Current JAIN-SIP RI (JSR 32)

Hello to the community of JAIN-SIP (JSR 32),
I'm a student currently working on a project utilizing
JAIN-SIP 1.2 RI in the implementation of the application.
In the application, there is an API from another project
being used that has a SipListener implementation. I would
like to develop my application using existing RI's or any
interface implementations as much as possible to avoid the
need of implementing interfaces required to run my
application.
The API mentioned earlier involves pure TCP transfer of data
only, and it serves as the network setup layer of my
applicaiton that establishes SIP conferencing. An exception
of "listening point not found for this provider" was observed
which failed the setup of the conference. It was discovered
later that the DefaultRouter class in gov.nist.javax.sip.stack
package defaults the transfer type to UDP which causes the
error. After this has been changed to TCP, things worked fine.
I'm trying to avoid packaging this "fix" into my application as
I would like people using the application to obtain their copy
of the JAIN-SIP RI. Fiddling with a public RI probably isn't a
good idea!
Is it possible to change the default transfer type in the default
router to TCP instead of UDP? Can this change be realized
by the JAIN-SIP community in the form of a new release of the RI?
Or...any suggestions for possible solutions the this problem?
Thanks for looking into this,
Any comment is greatly appreciated!

Hi,
I got error 500 Internal Server Error when i send a text message from jain-sip client, Client send the Text message to server but server return the above error.
There is no Text Message Handler Code in Server Proxy Class,
Also when i try to start Voice Connversation, Client sent a INVITE request to server but server return 500 Server Intnal Error. also there is no INVITE handler code in Server Proxy Class.
I am using jain-sip-presence-proxy SIP server and jain-sip-applet-phone.
please help me .......................................Its Urgent
why server could not handle the text or voice messages.
How i make the text and voice conversation.
with regards

Similar Messages

Jain SIP sample code to do a VOIP...

Hi,
Right now I have a problem with my current project, can somebody give me some sample codes/can help me on how to make an client VOIP application using Jain SIP API to do a VOIP call... I have an PBX Server but i dont know how to start writing my application to connect to my server...
thanks,
dOnie

[https://jain-sip.dev.java.net/]
Edited by: cotton.m on 10-Jun-2010 10:05 AM

How to continually reregister using Jain SIP?

