HELP!! Voice Connection During A SIP Call

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My problem was no voice was transmitted during a phone call that's why I can't hear the voice of the person on the other network. I think someone had faced this problem before

here is the link..


http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2008-June/003452.html


Im sorry but i dont have any idea on how to transmit voice.

This is what I did.

Call the pjsua_call_make_call function to begin the invite Sip Call

Then on the other end, call the function pjsua_call_answer then the call is successfully established but no audio is enable. How can I enable the audio during the call? this is what I wanted to know.. Hope you can help me guys, a sample code on how to make it will be very useful.. 

Thanks in advance..


BR,

Joseph

--- On Mon, 7/28/08, Benny Prijono <bennylp at pjsip.org> wrote:
From: Benny Prijono <bennylp@xxxxxxxxx>
Subject: Re: HELP!! Voice Connection During A SIP Call
To: "pjsip list" <pjsip at lists.pjsip.org>
Date: Monday, July 28, 2008, 1:54 PM

On Mon, Jul 28, 2008 at 5:28 PM, Fabio Pietrosanti (naif) <lists at infosecurity.ch> wrote:

By migrating to APS for symbian_sound.c low level driver this will be

automatically fixed.




How come?
?With Audio Proxy Server the Audio priorities it's automatically managed


by Symbian .



I think it's still way too early to conclude that this is a priority issue. There are many other reasons that could cause the loss of audio (please see the audio troubleshooting link).


Cheers
?Benny


?
Regards



Fabio Pietrosanti



Benny Prijono wrote:

> I'm afraid the description is still too vague to be useful. For

> general audio troubleshooting ideas, please see

> http://trac.pjsip.org/repos/wiki/sound-problems. Before trying on the

> device, I would suggest to use pjsua to try out your configuration first.

>

> Cheers

> ?Benny



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