我正在使用JAIN SIP在Java中创建SIP客户端。
我已经成功注册并发送INVITE，但是将ACK发送回服务器时却收到错误消息：
Cannot Create ACK - no remote Target
我检查了Dialog的值，它为null
在processResponce（）中，我得到的值也为null
this.dialog = responseEvent.getClientTransaction().getDialog();
我将其传递给ack（responseEvent.getResponse（），对话框）
request =this.dialog.createAck(((CSeqHeader)response.getHeader("CSeq")).getSeqNumber());
dialog.sendAck(request);
同样在Register（）和Call（）中
this.dialog = inviteTid.getDialog();
这里的对话框的值也为空
我也试过
dialog = sipProvider.getNewDialog(inviteID);
但它给出了错误
AUTOMATIC_DIALOG_SUPPORT is on
我是否必须初始化对话框或进行更多调用以设置其值？
如何实施ACK？
REGISTER sip:SipIP SIP/2.0
Call-ID: b83eb80b195f6802c283c5927fb6415a@localHost
CSeq: 1 REGISTER
From: <sip:username@SipIP>;tag=1626086046
To: <sip:username@SipIP>
Via: SIP/2.0/UDP localHost:52216;rport;branch=z9hG4bK-363935-33f876b2e3720a123b62c68fc23cfa68
Max-Forwards: 70
Contact: <sip:username@localHost:52216;transport=UDP>
Content-Length: 0
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP localHost:52216;branch=z9hG4bK-363935-33f876b2e3720a123b62c68fc23cfa68;received=localHost;rport=52216
From: <sip:username@SipIP>;tag=1626086046
To: <sip:username@SipIP>;tag=as3c3695d2
Call-ID: b83eb80b195f6802c283c5927fb6415a@localHost
CSeq: 1 REGISTER
Server: Asterisk PBX 10.5.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest algorithm=MD5,realm="xyz.com",nonce="65dfb3ad"
Content-Length: 0
REGISTER sip:SipIP SIP/2.0
Call-ID: b83eb80b195f6802c283c5927fb6415a@localHost
CSeq: 2 REGISTER
From: <sip:username@SipIP>;tag=1626086046
To: <sip:username@SipIP>
Via: SIP/2.0/UDP localHost:52216;rport;branch=z9hG4bK-363935-150b07c1e9409d05aafaa7652859024a
Max-Forwards: 70
Contact: <sip:username@localHost:52216;transport=UDP>
Authorization: Digest username="username",realm="xyz.com",nonce="65dfb3ad",uri="sip:SipIP",algorithm=MD5,response="b34005eb8ded9180fb5f5667f1ee842d"
Content-Length: 0
---------------------------Registered200--------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP localHost:52216;branch=z9hG4bK-363935- 150b07c1e9409d05aafaa7652859024a;received=localHost;rport=52216
From: <sip:username@SipIP>;tag=1626086046
To: <sip:username@SipIP>;tag=as3c3695d2
Call-ID: b83eb80b195f6802c283c5927fb6415a@localHost
CSeq: 2 REGISTER
Server: Asterisk PBX 10.5.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Expires: 240
Contact: <sip:username@localHost:52216;transport=UDP>;expires=240
Date: Wed, 22 Apr 2015 23:52:44 GMT
Content-Length: 0
Dialog created: gov.nist.javax.sip.stack.SIPDialog@d15ad713
Dialog: gov.nist.javax.sip.stack.SIPDialog@d15ad713
INVITE sip:SipIP SIP/2.0
Call-ID: b83eb80b195f6802c283c5927fb6415a@localHost
CSeq: 3 INVITE
From: <sip:username@SipIP>;tag=1626086046
To: <sip:160@SipIP>
Via: SIP/2.0/UDP localHost:52216;rport;branch=z9hG4bK-363935-3d3a34f99499b96c6c1f709065ed4c85
Max-Forwards: 70
Content-Type: application/sdp
Contact: <sip:username@localHost:52216;transport=UDP>
Content-Length: 300
v=0
o=fraunhofer 392867480 292042336 IN IP4 localHost
s=-
c=IN IP4 localHost
t=0 0
m=audio 8000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
------------------Invite 401--------------------
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP localHost:52216;branch=z9hG4bK-363935- 3d3a34f99499b96c6c1f709065ed4c85;received=localHost;rport=52216
From: <sip:username@SipIP>;tag=1626086046
To: <sip:160@SipIP>;tag=as3c3695d2
Call-ID: b83eb80b195f6802c283c5927fb6415a@localHost
CSeq: 3 INVITE
Server: Asterisk PBX 10.5.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest algorithm=MD5,realm="xyz.com",nonce="4c1d89c8"
Content-Length: 0
-------------------------Dialog:gov.nist.javax.sip.stack.SIPDialog@d15ad713
Dialog get Remote Target: null
javax.sip.SipException: Cannot create ACK - no remote Target!