I currently use Jain-Sip to successfully send a REGISTER request and get a 401 UnAuth back, I then use the nonce to generate the response and send it back and I successfully get a 200 OK response.
However when I go to re register again I cant seem to be able to send the request again, I keep getting null pointer exceptions.
Here is my code:
public class CSip extends Activity implements javax.sip.SipListener {
     private static AddressFactory addressFactory;
    @Override
    public void onCreate(Bundle savedInstanceState) {
        super.onCreate(savedInstanceState);
        digest = new DigestClientAuthenticationMethod();
        myAddress = getLocalIpAddress();
        tv = new TextView(this);
        setContentView(tv);
             try {
                init(tv);
                register();
             } catch (Exception ex) {
                tv.append("Unxpected exception " + ex.getMessage());
                ex.printStackTrace();
    super.finish();
    public void init(TextView tv) throws Exception {
          SipFactory sipFactory = null;
          sipStack = null;
          sipFactory = SipFactory.getInstance();
          sipFactory.setPathName("gov.nist");
          Properties properties = new Properties();
          properties.setProperty("javax.sip.STACK_NAME", "Sip_Test");
          // Create SipStack object
          sipStack = sipFactory.createSipStack(properties);
          tv.setText("sipStack = " + sipStack);
          headerFactory = sipFactory.createHeaderFactory();
          addressFactory = sipFactory.createAddressFactory();
          messageFactory = sipFactory.createMessageFactory();
          lp = sipStack.createListeningPoint(getLocalIpAddress(),
                    8002, ListeningPoint.UDP);
          if(sipProvider == null){
          sipProvider = sipStack.createSipProvider(lp);
          sipOnOffFlag = true;
          tv.append("\n jain sip stack started on " + getLocalIpAddress() + ":" + myPort + "/" + ListeningPoint.UDP + " ");
          sipProvider.addSipListener(this);
          Log.d("INIT", "SipProvider = : " + sipProvider.toString());
    public void register()throws Exception{
          String fromName = "xxxxxxxx";
          String fromSipAddress = "sip.network.com";
          String toSipAddress = "sip.network.com";
          String toUser = "xxxxxxxx";
          SipURI fromAddress = addressFactory.createSipURI(fromName,
                    fromSipAddress);
          Address fromNameAddress = addressFactory.createAddress(fromAddress);
          FromHeader fromHeader = headerFactory.createFromHeader(
                    fromNameAddress, null);
          SipURI toAddress = addressFactory
                    .createSipURI(toUser, toSipAddress);
          Address toNameAddress = addressFactory.createAddress(toAddress);
          ToHeader toHeader = headerFactory.createToHeader(toNameAddress,
                    null);
          URI requestURI = addressFactory.createURI(
                    "sip:" + "sip.network.com");
          List<ViaHeader> viaHeaders = new ArrayList<ViaHeader>();
          String ipAddress = lp.getIPAddress();
          ViaHeader viaHeader = headerFactory.createViaHeader(ipAddress,
                    lp.getPort(),
                    lp.getTransport(), null);
          viaHeaders.add(viaHeader);
          CallIdHeader callIdHeader = sipProvider.getNewCallId();
          CSeqHeader cSeqHeader = headerFactory.createCSeqHeader(1L,
                    Request.REGISTER);
          MaxForwardsHeader maxForwards = headerFactory
                    .createMaxForwardsHeader(70);
          Request request = messageFactory.createRequest(requestURI,
                    Request.REGISTER, callIdHeader, cSeqHeader, fromHeader,
                    toHeader, viaHeaders, maxForwards);
          SipURI contactUrl = addressFactory.createSipURI(fromName, fromSipAddress);
          contactUrl.setPort(8002);
          contactUrl.setLrParam();
          SipURI contactURI = addressFactory.createSipURI(fromName, "sip.network.com");
          contactURI.setPort(sipProvider.getListeningPoint(lp.getTransport())
                    .getPort());
          Address contactAddress = addressFactory.createAddress(contactURI);
          contactHeader = headerFactory.createContactHeader(contactAddress);
          request.addHeader(contactHeader);
          Header extensionHeader = headerFactory.createHeader("Expires",
               "120");
          request.addHeader(extensionHeader);
          Log.d("SIP", "" + request.toString());
          // Create the client transaction.
          Log.d("BEFORE TID", "SipProvider = : " + sipProvider.toString());
          inviteTid = sipProvider.getNewClientTransaction(request);
          inviteTid.sendRequest();
          Log.d("AFTERSENDREQUEST", "SipProvider = : " + sipProvider.toString());
          Log.d("INVITETID", "inviteTid = " + inviteTid.getState());
          dialog = inviteTid.getDialog();
     public void processResponse(ResponseEvent responseEvent) {
          Log.d("RESPONSE", "response " + responseEvent.getResponse());
          Message message = Message.obtain();
          message.obj = "received response "+responseEvent.getResponse();
          handler.sendMessage(message);
          Response response = (Response) responseEvent.getResponse();
          ClientTransaction tid = responseEvent.getClientTransaction();
          CSeqHeader cseq = (CSeqHeader) response.getHeader(CSeqHeader.NAME);
          Log.d("STATUS CODE", "status code = " + response.getStatusCode());
          Log.d("STATUS CODE", "header code = " + response.getHeader("WWW-Authenticate"));
          Log.d("STATUS CODE", "reason code = " + response.getReasonPhrase());
          CallIdHeader callid = (CallIdHeader) response.getHeader("Call-ID");
          WWWAuthenticate wwwAuth = (WWWAuthenticate) response.getHeader("WWW-Authenticate");
          Log.d("STATUS NONCE", ", nonce : " + wwwAuth.getNonce());
          Log.d("STATUS NONCE", "nonce : " + wwwAuth.getNonce());
          String nonce = wwwAuth.getNonce();
          String cNonce = wwwAuth.getCNonce();
          Log.d("STATUS ", "cnonce : " + cNonce);
          String realm = wwwAuth.getRealm();
          Log.d("STATUS ", "realm : " + realm);
          String method = wwwAuth.getQop();
          Log.d("STATUS", "method : " + method);
          String alg = wwwAuth.getAlgorithm();
          Log.d("STATUS", "alg : " + alg);
          String authResponse = null;
          AuthorizationHeader authHeader = null;
          try {
               digest.initialize(realm,"xxxxxxxx","sip:sip.network.com",
                         nonce,"xxxxxxx", "REGISTER",null, alg);
               Log.d("GENRESPONSE", "" + digest.generateResponse());
               authResponse = digest.generateResponse();
          } catch (Exception e) {
               // TODO Auto-generated catch block
               e.printStackTrace();
          try {
                authHeader = headerFactory.createAuthorizationHeader("Digest");
                authHeader.setUsername("xxxxxxxx");
                authHeader.setRealm(realm);
                authHeader.setNonce(nonce);
                authHeader.setResponse(authResponse);
                authHeader.setAlgorithm(alg);
                URI authURI = addressFactory.createURI(
                              "sip:" + "sipnetworks.com");
                authHeader.setURI(authURI);
          } catch (ParseException e) {
               e.printStackTrace();
          Log.d("AUTH HEADER", "Auth Header = " + authHeader);
          if (response.getStatusCode() == Response.UNAUTHORIZED){
          try {
               authHeader.setNonceCount(1);
               createAuthReply(authHeader, callid);
          } catch (Exception e) {
               e.printStackTrace();
          scheduleReRegistration(120, callid);
     }

