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Thread: Jain SIP for Voip aplication !

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    Lightbulb Jain SIP for Voip aplication !

    Hi,

    How can i use the JAIN SIP package in my project?

    i have searched a lot in the internet i never found a good answer !

    i trust the member of this great Forum

    so anything you say in the topic of SIP ,JMF and JAIN SIP i really appreciate


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    Default Re: Jain SIP for Voip aplication !

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    khalilghenimi (May 10th, 2011)

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    Default Re: Jain SIP for Voip aplication !

    I read them all before, thanks anyway, but i never found there a real guidance,

    So for example i have a variable "varSIP" that have the number of a sip user to call, in my program, what to do exactly with it? What are the first steps? And what are the main methods i have to use?

    i guess this forum is full of experts that can easily answer, so thanks for each one.
    Last edited by khalilghenimi; April 26th, 2011 at 06:16 PM.

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    Arrow Re: Jain SIP for Voip aplication !

    no answer !

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    Default Re: Jain SIP for Voip aplication !

    I don't think this is an area many people are familiar with hence the low number of replies I'm afraid

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    khalilghenimi (May 10th, 2011)

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    Thumbs up Re: Jain SIP for Voip aplication !

    No problem i have resolve it alone, now am nearly finished my SIP client.
    This is my first application with JAVA, really amazing the world of JAVA, i love it.

    Anyway! To share Knowles with all people, spicily on this forum, just advices to develop client SIP.
    So for all new developers VOIP application, there is many way to build it with deferent protocol, the best is a client H.323 in my opinion, I know it after my project :S

    My application build with the SIP protocol, it is easy when you think of API like the JFM API to use media "microphone and speaker" and the API JAIN SIP, very easy just create your interface then call the Methods that you will found them on the Javadoc.


    So another advice, install a SIP server in your pc, i like the 3CX "client and server"
    Try to test some call and request with others client like c-lite you will understand the main idea.

    SIP server will not send the sound, it just to open the port with caller and callee that is it. So then when the port is open the protocol SDP and RTP will take the place and exchange the sound.

    That is the main idea, i think i will soon publish my application for free and open source in this forum, maybe am not sure.


    Have a good time, your brother from Tunisia