at gov.nist.javax.sip.stack.SIPDialog.createAck(SIPDialog.java:3021)
at test.ack(test.java:507)
at test.processResponse(test.java:214)
at gov.nist.javax.sip.EventScanner.deliverEvent(EventScanner.java:296)
at gov.nist.javax.sip.EventScanner.run(EventScanner.java:519)
at java.lang.Thread.run(Thread.java:745)
具有更改的呼叫ID的新响应
REGISTER sip:SIPIP SIP/2.0
Call-ID: 01bb8e9788fc251f8d44f2c709542fa5@LOCALIP
CSeq: 1 REGISTER
From: <sip:username@SIPIP>;tag=2125326583
To: <sip:username@SIPIP>
Via: SIP/2.0/UDP LOCALIP:52216;rport;branch=z9hG4bK-3833-142aff6ad1e359c2b618eba23fa04453
Max-Forwards: 70
Contact: <sip:username@LOCALIP:52216;transport=UDP>
Content-Length: 0
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP LOCALIP:52216;branch=z9hG4bK-3833-142aff6ad1e359c2b618eba23fa04453;received=LOCALIP;rport=52216
From: <sip:username@SIPIP>;tag=2125326583
To: <sip:username@SIPIP>;tag=as2e8e5e8d
Call-ID: 01bb8e9788fc251f8d44f2c709542fa5@LOCALIP
CSeq: 1 REGISTER
Server: Asterisk PBX 10.5.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest algorithm=MD5,realm="xyz.com",nonce="6223df47"
Content-Length: 0
REGISTER sip:SIPIP SIP/2.0
Call-ID: 01bb8e9788fc251f8d44f2c709542fa5@LOCALIP
CSeq: 2 REGISTER
From: <sip:username@SIPIP>;tag=2125326583
To: <sip:username@SIPIP>
Via: SIP/2.0/UDP LOCALIP:52216;rport;branch=z9hG4bK-3833-36d48a4c062bbb9e83db9a12f36414b3
Max-Forwards: 70
Contact: <sip:username@LOCALIP:52216;transport=UDP>
Authorization: Digest username="username",realm="xyz.com",nonce="6223df47",uri="sip:SIPIP",algorithm=MD5,response="d4ac55bbc8dacb87f66cf9f4041af03c"
Content-Length: 0
---------------------------Registered200--------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP LOCALIP:52216;branch=z9hG4bK-3833-36d48a4c062bbb9e83db9a12f36414b3;received=LOCALIP;rport=52216
From: <sip:username@SIPIP>;tag=2125326583
To: <sip:username@SIPIP>;tag=as2e8e5e8d
Call-ID: 01bb8e9788fc251f8d44f2c709542fa5@LOCALIP
CSeq: 2 REGISTER
Server: Asterisk PBX 10.5.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Expires: 240
Contact: <sip:username@LOCALIP:52216;transport=UDP>;expires=240
Date: Thu, 23 Apr 2015 08:38:28 GMT
Content-Length: 0
Dialog: createdgov.nist.javax.sip.stack.SIPDialog@977c31a5
INVITE sip:SIPIP SIP/2.0
Call-ID: a86cc90138ad50d9716664b7925cb205@LOCALIP
CSeq: 3 INVITE
From: <sip:username@SIPIP>;tag=2125326583
To: <sip:160@SIPIP>
Via: SIP/2.0/UDP LOCALIP:52216;rport;branch=z9hG4bK- 3833-0db5ef4b4e463aee4fa22c3379915d5e
Max-Forwards: 70
Content-Type: application/sdp
Contact: <sip:username@LOCALIP:52216;transport=UDP>
Content-Length: 300
v=0
o=fraunhofer 392867480 292042336 IN IP4 LOCALIP
s=-
c=IN IP4 LOCALIP
t=0 0
m=audio 8000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
------------------Invite 401--------------------
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP LOCALIP:52216;branch=z9hG4bK- 3833-0db5ef4b4e463aee4fa22c3379915d5e;received=LOCALIP;rport=52216
From: <sip:username@SIPIP>;tag=2125326583
To: <sip:160@SIPIP>;tag=as54bd3315
Call-ID: a86cc90138ad50d9716664b7925cb205@LOCALIP
CSeq: 3 INVITE
Server: Asterisk PBX 10.5.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest algorithm=MD5,realm="xyz.com",nonce="1e960014"
Content-Length: 0
通话（响应）
enterpublic void call(Response response) {
try {
cseq++;
String callee = "160";
current_process = cseq + "INVITE";
ArrayList viaHeaders = new ArrayList();
ViaHeader viaHeader = headerFactory.createViaHeader(localIP,
rport, "udp", null);
viaHeader.setRPort();
viaHeaders.add(viaHeader);
// The "Max-Forwards" header.