I have tried putting in my register() method several times and if I call the different register() methods,
so call register() then register2() then register3() it works this way.
But not if I simply put register() in a timer looping.
So there must be something I have to terminate and restart or something each time I run the method?
Edited by: draffodx on Feb 11, 2010 10:12 AM

Memory leak in Jain-SIP ?

Hi there.
We implemented a SIP gateway to MS OCS 2007 using
Jain-SIP version 1.2.1906.
One problem we have is memory consumption.
Using jconsole we found out that memory is not
completely released when SIP clients log out.
E.g. the following classes keep on accumulating:
53134 instances of class gov.nist.core.NameValueList
41730 instances of class gov.nist.core.NameValue
20330 instances of class gov.nist.core.Host
20330 instances of class gov.nist.core.HostPort
19767 instances of class gov.nist.core.DuplicateNameValueList
19767 instances of class gov.nist.core.MultiValueMapImpl
16162 instances of class gov.nist.javax.sip.address.Authority
16162 instances of class gov.nist.javax.sip.address.SipUri
13570 instances of class gov.nist.javax.sip.address.UserInfo
12261 instances of class gov.nist.javax.sip.address.AddressImpl
5210 instances of class gov.nist.javax.sip.header.Protocol
5210 instances of class gov.nist.javax.sip.header.Via
4168 instances of class gov.nist.javax.sip.header.CallID
4168 instances of class gov.nist.javax.sip.header.CallIdentifier
3901 instances of class gov.nist.javax.sip.header.CSeq
3901 instances of class gov.nist.javax.sip.header.ContentLength
3901 instances of class gov.nist.javax.sip.header.From
3901 instances of class gov.nist.javax.sip.header.MaxForwards
3901 instances of class gov.nist.javax.sip.header.RequestLine
3901 instances of class gov.nist.javax.sip.header.To
3901 instances of class gov.nist.javax.sip.header.ViaList
3901 instances of class gov.nist.javax.sip.message.SIPRequest
3901 instances of class gov.nist.javax.sip.stack.SIPTransaction$TransactionSemaphore
2858 instances of class gov.nist.javax.sip.DialogFilter
2618 instances of class gov.nist.javax.sip.header.Route
2618 instances of class gov.nist.javax.sip.header.RouteList
2592 instances of class gov.nist.javax.sip.stack.SIPServerTransaction
2590 instances of class gov.nist.javax.sip.header.ContentDisposition
2590 instances of class gov.nist.javax.sip.header.ContentType
2590 instances of class gov.nist.javax.sip.header.MediaRange
1579 instances of class org.apache.xmlbeans.SchemaType$Ref
1309 instances of class gov.nist.javax.sip.header.UserAgent
1309 instances of class gov.nist.javax.sip.parser.Pipeline
1309 instances of class gov.nist.javax.sip.parser.PipelinedMsgParser
1309 instances of class gov.nist.javax.sip.parser.StringMsgParser
1309 instances of class gov.nist.javax.sip.stack.HopImpl
1309 instances of class gov.nist.javax.sip.stack.SIPClientTransaction
1309 instances of class gov.nist.javax.sip.stack.SIPClientTransaction$TransactionTimer
1309 instances of class gov.nist.javax.sip.stack.SIPDialog
These all are classes from the SIP stack aren't they ?
Are there any known issues with memory leaks
in Jain-SIP ?
Is it probably a configuration issue with
the properties/parameters for the SIP stack ?
(There is a property gov.nist.javax.sip.AGGRESSIVE_CLEANUP which we set
to TRUE but it didn't help)
Do we miss to release/initialize anything in the SIP stack manually ?
Thanks a lot in advance,
Fred

