We have a client who want to use late offer, so no sdp sent with INVITE.
Connection is TLS
When Asterisk reply with 200 OK it sends insecure RTP / AVP profile in sdp and client reject it sending BYE with “488 Not Acceptable Here”.
How to configure Asterisk to send SRTP offer in 200 OK sdp for TLS connections?
TLS and SRTP are two separate things. TLS secures the signaling while SRTP secures the media. SRTP has to be explicitly enabled in chan_sip using the “encryption” option.
“Whether to offer SRTP encrypted media (and only SRTP encrypted media) on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if the peer does not support SRTP. Defaults to no.”
Isn’t it for outgoing calls?
Here we are talking about late offer negotiation (No sdp in INVITE from client and RTP instead of SRTP in 200 OK from Asterisk)
What is your reason for not using PJSIP? Even if this is due to a limitation in chan_sip, it is extremely unlikely that anyone in the community will remove it.
You could try turning up debugging, to see if it logs a reason for not using SRTP. I’d expect that late offer SDP would be treated pretty much the same as an outgoing call, but it would take far too long the check what the code actually does.
Can you please give me an idea if that was ever fixed?
We are on asterisk 16.2.1
And as bug descriptions says SRTP works well for us with an early offer but doesn’t with late offer
Also I enabled debug logs and don’t see anything about why asterisk doesn’t want to use SRTP