Intermittent one way audio when transferring calls or resuming from hold PJSIP SRTP

I’ve been dealing with this issue for several months.

Free PBX 16 running Asterisk 20 (also occurred on v18)
Endpoints are Polycom VVX 410s
Using PJSIP w/ TLS and SRTP
SRTP Replay Protection: No

When a phone places a call on hold, we will occasionally see the following errors in the PBX logs:
VERBOSE[19720][C-00000a1d] res_srtp.c: SRTCP unprotect failed on SSRC 1875483051 because of authentication failure

The phone logs show:

0424113855|srtp |4|00|srtp: srtp_unprotect_rtcp: error: 10 - failed seq number 1 for replays, SSRC=0xdbba98d
0424113900|srtp |4|00|srtp: srtp_unprotect_rtcp: error: 10 - failed seq number 2 for replays, SSRC=0xdbba98d
0424113905|srtp |4|00|srtp: srtp_unprotect_rtcp: error: 10 - failed seq number 3 for replays, SSRC=0xdbba98d
0424113910|srtp |4|00|srtp: srtp_unprotect_rtcp: error: 10 - failed seq number 4 for replays, SSRC=0xdbba98d
0424113914|srtp |4|00|srtp: srtp_unprotect: auth failure - seq=0x8f91, ROC=1

This happened a lot more often before we disabled SRTP Relay Protection, but the issue still occurs.

Surely I’m not the only one who has encountered this?

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