Nobody else seems ton have this problem. I haven't run my SIP code for as long as a week at a time but i didn't notice any undue memory usage, and I also trawled inside the source a lot without seeing anything odd. So is it your code leaking? Are you sure you're releasing everything to do with a conversation when it ends? Rather than say accumulating things in some static data structure?

Jain SIP Rookie needs help!

Ok here is my story....i was given a project to develop a client application that can do voip calls...so i started researching and it led me to jain-sip...i've been researching on it for weeks now and still i do not have a clue on how to start this....so what i did first was build my GUI using swing.. but now that im done with that i do not know where to start....So far this is what i know....We here have a SIP Proxy server waiting for my application to start registering...but thats the thing i even dont know what is needed to code this application...I already downloaded the nist-sip.jar package that lets me use the javax.sip.* package...thats it....I also know that i should implement the SipListener and use its abstract method..but that as far as i got....please anyone please...this is my first post so please be patient with me...thanks guys

so far here is my code.....
package rurutalk;
public void start()
System.out.println("Inside Start Function");
try
properties.setProperty("javax.sip.IP_ADDRESS","202.124.134.44");
sipFactory = SipFactory.getInstance();
sipFactory.setPathName("gov.nist");
sipStack = sipFactory.createSipStack(properties);
headerFactory = sipFactory.createHeaderFactory();
addressFactory = sipFactory.createAddressFactory();
messageFactory = sipFactory.createMessageFactory();
catch(Exception e)
e.printStackTrace();
public void stop()
System.out.println("Inside Stop Function");
public static void main(String args[])
new SipExample ();
/** Sip Listener Interface method.*/
public void processTimeout(TimeoutEvent timeOutEvent)
System.out.println("Inside processTimeout Function");
/** Sip Listener Interface method.*/
public void processResponse(ResponseEvent responseEvent)
//Response response = responseEvent.getResponse();
System.out.println("Inside processResponse Function");
/** Sip Listener Interface method.*/
public void processRequest(RequestEvent requestEvent)
System.out.println("Inside processRequest Function");
Message was edited by:
iromswa
Message was edited by:
iromswa
null

Jain-sip-proxy server does not display in browser

Hi JCP
I have downloaded the codes for JAin-sip-Proxy Server, Jain-sip-applet-phone.my problems are:-
1. I have downloaded all libraries and don't find any error in boiuding the code, but when I deploy using Sailfin it just display Hello World in browser- where else the
Netbeans Ide tells me deployment was successful.
2. When I use Tomcat 6.26 Server, the browser just displays JavaFaces(Welcoming page of jsp framework)
I can't see both the server site and the applet phone. My O.S is Windows 7
(I have followed all instructions in setup including in control panels->Java Plug in)
thanx. Can someone help me implementing this two codes( I downloaded them form NIST JAIN SIP WEB SIT)
Edited by: FihlaTV on Aug 6, 2010 10:54 AM

If I try to look at your XML file directly, I get an error.
http://www.ericaeschmann.com/photos.xml
That is likely the cause. Fix that closing tag issue and check it...

Top of JAIN SIP

Hi!
I try to write an application based on JAIN SIP (It's call 2 voipbuster account). But it's too complicated, anybody know a top level API based on JAIN SIP? I want only set the server host and port, the caller user name and password and the called user data (and maybe the Session Descriptor). After the call successfuly I want to connect the two account RTP communication, and if somebody disconnect send a Bye to the another user.
With JAIN SIP I can call authenticated but after one min VOIPBUSTER drop the connection (nothing happened, the remote side sent 3 "Transaction terminated event" and after than a bye...).
I think jain sip is very good but a little bit too low level.