MaxForwardsHeader maxForwardsHeader = headerFactory.createMaxForwardsHeader(70);
// The "Call-Id" header.
CallIdHeader callIdHeader = this.sipProvider.getNewCallId();;
// The "CSeq" header.
CSeqHeader cSeqHeader = headerFactory.createCSeqHeader(cseq, "INVITE");
Address fromAddress = addressFactory.createAddress("sip:"
+ username + '@' + server);
Address toAddress = addressFactory.createAddress("sip:"+callee+'@'+sipIP);
FromHeader fromHeader = headerFactory.createFromHeader(
fromAddress, String.valueOf(this.tag));
// The "To" header.
ToHeader toHeader = headerFactory.createToHeader(toAddress,
null);
ContentLengthHeader contentLength = headerFactory.createContentLengthHeader(211);
ContentTypeHeader contentType = headerFactory.createContentTypeHeader("application", "sdp");
String sdpData = "v=0\n" +
"o=user1 795808818 480847547 IN IP4 10.99.70.106\n" +
"s=-\n" +
"c=IN IP4 10.99.70.106\n" +
"t=0 0\n" +
"m=audio 8000 RTP/AVP 0 8 101\n" +
"a=rtpmap:0 PCMU/8000\n" +
"a=rtpmap:8 PCMA/8000\n" +
"a=rtpmap:101 telephone-event/8000\n" +
"a=sendrecv";
byte[] contents = sdpData.getBytes();
this.contactHeader = this.headerFactory
.createContactHeader(contactAddress);
URI requestURI = addressFactory.createURI("sip:"
+callee+ '@'+ server);
request = this.messageFactory.createRequest(requestURI, Request.INVITE,
callIdHeader,cSeqHeader, fromHeader, toHeader, viaHeaders, maxForwardsHeader, contentType, contents);
request.addHeader(contactHeader);
request.addHeader(contentLength);
test listener =this;
if (response != null) {
boolean retry = true;
System.out.println("DEBUG: Response: "+response);
}
listener.inviteTid = sipProvider.getNewClientTransaction(request);
if(dialog!= null && logger.isDebugEnabled()){
logger.debug("Obtain dialog from ClientTransaction: automatic dialog support on");
// System.out.println("Obtain dialog from ClientTransaction: automatic dialog support on");
}
if(dialog == null){
//Automatic Dialog support turned off
dialog = sipProvider.getNewDialog(inviteTid);
}
System.out.println("Dialog: created" + dialog);
// send the request out.
listener.inviteTid.sendRequest();
this.dialog = this.inviteTid.getDialog();
// System.out.println("Dialog:" + dialog);
// Send the request statelessly through the SIP provider.
