实施取消请求
Implement cancelRequest
我正在为正在进行的通话实施取消请求。会话建立后，我尝试发送请求。取消功能在我第一次实现它时有效，但现在它说交易存在。我该如何解决？
else if(currentREsponse.equals("Session Progress")){
System.out.println("------------------- Status Code: "+ statusCode+ "--------------------------");
System.out.println(responseEvent.getResponse());
try {
Request cancelRequest;
cancelRequest = inviteTid.createCancel();
ClientTransaction canceltid = sipProvider.getNewClientTransaction(cancelRequest);
canceltid.sendRequest();
System.out.println("Cancel Requst:" + cancelRequest);
} catch (SipException e) {
// TODO Auto-generated catch block
e.printStackTrace();
}
}
回复：
-----------StatusCode:100----------------------------
-----------currentREsponse:Trying----------------------------
------------------- Status Code: 100--------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP localIP:52216;branch=z9hG4bK-353435-03dc537ef691bb00e1e5512b034ea431;received=localIP;rport=52216
From: <sip:username@sipIP>;tag=-2016245219
To: <sip:86940160@sipIP>
Call-ID: fab08b0be8c74b236c9eced419dec3ae@localIP
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:86940160@sipIP:5060>
Content-Length: 0
------------------- Phone is RINGING hangup --------------------------
Request OPTIONSreceived at stackwith server transaction idnull
-----------StatusCode:183----------------------------
-----------currentREsponse:Session Progress----------------------------
------------------- Status Code: 183--------------------------
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP localIP:52216;branch=z9hG4bK-353435-03dc537ef691bb00e1e5512b034ea431;received=localIP;rport=52216
From: <sip:username@sipIP>;tag=-2016245219
To: <sip:86940160@sipIP>;tag=as06d748b5
Call-ID: fab08b0be8c74b236c9eced419dec3ae@localIP
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:86940160@sipIP:5060>
Content-Type: application/sdp
Content-Length: 254
v=0
o=root 16865513 16865513 IN IP4 sipIP
s=Asterisk PBX 10.5.1
c=IN IP4 sipIP
t=0 0
m=audio 16188 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
Cancel Requst:CANCEL sip:86940160@sipIP SIP/2.0
Call-ID: fab08b0be8c74b236c9eced419dec3ae@localIP
To: <sip:86940160@sipIP>
CSeq: 5 CANCEL
From: <sip:username@sipIP>;tag=-2016245219
Via: SIP/2.0/UDP localIP:52216;rport;branch=z9hG4bK-353435-32ce00c8a1c2efc96caa44a14bf3b910
Max-Forwards: 70
Content-Length: 0
Request OPTIONSreceived at stackwith server transaction idnull
-----------StatusCode:401----------------------------
-----------currentREsponse:Unauthorized----------------------------
---------------UNAUTHORIZED--------SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP localIP:52216;branch=z9hG4bK-353435-32ce00c8a1c2efc96caa44a14bf3b910;received=localIP;rport=52216
From: <sip:username@sipIP>;tag=-2016245219
To: <sip:86940160@sipIP>;tag=as647be936
Call-ID: fab08b0be8c74b236c9eced419dec3ae@localIP
CSeq: 5 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="69e1f4c2"
Content-Length: 0
-----------StatusCode:481----------------------------
-----------currentREsponse:Call/Transaction Does Not Exist----------------------------
-------------------Status Code: 481 ----------------------
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.99.136.131:52216;branch=z9hG4bK-353435-32ce00c8a1c2efc96caa44a14bf3b910;received=10.99.136.131;rport=52216
From: <sip:username@sipIP>;tag=-2016245219
To: <sip:86940160@sipIP>;tag=as647be936
Call-ID: fab08b0be8c74b236c9eced419dec3ae@10.99.136.131
CSeq: 5 CANCEL
Server: Asterisk PBX 10.5.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Content-Length: 0
Request OPTIONSreceived at stackwith server transaction idnull
处理响应
public void processResponse(ResponseEvent responseEvent) {
// TODO Auto-generated method stub
int statusCode = responseEvent.getResponse().getStatusCode();
String currentResponse = responseEvent.getResponse().getReasonPhrase();
// System.out.println("-----------StatusCode:"+ statusCode+"----------------------------");
// System.out.println("-----------currentREsponse:"+ currentResponse+"----------------------------");
CSeqHeader cseqtemp = (CSeqHeader) responseEvent.getResponse().getHeader(CSeqHeader.NAME);
// System.out.println(responseEvent.getResponse().toString());
System.out.println("................cseq" + cseqtemp);
if(cseqtemp.getMethod().equals(Request.REGISTER)){
processRegisterResponse(responseEvent);
}else if(cseqtemp.getMethod().equals(Request.INVITE)){
countInviteResponse ++;
System.out.println(".....countInviteResponse: "+ countInviteResponse+" ......" );
if(countInviteResponse == 6){
System.out.println(responseEvent.getResponse());
// functions.cancel(responseEvent.getResponse());
}
try {
processInviteResponse(responseEvent);
} catch (SipException | InvalidArgumentException e) {
// TODO Auto-generated catch block
e.printStackTrace();
}
}
else {
System.out.println("-------------------Status Code: "+ statusCode+" ----------------------");
System.out.println(responseEvent.getResponse().toString());
}
ProcessInvite 响应：
private void processInviteResponse(ResponseEvent responseReceivedEvent) throws SipException, InvalidArgumentException {
int statusCode = responseReceivedEvent.getResponse().getStatusCode();
