Hi Senthil,
Which version of PjSIP are you using ? PjSIP 1.0.x series or PjSIP 1.x
series? In earlier one PCM is the default codec and in later one you can
change the codecs in runtime. Also check bold lines in below method.
/* Callback called by the library when call's media state has changed */
static void on_call_media_state(pjsua_call_id call_id)
{
pjsua_call_info ci;
pjsua_call_get_info(call_id, &ci);
if (ci.media_status == PJSUA_CALL_MEDIA_ACTIVE) {
// When media is active, connect call to sound device.
* pjsua_conf_connect(ci.conf_slot, 0);
pjsua_conf_connect(0, ci.conf_slot);*
}
}
To establish full duplex audio you need to connect with remote port and
remote port has to be connect to you.
Regards,
Srivatsan D
Senthil wrote:
> Hello All,
>
> In my application I'm facing an issue with audio. Once the call gets
> connected I have only one way of audio, (i.e) Remote end can hear but
> I can't hear what they speak. Can anybody tell me the fix for this issue?
>
> Note: I went through the pjsip mail list and one suggestion is to
> change the codec priority to PCMU as default instead of GSM. If this
> is the fix for the issue, could you please tell me the way(s) to make
> it happen.
>
> Any help is greatly appreciated!!
>
> Thanks & Regards,
> - Senthil.
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Visit our blog: http://blog.pjsip.org
>
> pjsip mailing list
> pjsip at lists.pjsip.org
> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
>
