Hi, I am looking for a little bit of help from the community if possible please? I’m experimenting with Chrome (JSSIP), WebRTC and Asterisk 13.2.0-rc1. I currently have everything configured as per the latest recommendations and so far have made some promising progress.
I have two peers defined in Asterisk, which I am able to successfully register with. Signalling seems to be working nicely, as I am able to initiate and reject calls without any problems.
However, when I answer a call which is initiated, it terminates immediately. In the Asterisk logs, I see the following:-
[2015-05-12 13:03:33] WARNING[C-00000009] res_rtp_asterisk.c: Could not set policies when setting up DTLS-SRTP on ‘0x7f752c001320’
[2015-05-12 13:03:33] WARNING[C-00000009] res_rtp_asterisk.c: RTP Read error: Unspecified. Hanging up.
I have generated keys/certificates for Asterisk and they are declared in sip.conf (example):-
Please could anyone shed some light on what I’m doing wrong? Thank you