// this.sipProvider.sendRequest(request);
System.out.println(request.toString());
// Display the message in the text area.
logger.debug("Request sent:\n" + "\n\n");
} catch (Exception e) {
// If an error occurred, display the error.
e.printStackTrace();
logger.debug("Request sent failed: " + e.getMessage() + "\n");
}
}
无法创建ACK的日志-没有远程目标
enter REGISTER sip:localIP SIP/2.0
Call-ID: [email protected]
CSeq: 1 REGISTER
From: <sip:username@localIP>;tag=1825098223
To: <sip:username@localIP>
Via: SIP/2.0/UDP 10.99.136.136:52216;rport;branch=z9hG4bK-363735-0dd0a58b853bb23070f706fc3b058461
Max-Forwards: 70
Contact: <sip:[email protected]:52216;transport=UDP>
Content-Length: 0
-----------StatusCode:401----------------------------
-----------currentREsponse:Unauthorized----------------------------
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.99.136.136:52216;branch=z9hG4bK- 363735-0dd0a58b853bb23070f706fc3b058461;received=10.99.136.136;rport=52216
From: <sip:username@localIP>;tag=1825098223
To: <sip:username@localIP>;tag=as78941717
Call-ID: [email protected]
CSeq: 1 REGISTER
Server: Asterisk PBX 10.5.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest algorithm=MD5,realm="xyz.com",nonce="49a39764"
Content-Length: 0
REGISTER sip:localIP SIP/2.0
Call-ID: [email protected]
CSeq: 2 REGISTER
From: <sip:username@localIP>;tag=1825098223
To: <sip:username@localIP>
Via: SIP/2.0/UDP 10.99.136.136:52216;rport;branch=z9hG4bK-363735-0a214e6d4a1bc50c0faf1657cf109e31
Max-Forwards: 70
Contact: <sip:[email protected]:52216;transport=UDP>
Authorization: Digest username="username",realm="xyz.com",nonce="49a39764",uri="sip:localIP",algorithm=MD5,response="1af91bd8169339ec8c77decfab59fd21"
Content-Length: 0
-----------StatusCode:200----------------------------
-----------currentREsponse:OK----------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.99.136.136:52216;branch=z9hG4bK- 363735-0a214e6d4a1bc50c0faf1657cf109e31;received=10.99.136.136;rport=52216
From: <sip:username@localIP>;tag=1825098223
To: <sip:username@localIP>;tag=as78941717
Call-ID: [email protected]
CSeq: 2 REGISTER
Server: Asterisk PBX 10.5.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Expires: 240
Contact: <sip:[email protected]:52216;transport=UDP>;expires=240
Date: Fri, 24 Apr 2015 16:58:50 GMT
Content-Length: 0
---------------------------Registered: 200--------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.99.136.136:52216;branch=z9hG4bK-363735-0a214e6d4a1bc50c0faf1657cf109e31;received=10.99.136.136;rport=52216
From: <sip:username@localIP>;tag=1825098223
To: <sip:username@localIP>;tag=as78941717
Call-ID: [email protected]
CSeq: 2 REGISTER
Server: Asterisk PBX 10.5.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Expires: 240
Contact: <sip:[email protected]:52216;transport=UDP>;expires=240
Date: Fri, 24 Apr 2015 16:58:50 GMT
Content-Length: 0
DEBUG: Response: SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.99.136.136:52216;branch=z9hG4bK- 363735-0a214e6d4a1bc50c0faf1657cf109e31;received=10.99.136.136;rport=52216
From: <sip:username@localIP>;tag=1825098223
To: <sip:username@localIP>;tag=as78941717
Call-ID: [email protected]
CSeq: 2 REGISTER
Server: Asterisk PBX 10.5.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Expires: 240
Contact: <sip:[email protected]:52216;transport=UDP>;expires=240
Date: Fri, 24 Apr 2015 16:58:50 GMT
Content-Length: 0
Dialog: createdgov.nist.javax.sip.stack.SIPDialog@393895ed
INVITE sip:160@localIP SIP/2.0
Call-ID: [email protected]
CSeq: 3 INVITE
From: <sip:username@localIP>;tag=1825098223
To: <sip:160@localIP>
Via: SIP/2.0/UDP 10.99.136.136:52216;rport;branch=z9hG4bK-363735-312eacb8db309d6c6794e2eb7adf8b92
Max-Forwards: 70
Content-Type: application/sdp
Contact: <sip:[email protected]:52216;transport=UDP>
Authorization: Digest username="username",realm="xyz.com",nonce="49a39764",uri="sip:localIP",algorithm=MD5,response="1af91bd8169339ec8c77decfab59fd21"
Content-Length: 211
v=0
o=user1 795808818 480847547 IN IP4 localIP
s=-
c=IN IP4 localIP
t=0 0