String currentResponse = responseReceivedEvent.getResponse().getReasonPhrase();
Response response = responseReceivedEvent.getResponse();
ClientTransaction tid = responseReceivedEvent.getClientTransaction();
CSeqHeader cseqtemp = (CSeqHeader) response.getHeader(CSeqHeader.NAME);
if(statusCode> 400 && statusCode <410){
if((statusCode ==401 && cseqtemp.getMethod().equals(Request.INVITE))
|| response.getStatusCode() == Response.PROXY_AUTHENTICATION_REQUIRED){
System.out.println(responseReceivedEvent.getResponse().toString());
// if(functions.cseq<4){
System.out.println("------------------Invite 401--------------------\n");
AuthenticationHelper authenticationHelper =
((SipStackExt) functions.sipStack).getAuthenticationHelper(new AccountManagerImpl(), functions.headerFactory);
try {
functions.inviteTid = authenticationHelper.handleChallenge(response, functions.inviteTid, functions.sipProvider, 5);
functions.inviteTid.sendRequest();
System.out.println("/n" + functions.inviteTid.getRequest());
} catch (NullPointerException e) {
// TODO Auto-generated catch block
e.printStackTrace();
} catch (SipException e) {
// TODO Auto-generated catch block
e.printStackTrace();
}
// functions.call(responseReceivedEvent.getResponse());
// }
}
}else if(statusCode == 200){
try {
System.out.println(responseReceivedEvent.getResponse().toString());
response = (Response) responseReceivedEvent.getResponse();
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);
}
}
catch (InvalidArgumentException e) {
// TODO Auto-generated catch block
e.printStackTrace();
} catch (SipException e) {
// TODO Auto-generated catch block
e.printStackTrace();
}
}else if(responseReceivedEvent.getResponse().getReasonPhrase().equals("Trying")){
System.out.println("------------------- Status Code: "+ statusCode+ "--------------------------");
System.out.println(responseReceivedEvent.getResponse());
}else if(responseReceivedEvent.getResponse().getReasonPhrase().equals("Ringing")){
System.out.println("------------------- Status Code: "+ statusCode+ "--------------------------");
System.out.println(responseReceivedEvent.getResponse());
System.out.println("------------------- Phone is RINGING --------------------------");
}else if(responseReceivedEvent.getResponse().getReasonPhrase().equals("Session Progress")){
System.out.println("------------------- Status Code: "+ statusCode+ "--------------------------");
System.out.println(responseReceivedEvent.getResponse());
// functions.cancel(response);
Request cancelRequest= responseReceivedEvent.getClientTransaction().createCancel();
functions.sipProvider.sendRequest(cancelRequest);
System.out.println(cancelRequest.toString());
}else {
System.out.println("------------------- Status Code: "+ statusCode+ "--------------------------");
System.out.println(responseReceivedEvent.getResponse());
}
}
邀请：
public void call(Response response) {
try {
Request request;
cseq++;
String callee = "86940160";
// 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;
if(!retry){
callIdHeader = this.sipProvider.getNewCallId();
callIdHeaderinit = callIdHeader;
}else{
callIdHeader = callIdHeaderinit;
}
// The "CSeq" header.
CSeqHeader cSeqHeader = headerFactory.createCSeqHeader(cseq, Request.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);
// Create the contact address used for all SIP messages.
contactAddress = addressFactory.createAddress("sip:" + username + "@"+ localIP +":"+rport+ ";"+ "transport=UDP");
// Create the contact header used for all SIP messages.
contactHeader = headerFactory.createContactHeader(contactAddress);
ContentLengthHeader contentLength = headerFactory.createContentLengthHeader(211);
ContentTypeHeader contentType = headerFactory.createContentTypeHeader("application", "sdp");
String sdpData = "v=0\n" +
"o=user1 795808818 480847547 IN IP4 "+localIP+"\n" +
"s=-\n" +
"c=IN IP4 "+localIP+"\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 = messageFactory.createRequest(requestURI, Request.INVITE,
callIdHeader,cSeqHeader, fromHeader, toHeader, viaHeaders, maxForwardsHeader);
request.setContent(contents, contentType);
request.addHeader(contactHeader);
request.addHeader(contentLength);
if (response != null) {
AuthorizationHeader authHeader = null;
if(retry){
authHeader = getAuthorizationHeader();
retry = false;
}else{
authHeader = makeAuthHeader(headerFactory, response, request, username, password);
retry = true;
}
request.addHeader(authHeader);
// AuthenticationHelper authenticationHelper =
// ((SipStackExt) sipStack).getAuthenticationHelper(new AccountManagerImpl(), headerFactory);
// try {
// inviteTid = authenticationHelper.handleChallenge(response, tid, sipProvider, 5);
// inviteTid.sendRequest();
// } catch (NullPointerException e) {
// // TODO Auto-generated catch block
// e.printStackTrace();
// } catch (SipException e) {
// // TODO Auto-generated catch block
// e.printStackTrace();
// }
}
// ExpiresHeader expiresHeader = headerFactory.createExpiresHeader(1);
//
// request.addHeader(expiresHeader);
// test listener =this;
if (response != null) {
boolean retry = true;
// System.out.println("DEBUG: Response: "+response);
}
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.
