Hi there, I need to measure handover delays in wireless environments where VoIP services are offered and I thought the PJSIP project was powerful enough to let me tweak the given sample programs to achieve what I need. As I'm completely new in PJSIP (although I've read some project documentation and source code) I don't know if this can really be done, so I would like to get some help from any of you. AFAIK, the "Simple UA" (simpleua) sample application establishes a SIP/RTP/RTCP session, gets audio from the sound input (mic) on one side and dumps it on the sound ouput (speaker) on the other side, doesn't it? What I would like to do is to stream an audio file (i.e. an mp3 file) instead of real-time voice. I know I can stream WAV files with PJMEDIA's "Remote Streaming" sample application, but I would like to do it within a SIP session. So, I don't know how I can tweak the source code to get the audio source from a file instead audio input hardware. Could anybody tell me if this can be done and how to get it working? Anyway, if there is any better way to do this, please, tell me. BTW, I tried "Remote Streaming" (streamutil) sample application but when trying to stream on the sender side I get an error: streamutil: ../src/pjmedia/master_port.c:67: pjmedia_master_port_create: Assertion 'u_port->info.clock_rate == d_port->info.clock_rate' failed Searching on the mailing list archive I read<http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2008-November/005587.html>this is because streamutil does no resampling or channel number adjustment, so I need a mono WAV file with 8kHz clock rate. My question here is where can I get an encoder in order to transcode an audio file to this format. I'm using Debian Lenny 2.6.26-2. Any help would be much appreciated. Thank you for your time. Regards, Javi -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20090519/4a0992b1/attachment.html>
