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Thread: Sending RTP

  1. #1
    Registered User
    Join Date
    May 2007
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    45

    Sending RTP

    Hi all,
    i'm developing a voip application, and i'm not satisfied in sending RTP.

    I create my RTP stack because native Symbian stack didn't work fine.

    I have no problem in receiveing RTP packets (using active object): i receive UDP packet and in RunL() i parse packet and write payload to audio interface (CMdaAudioOutputStream); quality is very good!

    Problems start in sending RTP: i tried 2 solutions, but both ones don't work well.
    1) I read input stream and then i send RTP packet (using active object): in RunL i recall function that reads stream and sends packet.
    This solution doesn't work fine because i send few packets and quality is very very low (for example, if i speak a word, i can listen to only one or two letters).

    2) In MaiscBufferComplete i call function for sending rtp packet and then i call function for new stream reading.

    iProvaRTP->PreparePacketL(frame);
    Record();

    This solution send a good number of packets, but quality is not very good, probably because i set timestamp in a static way: i record in PCM and then every packet has a difference in timestamp of 160 units, but i cannot have a constant sample time.

    How could i do?

    thanks in advance,

    Alessio

  2. #2
    Registered User
    Join Date
    Aug 2007
    Posts
    14

    Re: Sending RTP

    I don't know whether it helps, but you probably should set the "Synchronous Send" socket option for UDP sockets that you use for sending.

    The Symbian default is (hopefully it will be changed sometime) inconvenient. When synchronous send is OFF, the datagram write will drop packets, if the interface is not accepting packets at that point (interface is busy sending). When ON, the the write completes when the packet has been accepted by the interface.

    Code:
    err = iSocket.SetOpt(KSoUdpSynchronousSend , KSolInetUdp, 1);
    Last edited by msa22; 2007-08-28 at 08:23.

  3. #3
    Registered User
    Join Date
    May 2007
    Posts
    45

    Re: Sending RTP

    Hi msa22,
    thank you for reply.

    I've added code you suggested me, but this doesn't change anything.

    Alessio

  4. #4
    Super Contributor
    Join Date
    Mar 2004
    Location
    Bangalore,India
    Posts
    2,146

    Re: Sending RTP

    The MR/FP1 extension has RTP/RTCP extension plugin..maybe u can look at that and see if that provides something yu dont have already..

    http://www.forum.nokia.com/info/sw.n...Pack1.zip.html

    Cheers
    mayank

  5. #5
    Registered User
    Join Date
    May 2007
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    45

    Re: Sending RTP

    Thank you for your reply mayankkedia,
    but that extension is only for device building, while i'm developing app on emulator, and when it works on it, i can try to test app on phone.

    Regards,

    Alessio

  6. #6
    Registered User
    Join Date
    Oct 2007
    Posts
    1

    Re: Sending RTP

    Quote Originally Posted by ecio83 View Post
    Hi all,
    i'm developing a voip application, and i'm not satisfied in sending RTP.

    I create my RTP stack because native Symbian stack didn't work fine.

    I have no problem in receiveing RTP packets (using active object): i receive UDP packet and in RunL() i parse packet and write payload to audio interface (CMdaAudioOutputStream); quality is very good!

    Problems start in sending RTP: i tried 2 solutions, but both ones don't work well.
    1) I read input stream and then i send RTP packet (using active object): in RunL i recall function that reads stream and sends packet.
    This solution doesn't work fine because i send few packets and quality is very very low (for example, if i speak a word, i can listen to only one or two letters).

    2) In MaiscBufferComplete i call function for sending rtp packet and then i call function for new stream reading.

    iProvaRTP->PreparePacketL(frame);
    Record();

    This solution send a good number of packets, but quality is not very good, probably because i set timestamp in a static way: i record in PCM and then every packet has a difference in timestamp of 160 units, but i cannot have a constant sample time.

    How could i do?

    thanks in advance,

    Alessio
    Hi Alessio, my Name is Nadia and I'm developing a voip application but I have some problem, the first one is RTP Protocol.
    I downloaded extended api for SDK 3rd and I tried it but not work yet.
    Please help me it's very important!

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