Opus codec deploy in asterik made unable to call establish

Did you ran the command: sip reload before test that again?

I have did below:

  1. module unload res_srtp

  2. in sip.conf
    ;encryption=yes

  3. asterisk cli > sip reload

Made call and found the SRTP error is gone but the error as below:

== Using SIP RTP CoS mark 5
[Oct 1 03:33:18] WARNING[5303][C-00000006]: chan_sip.c:11155 process_sdp_a_audio: Got Opus minptime=10
[Oct 1 03:33:18] WARNING[5303][C-00000006]: chan_sip.c:10492 process_sdp: Rejecting secure audio stream without encryption details: audio 17013 RTP/SAVPF 111 103 104 0 8 106 105 13 126

Any missed step?

You haven’t turned off encryption on the client. It is still requesting SAVFP.