Currently I am trying to setup asterisk for SIP. I want to use G723.1 codec for testing purpose. My SIP provider asked me about it’s annex and has given me these options:
g723ar53 G.723.1 ANNEX-A 5300 bps
g723ar63 G.723.1 ANNEX-A 6300 bps
g723r53 G.723.1 5300 bps
g723r63 G.723.1 6300 bps
Q1. Which annex does G723.1 codec use in asterisk?
Q2. Currently I am told them to setup the last one for me. After I initiate a call, I get “183: SIP Progress SDP” packet but asterisk sends a CANCEL call request to the provider after receiving the SDP. Is it because of the wrong codec?
Thank you in advance,