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  1. #1
    Registered User ramyakk's Avatar
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    Apr 2008
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    Hi All..

    I am developing a J2ME Voip Client,

    For testing purposes I am using a soft phone(Zoiper) on the other end.
    I am able to receive RTP packets,encoded using G711-PCMA from the Soft phone in my J2ME client and I am able to extract useful data like payload type,SSRC etc.

    But I am unable to play the RTP payload.
    Here is part of the code.

    RTPPacket rtpPacket = getRTPPacket(dg, dg.getData());
    buffer = rtpPacket.getData();
    form.append(rtpPacket + " with media length: " + buffer.length);
    ByteArrayInputStream input1 = new ByteArrayInputStream(buffer);
    form.append("\n Receiving RTP Packets...");
    try {
    form.append("\n Trying to create player...");
    player= Manager.createPlayer(input1,"audio/x-wav");
    form.append("\n Player created");
    player.prefetch();
    player.realize();
    player.start();
    form.append("\n Player started");
    form.append("Player duration:"+player.getDuration());

    } catch(Exception e) {
    e.printStackTrace();
    }




    Thanks in advance.

    Ramya

  2. #2
    what is your RTP packet size? If you use PCMA/PCMU, with 12 bytes RTP header your RTP packet size will be 172(160+12) bytes. If you are able to extract 12 bytes rtp header from rtp packet accurately the rtp packet should play.
    Check closely your rtp header add and extraction.

    BR
    Javed

  3. #3
    Registered User ramyakk's Avatar
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    Hello Javed

    Thank you for your reply.

    Yes, I am receiving 172 byte RTP packets,and I extract the 12 byte header and passing the rest 160 bytes to the player.

    It is not throwing any exception while creating the player.
    But played is not being started.


    Ramya

  4. #4
    Regular Contributor raj_rr7's Avatar
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    May 2007
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    India
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    136
    Hi Ramya,

    You need to add 0x01 and 0x00 (2 Bytes) as a prefix of 160 bytes of data and then provide it to player. Hope this will solve your problem.


    With Regards
    Rajat

  5. #5
    Registered User mihirsgandhi's Avatar
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    Jul 2008
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    Hello Ramya,

    I am also making J2ME VOIP client. Can you give me piece of code to send the captured audio data over RTP from the J2ME client to other end ?

    - Mihir

  6. #6
    Registered User ipitts's Avatar
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    Aug 2008
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    Ramya,
    I am very interested to see if you got this to work?
    We are creating an application for the eseries where we need to play the RTP audio received from multicast stream.

    Ian

  7. #7
    Registered User muneer.essa's Avatar
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    Feb 2009
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    Can you please guide me,

    what RTP library you have used for J2ME ?

  8. #8
    Registered User metla12's Avatar
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    Apr 2010
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    Respected Ramya
    Can you pls tell me how u can make RTP packets and send to the other client......
    Help me with some peace of code of
    on my mail id metla12@gmail.com

    pls
    thnks in advance

  9. #9
    Quote Originally Posted by metla12 View Post
    Respected Ramya
    Can you pls tell me how u can make RTP packets and send to the other client......
    Help me with some peace of code of
    on my mail id metla12@gmail.com

    pls
    thnks in advance
    As i can remember there is a recorder api for J2ME using which you can get the payload, add it with your RTP header and send to the destination.

    FYI, full duplex audio is not possible in J2ME. You can try Symbian if you need a dialer/softphone for NOKIA devices.

    Regards
    salauddinrubelbd@gmail.com

  10. #10
    Registered User ashraf_arifin's Avatar
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    Mar 2009
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    I wonder how UNOS can implement their VOIP mobile application in J2ME if RTP is not possible. UNOS claimed Voice is able to operate on MIDP 1.0 phones. To achieve VOIP on mobile we need to handel RTP packet for incoming and outgoing directions. Can anyone help me to find the answer?

  11. #11
    raj_rr7 or someone can explain me how to do this: "You need to add 0x01 and 0x00 (2 Bytes) as a prefix of 160 bytes of data and then provide it to player. Hope this will solve your problem"....

    I dont play the packets rtp into player J2ME

  12. #12
    Registered User Fritzinger's Avatar
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    May 2010
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    Quote Originally Posted by ashraf_arifin View Post
    I wonder how UNOS can implement their VOIP mobile application in J2ME if RTP is not possible. UNOS claimed Voice is able to operate on MIDP 1.0 phones. To achieve VOIP on mobile we need to handel RTP packet for incoming and outgoing directions. Can anyone help me to find the answer?
    UNOS Voice uses a technique called Callback. The software on your phone connects to a server and tells it which number you want to speak to. The server then uses its own VOIP connection to call the other number, and calls you at your number, then connects the two conversations. In other words your phone is just acting as a normal GSM phone receiving a call, it is not initiating a VOIP call from your phone.

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