap:109 H264/90000 a=fmtp:109 profile-level-id=42800d a=rtpmap:96 H263-1998/90000 a=fmtp:96 CIF=1;QCIF=1;SQCIF=1;K=1;N=1 a=rtpmap:34 H263/90000 a=fmtp:34 CIF=1;QCIF=1;SQCIF=1 a=rtpmap:31 H261/90000 And this is what is being sent by PBXNSIP to the Called Station [8] 2010/03/12 09:32:40: SIP Tx udp:192.168.11.10:5060: <<<< ---- This is the Called Station INVITE sip:205@192.168.11.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.123:5060;branch=z9hG4bK-f2e5320e5c94448f484558ea7ca404aa;rport From: <sip:204@localhost>;tag=58537 To: <sip:205@localhost> Call-ID: 77b8805c@pbx CSeq: 1732 INVITE Max-Forwards: 70 Contact: <sip:205@192.168.11.123:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.4.0.3201 Alert-Info: Internal Content-Type: application/sdp Content-Length: 294 v=0 o=- 17706 17706 IN IP4 192.168.11.123 s=- c=IN IP4 192.168.11.123 t=0 0 m=audio 51042 RTP/AVP 0 8 9 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv m=video 2244 RTP/AVP 109 96 34 31 <<<< ---- It isn't there Can anyone shed a little light on this?? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted March 14, 2010 Report Share Posted March 14, 2010 You first have to establish a voice session, then you can turn video on. That's the way it is in 4.0. Quote Link to comment Share on other sites More sharing options...
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