I am not sure...But maybe http://www.mjsip.org/index.html is what you are searching for.

JAIN SIP Android Problem

I use this tutorial :
[http://www.oracle.com/technology/pub/articles/dev2arch/2007/10/introduction-jain-sip.html]
to devlope an JAIN SIP Application the code works fine, but whene I use the same code in an Android application I get this message :
The Peer SIP Stack: gov.nist.javax.sip.SipStackImpl could not be instantiated. Ensure the Path Name has been set.

I use this tutorial :
[http://www.oracle.com/technology/pub/articles/dev2arch/2007/10/introduction-jain-sip.html]
to devlope an JAIN SIP Application the code works fine, but whene I use the same code in an Android application I get this message :
The Peer SIP Stack: gov.nist.javax.sip.SipStackImpl could not be instantiated. Ensure the Path Name has been set.

JAIN SIP PRESENCE PROXY PROBLEM

I was download the NIST JAIN-SIP-PRESENCE-PROXY project.
i use the ant to run and compile the project is fine.
but IM cannot sign in to the presence proxy
i use two command prompt to open IM and presence proxy. i have change the domain as my computer name. but the IM also cannot sign in to the presence proxy. i use the default user inside the IM and proxy server. the user is Deruelle, mranga.
i also use the window messenger 5.1 to sign in the presence proxy but it also cannot sign in.
what happen and how to solve it
Please help me as soon as possible.

but i need use the jain sip to develop the SIP
presence server.
so i hope can use the jain-sip-presence-proxy as a
example.
i already download the newer cvs for
jain-sip-presence-proxy but the messaging function
still not yet fuction.
so i dont know how.
can help me pls!!
now i learn jsip 1.2 api.
any other material and sourceHi
Not sure if you read my first 2 replies, but the website says the proxy server isnt functional. It hasnt been worked on for a while, and I doubt there will be any work on it for a while longer...

Where is JainSipApi.jar in bea sip server?

hi,
to my understanding JSR-116 reuse the sip stack from JSR-32
but I do not find any JainSipApi.jar in bea sip server.
i am a bit confused. anyone can help?

Hi,
The WebLogic SIP Server does not provide a simple ava langunage binding to the SIP protocol, as you would get with the JAIN SIP API. Our SIP message parser/assembler and transaction layer is fully integrated with our Servlet container and is not visible as a .jar file.
The SIP Servlet API is part of the JAIN initiative within the JCP but it does not have any direct dependency on, or relationship with, the JAIN SIP specification.
BR,
-Mike

SIP for J2ME API

Anyone out there ever get the SIP for J2ME API (JSR-180) working on an actual device, like a PDA ?
I can get it to run with the nokia emulator but other then that it's been pretty much useless to me.
Would the full blown JAIN SIP API be better for high end PDA usage ?
TIA
Anoop

Hi Anoop,
Did you get the SIP application working on device. I am trying to run Wtk2.5 "GoSip" App on the device but didn't succeed but was able to run on emulator. While running on emulator I started the SIP server from File->Utilities and entered the local machine name as a proxy host and it ran successfully on emulator. Can you tell me what server to use, which proxy host to enter and what all other changes do I need to do while running the same example on the device. Or were u able to run any SIP application on device? Please help.
Regards,
Pankaj Hotwani.

Get msg with SIP

Hi,
How can i poll for incoming msgs with SIP in JAVA? I understand that one have to use use JAIN SIP lib to do it.
and implement a SipListener, but how do i get from here. Are there any examples i can look at or can someone give me a hand?
what i am trying to make is a client who connects to my sip server with a user:[email protected] and listen for a incoming message...

And do someone know how to get the lib:
JSR-165 SIMPLE Instant Messaging
http://www.jcp.org/en/jsr/detail?id=165

Help needed in receiving two consecutive msges from SIP Ser in UDP server!

