How to replace real-time voice with (e.g.) a mp3 file in PJSIP samples

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Any suggestion on this? Am I the only one having such problem?

Regards,
Javi

El 21 de mayo de 2009 10:24, Javier G?lvez Guerrero <
javier.galvez.guerrero at gmail.com> escribi?:

> 2009/5/20 Benny Prijono <bennylp at teluu.com>
>
>> 2009/5/20 Javier G?lvez Guerrero <javier.galvez.guerrero at gmail.com>
>>
>>> Hi,
>>>
>>> Thanks a lot for your answers. They have been very helpful.
>>>
>>> I tried pjsua and recommended options as Roman suggested and it worked
>>> for what I needed (with an issue I'll comment next). I assume that if I'd
>>> like to implement a program myself which allowed me to do what I asked for,
>>> then I should use PJSUA-LIB API. Anyway, in case I needed to use simpleua, I
>>> guess I could see in the pjsua source code what is used to map a file as the
>>> audio source (pjmedia master port?).
>>>
>>> So, although I succeeded in establishing a SIP/RTP session between two
>>> hosts and stream a wav audio file, in the receiver side I get sound in a
>>> very choppy way. I tried changing sender and receiver hosts, but got the
>>> same results. Wireshark show me that all RTP packets are sent (all sequence
>>> numbers are received). However, some RTP packets are marked. Could this be
>>> the problem?
>>>
>>>
>> RTP packets are marked if they are the start of talksprut, it shouldn't be
>> a problem.
>>
>
> Maybe a stupid question but... what is 'talksprut'?
>
>
>>
>>
>>
>>> Sender side
>>> # ./pjsua-i686-pc-linux-gnu --play-file
>>> /home/dulceangustia/Desktop/file.wav --auto-answer 200 --auto-play
>>> --auto-loop --no-tcp
>>>
>>> Receiver side
>>> # ./pjsua-i686-pc-linux-gnu sip:147.83.47.179:5060
>>>
>>> I tried with mono 8kHz and 16 kHz sampled audio files (WAV) and specyfing
>>> it in the command line but got the same choppy audio. Note that the file is
>>> correctly encoded as it's properly played in the local host. Also note that
>>> the sound gaps appear randomly, in every try but not in the same place in
>>> the audio file. Both computers are connected through a LAN network with
>>> UTP/Ethernet cables.
>>>
>>> Any idea about what this issue can be caused for?
>>>
>>>
>> It could be the jitter. Try with adding --null-audio on the sender side to
>> see if it helps (and this would be more similar to your requirement too).
>>
>
> This didn't solve it. I've also tried changing the jitter buffer size but
> it neither worked (in fact, I don't know which buffer size I should use, in
> case this is the way to solve it). Dropping --auto-loop neither worked.
>
> But, is this issue just happening to me or is a general problem (I mean
> playing a WAV file in a pjsua session and getting choppy audio)?
>
> Sorry if my questions seem so stupid, but I'm a completely newbie (as you
> may have already noted). Any help or suggestion is so much appreciated.
>
>
> Thanks for your time,
> Javi
>
>
>
>>
>> cheers
>>  Benny
>>
>>
>>
>>>
>>> Thanks for your help.
>>>
>>> Regards,
>>> Javi
>>>
>>>
>>
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>>
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>>
>>
>
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