RTP Packet Loss in Bridge after playback greeting

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
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.

Also posted as Asterisk misses RTP outgoing packets in Bridge - #2 by david551 - Asterisk SIP - Asterisk Community

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