Hi , I am attaching the sip debug messages enabled on asterisk server .the present status is that when i call from Mirial softphone ,the call gets established and gets disconnected after a while .i can't see video on mcuWeb page .i think i have done all things correctly but i am stuck with this thing .please help me out of this problem .
INVITE sip:300@192.168.115.29:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.115.53:5060;branch= z9hG4bK504e1a07 Max-Forwards: 70 From: "205" <sip:205@192.168.115.53>;tag=as03ab7ca9 To: <sip:300@192.168.115.29:5060> Contact: <sip:205@192.168.115.53:5060> Call-ID: 687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.6.0 Date: Tue, 11 Oct 2011 06:54:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 363 v=0 o=root 1739466419 1739466419 IN IP4 192.168.115.53 s=Asterisk PBX 1.8.6.0 c=IN IP4 192.168.115.53 b=CT:384 t=0 0 m=audio 14856 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 15692 RTP/AVP 99 a=rtpmap:99 H264/90000 a=sendrecv --- <--- SIP read from UDP:192.168.115.29:5060 ---> SIP/2.0 100 Trying Content-Length: 0 To: <sip:300@192.168.115.29:5060> Cseq: 102 INVITE Via: SIP/2.0/UDP 192.168.115.53:5060;branch=z9hG4bK504e1a07 Server: Glassfish_SIP_1.0.0 Call-Id: 687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060 From: "205" <sip:205@192.168.115.53>;tag=as03ab7ca9 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:192.168.115.29:5060 ---> SIP/2.0 180 Ringing Content-Length: 0 To: <sip:300@192.168.115.29:5060>;tag=gtmjcomk-b Contact: <sip:192.168.115.29:5070;fid=server_1> Cseq: 102 INVITE Via: SIP/2.0/UDP 192.168.115.53:5060;branch=z9hG4bK504e1a07 From: "205"<sip:205@192.168.115.53>;tag=as03ab7ca9 Call-Id: 687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060 Server: Glassfish_SIP_1.0.0 <-------------> --- (9 headers 0 lines) --- <--- Transmitting (no NAT) to 192.168.115.40:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.115.40:5060 ;branch=z9hG4bK-d7358366c5-DL;received=192.168.115.40 From: "205" <sip:205@192.168.115.53>;tag=DLb62dc55e0d;epid=021776A8 To: <sip:300@192.168.115.53;user=phone>;tag=as6ba67fa9 Call-ID: DLd873cc4431-1078986205@fstl-312 CSeq: 1 INVITE Server: Asterisk PBX 1.8.6.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:300@192.168.115.53:5060> Content-Length: 0 <------------> <--- SIP read from UDP:192.168.115.29:5060 ---> SIP/2.0 200 Ok Content-Length: 245 To: <sip:300@192.168.115.29:5060>;tag=gtmjcomk-b Contact: <sip:192.168.115.29:5070;fid=server_1> Cseq: 102 INVITE Via: SIP/2.0/UDP 192.168.115.53:5060;branch=z9hG4bK504e1a07 Content-Type: application/sdp From: "205"<sip:205@192.168.115.53>;tag=as03ab7ca9 Call-Id: 687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060 Server: Glassfish_SIP_1.0.0 v=0 o=- 0 0 IN IP4 192.168.115.29 s=MediaMixerSession c=IN IP4 192.168.115.29 t=0 0 m=audio 60014 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 m=video 35814 RTP/AVP 99 a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=428021 <-------------> --- (10 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found RTP video format 99 Found video description format H264 for ID 99 Capabilities: us - 0x30000c (ulaw|alaw|h263p|h264), peer - audio=0xc (ulaw|alaw)/video=0x200000 (h264)/text=0x0 (nothing), combined - 0x20000c (ulaw|alaw|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.115.29:60014 Peer video RTP is at port 192.168.115.29:35814 list_route: hop: <sip:192.168.115.29:5070;fid=server_1> set_destination: Parsing <sip:192.168.115.29:5070;fid=server_1> for address/port to send to set_destination: set destination to 192.168.115.29:5070 Transmitting (no NAT) to 192.168.115.29:5070: ACK sip:192.168.115.29:5070;fid=server_1 SIP/2.0 Via: SIP/2.0/UDP 192.168.115.53:5060;branch=z9hG4bK07561f41 Max-Forwards: 70 From: "205" <sip:205@192.168.115.53>;tag=as03ab7ca9 To: <sip:300@192.168.115.29:5060>;tag=gtmjcomk-b Contact: <sip:205@192.168.115.53:5060> Call-ID: 687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.6.0 Content-Length: 0 --- Audio is at 5060 Video is at 192.168.115.53:5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding video codec 0x200000 (h264) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.115.40:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.115.40:5060 ;branch=z9hG4bK-d7358366c5-DL;received=192.168.115.40 From: "205" <sip:205@192.168.115.53>;tag=DLb62dc55e0d;epid=021776A8 To: <sip:300@192.168.115.53;user=phone>;tag=as6ba67fa9 Call-ID: DLd873cc4431-1078986205@fstl-312 CSeq: 1 INVITE Server: Asterisk PBX 1.8.6.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:300@192.168.115.53:5060> Content-Type: application/sdp Content-Length: 361 v=0 o=root 797835425 797835425 IN IP4 192.168.115.53 s=Asterisk PBX 1.8.6.0 c=IN IP4 192.168.115.53 b=CT:384 t=0 0 m=audio 10346 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 12196 RTP/AVP 99 a=rtpmap:99 H264/90000 a=sendrecv <------------> -- Locally bridging SIP/205-00000002 and SIP/mcuWeb-00000003 <--- SIP read from UDP:192.168.115.40:5060 ---> ACK sip:300@192.168.115.53:5060 SIP/2.0 CSeq: 1 ACK Via: SIP/2.0/UDP 192.168.115.40:5060;branch=z9hG4bK-818b523671-DL To: <sip:300@192.168.115.53;user=phone>;tag=as6ba67fa9 From: "205" <sip:205@192.168.115.53>;tag=DLb62dc55e0d;epid=021776A8 Call-ID: DLd873cc4431-1078986205@fstl-312 Max-Forwards: 70 Contact: "205" <sip:205@192.168.115.40:5060> Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:192.168.115.29:5060 ---> SIP/2.0 200 Ok Content-Length: 245 To: <sip:300@192.168.115.29:5060>;tag=gtmjcomk-b Contact: <sip:192.168.115.29:5070;fid=server_1> Cseq: 102 INVITE Via: SIP/2.0/UDP 192.168.115.53:5060;branch=z9hG4bK504e1a07 Content-Type: application/sdp From: "205"<sip:205@192.168.115.53>;tag=as03ab7ca9 Call-Id: 687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060 Server: Glassfish_SIP_1.0.0 v=0 o=- 0 0 IN IP4 192.168.115.29 s=MediaMixerSession c=IN IP4 192.168.115.29 t=0 0 m=audio 60014 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 m=video 35814 RTP/AVP 99 a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=428021 <-------------> --- (10 headers 11 lines) --- set_destination: Parsing <sip:192.168.115.29:5070;fid=server_1> for address/port to send to set_destination: set destination to 192.168.115.29:5070 Transmitting (no NAT) to 192.168.115.29:5070: ACK sip:192.168.115.29:5070;fid=server_1 SIP/2.0 Via: SIP/2.0/UDP 192.168.115.53:5060;branch=z9hG4bK4559c232 Max-Forwards: 70 From: "205" <sip:205@192.168.115.53>;tag=as03ab7ca9 To: <sip:300@192.168.115.29:5060>;tag=gtmjcomk-b Contact: <sip:205@192.168.115.53:5060> Call-ID: 687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.6.0 Content-Length: 0 <--- SIP read from UDP:192.168.115.29:5060 ---> BYE sip:205@192.168.115.53:5060 SIP/2.0 Max-Forwards: 69 Content-Length: 0 To: "205"<sip:205@192.168.115.53>; tag=as03ab7ca9 Contact: <sip:192.168.115.29:5070;fid=server_1> Cseq: 2 BYE Via: SIP/2.0/UDP 192.168.115.29:5070 ;branch=z9hG4bKdaaccb789521ce3e4a69b8b9f63068bbfa01 From: <sip:300@192.168.115.29:5060>;tag=gtmjcomk-b Call-Id: 687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060 <-------------> --- (9 headers 0 lines) --- Sending to 192.168.115.29:5070 (no NAT) Scheduling destruction of SIP dialog ' 687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060' in 32000 ms (Method: BYE) <--- Transmitting (no NAT) to 192.168.115.29:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.115.29:5070 ;branch=z9hG4bKdaaccb789521ce3e4a69b8b9f63068bbfa01;received=192.168.115.29 From: <sip:300@192.168.115.29:5060>;tag=gtmjcomk-b To: "205"<sip:205@192.168.115.53>;tag=as03ab7ca9 Call-ID: 687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060 CSeq: 2 BYE Server: Asterisk PBX 1.8.6.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'DLd873cc4431-1078986205@fstl-312' in 32000 ms (Method: OPTIONS) set_destination: Parsing <sip:205@192.168.115.40:5060> for address/port to send to set_destination: set destination to 192.168.115.40:5060 Reliably Transmitting (no NAT) to 192.168.115.40:5060: BYE sip:205@192.168.115.40:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.115.53:5060;branch=z9hG4bK22de4c9c Max-Forwards: 70 From: <sip:300@192.168.115.53;user=phone>;tag=as6ba67fa9 To: "205" <sip:205@192.168.115.53>;tag=DLb62dc55e0d;epid=021776A8 Call-ID: DLd873cc4431-1078986205@fstl-312 CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.6.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- <--- SIP read from UDP:192.168.115.40:5060 ---> SIP/2.0 200 OK From: <sip:300@192.168.115.53;user=phone>;tag=as6ba67fa9;epid=021776A8 Call-ID: DLd873cc4431-1078986205@fstl-312 CSeq: 102 BYE Via: SIP/2.0/UDP 192.168.115.53:5060 ;branch=z9hG4bK22de4c9c;received=192.168.115.53 To: "205" <sip:205@192.168.115.53>;tag=DLb62dc55e0d;epid=021776A8 Allow: INVITE,CANCEL,ACK,OPTIONS,INFO,SUBSCRIBE,NOTIFY,BYE,MESSAGE,UPDATE,REFER Contact: "205" <sip:205@192.168.115.40:5060> User-Agent: Dylogic Mirial 7.0.54 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog 'DLd873cc4431-1078986205@fstl-312' Method: OPTIONS <--- SIP read from UDP:192.168.115.29:5060 ---> BYE sip:205@192.168.115.53:5060 SIP/2.0 Max-Forwards: 69 Content-Length: 0 Contact: <sip:192.168.115.29:5070;fid=server_1> To: "205"<sip:205@192.168.115.53>;tag=as03ab7ca9 Cseq: 2 BYE Via: SIP/2.0/UDP 192.168.115.29:5070 ;branch=z9hG4bKdaaccb789521ce3e4a69b8b9f63068bbfa01 Call-Id: 687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060 From: <sip:300@192.168.115.29:5060>;tag=gtmjcomk-b <-------------> --- (9 headers 0 lines) --- Sending to 192.168.115.29:5070 (no NAT) <--- Transmitting (no NAT) to 192.168.115.29:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.115.29:5070 ;branch=z9hG4bKdaaccb789521ce3e4a69b8b9f63068bbfa01;received=192.168.115.29 From: <sip:300@192.168.115.29:5060>;tag=gtmjcomk-b To: "205"<sip:205@192.168.115.53>;tag=as03ab7ca9 Call-ID: 687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060 CSeq: 2 BYE Server: Asterisk PBX 1.8.6.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> <--- SIP read from UDP:192.168.115.29:5060 ---> BYE sip:205@192.168.115.53:5060 SIP/2.0 Max-Forwards: 69 Content-Length: 0 Contact: <sip:192.168.115.29:5070;fid=server_1> To: "205"<sip:205@192.168.115.53>;tag=as03ab7ca9 Cseq: 2 BYE Via: SIP/2.0/UDP 192.168.115.29:5070 ;branch=z9hG4bKdaaccb789521ce3e4a69b8b9f63068bbfa01 Call-Id: 687d923112d7df683bf0cb08115c1d7d@192.168.115.53:5060 From: <sip:300@192.168.115.29:5060>;tag=gtmjcomk-b
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