Re: [asterisk-users] Problem with video phone call, error in sdp media handling?
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" ;tag=1171101891. > To: . > Call-ID: 1555625029@192.168.10.106. > CSeq: 1 INVITE. > Contact: . > 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" ;tag=as6e33f30b. > To: . > Contact: . > 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" . > 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 H261/9. > a=rtpmap:34 H263/9. > a=rtpmap:98 h263-1998/9. > a=rtpmap:99 H264/9. > a=rtpmap:104 MP4V-ES/9. > 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" ;tag=as6e33f30b. > To: ;tag=30873f0b0ea954d6. > Call-ID: 73a216f9167396885e099d0f2e5d4ca2@192.168.10.75. > CSeq: 102 INVITE. > User-Agent: Ninja GlobalIPTel. > Max-Forwards: 70. > Contact: . > Content-Type: application/sdp. > Content-Length: 322. > . > v=0. > o=- 3528024652 3528024652 IN IP4 192.168.10.141. > s=SIPCall.
Re: [asterisk-users] Problem with video phone call, error in sdp media handling?
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" ;tag=1171101891. To: . Call-ID: 1555625029@192.168.10.106. CSeq: 1 INVITE. Contact: . 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" ;tag=as6e33f30b. To: . Contact: . 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" . 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 H261/9. a=rtpmap:34 H263/9. a=rtpmap:98 h263-1998/9. a=rtpmap:99 H264/9. a=rtpmap:104 MP4V-ES/9. 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" ;tag=as6e33f30b. To: ;tag=30873f0b0ea954d6. Call-ID: 73a216f9167396885e099d0f2e5d4ca2@192.168.10.75. CSeq: 102 INVITE. User-Agent: Ninja GlobalIPTel. Max-Forwards: 70. Contact: . 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 H263/9. 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" ;tag=1171101891. To: ;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.
[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" ;tag=1171101891. To: . Call-ID: 1555625029@192.168.10.106. CSeq: 1 INVITE. Contact: . 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" ;tag=as6e33f30b. To: . Contact: . 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" . 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 H261/9. a=rtpmap:34 H263/9. a=rtpmap:98 h263-1998/9. a=rtpmap:99 H264/9. a=rtpmap:104 MP4V-ES/9. 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" ;tag=as6e33f30b. To: ;tag=30873f0b0ea954d6. Call-ID: 73a216f9167396885e099d0f2e5d4ca2@192.168.10.75. CSeq: 102 INVITE. User-Agent: Ninja GlobalIPTel. Max-Forwards: 70. Contact: . 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 H263/9. 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" ;tag=1171101891. To: ;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: . 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. The