m=audio 8000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
Request OPTIONSreceived at stackwith server transaction idnull
-----------StatusCode:401----------------------------
-----------currentREsponse:Unauthorized----------------------------
------------------Invite 401--------------------
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.99.136.136:52216;branch=z9hG4bK- 363735-312eacb8db309d6c6794e2eb7adf8b92;received=10.99.136.136;rport=52216
From: <sip:username@localIP>;tag=1825098223
To: <sip:160@localIP>;tag=as4f6f7f78
Call-ID: [email protected]
CSeq: 3 INVITE
Server: Asterisk PBX 10.5.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest algorithm=MD5,realm="xyz.com",nonce="7e048345"
Content-Length: 0
DEBUG: Response: SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.99.136.136:52216;branch=z9hG4bK-363735-312eacb8db309d6c6794e2eb7adf8b92;received=10.99.136.136;rport=52216
From: <sip:username@localIP>;tag=1825098223
To: <sip:160@localIP>;tag=as4f6f7f78
Call-ID: [email protected]
CSeq: 3 INVITE
Server: Asterisk PBX 10.5.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest algorithm=MD5,realm="xyz.com",nonce="7e048345"
Content-Length: 0
Dialog: createdgov.nist.javax.sip.stack.SIPDialog@393895ed
INVITE sip:160@localIP SIP/2.0
Call-ID: [email protected]
CSeq: 4 INVITE
From: <sip:username@localIP>;tag=1825098223
To: <sip:160@localIP>
Via: SIP/2.0/UDP 10.99.136.136:52216;rport;branch=z9hG4bK-363735-dae4419c9405bb40dda573fe9c276518
Max-Forwards: 70
Content-Type: application/sdp
Contact: <sip:[email protected]:52216;transport=UDP>
Authorization: Digest username="username",realm="xyz.com",nonce="49a39764",uri="sip:localIP",algorithm=MD5,response="1af91bd8169339ec8c77decfab59fd21"
Content-Length: 211
v=0
o=user1 795808818 480847547 IN IP4 localIP
s=-
c=IN IP4 localIP
t=0 0
m=audio 8000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
-----------StatusCode:401----------------------------
-----------currentREsponse:Unauthorized----------------------------
------------------Invite 401--------------------
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.99.136.136:52216;branch=z9hG4bK-363735-1a2fd8de5c389de950ecbb1a3d9a7d3e;received=10.99.136.136;rport=52216
From: <sip:username@localIP>;tag=1825098223
To: <sip:160@localIP>;tag=as22472833
Call-ID: [email protected]
CSeq: 4 INVITE
Server: Asterisk PBX 10.5.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest algorithm=MD5,realm="xyz.com",nonce="599cd63b"
Content-Length: 0
DEBUG: Response: SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.99.136.136:52216;branch=z9hG4bK-363735-1a2fd8de5c389de950ecbb1a3d9a7d3e;received=10.99.136.136;rport=52216
From: <sip:username@localIP>;tag=1825098223
To: <sip:160@localIP>;tag=as22472833
Call-ID: [email protected]
CSeq: 4 INVITE
Server: Asterisk PBX 10.5.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest algorithm=MD5,realm="xyz.com",nonce="599cd63b"
Content-Length: 0
Dialog: createdgov.nist.javax.sip.stack.SIPDialog@393895ed
INVITE sip:160@localIP SIP/2.0
Call-ID: [email protected]
CSeq: 5 INVITE
From: <sip:username@localIP>;tag=1825098223
To: <sip:160@localIP>
Via: SIP/2.0/UDP 10.99.136.136:52216;rport;branch=z9hG4bK-363735-633d808b788771fa1e211672ddc3e903
Max-Forwards: 70
Content-Type: application/sdp
Contact: <sip:[email protected]:52216;transport=UDP>
Authorization: Digest username="username",realm="xyz.com",nonce="49a39764",uri="sip:localIP",algorithm=MD5,response="1af91bd8169339ec8c77decfab59fd21"
Content-Length: 211
v=0
o=user1 795808818 480847547 IN IP4 localIP
s=-
c=IN IP4 localIP
t=0 0
m=audio 8000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
-----------StatusCode:401----------------------------
-----------currentREsponse:Unauthorized----------------------------
-----------StatusCode:100----------------------------
-----------currentREsponse:Trying----------------------------
------------------- Status Code: 100--------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.99.136.136:52216;branch=z9hG4bK-363735-a872936256c40065a4b147d4981cc74a;received=10.99.136.136;rport=52216
