Hi, thanks for Your quick response. But as You can see in the commented SIP-Messages, asterisk gets a voice call and sends out a INVITE with two media attributes for video and voice towards the destination.
Karsten Am Mittwoch, den 19.10.2011, 10:40 -0500 schrieb Danny Nicholas: > Just a WAG - if you start the call in voice-mode, the video codecs aren't > loaded. > > -----Original Message----- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karsten > Wemheuer > Sent: Wednesday, October 19, 2011 10:37 AM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Problem with video phone call, error in sdp media > handling? > > Hi, > > I try to setup a video call and I sometimes get no video. > > I set up a Yealink VP 2009 and a Ninja Softphone. Both a in the same LAN. > Asterisk release is 1.8.7.0. > > Call from Yealink to the Ninja is working fine, if I start the call in video > mode. In this case I can switch between voice-only and video and back > without any problem. > > If I try the opposite direction there is no video. The Ninja starts the call > in voice-mode and try to add video in an second invite. The same happens, if > I start the call in voice-mode from the Yealink phone. > > As far as I can see there seems to be something broken in SDP handling. > > In the following test phone1 is calling extension 200, which is extension of > phone2. > > In case of failure phone1 sends: > INVITE sip:200@192.168.10.75 SIP/2.0. > Via: SIP/2.0/UDP 192.168.10.106:5062;branch=z9hG4bK1784123944. > From: "Karsten" <sip:phone1@192.168.10.75>;tag=1171101891. > To: <sip:200@192.168.10.75>. > Call-ID: 1555625029@192.168.10.106. > CSeq: 1 INVITE. > Contact: <sip:phone1@192.168.10.106:5062>. > Content-Type: application/sdp. > Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, > REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE. > Max-Forwards: 70. > User-Agent: VideoPhone-V8438 22.30.0.60 00:15:65:1b:20:3f. > Supported: replaces,100rel. > Allow-Events: talk,hold,conference,refer. > Content-Length: 274. > . > v=0. > o=- 20006 20006 IN IP4 192.168.10.106. > s=SDP data. > c=IN IP4 192.168.10.106. > t=0 0. > m=audio 10020 RTP/AVP 0 8 18 101. > a=rtpmap:0 PCMU/8000. > a=rtpmap:8 PCMA/8000. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=fmtp:101 0-15. > a=rtpmap:101 telephone-event/8000. > a=sendrecv. > > Asterisk sends to the second phone: > INVITE sip:phone2@192.168.10.141:1116 SIP/2.0. > Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK3aa00bba. > Max-Forwards: 70. > From: "User1" <sip:phone1@192.168.10.75>;tag=as6e33f30b. > To: <sip:phone2@192.168.10.141:1116>. > Contact: <sip:phone1@192.168.10.75:5060>. > Call-ID: 73a216f9167396885e099d0f2e5d4ca2@192.168.10.75. > CSeq: 102 INVITE. > User-Agent: IPTAM PBX. > Date: Wed, 19 Oct 2011 14:49:17 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, > NOTIFY, INFO, PUBLISH. > Supported: replaces, timer. > P-Asserted-Identity: "User1" <sip:phone1@192.168.10.75>. > Content-Type: application/sdp. > Content-Length: 454. > . > v=0. > o=root 1873948927 1873948927 IN IP4 192.168.10.75. > s=Asterisk PBX 1.8.7.0-1. > c=IN IP4 192.168.10.75. > b=CT:384. > t=0 0. > m=audio 18858 RTP/AVP 8 0 101. > a=rtpmap:8 PCMA/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:20. > a=sendrecv. > m=video 16964 RTP/AVP 31 34 98 99 104. > a=rtpmap:31 H261/90000. > a=rtpmap:34 H263/90000. > a=rtpmap:98 h263-1998/90000. > a=rtpmap:99 H264/90000. > a=rtpmap:104 MP4V-ES/90000. > a=sendrecv. > > So asterisks adds a second media attribute for video. > > The OK from the second phone looks like this: > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK3aa00bba. > From: "User1" <sip:phone1@192.168.10.75>;tag=as6e33f30b. > To: <sip:phone2@192.168.10.141:1116>;tag=30873f0b0ea954d6. > Call-ID: 73a216f9167396885e099d0f2e5d4ca2@192.168.10.75. > CSeq: 102 INVITE. > User-Agent: Ninja GlobalIPTel. > Max-Forwards: 70. > Contact: <sip:phone2@192.168.10.141:1116>. > Content-Type: application/sdp. > Content-Length: 322. > . > v=0. > o=- 3528024652 3528024652 IN IP4 192.168.10.141. > s=SIPCall. > i=VoIPCall. > c=IN IP4 192.168.10.141. > t=0 0. > m=audio 24608 RTP/AVP 8 0 101. > a=rtpmap:8 PCMA/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-15. > a=ptime:20. > a=sendrecv. > m=video 24610 RTP/AVP 34. > a=rtpmap:34 H263/90000. > a=sendrecv. > > There is also a m=video attribute. > > Asterisk sends the OK to the initiating device: > SIP/2.0 200 OK. > Via: SIP/2.0/UDP > > 192.168.10.106:5062;branch=z9hG4bK1387721920;received=192.168.10.106. > From: "Karsten" <sip:phone1@192.168.10.75>;tag=1171101891. > To: <sip:200@192.168.10.75>;tag=as5d003051. > Call-ID: 1555625029@192.168.10.106. > CSeq: 2 INVITE. > Server: IPTAM PBX. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, > NOTIFY, INFO, PUBLISH. > Supported: replaces, timer. > Contact: <sip:200@192.168.10.75:5060>. > Content-Type: application/sdp. > Content-Length: 262. > . > v=0. > o=root 212893361 212893361 IN IP4 192.168.10.75. > s=Asterisk PBX 1.8.7.0-1. > c=IN IP4 192.168.10.75. > t=0 0. > m=audio 17248 RTP/AVP 8 0 101. > a=rtpmap:8 PCMA/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:20. > a=sendrecv. > > There is no m=video attribute. > > Now, when switching to video, the initiating phone sends: > INVITE sip:200@192.168.10.75:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.10.106:5062;branch=z9hG4bK689886900. > From: "Karsten" <sip:phone1@192.168.10.75>;tag=1171101891. > To: <sip:200@192.168.10.75>;tag=as5d003051. > Call-ID: 1555625029@192.168.10.106. > CSeq: 3 INVITE. > Contact: <sip:phone1@192.168.10.106:5062>. > Proxy-Authorization: Digest username="phone1", realm="asterisk", > nonce="63635ca5", uri="sip:200@192.168.10.75:5060", > response="06f969b3a3555c5b9428a75fa27fc9ae", algorithm=MD5. > Content-Type: application/sdp. > Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, > REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE. > Max-Forwards: 70. > User-Agent: VideoPhone-V8438 22.30.0.60 00:15:65:1b:20:3f. > Allow-Events: talk,hold,conference,refer. > Supported: 100rel. > Content-Length: 361. > . > v=0. > o=- 20006 20008 IN IP4 192.168.10.106. > s=SDP data. > c=IN IP4 192.168.10.106. > t=0 0. > m=audio 10020 RTP/AVP 0 8 18 101. > a=rtpmap:0 PCMU/8000. > a=rtpmap:8 PCMA/8000. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=fmtp:101 0-15. > a=rtpmap:101 telephone-event/8000. > a=sendrecv. > m=video 10022 RTP/AVP 34. > a=rtpmap:34 H263/90000. > a=fmtp:34 CIF=1; QCIF=1. > a=sendrecv. > > Now with a second media line. > > Asterisk sends a 200 OK to the initiator > SIP/2.0 200 OK. > Via: SIP/2.0/UDP > 192.168.10.106:5062;branch=z9hG4bK689886900;received=192.168.10.106. > From: "Karsten" <sip:phone1@192.168.10.75>;tag=1171101891. > To: <sip:200@192.168.10.75>;tag=as5d003051. > Call-ID: 1555625029@192.168.10.106. > CSeq: 3 INVITE. > Server: IPTAM PBX. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, > NOTIFY, INFO, PUBLIS > H. > Supported: replaces, timer. > Contact: <sip:200@192.168.10.75:5060>. > Content-Type: application/sdp. > Content-Length: 336. > . > v=0. > o=root 212893361 212893363 IN IP4 192.168.10.141. > s=Asterisk PBX 1.8.7.0-1. > c=IN IP4 192.168.10.141. > b=CT:384. > t=0 0. > m=audio 24608 RTP/AVP 8 0 101. > a=rtpmap:8 PCMA/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:20. > a=sendrecv. > m=video 24610 RTP/AVP 34. > a=rtpmap:34 H263/90000. > a=sendrecv. > > There is a media attribute for video. But now asterisk sends the INVITE to > second phone: > INVITE sip:phone2@192.168.10.141:1116 SIP/2.0. > Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK2df93523. > Max-Forwards: 70. > From: "User1" <sip:phone1@192.168.10.75>;tag=as6e33f30b. > To: <sip:phone2@192.168.10.141:1116>;tag=30873f0b0ea954d6. > Contact: <sip:phone1@192.168.10.75:5060>. > Call-ID: 73a216f9167396885e099d0f2e5d4ca2@192.168.10.75. > CSeq: 105 INVITE. > User-Agent: IPTAM PBX. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, > NOTIFY, INFO, PUBLISH. > Supported: replaces, timer. > P-Asserted-Identity: "User1" <sip:phone1@192.168.10.75>. > Content-Type: application/sdp. > Content-Length: 266. > . > v=0. > o=root 1873948927 1873948930 IN IP4 192.168.10.106. > s=Asterisk PBX 1.8.7.0-1. > c=IN IP4 192.168.10.106. > t=0 0. > m=audio 10020 RTP/AVP 8 0 101. > a=rtpmap:8 PCMA/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:20. > a=sendrecv. > Now there is no video attribute. > > Is this a known issue or am I doing something wrong? Should I open an issue > on jira? > > Thanks, > > Karsten > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to > Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? 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