Hi ,
Iam facing an issue with Cancel. Please find the Cancel.txt which has the
call cancel sequence..
Voip-1 makes a call throough Server ,before Voip2 accepts it cancels call.
1)Voip-1 sends Cancel ,Server responds with 200 Ok.
2)Server sends 487 Request Termination ,But Voip1 is not responding .Whats
exactly the issue here .?Please Can some one clarify here.
3) Voip1 sends Cancel request for which server responds with 481 call doesn't
exist
4) Voip1 Sends ACK .Even After this Server keep sending 481 call doesn't exist.
Voip 1: 10.32.140.122
Voip 2: 10.32.140.139
Server: 10.32.128.20
Please Can some one let me know where excatly the issue is.
Best Regards,
Ckumar.
Voip1:10.32.140.122
INVITE sip:[email protected];transport=udp SIP/2.0
From: "AHTV Ric
1"<sip:[email protected]>;tag=13e7858-7a8c200a-13c4-5506-38-3625c56-38
To: <sip:[email protected];transport=udp>
Call-ID: 14800c0-7a8c200a-13c4-5506-38-1b91bc30-38
CSeq: 1 INVITE
Via: SIP/2.0/UDP 10.32.140.122:5060;rport;branch=z9hG4bK-38-dc39-6b4deb27
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,REFER,INFO,MESSAGE,NOTIFY,PRACK,UPDATE
Priority: normal
Subject: calltype=standard
Max-Forwards: 70
Supported: 100rel,timer,replaces
Contact: <sip:[email protected]:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 355
v=0
o=200 139924350 1597803496 IN IP4 10.32.140.122
s=ADI VoIP Phone
c=IN IP4 10.32.140.122
b=AS:1024
t=0 0
m=audio 8000 RTP/AVP 8 18 4 0 101
a=fmtp:101 0-15
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=fmtp:18 annexb=no
a=fmtp:4 annexa=no
a=ptime:40
a=sendrecv
a=rtpmap:101 telephone-event/8000
Server:10.32.128.20
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.32.140.122:5060;rport;branch=z9hG4bK-38-dc39-6b4deb27
From: "AHTV Ric
1"<sip:[email protected]>;tag=13e7858-7a8c200a-13c4-5506-38-3625c56-38
To: <sip:[email protected];transport=udp>
Call-ID: 14800c0-7a8c200a-13c4-5506-38-1b91bc30-38
CSeq: 1 INVITE
Contact: <sip:10.32.128.20>
Allow: INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO
Allow-Events: dialog
Supported: gruu
Content-Length: 0
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.32.140.122:5060;rport;branch=z9hG4bK-38-dc39-6b4deb27
From: "AHTV Ric
1"<sip:[email protected]>;tag=13e7858-7a8c200a-13c4-5506-38-3625c56-38
To: <sip:[email protected];transport=udp>;tag=2406394696
Call-ID: 14800c0-7a8c200a-13c4-5506-38-1b91bc30-38
CSeq: 1 INVITE
Contact: <sip:10.32.128.20>
Allow: INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO
Allow-Events: dialog
Supported: gruu
Content-Length: 0
Voip2:10.32.140.139
INVITE sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.32.128.20:5060;branch=z9hG4bK-2625200056;rport
From: "AHTV Ric1 200" <sip:[email protected]>;tag=2625275176
To: <sip:[email protected];transport=udp>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]:5060>
Accept: application/sdp
Alert-Info: bellcore-dr1
Allow: INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO
Allow-Events: dialog
Max-Forwards: 70
Priority: normal
Subject: calltype=standard;entity=phonedev.2017
Supported: gruu
User-Agent: CYTEL.iBX SIP Client
Content-Type: application/sdp
Content-Length: 347
v=0
o=cytelibx 356350691 356350691 IN IP4 10.32.128.20
s=call
c=IN IP4 10.32.128.20
t=0 0
m=audio 8008 RTP/AVP 8 0 2 3 18 97 101
a=rtpmap:8 pcma/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:97 ilbc/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:40
a=sendrecv
SIP/2.0 100 Trying
From: "AHTV Ric1 200"<sip:[email protected]>;tag=2625275176
