SIP Callee는 통화가 끝난 통지를 얻지 못합니다

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

문제

이전의 질문을 삭제하고 업데이트 된 업데이트 된 게시

호출자와 Calle에서 B2Bua에서 벨소리를 받았고 호출자가 울리는 동안 호출자가 통화를 끊는 일단 호출자가 전화를 끊습니다 (나는 취소 요청을 보내고 "요청을 보내고" "발신자 측면에서) 호출자가 호출자가 통화가 종료되었다는 알림을받지 못합니다.

그러나 Callee가 전화를 거부하면 발신자가 여기서 바쁠 것입니다.

여기 callee 측면이 있습니다. 

/-----------------------   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   -----------------------------/
.

요청에 응답하지 않으면 CALLE 계정이 루프에 멈추지 않으면 클라이언트가 영원히 바쁘게 되돌아 오거나 요청이 클라이언트에 도달하지 않거나 적어도 사용할 때까지 계정이 삭제됩니다.

다른 것이 있으며, B2Bua는 쇠퇴 응답으로 되돌아 가지 않아야합니다. 서버에서 ACK를 얻지 않아야합니까?

와 여기 호출자 측면 : 

/-----------------------   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   -----------------------------/
.

프랭크, 나는 당신의 세부 사항에주의를 기울이려고 노력했습니다. 아마도 다른 쪽이 여전히 일찍 끊어지는 통지를받지 못하면서 뭔가를 놓쳤을 것입니다.

왜 그 이유는 무엇입니까?

미리 감사드립니다

아담.

도움이 되었습니까?

해결책

1) 6xx는 특이합니다.일반적으로 전화를 거부하는 것은 4xx (보통 "바쁜 여기")로 수행됩니다.

2) 대상에 대한 취소 부족은 SIP 서버의 버그입니다.(확인, 필요 없음 은 취소되면 취소를 보내지 만 실제로해야합니다.)

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