We have the following scenario which works fine in many of our PBX deployments and I cannot understand what setting we are missing.
For inbound calls:
Provider > FreePBX (trunk to SBC uses SRTP) > SBC > MS Teams
For outbound calls:
MS Teams > SBC > FreePBX (receives SRTP media from the SBC but sends to the provider as RTP) > Provider
On the outbound leg, MS Teams sends to our SBC using SRTP which is enabled on the SBC trunk. On all other installs, FreePBX sends the call to the provider as RTP (and does some sort of decryption before it is sent out of the SIP trunk)
On this particular install and all other installs, inbound calls work fine with two way audio.
Upon checking asterisk verbose logging, the PBX cannot handle the SRTP received from the SBC and thus tries to send it as SRTP to the provider.
ERROR: res_pjsip_session.c:938 handle_incoming_sdp: to_provider: Couldn’t negotiate stream 0:audio-0:audio:sendrecv (nothing)
I notice on the extension settings there is the option for Media Use Received Transport: Determines whether res_pjsip will use the media transport received in the offer SDP in the corresponding answer SDP. however is there some setting like this that I can enable globally?
Otherwise I am out of ideas as to why the PBX does not handle SRTP like other deployments