SIP Callee no obtiene la notificación de que la llamada terminó

StackOverflow https://stackoverflow.com/questions/3221752

Pregunta

He eliminado mi pregunta anterior y publicé esto actualizado:

Tengo un problema con mi SIP UAC, una vez que recibí un timbre del B2BUA tanto en la persona que llama como en el callee, y la persona que llama cuelga la llamada mientras la llamada está sonando (envío la solicitud de cancelación y recibo "Solicitud terminada "En el lado de la persona que llama), el callee no recibe ninguna notificación de que la llamada haya sido terminada por la persona que llama.

Pero cuando el callee declina la llamada, la persona que llama se ocupa aquí.

Aquí está el lado callee : 

/-----------------------   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   -----------------------------/

Debo disminuir en el extremo de Callee, porque si no respondo a la solicitud, la cuenta de Callee se atasca en un bucle y luego el cliente regresa para siempre, y las solicitudes no llegan a ese cliente, o al menos hasta que La cuenta se elimina.

Y hay otra cosa, el B2BUA no envía nada a la respuesta de declive, ¿no debería obtener un ACK del servidor?

y aquí está el lado llamador : 

/-----------------------   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: 97ee019923d6a6d11a9476d71880e289@10.0.0.1
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: 97ee019923d6a6d11a9476d71880e289@10.0.0.1
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: 97ee019923d6a6d11a9476d71880e289@10.0.0.1
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: 97ee019923d6a6d11a9476d71880e289@10.0.0.1
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: 97ee019923d6a6d11a9476d71880e289@10.0.0.1
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: 97ee019923d6a6d11a9476d71880e289@10.0.0.1
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: 97ee019923d6a6d11a9476d71880e289@10.0.0.1
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: 97ee019923d6a6d11a9476d71880e289@10.0.0.1
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: 97ee019923d6a6d11a9476d71880e289@10.0.0.1
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: 97ee019923d6a6d11a9476d71880e289@10.0.0.1
Content-Length: 0

/----------------------------   END   -----------------------------/

Frank, intenté prestar atención a sus datos, quizás me perdí algo, ya que el otro lado aún no recibe una notificación en una cuelga temprana.

¿Alguna idea de por qué?

gracias de antemano,

adam.

¿Fue útil?

Solución

1) 6xx es inusual;Normalmente, rechazar una llamada se realiza con un 4xx (generalmente "ocupado aquí")

2) La falta de cancelación al destino es un error en el servidor SIP.(OK, no se requieren para enviar una cancelación si cancela, pero realmente deberían).

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