inviteTid.sendRequest();
dialog = 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");
}
}
回复：
REGISTER sip:sipIP SIP/2.0
Call-ID: cdf4ef0fa1a145b1b07b632797f53fdb@localIP
CSeq: 1 REGISTER
From: <sip:username@sipIP>;tag=-1436906198
To: <sip:username@sipIP>
Via: SIP/2.0/UDP localIP:52216;rport;branch=z9hG4bK-3230-3d42d8c86f8aa3d6d0194a4846511464
Max-Forwards: 70
Contact: <sip:username@localIP:52216;transport=UDP>
Content-Length: 0
log4j:WARN No appenders could be found for logger (test).
log4j:WARN Please initialize the log4j system properly.
log4j:WARN See http://logging.apache.org/log4j/1.2/faq.html#noconfig for more info.
................cseqCSeq: 1 REGISTER
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP localIP:52216;branch=z9hG4bK-3230-3d42d8c86f8aa3d6d0194a4846511464;received=localIP;rport=52216
From: <sip:username@sipIP>;tag=-1436906198
To: <sip:username@sipIP>;tag=as24ab6bcf
Call-ID: cdf4ef0fa1a145b1b07b632797f53fdb@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="6ac7ff34"
Content-Length: 0
REGISTER sip:sipIP SIP/2.0
Call-ID: cdf4ef0fa1a145b1b07b632797f53fdb@localIP
CSeq: 2 REGISTER
From: <sip:username@sipIP>;tag=-1436906198
To: <sip:username@sipIP>
Via: SIP/2.0/UDP localIP:52216;rport;branch=z9hG4bK-3230-a94c094a3aacfc40d0455de777b0ae26
Max-Forwards: 70
Contact: <sip:username@localIP:52216;transport=UDP>
Authorization: Digest username="username",realm="xyz.com",nonce="6ac7ff34",uri="sip:sipIP",algorithm=MD5,response="53231e4a2cd26604d8d2bc867b0e43e9"
Content-Length: 0
................cseqCSeq: 2 REGISTER
---------------------------Registered: 200--------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP localIP:52216;branch=z9hG4bK-3230-a94c094a3aacfc40d0455de777b0ae26;received=localIP;rport=52216
From: <sip:username@sipIP>;tag=-1436906198
To: <sip:username@sipIP>;tag=as24ab6bcf
Call-ID: cdf4ef0fa1a145b1b07b632797f53fdb@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, 30 Apr 2015 14:46:46 GMT
Content-Length: 0
Dialog: createdgov.nist.javax.sip.stack.SIPDialog@79f65165
INVITE sip:86940160@sipIP SIP/2.0
Call-ID: cdf4ef0fa1a145b1b07b632797f53fdb@localIP
CSeq: 3 INVITE
From: <sip:username@sipIP>;tag=-1436906198
To: <sip:86940160@sipIP>
Via: SIP/2.0/UDP localIP:52216;rport;branch=z9hG4bK-3230-c5512f65af9890c0798f1e5757ef1c57
Max-Forwards: 70
Content-Type: application/sdp
Contact: <sip:username@localIP:52216;transport=UDP>
Authorization: Digest username="username",realm="xyz.com",nonce="6ac7ff34",uri="sip:sipIP",algorithm=MD5,response="53231e4a2cd26604d8d2bc867b0e43e9"
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
Request OPTIONSreceived at stackwith server transaction idnull
................cseqCSeq: 3 INVITE
.....countInviteResponse: 1 ......
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP localIP:52216;branch=z9hG4bK-3230-c5512f65af9890c0798f1e5757ef1c57;received=localIP;rport=52216
From: <sip:username@sipIP>;tag=-1436906198
To: <sip:86940160@sipIP>;tag=as43ee81b7
Call-ID: cdf4ef0fa1a145b1b07b632797f53fdb@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="6ad3bcf0"
Content-Length: 0
删除标签后的响应：
------------------Invite 401--------------------
/nINVITE sip:86940160@sipIP:5060;maddr=sipIP SIP/2.0
Call-ID: 7128d7fc7109b0c1293dce9f6b2b9c7b@localIP.131
CSeq: 4 INVITE
From: <sip:username@sipIP>;tag=-515507368
To: <sip:86940160@sipIP>
Via: SIP/2.0/UDP localIP.131:52216;rport;branch=z9hG4bK-313339-453b043ba57ac51c5293cf17f8823071
Max-Forwards: 70
Content-Type: application/sdp
Contact: <sip:username@localIP.131:52216;transport=UDP>
Authorization: Digest username="username",realm="xyz.com",nonce="3cdd5016",uri="sip:86940160@sipIP:5060;maddr=sipIP",response="38b50cad69f74ba89fd63f78a1968acc",algorithm=MD5
Content-Length: 211
v=0
o=user1 795808818 480847547 IN IP4 localIP.131
s=-
c=IN IP4 localIP.131
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
................cseqCSeq: 4 INVITE
.....countInviteResponse: 2 ......
