I am developing a VoIP application using Speex audio codec and Asterisk 188.8.131.52. Due to certain limitations its very important for me to send and receive 200ms of audio data in one RTP packet instead of default 20ms which Asterisk sends.
I tried the changes in sip.conf to make Asterisk send more than 20ms (tried 40 & 60) of audio data but that didnt work.
I read somewhere if you send more than 20ms of encoded data (say 200ms) to asterisk it will process it without any issues. But in my case I hear choppy audio on the other end. Which means asterisk is not properly decoding 200ms of data.
Do I need to make some changes in the config files for it to process more than 20ms of data or do i need to change in the code and recompile everything.
Any help is greatly appreciated.