Hi!
I got strange behaviour of FreePBX (asterisk 13.38.1).
I have two PBX-s, connected with sip trunk (PBX-1 and GATEWAY-1).
When user, registered on PBX-1, calls to external Number, SIP and RTP traffic goes through GATEWAY-1. First, the Playback greeting sounds to callee. After it caller answers. RTP SSRC changed and near first 10 seconds gaps in RTP voice appear (300-350 RTP packets lost). On call leg from PBX-1 all RTP packets arrives into GATEWAY-1. But on call leg into ISP provider from GATEWAY-1 there are packet loss.
Asterisk debug says (near 300-350 such records in log):
res_rtp_asterisk.c: Received frame with no data for RTP instance '0x7fb248a2b578' so dropping frame
When recording is enabled, additional debug info are presented:
audiohook.c: Flushing audiohook 0x7fb2489f5c00 so it remains in sync
I was looking for similar issue
ASTERISK-25734
But in that, author got issue when recording calls.
In my case, whether or not recording, issue exists anyway.
Such a behaviour is not constant. Sometimes all working stable, but in most cases issue appears.