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP localIP.131:52216;branch=z9hG4bK-313339-453b043ba57ac51c5293cf17f8823071;received=localIP.131;rport=52216
From: <sip:username@sipIP>;tag=-515507368
To: <sip:86940160@sipIP>;tag=as5f585fd4
Call-ID: 7128d7fc7109b0c1293dce9f6b2b9c7b@localIP.131
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="6aa75c9f"
Content-Length: 0
------------------Invite 401--------------------
/nINVITE sip:86940160@sipIP:5060;maddr=sipIP SIP/2.0
Call-ID: 7128d7fc7109b0c1293dce9f6b2b9c7b@localIP.131
CSeq: 5 INVITE
From: <sip:username@sipIP>;tag=-515507368
To: <sip:86940160@sipIP>
Via: SIP/2.0/UDP localIP.131:52216;rport;branch=z9hG4bK-313339-5ed3ab3afc3b791c0af19470ae58813d
Max-Forwards: 70
Content-Type: application/sdp
Contact: <sip:username@localIP.131:52216;transport=UDP>
Authorization: Digest username="username",realm="xyz.com",nonce="6aa75c9f",uri="sip:86940160@sipIP:5060;maddr=sipIP",response="092fc083a66f19b5b995c8cbc8d5a8bc",algorithm=MD5
Content-Length: 211
v=0
o=user1 795808818 480847547 IN IP4 localIP.131
s=-
c=IN IP4 localIP.131
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
................cseqCSeq: 5 INVITE
.....countInviteResponse: 3 ......
------------------- Status Code: 100--------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP localIP.131:52216;branch=z9hG4bK-313339-5ed3ab3afc3b791c0af19470ae58813d;received=localIP.131;rport=52216
From: <sip:username@sipIP>;tag=-515507368
To: <sip:86940160@sipIP>
Call-ID: 7128d7fc7109b0c1293dce9f6b2b9c7b@localIP.131
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:86940160@sipIP:5060>
Content-Length: 0
................cseqCSeq: 5 INVITE
.....countInviteResponse: 4 ......
------------------- Status Code: 183--------------------------
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP localIP.131:52216;branch=z9hG4bK-313339-5ed3ab3afc3b791c0af19470ae58813d;received=localIP.131;rport=52216
From: <sip:username@sipIP>;tag=-515507368
To: <sip:86940160@sipIP>;tag=as7fc2a07d
Call-ID: 7128d7fc7109b0c1293dce9f6b2b9c7b@localIP.131
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:86940160@sipIP:5060>
Content-Type: application/sdp
Content-Length: 256
v=0
o=root 163700690 163700690 IN IP4 sipIP
s=Asterisk PBX 10.5.1
c=IN IP4 sipIP
t=0 0
m=audio 25036 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
Invite ID:z9hG4bK-313339-5ed3ab3afc3b791c0af19470ae58813d
CANCEL sip:86940160@sipIP:5060;maddr=sipIP SIP/2.0
Call-ID: 7128d7fc7109b0c1293dce9f6b2b9c7b@localIP.131
To: <sip:86940160@sipIP>
CSeq: 5 CANCEL
From: <sip:username@sipIP>;tag=-515507368
Via: SIP/2.0/UDP localIP.131:52216;rport;branch=z9hG4bK-313339-5ed3ab3afc3b791c0af19470ae58813d
Max-Forwards: 70
Content-Length: 0
................cseqCSeq: 5 CANCEL
-------------------Status Code: 481 ----------------------
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP localIP.131:52216;branch=z9hG4bK-313339-5ed3ab3afc3b791c0af19470ae58813d;received=localIP.131;rport=52216
From: <sip:username@sipIP>;tag=-515507368
To: <sip:86940160@sipIP>;tag=as23aeabf5
Call-ID: 7128d7fc7109b0c1293dce9f6b2b9c7b@localIP.131
CSeq: 5 CANCEL
Server: Asterisk PBX 10.5.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Content-Length: 0
INVITE 交易 ID 与 CANCEL 交易 ID 不同。邀请
z9hG4bK-353435-03dc537ef691bb00e1e5512b034ea431
取消是
z9hG4bK-353435-32ce00c8a1c2efc96caa44a14bf3b910
它们应该是一样的。
取消的正确方法是使用原始邀请交易并使用提供商
cancelRequest = inviteClientTransaction.createCancel();
sipProvider.sendRequest(cancelRequest);
我添加了一个 5 秒的线程，现在可以使用了。防火墙是如何延迟交易的。
我正在为正在进行的通话实施取消请求。会话建立后，我尝试发送请求。取消功能在我第一次实现它时有效，但现在它说交易存在。我该如何解决？
else if(currentREsponse.equals("Session Progress")){
System.out.println("------------------- Status Code: "+ statusCode+ "--------------------------");
System.out.println(responseEvent.getResponse());
try {
Request cancelRequest;
cancelRequest = inviteTid.createCancel();
ClientTransaction canceltid = sipProvider.getNewClientTransaction(cancelRequest);
canceltid.sendRequest();
System.out.println("Cancel Requst:" + cancelRequest);
} catch (SipException e) {
// TODO Auto-generated catch block
e.printStackTrace();
}
}
回复：
-----------StatusCode:100----------------------------
-----------currentREsponse:Trying----------------------------
------------------- Status Code: 100--------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP localIP:52216;branch=z9hG4bK-353435-03dc537ef691bb00e1e5512b034ea431;received=localIP;rport=52216
From: <sip:username@sipIP>;tag=-2016245219
To: <sip:86940160@sipIP>
Call-ID: fab08b0be8c74b236c9eced419dec3ae@localIP
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:86940160@sipIP:5060>
