Im running asterisk Asterisk 13.9.1 with PJSIP and was trying to let our users dial onto cisco Webex conferences directly over SIP. The connection works well (had some issues with SRV records not being used so had to setup some entries in /etc/hosts) and the connection to the conference room works until a web client connects when the Webex server sends a request to move the call to SRTP.
The connection is not using TLS but only SRTP with SDES. I have the full log including the SDP and the debug trace which shows that only one of the keys sent in the SDP is accepted. (the rest are in unsupported formats). The SDP includes multiple video streams and audio (I am only joining the audio part) and the SDP describes it as RTP/AVP when, as far as I understand, it should be RTP/SAVP (if requesting SRTP).
It then errors out as follows
[2017-05-31 14:18:38] WARNING[C-0000000f]: chan_sip.c:10715 process_sdp: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio
The full log is attached. Any ideas