hi everyone, i use siprtp to send my captured audio which is done using DIRECTSOUND directly to a sip phone xlite in windows. right now everything from capturing audio works just fine because i have tested it. SIP part works just fine too and i see on second laptop the aswer/deny button poped up on xlite and i click answer. in first laptop which made the call with siprtp i get ack and "call_on_media_update(...) function of siprtp is called. then in another function media_thread1(...) just slightly different from media_thread(...) function in sirtp i call pjsip function (pjmedia_rtp_encode_rtp) in order to get the header for the RTP packet and then i copy my PCM audio which is 160 bytes to the back of this header to make a packet of size 172 bytes and then i send it to the xlite sip phone with pjsip function call (pjmedia_transport_send_rtp(...) ) . so far so good but problems start here that i hear my voce in xlite with a lot of noise. i suspect it because i dont encode it. i have tried different things to encode the PCM audio but it does not work. xlite is setup to function on alaw and ulaw codec. i tried to encode my pcm raw data with PCMU codec but i cannot even hear my voice anymore and noise is even worse that without encoding. how to encode the RTP packet so i hear my voice without noice just like the one in simpleua example. BR sag -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20140217/7dbbc302/attachment-0001.html>
