On Mon, May 11, 2009 at 7:23 AM, S. M. Nazmul Hasan (Opu)
<apus29 at gmail.com>wrote:
> Hi Benny/Nanang
>
> I have experienced a peculiar problem that was not in previous release but
> in 1.1 APS Direct release..I am using symbian_ua_gui application with direct
> APS.
>
> I have modified the function like this
>
> int symbian_ua_answercall()
> {
> PJ_ASSERT_RETURN (g_call_id != PJSUA_INVALID_ID, PJ_EINVAL);
>
> ////////////////2nd call////////////
> if (callCounter() == 2) {
> pjsua_call_set_hold(first_call_id, NULL);
> }
> ////////////////2nd call////////////
>
> return pjsua_call_answer(g_call_id, 200, NULL, NULL);
> }
>
> It hold the first call and answer the second call. But the problem is for
> the second call X-Lite end can hear voice and no voice in pjsip end.
>
> Experiment was like this:
>
> 1. call from X-Lite to pjsip
> 2. call from another X-Lite to pjsip
>
> Can anyone please tell me the solution please...
>
>
Hi Nazmul,
As mentioned in the doc, with APS-Direct we loose the audio mixing
capability. While it may not be obvious, you are potentially creating audio
mixing situation with your scenario above. Here's what probably happened:
- pjsua_call_set_hold() is called. But this will not immediately hold the
call (it waits until it receives 200/OK for the re-INVITE).
- pjsua_call_answer() is called, this will immediately activate the second
stream. Your on_call_media_state() is called, and typically you will connect
the stream to audio device
- your pjsua_conf_connect(call2, 0) would fail (do you check the
pjsua_conf_connect() return value?), hence you won't hear audio in the
handset
- your pjsua_conf_connect(0, call2) could succeed as long as it uses the
same codec as the first call, that's probably why you hear audio in X-lite.
So the solution, disconnect the first call from the sound device before you
call pjsua_call_set_hold(). That should work.
cheers
Benny
--
> S. M. Nazmul Hasan Opu
> Software Engineer
> R & D Application
> Dhaka, Bangladesh
> Mob: +880 1712 901 764
>
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