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

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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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