Content-Length: 0
------------------- Phone is RINGING hangup --------------------------
Request OPTIONSreceived at stackwith server transaction idnull
-----------StatusCode:183----------------------------
-----------currentREsponse:Session Progress----------------------------
------------------- Status Code: 183--------------------------
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP localIP:52216;branch=z9hG4bK-353435-03dc537ef691bb00e1e5512b034ea431;received=localIP;rport=52216
From: <sip:username@sipIP>;tag=-2016245219
To: <sip:86940160@sipIP>;tag=as06d748b5
Call-ID: fab08b0be8c74b236c9eced419dec3ae@localIP
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:86940160@sipIP:5060>
Content-Type: application/sdp
Content-Length: 254
v=0
o=root 16865513 16865513 IN IP4 sipIP
s=Asterisk PBX 10.5.1
c=IN IP4 sipIP
t=0 0
m=audio 16188 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
Cancel Requst:CANCEL sip:86940160@sipIP SIP/2.0
Call-ID: fab08b0be8c74b236c9eced419dec3ae@localIP
To: <sip:86940160@sipIP>
CSeq: 5 CANCEL
From: <sip:username@sipIP>;tag=-2016245219
Via: SIP/2.0/UDP localIP:52216;rport;branch=z9hG4bK-353435-32ce00c8a1c2efc96caa44a14bf3b910
Max-Forwards: 70
Content-Length: 0
Request OPTIONSreceived at stackwith server transaction idnull
-----------StatusCode:401----------------------------
-----------currentREsponse:Unauthorized----------------------------
---------------UNAUTHORIZED--------SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP localIP:52216;branch=z9hG4bK-353435-32ce00c8a1c2efc96caa44a14bf3b910;received=localIP;rport=52216
From: <sip:username@sipIP>;tag=-2016245219
To: <sip:86940160@sipIP>;tag=as647be936
Call-ID: fab08b0be8c74b236c9eced419dec3ae@localIP
CSeq: 5 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="69e1f4c2"
Content-Length: 0
-----------StatusCode:481----------------------------
-----------currentREsponse:Call/Transaction Does Not Exist----------------------------
-------------------Status Code: 481 ----------------------
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.99.136.131:52216;branch=z9hG4bK-353435-32ce00c8a1c2efc96caa44a14bf3b910;received=10.99.136.131;rport=52216
From: <sip:username@sipIP>;tag=-2016245219
To: <sip:86940160@sipIP>;tag=as647be936
Call-ID: fab08b0be8c74b236c9eced419dec3ae@10.99.136.131
CSeq: 5 CANCEL
Server: Asterisk PBX 10.5.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Content-Length: 0
Request OPTIONSreceived at stackwith server transaction idnull
处理响应
public void processResponse(ResponseEvent responseEvent) {
// TODO Auto-generated method stub
int statusCode = responseEvent.getResponse().getStatusCode();
String currentResponse = responseEvent.getResponse().getReasonPhrase();
// System.out.println("-----------StatusCode:"+ statusCode+"----------------------------");
// System.out.println("-----------currentREsponse:"+ currentResponse+"----------------------------");
CSeqHeader cseqtemp = (CSeqHeader) responseEvent.getResponse().getHeader(CSeqHeader.NAME);
// System.out.println(responseEvent.getResponse().toString());
System.out.println("................cseq" + cseqtemp);
if(cseqtemp.getMethod().equals(Request.REGISTER)){
processRegisterResponse(responseEvent);
}else if(cseqtemp.getMethod().equals(Request.INVITE)){
countInviteResponse ++;
System.out.println(".....countInviteResponse: "+ countInviteResponse+" ......" );
if(countInviteResponse == 6){
System.out.println(responseEvent.getResponse());
// functions.cancel(responseEvent.getResponse());
}
try {
processInviteResponse(responseEvent);
} catch (SipException | InvalidArgumentException e) {
// TODO Auto-generated catch block
e.printStackTrace();
}
}
else {
System.out.println("-------------------Status Code: "+ statusCode+" ----------------------");
System.out.println(responseEvent.getResponse().toString());
}
ProcessInvite 响应：
private void processInviteResponse(ResponseEvent responseReceivedEvent) throws SipException, InvalidArgumentException {
int statusCode = responseReceivedEvent.getResponse().getStatusCode();
String currentResponse = responseReceivedEvent.getResponse().getReasonPhrase();
Response response = responseReceivedEvent.getResponse();
ClientTransaction tid = responseReceivedEvent.getClientTransaction();
CSeqHeader cseqtemp = (CSeqHeader) response.getHeader(CSeqHeader.NAME);
if(statusCode> 400 && statusCode <410){
if((statusCode ==401 && cseqtemp.getMethod().equals(Request.INVITE))
|| response.getStatusCode() == Response.PROXY_AUTHENTICATION_REQUIRED){
System.out.println(responseReceivedEvent.getResponse().toString());
// if(functions.cseq<4){
