I have an Asterisk 12 installation talking to an Avaya over TLS but I need to secure the audio streams using SRTP.

The SDP in the INVITE from the Avaya contains:
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:a84nzjof8LyBy0xcGEi1fGkkEv8RCLG5E9IcofL9 UNENCRYPTED_SRTCP

And the Asterisk logging shows:
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 127
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 127
[Mar 21 17:39:18] WARNING[2236][C-00000035]: sdp_srtp.c:240 ast_sdp_crypto_process: Unsupported crypto parameters: UNENCRYPTED_SRTCP
[Mar 21 17:39:18] WARNING[2236][C-00000035]: chan_sip.c:10620 process_sdp: Rejecting secure audio stream without encryption details: audio 2076 RTP/SAVP 8 18 127

<— Reliably Transmitting (no NAT) to —>
SIP/2.0 488 Not acceptable here

Now Asterisk is only reporting warnings but the call doesn’t get set up. Is lack of support for UNENCRYPTED_SRTCP the problem or is there something more fundamental wrong?

Thanks in advance,

Lack of support is the problem. Asterisk is unable to establish an encryption key but has been told it must use one.

Actually, that rather sounds like a statement that the RTP won’t be encrypted!