[OpenSIPS-Users] OpenSIPS 3.1 & NAT issues

[Mark Allen]

Thanks for the responses. They helped me exclude some things. I've managed
to make progress and pinned down the lack of audio to a misconfiguration of
Mediaproxy. Two-way audio through double-nat / firewall is working but goes
silent after about 60 seconds connected and Asterisk kills the connection
31 seconds later due to lack of RTP activity for the last 31 seconds

On Thu, 14 Jan 2021 at 12:00, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Check out what IPs are offered in the SDPs in asterisk. Make sure they’re
> both public IPs.
> If you only have 1 asterisk, forwarding the rtp port range configured in
> asterisk from the firewall to asterisk should do it.
>
>
> On Thu, 14 Jan 2021 at 08:23, Mark Allen <mark at allenclan.co.uk> wrote:
>
>> Thanks Adrian
>>
>> The firewall has SIP-ALG disabled and just forwards ports from externally
>> to where they need to be internally - so ports 5060 and 10000 - 65535 of
>> 46.x.x.x are mapped to 192.168.x.x (the OpenSIPS box)
>>
>> On Wed, 13 Jan 2021 at 17:32, Adrian Georgescu <ag at ag-projects.com>
>> wrote:
>>
>>> 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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>>>
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> --
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
> _______________________________________________
> Users mailing list
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>
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