System.out.println("------------------Invite 401--------------------\n");
AuthenticationHelper authenticationHelper =
((SipStackExt) functions.sipStack).getAuthenticationHelper(new AccountManagerImpl(), functions.headerFactory);
try {
functions.inviteTid = authenticationHelper.handleChallenge(response, functions.inviteTid, functions.sipProvider, 5);
functions.inviteTid.sendRequest();
System.out.println("/n" + functions.inviteTid.getRequest());
} catch (NullPointerException e) {
// TODO Auto-generated catch block
e.printStackTrace();
} catch (SipException e) {
// TODO Auto-generated catch block
e.printStackTrace();
}
// functions.call(responseReceivedEvent.getResponse());
// }
}
}else if(statusCode == 200){
try {
System.out.println(responseReceivedEvent.getResponse().toString());
response = (Response) responseReceivedEvent.getResponse();
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);
}
}
catch (InvalidArgumentException e) {
// TODO Auto-generated catch block
e.printStackTrace();
} catch (SipException e) {
// TODO Auto-generated catch block
e.printStackTrace();
}
}else if(responseReceivedEvent.getResponse().getReasonPhrase().equals("Trying")){
System.out.println("------------------- Status Code: "+ statusCode+ "--------------------------");
System.out.println(responseReceivedEvent.getResponse());
}else if(responseReceivedEvent.getResponse().getReasonPhrase().equals("Ringing")){
System.out.println("------------------- Status Code: "+ statusCode+ "--------------------------");
System.out.println(responseReceivedEvent.getResponse());
System.out.println("------------------- Phone is RINGING --------------------------");
}else if(responseReceivedEvent.getResponse().getReasonPhrase().equals("Session Progress")){
System.out.println("------------------- Status Code: "+ statusCode+ "--------------------------");
System.out.println(responseReceivedEvent.getResponse());
// functions.cancel(response);
Request cancelRequest= responseReceivedEvent.getClientTransaction().createCancel();
functions.sipProvider.sendRequest(cancelRequest);
System.out.println(cancelRequest.toString());
}else {
System.out.println("------------------- Status Code: "+ statusCode+ "--------------------------");
System.out.println(responseReceivedEvent.getResponse());
}
}
邀请：
public void call(Response response) {
try {
Request request;
cseq++;
String callee = "86940160";
// 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;
if(!retry){
callIdHeader = this.sipProvider.getNewCallId();
callIdHeaderinit = callIdHeader;
}else{
callIdHeader = callIdHeaderinit;
}
// The "CSeq" header.
CSeqHeader cSeqHeader = headerFactory.createCSeqHeader(cseq, Request.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);
// Create the contact address used for all SIP messages.
contactAddress = addressFactory.createAddress("sip:" + username + "@"+ localIP +":"+rport+ ";"+ "transport=UDP");
// Create the contact header used for all SIP messages.
contactHeader = headerFactory.createContactHeader(contactAddress);
ContentLengthHeader contentLength = headerFactory.createContentLengthHeader(211);
ContentTypeHeader contentType = headerFactory.createContentTypeHeader("application", "sdp");
String sdpData = "v=0\n" +
"o=user1 795808818 480847547 IN IP4 "+localIP+"\n" +
"s=-\n" +
"c=IN IP4 "+localIP+"\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 = messageFactory.createRequest(requestURI, Request.INVITE,
callIdHeader,cSeqHeader, fromHeader, toHeader, viaHeaders, maxForwardsHeader);
request.setContent(contents, contentType);
request.addHeader(contactHeader);
request.addHeader(contentLength);
if (response != null) {
AuthorizationHeader authHeader = null;
if(retry){
authHeader = getAuthorizationHeader();
retry = false;
}else{
authHeader = makeAuthHeader(headerFactory, response, request, username, password);
retry = true;
}
request.addHeader(authHeader);
// AuthenticationHelper authenticationHelper =
// ((SipStackExt) sipStack).getAuthenticationHelper(new AccountManagerImpl(), headerFactory);
// try {
// inviteTid = authenticationHelper.handleChallenge(response, tid, sipProvider, 5);
// inviteTid.sendRequest();
// } catch (NullPointerException e) {
// // TODO Auto-generated catch block
// e.printStackTrace();
// } catch (SipException e) {
// // TODO Auto-generated catch block
// e.printStackTrace();
// }
}
// ExpiresHeader expiresHeader = headerFactory.createExpiresHeader(1);
//
// request.addHeader(expiresHeader);
// test listener =this;
if (response != null) {
boolean retry = true;
// System.out.println("DEBUG: Response: "+response);
}
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.
