SIP CALLEE non riceve la notifica che la chiamata è finita

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

Domanda

Ho cancellato la mia domanda precedente e pubblica questo aggiornato:

Ho un problema con il mio SIP UAC, una volta ricevuto un squillo dalla B2Bua sia sul chiamante che sul Callee, e il chiamante riaggancia la chiamata mentre la chiamata è squillando (invio di richiesta di annullamento e ricezione "richiesta "Sul lato chiamante), il Callee non ottiene alcuna notifica che la chiamata è stata terminata dal chiamante.

Ma quando il callee declina la chiamata, il chiamante ottiene trafficato qui.

Ecco il 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   -----------------------------/
.

Devo rifiutare l'estremità di Callee, perché se non rispondo alla richiesta, l'account Callee si blocca in un loop e quindi il client restituisce occupato per sempre, e le richieste non raggiungono quel cliente, o almeno fino a quando L'account è cancellato.

E c'è un'altra cosa, il B2Bua non invia nulla alla risposta del declino, non dovrei ottenere un ACK dal server?

Ed Ecco il Caller : 

/-----------------------   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, ho cercato di prestare attenzione ai tuoi dettagli, forse ho perso qualcosa, dal momento che l'altro lato non riceve ancora una notifica su un riaggancio presto.

Qualche idea del perché?

Grazie in anticipo,

adam.

È stato utile?

Soluzione

1) 6xx è insolito;Normalmente respingere una chiamata viene eseguita con un 4xx (di solito "occupato qui")

2) La mancanza di annullamento alla destinazione è un bug nel server SIP.(OK, non sono richiesto per inviare un annullamento se si annulla, ma dovrebbero davvero.)

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