Would someone mind pointing me in the right direction for capturing audio to a buffer? I have multiple inbound calls and I would like to check for in-band DTMF (no rfc2833). What is the API order for setting up PJSIP to get the audio data? I just need some framework help -- I have reviewed the sample files on the web site but I'm having trouble grasping which APIs to use. I do know how to do play audio files, it's the capturing to a buffer that has me confused. I mostly use the PJSUA APIs. Global --------- pjsua_pool_create [Shared pool used for all calls] Call comes in (for each call do below) ------------- pjmedia_mem_capture_create [What size buffer should I use? Does it matter?] [How do I get the call information such clockrate, etc, so I can set this API to use the same values as the call? Does it matter? Or should I use the values in the sample programs?] pjmedia_mem_capture_set_eof_cb [How do I connect the call to this media port for so the media port is "listening" and the callback works] Call ends ---------- pjmedia_port_destroy Thank you in advance, Archie -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20090915/1d24fee7/attachment.html>