inviteTid.sendRequest();
dialog = 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");
}
}
回复：
REGISTER sip:sipIP SIP/2.0
Call-ID: cdf4ef0fa1a145b1b07b632797f53fdb@localIP
CSeq: 1 REGISTER
From: <sip:username@sipIP>;tag=-1436906198
To: <sip:username@sipIP>
Via: SIP/2.0/UDP localIP:52216;rport;branch=z9hG4bK-3230-3d42d8c86f8aa3d6d0194a4846511464
Max-Forwards: 70
Contact: <sip:username@localIP:52216;transport=UDP>
Content-Length: 0
log4j:WARN No appenders could be found for logger (test).
log4j:WARN Please initialize the log4j system properly.
log4j:WARN See http://logging.apache.org/log4j/1.2/faq.html#noconfig for more info.
................cseqCSeq: 1 REGISTER
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP localIP:52216;branch=z9hG4bK-3230-3d42d8c86f8aa3d6d0194a4846511464;received=localIP;rport=52216
From: <sip:username@sipIP>;tag=-1436906198
To: <sip:username@sipIP>;tag=as24ab6bcf
Call-ID: cdf4ef0fa1a145b1b07b632797f53fdb@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="6ac7ff34"
Content-Length: 0
REGISTER sip:sipIP SIP/2.0
Call-ID: cdf4ef0fa1a145b1b07b632797f53fdb@localIP
CSeq: 2 REGISTER
From: <sip:username@sipIP>;tag=-1436906198
To: <sip:username@sipIP>
Via: SIP/2.0/UDP localIP:52216;rport;branch=z9hG4bK-3230-a94c094a3aacfc40d0455de777b0ae26
Max-Forwards: 70
Contact: <sip:username@localIP:52216;transport=UDP>
Authorization: Digest username="username",realm="xyz.com",nonce="6ac7ff34",uri="sip:sipIP",algorithm=MD5,response="53231e4a2cd26604d8d2bc867b0e43e9"
Content-Length: 0
................cseqCSeq: 2 REGISTER
---------------------------Registered: 200--------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP localIP:52216;branch=z9hG4bK-3230-a94c094a3aacfc40d0455de777b0ae26;received=localIP;rport=52216
From: <sip:username@sipIP>;tag=-1436906198
To: <sip:username@sipIP>;tag=as24ab6bcf
Call-ID: cdf4ef0fa1a145b1b07b632797f53fdb@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, 30 Apr 2015 14:46:46 GMT
Content-Length: 0
Dialog: createdgov.nist.javax.sip.stack.SIPDialog@79f65165
INVITE sip:86940160@sipIP SIP/2.0
Call-ID: cdf4ef0fa1a145b1b07b632797f53fdb@localIP
CSeq: 3 INVITE
From: <sip:username@sipIP>;tag=-1436906198
To: <sip:86940160@sipIP>
Via: SIP/2.0/UDP localIP:52216;rport;branch=z9hG4bK-3230-c5512f65af9890c0798f1e5757ef1c57
Max-Forwards: 70
Content-Type: application/sdp
Contact: <sip:username@localIP:52216;transport=UDP>
Authorization: Digest username="username",realm="xyz.com",nonce="6ac7ff34",uri="sip:sipIP",algorithm=MD5,response="53231e4a2cd26604d8d2bc867b0e43e9"
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
Request OPTIONSreceived at stackwith server transaction idnull
................cseqCSeq: 3 INVITE
.....countInviteResponse: 1 ......
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP localIP:52216;branch=z9hG4bK-3230-c5512f65af9890c0798f1e5757ef1c57;received=localIP;rport=52216
From: <sip:username@sipIP>;tag=-1436906198
To: <sip:86940160@sipIP>;tag=as43ee81b7
Call-ID: cdf4ef0fa1a145b1b07b632797f53fdb@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="6ad3bcf0"
Content-Length: 0
删除标签后的响应：
------------------Invite 401--------------------
/nINVITE sip:86940160@sipIP:5060;maddr=sipIP SIP/2.0
Call-ID: 7128d7fc7109b0c1293dce9f6b2b9c7b@localIP.131
CSeq: 4 INVITE
From: <sip:username@sipIP>;tag=-515507368
To: <sip:86940160@sipIP>
Via: SIP/2.0/UDP localIP.131:52216;rport;branch=z9hG4bK-313339-453b043ba57ac51c5293cf17f8823071
Max-Forwards: 70
Content-Type: application/sdp
Contact: <sip:username@localIP.131:52216;transport=UDP>
Authorization: Digest username="username",realm="xyz.com",nonce="3cdd5016",uri="sip:86940160@sipIP:5060;maddr=sipIP",response="38b50cad69f74ba89fd63f78a1968acc",algorithm=MD5
Content-Length: 211
v=0
o=user1 795808818 480847547 IN IP4 localIP.131
s=-
c=IN IP4 localIP.131
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
................cseqCSeq: 4 INVITE
.....countInviteResponse: 2 ......
