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

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              String toSipAddress = "sip.network.com";
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              Log.d("STATUS NONCE", "nonce : " + wwwAuth.getNonce());
              String nonce = wwwAuth.getNonce();
              String cNonce = wwwAuth.getCNonce();
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              String realm = wwwAuth.getRealm();
              Log.d("STATUS ", "realm : " + realm);
              String method = wwwAuth.getQop();
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              String alg = wwwAuth.getAlgorithm();
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    I have tried putting in my register() method several times and if I call the different register() methods,
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  • Memory leak in Jain-SIP ?

    Hi there.
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    Message was edited by:
    iromswa
    Message was edited by:
    iromswa
    null

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         * 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();
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              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:
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    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?
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    Our current Lync SIP addresses are setup like so:
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    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.
    Please remember, if you see a post that helped you please click "Vote As Helpful" and if it answered your question please click "Mark As Answer".
    SWC Unified Communications

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