I managed to solve it adding --add-codec=pcma option to the 'client' side, then the A-law PCM codec is used instead speex and audio is received in a gapless way. Regards, Javi El 25 de mayo de 2009 9:52, Javier G?lvez Guerrero < javier.galvez.guerrero at gmail.com> escribi?: > 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 >>>> >>>> >>> >>> _______________________________________________ >>> 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 >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20090526/7c66b1fc/attachment.html>