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP localIP.131:52216;branch=z9hG4bK-313339-453b043ba57ac51c5293cf17f8823071;received=localIP.131;rport=52216
From: <sip:username@sipIP>;tag=-515507368
To: <sip:86940160@sipIP>;tag=as5f585fd4
Call-ID: 7128d7fc7109b0c1293dce9f6b2b9c7b@localIP.131
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="6aa75c9f"
Content-Length: 0
------------------Invite 401--------------------
/nINVITE sip:86940160@sipIP:5060;maddr=sipIP SIP/2.0
Call-ID: 7128d7fc7109b0c1293dce9f6b2b9c7b@localIP.131
CSeq: 5 INVITE
From: <sip:username@sipIP>;tag=-515507368
To: <sip:86940160@sipIP>
Via: SIP/2.0/UDP localIP.131:52216;rport;branch=z9hG4bK-313339-5ed3ab3afc3b791c0af19470ae58813d
Max-Forwards: 70
Content-Type: application/sdp
Contact: <sip:username@localIP.131:52216;transport=UDP>
Authorization: Digest username="username",realm="xyz.com",nonce="6aa75c9f",uri="sip:86940160@sipIP:5060;maddr=sipIP",response="092fc083a66f19b5b995c8cbc8d5a8bc",algorithm=MD5
Content-Length: 211
v=0
o=user1 795808818 480847547 IN IP4 localIP.131
s=-
c=IN IP4 localIP.131
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
................cseqCSeq: 5 INVITE
.....countInviteResponse: 3 ......
------------------- Status Code: 100--------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP localIP.131:52216;branch=z9hG4bK-313339-5ed3ab3afc3b791c0af19470ae58813d;received=localIP.131;rport=52216
From: <sip:username@sipIP>;tag=-515507368
To: <sip:86940160@sipIP>
Call-ID: 7128d7fc7109b0c1293dce9f6b2b9c7b@localIP.131
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:86940160@sipIP:5060>
Content-Length: 0
................cseqCSeq: 5 INVITE
.....countInviteResponse: 4 ......
------------------- Status Code: 183--------------------------
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP localIP.131:52216;branch=z9hG4bK-313339-5ed3ab3afc3b791c0af19470ae58813d;received=localIP.131;rport=52216
From: <sip:username@sipIP>;tag=-515507368
To: <sip:86940160@sipIP>;tag=as7fc2a07d
Call-ID: 7128d7fc7109b0c1293dce9f6b2b9c7b@localIP.131
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:86940160@sipIP:5060>
Content-Type: application/sdp
Content-Length: 256
v=0
o=root 163700690 163700690 IN IP4 sipIP
s=Asterisk PBX 10.5.1
c=IN IP4 sipIP
t=0 0
m=audio 25036 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
Invite ID:z9hG4bK-313339-5ed3ab3afc3b791c0af19470ae58813d
CANCEL sip:86940160@sipIP:5060;maddr=sipIP SIP/2.0
Call-ID: 7128d7fc7109b0c1293dce9f6b2b9c7b@localIP.131
To: <sip:86940160@sipIP>
CSeq: 5 CANCEL
From: <sip:username@sipIP>;tag=-515507368
Via: SIP/2.0/UDP localIP.131:52216;rport;branch=z9hG4bK-313339-5ed3ab3afc3b791c0af19470ae58813d
Max-Forwards: 70
Content-Length: 0
................cseqCSeq: 5 CANCEL
-------------------Status Code: 481 ----------------------
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP localIP.131:52216;branch=z9hG4bK-313339-5ed3ab3afc3b791c0af19470ae58813d;received=localIP.131;rport=52216
From: <sip:username@sipIP>;tag=-515507368
To: <sip:86940160@sipIP>;tag=as23aeabf5
Call-ID: 7128d7fc7109b0c1293dce9f6b2b9c7b@localIP.131
CSeq: 5 CANCEL
Server: Asterisk PBX 10.5.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Content-Length: 0
INVITE 交易 ID 与 CANCEL 交易 ID 不同。邀请
z9hG4bK-353435-03dc537ef691bb00e1e5512b034ea431
取消是
z9hG4bK-353435-32ce00c8a1c2efc96caa44a14bf3b910
它们应该是一样的。
取消的正确方法是使用原始邀请交易并使用提供商
cancelRequest = inviteClientTransaction.createCancel(); sipProvider.sendRequest(cancelRequest);
我添加了一个 5 秒的线程，现在可以使用了。防火墙是如何延迟交易的。