Initially i had asterisk 1.6, instead of re-installing completely, i just compiled and upgraded to asterisk 11. I am using web phone with WebRTC as the medium for transfer of audio/video codec, Now i try to make a call from one extension to other(i.e. from 1060 to 1061) i am not getting any call to destination extension, but getting the error as
"CLI> == Using SIP RTP CoS mark 5
[Apr 26 17:18:51] WARNING[C-0000000e]: chan_sip.c:10454 process_sdp:
Rejecting secure audio stream without encryption details:
audio 5354 RTP/SAVPF 103 104 111 0 8 107 106 105 13 126"
It seems that WEBRTC always transfer audio in an encrypted format, But my asterisk pbx in not supporting the encrypted audio stream and so it throws the above mentioned error. While doing further research on this res_srtp.so is the module needed for this encryption to happen. i am not able to get any patch/file/module for including this.
could someone please assist me on this.