Hi everyone,
I am Mitul Gogoi,from Assam,North-east part of India.
I am writing a SIP proxy server,which is simly a UDP server and which will sit between the Client(X-Lite softphone) and Brekeke SIP Server and just receive and send messages.
Client---------------->My UDP Server-------------------------->SIP Server (This is for requests)
again,
SIP Server---------->My UDP Server-------------------------->Client (This is for responses)
(couldnot draw the arrows together)
My server will receive any msg coming from Client in port 7000 from Client and My server will send the msg to SIP Server to its IP address and port 5060.So, I just wanted to change the port number to which Client will send msges;i.e. instead of earlier port 5060,it will now send to port 7000.
The msg sending and receiving scenerio:
1.The first msg from Client is received by My Server.
2.My server sends the msg to SIP Server.
3.My server then waits for response from SIP server.
4.One msg comes to My server and received successfully. and it closes the socket.
5.This msg is sent to Client.
6.Second msg comes from SIP server to My server.But My server is unable to receive this second msg.maybe because My server closes the socket to receive the next msg.
I am using ResponseHandler Thread in main which will listen to any msges that may come from SIP server to My server.
My question is :
Is it not possible to receive two consecutive msg from SIP server ?
If one msg comes,it then closes the socket.
My code for sending and receiving :
udpClSocket = new DatagramSocket();
packet = new DatagramPacket(buf, buf.length,InetAddress.getByName(server), port);
udpClSocket.send(packet);//it will send to SIP Server as well as Client,by changing the server and port
byte resbuf[] = new byte[msgSize];
packet = new DatagramPacket(resbuf, resbuf.length);
udpClSocket.receive(packet);//it will receive msg from SIP server only
packet = null;
udpClSocket.close();
The first msg comes to My server successfully,but the second msg is not being received.
I have tried in so many ways,such as taking two different ports :one for receiving for Client and My server ,and another port for My server and SIP server.
If anyone can help me in this ,then I will be highly grateful.
regards,
Mitul

Why? Throw all this code away, and use the NIST reference implementation of JAIN-SIP. They've done all the hard work for you. All you have to do to write a stateless proxy is a little mild header processing.
Thank you ejp for your reply.Mine is a Outbound SIP proxy server,which uses a port other than default 5060 SIP server port.The client will REGISTER SIP server via my server.and moreover,I want to build my own server at least once for learning purpose.
maybe because My server closes the socket to receive the next msg.
Why?
I donot know why.may be I am very new to network programming or did not do much research in networking.But,my code should work;couldn't find out any fault.
Here is my code:
package com.ef;
import java.net.DatagramPacket;
import java.net.DatagramSocket;
import java.net.InetAddress;
import java.util.Date;
* It sends UDP packets to following destinations-
* 1. SIP Server
* 2. SIP UACs
public class OBUDPClient extends Thread {
     private DatagramSocket udpClSocket;
     private DatagramPacket packet;
     private String server = "192.168.1.2";//IP of Brekeke SIP server
     private int port = 5060;
//SIP Message Size
     private int msgSize = 2048;
     private byte buf[];
     private boolean serFlag = true;
     private OBMessage m;
     * Send Request to SIP Server
     * @param in_packet
     * @throws Exception
     public OBUDPClient(DatagramPacket in_packet) throws Exception{
          super("Client-"+String.valueOf((new Date()).getTime()));
          this.buf = in_packet.getData();
          m = new OBMessage();
          * Keep track of IP and Port of the source of this packet. Response
          * from SIP Server will be redirected to this IP and Port
          m.setTargetIP(in_packet.getAddress().getHostAddress());
          m.setTargetPort(in_packet.getPort());
          start();
     * Send Request to SIP UAC (). This constructor gets call from OBResponseHandler.java
     * @param server
     * @param port
     * @param buf
     * @throws Exception
     public OBUDPClient(String server, int port, byte buf[]) throws Exception{
          super(String.valueOf((new Date()).getTime()));
          this.server = server;
          this.port = port;
          this.buf = buf;
          serFlag = false;
          start();
     public void run(){
          try{
          System.out.println("OBUDPClient Redirecting packet");
          udpClSocket = new DatagramSocket();
          //Send Request
          packet = new DatagramPacket(buf, buf.length,InetAddress.getByName(server), port);
          udpClSocket.send(packet);
          //Receive Response
          byte resbuf[] = new byte[msgSize];
          packet = new DatagramPacket(resbuf, resbuf.length);
          udpClSocket.receive(packet); //could not receive two consecutive msg from SIP server
          if(serFlag){
          * If request is sent to SIP Server we are interested for the
          * response otherwise not
          System.out.println("<----------------Handle Response----------------->");
     System.out.println("++++++++++++++++++++++++++++++++++++++++++++++++++");
     System.out.println("Response from SIP Server: "+new String(packet.getData()).trim());
     System.out.println("++++++++++++++++++++++++++++++++++++++++++++++++++");
     //Read SIP reseponse sent by SIP Server
     m.setMessage(new String(packet.getData()).trim());
     //Store the Response message in Queue
     OBMain.q.push(m);
     }catch(Exception ex){
     ex.printStackTrace();
     packet = null;
     udpClSocket.close();
Note:Everything starts from the Client.First,Client makes a REGISTER request;it is passed through Outbound server to SIP server.Then,SIP server responds with 100 Trying;this is received successfully by my outbound server and sent to Client.Then,again,SIP server responds with 200 OK;this is not received by my outbound server;hence cannot reach Client,as a result of which Registration fails.
regards,
mitul

