Firewall is not sip aware, rtprelay via box in dmz

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________________________________
Van: Users <users-boun...@lists.opensips.org> namens Mark Allen 
<m...@allenclan.co.uk>
Verzonden: Wednesday, January 13, 2021 5:08:27 PM
Aan: OpenSIPS users mailling list <users@lists.opensips.org>
Onderwerp: [OpenSIPS-Users] OpenSIPS 3.1 & NAT issues

Hi all - I've been banging my head against this but not succeeding.

Our setup...

UAC           192.168.x.x
  |
Router        5.x.x.x
  |
(internet)
  |
Firewall      46.x.x.x maps
  |           directly to
OpenSIPS      192.168.x.x      Mid-registrar
  |
Asterisk      192.168.x.x


Current situation:
- UAC can register on Asterisk via OpenSIPS
- UAC can call destination registered on Asterisk on local n/w to Asterisk box
- Destination extension rings and can pick up call
- There is no audio either way & call drops after about 30 secs (Asterisk kills 
call with "Requested channel not available" because not RTP traffic is reaching 
destination)

I have tried passing audio through Mediaproxy on OpenSIPS box but with no 
success. Using Wireshark I can see RTP traffic initiated at both ends, but it 
doesn't reach the other end either way.

Is there some definitive guide to setting this up correctly or are there 
specific steps that I need to follow?

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