[OpenSIPS-Users] OpenSIPS 3.1 & NAT issues

[Adrian Georgescu]

Google search for SIP ALG problem to see if this is relevant for your case.

Regards,
Adrian


> On 13 Jan 2021, at 13:08, Mark Allen <mark at allenclan.co.uk> wrote:
> 
> 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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