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  1. #1
    Regular Contributor
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    Audio Proxy Server APS for SIP

    Hi,
    Can any one please explain me the following?

    1.Whats is the necessity of APS?
    2.How to Use APS in the application for getting voice and delivering voice?
    3.How to use RTP(real time protocol) for transferring audio packets.

    Thanks,
    Srivatsan

  2. #2
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    Re: Audio Proxy Server APS for SIP

    Hi,

    Have you read the articles on the wiki? They contain explanations and example code.
    Audio Proxy Server
    S60 RTP/RTC API

  3. #3
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    Question Re: Audio Proxy Server APS for SIP

    Hi Bogdan Galiceanu,

    Thanks for the reply.

    But i don't know how to use it in VOIP application.
    Can u pls explain me how to use APS for getting voice and transferring it via RTP?

    I tried to understand from the link provided by you, but it seems to be hard for me to understand.

    Thanks,
    Srivatsan.D

  4. #4
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    Thumbs up Re: Audio Proxy Server APS for SIP

    Hi Bogdan Galiceanu,
    Finally i understood the example (streamer), thanks a lot.
    And i am at the final stage of SIP VOIP implementation.
    Can u please explain me how to deliver APS sisx file along with the
    sisx file of my application. Nokia article says the APS sisx has to be delivered along with the sisx installation of the VOIP application.I dont know how to do it.
    Thanks,
    Srivatsan.D

  5. #5
    Nokia Developer Champion
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    Helsinki, Finland
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    Re: Audio Proxy Server APS for SIP

    Quote Originally Posted by srivatsan.d View Post
    ...
    Can u please explain me how to deliver APS sisx file along with the
    sisx file of my application. Nokia article says the APS sisx has to be delivered along with the sisx installation of the VOIP application.I dont know how to do it.
    Thanks,
    Srivatsan.D
    You need to embed APS sisx file in your original package.

    Check this Wiki article: Embedded SIS File
    Nokia Developer Wiki Moderation team

  6. #6
    Regular Contributor
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    Re: Audio Proxy Server APS for SIP

    Hi,

    At last with use of RTP and APS with SIP i developed a SIP voip application,which is working fine with two users.

    Now i have to make conference call.. if i use same RTP , the conference call will be initiated?

    And also RTP will work only locally? or else it can work in cross network?

    What is the way for making conference call in SIP?

    I am using S60 3rd ed(N 95) for developing application.

    Please some one clarify my doubts....

    Thanks,
    Srivatsan.D

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