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 INVITE
Via: SIP/2.0/UDP 10.32.128.20:5060;rport=5060;branch=z9hG4bK-2625200056
Supported: 100rel,timer,replaces
Contact: <sip:[email protected]:5060>
Content-Length: 0
SIP/2.0 180 Ringing
From: "AHTV Ric1 200"<sip:[email protected]>;tag=2625275176
To: <sip:[email protected]>;tag=13d9a18-8b8c200a-13c4-5506-2f1-23e9b9c3-2f1
Call-ID: [email protected]
CSeq: 1 INVITE
Via: SIP/2.0/UDP 10.32.128.20:5060;rport=5060;branch=z9hG4bK-2625200056
Supported: 100rel,timer,replaces
Contact: <sip:[email protected]:5060>
Content-Length: 0
Voip-1 Sends Cancel
CANCEL sip:[email protected];transport=udp SIP/2.0
From: "AHTV Ric
1"<sip:[email protected]>;tag=13e7858-7a8c200a-13c4-5506-38-3625c56-38
To: <sip:[email protected];transport=udp>
Call-ID: 14800c0-7a8c200a-13c4-5506-38-1b91bc30-38
CSeq: 1 CANCEL
Via: SIP/2.0/UDP 10.32.140.122:5060;rport;branch=z9hG4bK-38-dc39-6b4deb27
Max-Forwards: 70
Supported: 100rel,timer,replaces
Content-Length: 0
Server responds with 200 Ok
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.32.140.122:5060;rport;branch=z9hG4bK-38-dc39-6b4deb27;rport=5060
From: "AHTV Ric
1"<sip:[email protected]>;tag=13e7858-7a8c200a-13c4-5506-38-3625c56-38
To: <sip:[email protected];transport=udp>;tag=2406394696
Call-ID: 14800c0-7a8c200a-13c4-5506-38-1b91bc30-38
CSeq: 1 CANCEL
Contact: <sip:10.32.128.20>
Allow: INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO
Allow-Events: dialog
Supported: gruu
Content-Length: 0
Server sends 487
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP
10.32.140.122:5060;rport;branch=z9hG4bK-38-dc39-6b4deb27;rport=5060
From: "AHTV Ric
1"<sip:[email protected]>;tag=13e7858-7a8c200a-13c4-5506-38-3625c56-38
To: <sip:[email protected];transport=udp>;tag=2406394696
Call-ID: 14800c0-7a8c200a-13c4-5506-38-1b91bc30-38
CSeq: 1 INVITE
Contact: <sip:10.32.128.20>
Allow: INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO
Allow-Events: dialog
Supported: gruu
Content-Length: 0
Voip1 is not responding to 487
Sequence between Server & Voip2 works fine.
Voip1 keep sends Cancel
CANCEL sip:[email protected];transport=udp SIP/2.0
From: "AHTV Ric
1"<sip:[email protected]>;tag=13e7858-7a8c200a-13c4-5506-38-3625c56-38
To: <sip:[email protected];transport=udp>
Call-ID: 14800c0-7a8c200a-13c4-5506-38-1b91bc30-38
CSeq: 1 CANCEL
Via: SIP/2.0/UDP 10.32.140.122:5060;rport;branch=z9hG4bK-38-dc39-6b4deb27
Max-Forwards: 70
Supported: 100rel,timer,replaces
Content-Length: 0
Server Sends 481 call leg doesn't exist.
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.32.140.122:5060;rport;branch=z9hG4bK-38-dc39-6b4deb27
From: "AHTV Ric
1"<sip:[email protected]>;tag=13e7858-7a8c200a-13c4-5506-38-3625c56-38
To: <sip:[email protected];transport=udp>
Call-ID: 14800c0-7a8c200a-13c4-5506-38-1b91bc30-38
CSeq: 1 CANCEL
Contact: <sip:10.32.128.20>
Allow: INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO
Allow-Events: dialog
Supported: gruu
Content-Length: 0
Voip sends ACK for 481 Response
ACK sip:[email protected];transport=udp SIP/2.0
From: "AHTV Ric
1"<sip:[email protected]>;tag=13e7858-7a8c200a-13c4-5506-38-3625c56-38
To: <sip:[email protected];transport=udp>;tag=2406394696
Call-ID: 14800c0-7a8c200a-13c4-5506-38-1b91bc30-38
CSeq: 1 ACK
Via: SIP/2.0/UDP 10.32.140.122:5060;rport;branch=z9hG4bK-38-dc39-6b4deb27
Max-Forwards: 70
Contact: <sip:[email protected]:5060;transport=udp>
Content-Length: 0
Even after this ACk Server keep sending 481 call/leg doesn't exit
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