Hey,
I have deleted my previous question and post this updated:
I have a an issue with my SIP UAC, once I received a ringing from the B2BUA on both the caller and callee, and the caller hangs up the call while the call is ringing (I send cancel request and receive "request terminated" on the caller side), the callee does not get any notification that the call has been terminated by the caller.
But when the callee declines the call, the caller gets a busy here.
Here is the Callee side:
/----------------------- MEDIA SESSION ------------------------/
--- Multimedia-Session: Composed Audio ---
1. Media Session: "Audio" enabled=true
States:
[Disconnected]
Capturers: (1 in total)
Stream 1: audio device - DirectSoundCapture
Formats:
[PCMU/8000]
Connection Details:
My address: 10.0.0.2:52044
Participants: (1 in total)
Address 1: HostAddress:17364
/-------------------- END OF MEDIA SESSION --------------------/
/------------------------- BEGINNING --------------------------/
-------------------------------- Request: Test 2-->Me: INVITE#102 --------------------------------
INVITE sip:430@Host SIP/2.0
Via: SIP/2.0/UDP HostAddress:5060;branch=z9hG4bK1fd06834;rport=5060;received=HostAddress
From: "Test 2" <sip:410@HostAddress>;tag=as2b22eddf
To: <sip:430@Host>
Contact: <sip:410@HostAddress>
Call-ID: 35e0e8655b20ad886f137a0c0e563809@HostAddress
CSeq: 102 INVITE
User-Agent: Freeswitch 1.2.3
Max-Forwards: 70
Date: Sun, 11 Jul 2010 02:44:43 GMT
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 264
v=0
o=root 27669 27669 IN IP4 HostAddress
s=session
c=IN IP4 HostAddress
t=0 0
m=audio 17364 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
------------------------ Response: Me ==> Test 2: INVITE#102: 180 Ringing ------------------------
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP HostAddress:5060;branch=z9hG4bK1fd06834;rport=5060;received=HostAddress
From: "Test 2" <sip:410@HostAddress>;tag=as2b22eddf
To: <sip:430@Host>;tag=e125be76
Call-ID: 35e0e8655b20ad886f137a0c0e563809@HostAddress
CSeq: 102 INVITE
Content-Length: 0
------------------------ Response: Me ==> Test 2: INVITE#102: 603 Decline ------------------------
SIP/2.0 603 Decline
Via: SIP/2.0/UDP HostAddress:5060;branch=z9hG4bK1fd06834;rport=5060;received=HostAddress
From: "Test 2" <sip:410@HostAddress>;tag=as2b22eddf
To: <sip:430@Host>;tag=e125be76
Call-ID: 35e0e8655b20ad886f137a0c0e563809@HostAddress
CSeq: 102 INVITE
Content-Length: 0
/---------------------------- END -----------------------------/
I must decline at the callee end, because if I don't respond to the request, the callee account get stuck in a loop and then the client returns busy forever, and requests does not reach that client, or at least until the account is deleted.
And there is another thing, the B2BUA does not send anything back to the decline response, shouldn't I get an ACK from the server?
And here is the Caller side:
/----------------------- MEDIA SESSION ------------------------/
--- Multimedia-Session: Audio ---
1. Media Session: "Audio" enabled=true
States:
[Disconnected]
Capturers: (1 in total)
Stream 1: audio device - DirectSoundCapture
Formats:
[PCMU/8000]
[GSM/8000]
[G723/8000]
[DVI4/8000]
[MPA/-1]
[DVI4/11025]
[DVI4/22050]
Connection Details:
My address:
Participants: (0 in total)
/-------------------- END OF MEDIA SESSION --------------------/
/------------------------- BEGINNING --------------------------/
-------------------------- Request: Client 410-->Client 430: INVITE#81 --------------------------
INVITE sip:430@host SIP/2.0
Subject: Session Name: Nu-Art Software
Via: SIP/2.0/UDP host:5060;branch=z9hG4bK4dd6bdf707a85fb5a73faec9ff648f703236
Contact: "Client 410" <sip:410@host>
From: "Client 410" <sip:410@host>;tag=8f7b94cb
To: "Client 430" <sip:430@host>
Organization: Future Earth
Max-Forwards: 32
CSeq: 81 INVITE
Call-ID: [email protected]
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
Expires: 60
Content-Type: application/sdp
Content-Length: 324
v=0
o=Client 410 699719 699719 IN IP4 MyAddress
s=Audio
i=Made by: Nu-Art Software 07-2010
c=IN IP4 MyAddress
t=0 0
m=audio 2871 RTP/AVP 0 3 4 5 14 16 17
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:14 MPA/-1
a=rtpmap:16 DVI4/11025
a=rtpmap:17 DVI4/22050
------- Response: Client 430 ==> Client 410: INVITE#81: 407 Proxy Authentication Required -------
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP host:5060;branch=z9hG4bK4dd6bdf707a85fb5a73faec9ff648f703236;received=MyAddress
From: "Client 410" <sip:410@host>;tag=8f7b94cb
To: "Client 430" <sip:430@host>;tag=as78e28f4d
