[OpenSIPS-Users] OpenSIPS 3.1 & NAT issues

[Johan De Clercq]

Firewall is not sip aware, rtprelay via box in dmz

Outlook voor iOS<https://aka.ms/o0ukef> downloaden
________________________________
Van: Users <users-bounces at lists.opensips.org> namens Mark Allen <mark at allenclan.co.uk>
Verzonden: Wednesday, January 13, 2021 5:08:27 PM
Aan: OpenSIPS users mailling list <users at 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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