From: <sip:username@localIP>;tag=1825098223
To: <sip:160@localIP>
Call-ID: [email protected]
CSeq: 5 INVITE
Server: Asterisk PBX 10.5.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Contact: <sip:160@localIP:5060>
Content-Length: 0
-----------StatusCode:401----------------------------
-----------currentREsponse:Unauthorized----------------------------
Request OPTIONSreceived at stackwith server transaction idnull
-----------StatusCode:180----------------------------
-----------currentREsponse:Ringing----------------------------
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.99.136.136:52216;branch=z9hG4bK-363735-a872936256c40065a4b147d4981cc74a;received=10.99.136.136;rport=52216
From: <sip:username@localIP>;tag=1825098223
To: <sip:160@localIP>;tag=as78f593d4
Call-ID: [email protected]
CSeq: 5 INVITE
Server: Asterisk PBX 10.5.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Contact: <sip:160@localIP:5060>
Content-Length: 0
-
Request OPTIONSreceived at stackwith server transaction idnull
Request OPTIONSreceived at stackwith server transaction idnull
Request OPTIONSreceived at stackwith server transaction idnull
-----------StatusCode:200----------------------------
-----------currentREsponse:OK----------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.99.136.136:52216;branch=z9hG4bK-363735-a872936256c40065a4b147d4981cc74a;received=10.99.136.136;rport=52216
From: <sip:username@localIP>;tag=1825098223
To: <sip:160@localIP>;tag=as78f593d4
Call-ID: [email protected]
CSeq: 5 INVITE
Server: Asterisk PBX 10.5.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Contact: <sip:160@localIP:5060>
Content-Type: application/sdp
Content-Length: 256
v=0
o=root 698934329 698934329 IN IP4 localIP
s=Asterisk PBX 10.5.1
c=IN IP4 localIP
t=0 0
m=audio 23142 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
javax.sip.SipException: Cannot create ACK - no remote Target!
at gov.nist.javax.sip.stack.SIPDialog.createAck(SIPDialog.java:3021)
at test.processResponse(test.java:323)
at gov.nist.javax.sip.EventScanner.deliverEvent(EventScanner.java:296)
at gov.nist.javax.sip.EventScanner.run(EventScanner.java:519)
at java.lang.Thread.run(Thread.java:745)
-----------StatusCode:200----------------------------
-----------currentREsponse:OK----------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.99.136.136:52216;branch=z9hG4bK-363735-a872936256c40065a4b147d4981cc74a;received=10.99.136.136;rport=52216
From: <sip:username@localIP>;tag=1825098223
To: <sip:160@localIP>;tag=as78f593d4
Call-ID: [email protected]
CSeq: 5 INVITE
Server: Asterisk PBX 10.5.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Contact: <sip:160@localIP:5060>
Content-Type: application/sdp
Content-Length: 256
v=0
o=root 698934329 698934329 IN IP4 localIP
s=Asterisk PBX 10.5.1
c=IN IP4 localIP
t=0 0
m=audio 23142 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
javax.sip.SipException: Cannot create ACK - no remote Target!
at gov.nist.javax.sip.stack.SIPDialog.createAck(SIPDialog.java:3021)
at test.processResponse(test.java:323)
at gov.nist.javax.sip.EventScanner.deliverEvent(EventScanner.java:296)
at gov.nist.javax.sip.EventScanner.run(EventScanner.java:519)
at java.lang.Thread.run(Thread.java:745)
-----------StatusCode:200----------------------------
-----------currentREsponse:OK----------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.99.136.136:52216;branch=z9hG4bK-363735-a872936256c40065a4b147d4981cc74a;received=10.99.136.136;rport=52216
From: <sip:username@localIP>;tag=1825098223
To: <sip:160@localIP>;tag=as78f593d4
Call-ID: [email protected]
CSeq: 5 INVITE
Server: Asterisk PBX 10.5.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Contact: <sip:160@localIP:5060>
Content-Type: application/sdp
Content-Length: 256
v=0
o=root 698934329 698934329 IN IP4 localIP
s=Asterisk PBX 10.5.1
c=IN IP4 localIP
t=0 0
m=audio 23142 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
javax.sip.SipException: Cannot create ACK - no remote Target!