Call-ID: [email protected]
CSeq: 81 INVITE
User-Agent: Freeswitch 1.2.3
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5,realm="asterisk",nonce="574b3d49"
Content-Length: 0
---------------------------- Request: Client 410-->Client 430: ACK#81 ----------------------------
ACK sip:430@host SIP/2.0
Via: SIP/2.0/UDP host:5060;branch=z9hG4bK4dd6bdf707a85fb5a73faec9ff648f703236
From: "Client 410" <sip:410@host>;tag=8f7b94cb
To: "Client 430" <sip:430@host>
Max-Forwards: 32
CSeq: 81 ACK
Call-ID: [email protected]
Content-Length: 0
-------------------------- Request: Client 410-->Client 430: INVITE#82 --------------------------
INVITE sip:430@host SIP/2.0
Subject: Session Name: Nu-Art Software
Via: SIP/2.0/UDP host:5060;branch=z9hG4bK34c52041066f24c6ac4499af25a948b63236
Contact: "Client 410" <sip:410@host>
From: "Client 410" <sip:410@host>;tag=8f7b94cb
To: "Client 430" <sip:430@host>;tag=as78e28f4d
Organization: Future Earth
Max-Forwards: 32
CSeq: 82 INVITE
Call-ID: [email protected]
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
Expires: 60
Content-Type: application/sdp
Proxy-Authorization: Digest username="410",nonce="574b3d49",realm="asterisk",uri="sip:410@host",algorithm=MD5,response="e674e15de7b6dd05c7fe6da6c155befd"
Content-Length: 324
v=0
o=Client 410 699719 699719 IN IP4 MyAddress
s=Audio
i=Made by: Nu-Art Software 07-2010
c=IN IP4 MyAddress
t=0 0
m=audio 2871 RTP/AVP 0 3 4 5 14 16 17
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:14 MPA/-1
a=rtpmap:16 DVI4/11025
a=rtpmap:17 DVI4/22050
------------------- Response: Client 430 ==> Client 410: INVITE#82: 100 Trying -------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP host:5060;branch=z9hG4bK34c52041066f24c6ac4499af25a948b63236;received=MyAddress
From: "Client 410" <sip:410@host>;tag=8f7b94cb
To: "Client 430" <sip:430@host>;tag=as78e28f4d
Call-ID: [email protected]
CSeq: 82 INVITE
User-Agent: Freeswitch 1.2.3
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO
Supported: replaces
Contact: <sip:430@HostAddress>
Content-Length: 0
------------------ Response: Client 430 ==> Client 410: INVITE#82: 180 Ringing ------------------
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP host:5060;branch=z9hG4bK34c52041066f24c6ac4499af25a948b63236;received=MyAddress
From: "Client 410" <sip:410@host>;tag=8f7b94cb
To: "Client 430" <sip:430@host>;tag=as78e28f4d
Call-ID: [email protected]
CSeq: 82 INVITE
User-Agent: Freeswitch 1.2.3
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO
Supported: replaces
Contact: <sip:430@HostAddress>
Content-Length: 0
-------------------------- Request: Client 410-->Client 430: CANCEL#82 --------------------------
CANCEL sip:430@host SIP/2.0
Call-ID: [email protected]
To: "Client 430" <sip:430@host>;tag=as78e28f4d
CSeq: 82 CANCEL
From: "Client 410" <sip:410@host>;tag=8f7b94cb
Via: SIP/2.0/UDP host:5060;branch=z9hG4bK34c52041066f24c6ac4499af25a948b63236
Max-Forwards: 32
Content-Length: 0
--------------------- Response: Client 430 ==> Client 410: CANCEL#82: 200 OK ---------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP host:5060;branch=z9hG4bK34c52041066f24c6ac4499af25a948b63236;received=MyAddress
From: "Client 410" <sip:410@host>;tag=8f7b94cb
To: "Client 430" <sip:430@host>;tag=as78e28f4d
Call-ID: [email protected]
CSeq: 82 CANCEL
User-Agent: Freeswitch 1.2.3
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO
Supported: replaces
Content-Length: 0
------------- Response: Client 430 ==> Client 410: INVITE#82: 487 Request Terminated -------------
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP host:5060;branch=z9hG4bK34c52041066f24c6ac4499af25a948b63236;received=MyAddress
From: "Client 410" <sip:410@host>;tag=8f7b94cb
To: "Client 430" <sip:430@host>;tag=as78e28f4d
Call-ID: [email protected]
CSeq: 82 INVITE
User-Agent: Freeswitch 1.2.3
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO
Supported: replaces
Content-Length: 0
---------------------------- Request: Client 410-->Client 430: ACK#82 ----------------------------
ACK sip:430@host SIP/2.0
Via: SIP/2.0/UDP host:5060;branch=z9hG4bK34c52041066f24c6ac4499af25a948b63236
From: "Client 410" <sip:410@host>;tag=8f7b94cb
To: "Client 430" <sip:430@host>
Max-Forwards: 32
CSeq: 82 ACK
Call-ID: [email protected]
Content-Length: 0
/---------------------------- END -----------------------------/
Frank, I tried to pay attention to your details, perhaps I missed something, since the other side still does not receives notification on an early hang up.
Any idea why?
Thanks in advance,
Adam.