2

我已经删除了我之前的问题并发布了这个更新：

我的 SIP UAC 出现问题，一旦我收到来自 B2BUA 的呼叫者和被呼叫者的振铃，并且呼叫者在呼叫响铃时挂断了呼叫（我发送取消请求并在呼叫方），被呼叫方不会收到呼叫已被呼叫方终止的任何通知。

但是当被呼叫者拒绝呼叫时，呼叫者会在这里忙。

这是被调用方：

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

我必须在被调用者端拒绝，因为如果我不响应请求，被调用者帐户会陷入循环，然后客户端永远返回忙碌，并且请求不会到达该客户端，或者至少直到帐户被删除。

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: 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, 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.

4

1 回答 1

1

1) 6xx is unusual; normally rejecting a call is done with a 4xx (usually "Busy Here")

2) The lack of cancel to the destination is a bug in the sip server. (Ok, they're not required to send a CANCEL if you cancel, but they really should.)

于 2011-04-09T06:15:07.710 回答