Dear Srivatsan,
Very thanks for your valuable suggestion. It helped me. One more
clarification, I'm using PJSIP-1.2 and when I try to communication with
machine running X-lite it gives "Bad RTP pt" error, but not for some
other machine running the same X-lite version(2). Any suggestion on this
issue?
Once again thanks for your help.
Thanks & Regards,
- Senthil
Srivatsan Deenadayalan wrote:
> 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.
>>
>>
>> ------------------------------------------------------------------------
>>
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>>
>
>
>
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