I finished enabling TLS and SRTP on FreePBX. I am able to make and receive calls. When making a call from the configured phone to my cellphone, I get the following logs for the RTP connection
[2023-06-26 15:01:25] VERBOSE[30823] netsock2.c: Using SIP RTP Audio TOS bits 184
[2023-06-26 15:01:25] VERBOSE[30823] netsock2.c: Using SIP RTP Audio CoS mark 5
I would assume there is something wrong in my configuration, as the logs state the audio connection is made with RTP instead of SRTP. I just want to make sure this is true, I’m fairly new to using this system. Or do the asterisk logs show RTP regardless if I’m using RTP or SRTP. Sorry for the noob question.
My configuration:
Asterisk SIP Setting
PJSIP 5061 TLS, Let’s encrypt certificate SSL method tls 1.2
RTP Settings 10,000-20,000
Trunks
Trunk using 0.0.0.0/TLS, media encryption SRTP via in-SDP.
Extensions
Transport 0.0.0.0/tls
Media Use Received Transport NO
RTP Symmetric YES
Media Encryption SRTP Via in-SDP
Allow Non-Encrypted Media (Opportunistic SRTP) NO
Enable DTLS NO
Phone
Port 5061, Enable and Enforce SRTP, TLS/TCP enabled, certificate verification disabled.
ATA
TLS, port 5061, SRTP enabled and enforced, ciphers enabled, RTP 10,000-20,000