at gov.nist.javax.sip.stack.SIPDialog.createAck(SIPDialog.java:3021)
at test.processResponse(test.java:323)
at gov.nist.javax.sip.EventScanner.deliverEvent(EventScanner.java:296)
at gov.nist.javax.sip.EventScanner.run(EventScanner.java:519)
at java.lang.Thread.run(Thread.java:745)
Request OPTIONSreceived at stackwith server transaction idnull
-----------StatusCode:200----------------------------
-----------currentREsponse:OK----------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.99.136.136:52216;branch=z9hG4bK-363735-a872936256c40065a4b147d4981cc74a;received=10.99.136.136;rport=52216
From: <sip:username@localIP>;tag=1825098223
To: <sip:160@localIP>;tag=as78f593d4
Call-ID: [email protected]
CSeq: 5 INVITE
Server: Asterisk PBX 10.5.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Contact: <sip:160@localIP:5060>
Content-Type: application/sdp
Content-Length: 256
v=0
o=root 698934329 698934329 IN IP4 localIP
s=Asterisk PBX 10.5.1
c=IN IP4 localIP
t=0 0
m=audio 23142 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
javax.sip.SipException: Cannot create ACK - no remote Target!
at gov.nist.javax.sip.stack.SIPDialog.createAck(SIPDialog.java:3021)
at test.processResponse(test.java:323)
at gov.nist.javax.sip.EventScanner.deliverEvent(EventScanner.java:296)
at gov.nist.javax.sip.EventScanner.run(EventScanner.java:519)
at java.lang.Thread.run(Thread.java:745)
Request OPTIONSreceived at stackwith server transaction idnull
Request BYEreceived at stackwith server transaction idnull
----------------Received a BYE-----------------------
null TID
Request BYEreceived at stackwith server transaction idnull
----------------Received a BYE-----------------------
null TID
Request BYEreceived at stackwith server transaction idnull
----------------Received a BYE-----------------------
null TID
最佳答案
让我们尝试更详细一些。
我要做的是获取JAIN SIP 1.2下载的样本UAC代码，该软件位于JAIN SIP's Jenkins，并对其进行修改以涵盖REGISTER和初始INVITE。
我已经使用IMS核心进行了测试，因此它可能与您使用Asterisk PBX的体验略有不同，特别是我不必对INVITE进行身份验证。
在高层结构方面，我有：
一个单一的类（TestUAC）既是主要类，又实现了SipListener接口。
在主要方面，我初始化堆栈，设置侦听器，读取任何配置，最后我调用sendRegister()，它将发送第一个REGISTER并启动该过程。
在processResponse()回调上。根据响应的CSeq方法，我可以区分以下两者：
对注册的响应（processRegisterResponse()）
对INVITE（processInviteResponse()）的响应
对BYE的响应（processByeResponse()）
在processRequest()和其他回调中，我只有踪迹。
通常，我很少将信息作为对象属性：
堆栈内容：对sipFactory，sipProvider，sipStack，addressFactory，messageFactory，headerFactory的引用
有关我的身份以及我想与谁交谈的信息：
例如
private String callingURI;
private String calledURI;
private String username;
private String password;
private Address fromNameAddress;
private ContactHeader contactHeader;
另外，我为cseq和（遵循JAIN SIP示例）保留了一个递增计数器，并使用了ackRequest的副本来回复重新传输的200 OK。
// Save the created ACK request, to respond to retransmitted 2xx
private Request ackRequest;
private long cseq=1L;
如我所说，
processResponse()只是基于CSeq方法进行分配：public void processResponse(ResponseEvent responseReceivedEvent) {
System.out.println("Got a response");
Response response = (Response) responseReceivedEvent.getResponse();
CSeqHeader cseq = (CSeqHeader) response.getHeader(CSeqHeader.NAME);
System.out.println("Response received : Status Code = "
+ response.getStatusCode() + " " + cseq);
try {
if(cseq.getMethod().equals(Request.REGISTER)) {
processRegisterResponse(responseReceivedEvent);
