[asterisk-users] Delayed RTP start

[Jeff LaCoursiere]

Hi,

I am debugging an issue that unfortunately involves two NAT instances - 
the firewall at our customer site, and the firewall in front of their 
Amazon instance.

I have an HTEK phone at the customer site registering to the public 
address of the Amazon instance running asterisk (and FreePBX).  This 
seems to work fine, and it can call local services (like fpbx *65 to 
read back the extension) with no problems.

If it tries to make an outbound outside call, the remote phone (my cell 
for example) rings, I answer it, but there is no audio in either 
direction for nearly exactly 16 seconds, every time.  Then audio starts 
in both directions without issue.

I did a packet trace on the phone itself and see 16 seconds of outbound 
RTP with no inbound, then suddenly RTP in both directions until the call 
ends.

I did a packet trace on the asterisk side and see the call setup, then 
sixteen seconds of nothing (??), then RTP starts in both directions.

In the asterisk console I see this bit of interestingness:

[2019-07-24 13:21:02] DEBUG[1890]: chan_sip.c:29923 
__start_session_timer: Session timer started: 78 - 
710779684e62266a77b047b31e4
261da at 10.0.116.239:60060 1768000ms
     -- SIP/ast01-0000024b answered SIP/7222-0000024a

[.....snip.....]

[2019-07-24 13:21:02] DEBUG[17928][C-000001f1]: bridge_native_rtp.c:660 
native_rtp_bridge_compatible_check: Bridge '3bfbf253-d34f-
45e2-abc3-75e590d81739' can not use native RTP bridge as channel 
'SIP/ast01-0000024b' has DTMF hooks

[.....snip.....]

[2019-07-24 13:21:18] DEBUG[18003][C-000001f1]: res_rtp_asterisk.c:4179 
ast_rtp_write: Ooh, format changed from none to ulaw
[2019-07-24 13:21:18] DEBUG[18003][C-000001f1]: res_rtp_asterisk.c:4019 
rtp_raw_write: Starting RTCP transmission on RTP instan
ce '0x7fe17426e7c8'


So my main question is, what would cause a sixteen second delay before 
the codec could be decided?

This is Asterisk 13.25.0 on the customer Amazon instance... the "ast01" 
peer is ours also - one of our external gateways, also running 13.25.0.

Thanks,


-- 
Jeff LaCoursiere
StratusTalk, Inc.
703 496 4990 x108
815 546 6599 cell