Interworking Parlay and JAIN

I'm evaluating two scenarios (one using an IMS architecture and one without) for a postgraduate project. So I've two questions:
1)
In an environment that doesn't use the IMS architecture and Parlay is used as middleware that extends from the application to the parlay gateway (where I assume some sort of runtime-environment-equivalent to the JSLEE exists in the parlay gateway), how could one connect the parlay gateway to the physical layer using JAIN SIP? Are there alternative ways to do this apart from interfacing Parlay to JSLEE first using a resource adaptor and then again from JSLEE to JAIN SIP using another RA?
2)
I'm thinking of an IMS architecture where JAIN APIs are used for protocol stacks (JAIN SIP etc) and the CSCF runtime environments (JAIN SLEE). Now, in addition to a parlay gateway at the IMS Application Server layer that supports purely parlay applications (with the relevant parlay APIs), I'm trying to work out where purely JAIN applications will connect into IMS. Is this possible by using JAIN SLEE as a separate IMS Application Server and communicating from this AS to the JSLEE in S-CSCF using JAIN SIP? Is this a work-able model or would it be more advisable to interface JAIN applications to Parlay at a higher layer (using Parlay JAIN API) and use only one parlay gateway in the IMS Application Server layer?
I recognise that these could be ill-informed questions, but I would appreciate answers to either (or both) of them. Thank you very much in advance!

I'm assuming you placed service group 61 and 62 on the router LAN, WAN inbound directions. Did you apply inspection to LAN to WAN direction or WAN to LAN direction?
Did you also used WCCP and IOS firewall on ASR routers?
Thanks a lot

Changing SIP to match email address

Our current Lync SIP addresses are setup like so:
[email protected] - EXAMPLE: [email protected]
Our emails are setup like:
[email protected] - EXAMPLE: [email protected]
So hopefully you can see why we'd want to change the Lync SIP to match our email addresses. Obviously, our domain logins match the current SIP address - or domain/initials. To match the example above, it would be contoso/cas.
So if we were to change the Lync SIP addresses to match their emails, it would no longer match the domain login - which throws off Outlook. I've used Set-CsClientPolicy -DisableEmailComparisonCheck, but it seems like Lync still throws the Exchange sync error
for the first few minutes that the user is logged in - sometimes more.
We currently have 4 test users (AKA our IT department) set to the "new" SIP address which matches their emails. There isn't anything horribly wrong. DNS and internal certificates were setup appropriately so no certificate popups at sign-in, except
for at our branch locations.
So what I'm really asking is: Does this seem like a viable option? It would be nice for when we setup external/mobile access to just tell users they can sign in with their email/password rather than domain.company.net, but if that's the only benefit, then
I'm not sure if it's such a good idea to be playing with the SIP addresses.

Agree 100% with Ben. SIP address matching the email address is best practice and is absolutely viable, nearly 100% of deployments I've seen use this with the exception of a small handful of internal-only shops.
When users sign in with external/mobile they will need both the SIP address and AD address during sign in, but once entered it will be cached and will not need to be entered repeatedly.
As Ben mentioned, the large benefit comes from federated communications. You don't have to tell anyone your SIP address, they can just communicate with you because they know your email address.
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SWC Unified Communications

Current JAIN-SIP RI (JSR 32)

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