}
else if(cseq.getMethod().equals(Request.INVITE)) {
processInviteResponse(responseReceivedEvent);
}
else if(cseq.getMethod().equals(Request.BYE)) {
processByeResponse(responseReceivedEvent);
}
else {
System.out.println("Response to unexpected request");
}
} catch(Exception e) {
e.printStackTrace();
}
}
processRegisterResponse()区分401未经授权和200 OK响应。在第一种情况下，它将触发发送带有身份验证的REGISTER。在第二种情况下，它将请求发送INVITE：private void processRegisterResponse(ResponseEvent responseReceivedEvent) throws TransactionUnavailableException, ParseException, InvalidArgumentException, SipException, NoSuchAlgorithmException {
Response response = (Response) responseReceivedEvent.getResponse();
if(response.getStatusCode() == Response.UNAUTHORIZED) {
sendRegister(response);
}
else if (response.getStatusCode() == Response.OK) {
contactHeader=(ContactHeader)response.getHeader(ContactHeader.NAME);
sendInvite();
}
}
现在，对于sendInvite（）（我将尝试将其提炼到相关部分）。
private void sendInvite() throws ParseException, InvalidArgumentException, TransactionUnavailableException, SipException {
// create To Header
URI toAddress = addressFactory.createURI(calledURI);
Address toNameAddress = addressFactory.createAddress(toAddress);
ToHeader toHeader = headerFactory.createToHeader(toNameAddress,null);
FromHeader fromHeader = headerFactory.createFromHeader(fromNameAddress, "12345");
// Create ViaHeaders
ArrayList<ViaHeader> viaHeaders = getViaHeaders();
// Create a new CallId header
CallIdHeader callIdHeader = sipProvider.getNewCallId();
// Create a new Cseq header
CSeqHeader cSeqHeader = headerFactory.createCSeqHeader(cseq,Request.INVITE);
cseq++;
// Create a new MaxForwardsHeader
MaxForwardsHeader maxForwards = headerFactory.createMaxForwardsHeader(70);
// Create the request.
Request request = messageFactory.createRequest(toAddress,
Request.INVITE, callIdHeader, cSeqHeader, fromHeader,
toHeader, viaHeaders, maxForwards);
request.addHeader(contactHeader);
// at this point you should add the rest of the headers, content, etc.
// Create the client transaction.
ClientTransaction currentTid = sipProvider.getNewClientTransaction(request);
// send the request out.
currentTid.sendRequest();
}
最后，接下来会发生的事情是我们将收到INVITE的200 OK。这大部分是来自JAIN SLEE示例的代码，我已经将其移至
processInviteResponse()（同样，我将尝试将其提炼为基本代码）。private void processInviteResponse(ResponseEvent responseReceivedEvent) throws SipException, InvalidArgumentException {
Response response = (Response) responseReceivedEvent.getResponse();
ClientTransaction tid = responseReceivedEvent.getClientTransaction();
CSeqHeader cseq = (CSeqHeader) response.getHeader(CSeqHeader.NAME);
Dialog dialog = responseReceivedEvent.getDialog();
if (tid == null) {
// RFC3261: MUST respond to every 2xx
if (ackRequest!=null && dialog!=null) {
System.out.println("re-sending ACK");
dialog.sendAck(ackRequest);
}
return;
}
if (response.getStatusCode() == Response.OK) {
System.out.println("Dialog after 200 OK " + dialog);
System.out.println("Dialog State after 200 OK " + dialog.getState());
ackRequest = dialog.createAck(cseq.getSeqNumber() );
System.out.println("Sending ACK");
dialog.sendAck(ackRequest);
}
}
收到200 OK并发送ACK后，您可能应该考虑下一步该怎么做：例如，在JAIN SLEE UAC示例中，它会启动
TimerTask以在一段时间后发送BYE。您还可以在那里添加其他错误（或临时）响应的处理。但是请记住，堆栈将自动确认（无需代码）最终的错误响应。