[asterisk-users] Problem with video phone call, error in sdp media handling?
Karsten Wemheuer
kwem at gmx.de
Wed Oct 19 11:01:52 CDT 2011
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-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Karsten
> Wemheuer
> Sent: Wednesday, October 19, 2011 10:37 AM
> To: asterisk-users at 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 at 192.168.10.75 SIP/2.0.
> Via: SIP/2.0/UDP 192.168.10.106:5062;branch=z9hG4bK1784123944.
> From: "Karsten" <sip:phone1 at 192.168.10.75>;tag=1171101891.
> To: <sip:200 at 192.168.10.75>.
> Call-ID: 1555625029 at 192.168.10.106.
> CSeq: 1 INVITE.
> Contact: <sip:phone1 at 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 at 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 at 192.168.10.75>;tag=as6e33f30b.
> To: <sip:phone2 at 192.168.10.141:1116>.
> Contact: <sip:phone1 at 192.168.10.75:5060>.
> Call-ID: 73a216f9167396885e099d0f2e5d4ca2 at 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 at 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 at 192.168.10.75>;tag=as6e33f30b.
> To: <sip:phone2 at 192.168.10.141:1116>;tag=30873f0b0ea954d6.
> Call-ID: 73a216f9167396885e099d0f2e5d4ca2 at 192.168.10.75.
> CSeq: 102 INVITE.
> User-Agent: Ninja GlobalIPTel.
> Max-Forwards: 70.
> Contact: <sip:phone2 at 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 at 192.168.10.75>;tag=1171101891.
> To: <sip:200 at 192.168.10.75>;tag=as5d003051.
> Call-ID: 1555625029 at 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 at 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 at 192.168.10.75:5060 SIP/2.0.
> Via: SIP/2.0/UDP 192.168.10.106:5062;branch=z9hG4bK689886900.
> From: "Karsten" <sip:phone1 at 192.168.10.75>;tag=1171101891.
> To: <sip:200 at 192.168.10.75>;tag=as5d003051.
> Call-ID: 1555625029 at 192.168.10.106.
> CSeq: 3 INVITE.
> Contact: <sip:phone1 at 192.168.10.106:5062>.
> Proxy-Authorization: Digest username="phone1", realm="asterisk",
> nonce="63635ca5", uri="sip:200 at 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 at 192.168.10.75>;tag=1171101891.
> To: <sip:200 at 192.168.10.75>;tag=as5d003051.
> Call-ID: 1555625029 at 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 at 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 at 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 at 192.168.10.75>;tag=as6e33f30b.
> To: <sip:phone2 at 192.168.10.141:1116>;tag=30873f0b0ea954d6.
> Contact: <sip:phone1 at 192.168.10.75:5060>.
> Call-ID: 73a216f9167396885e099d0f2e5d4ca2 at 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 at 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
>
>
> --
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>
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