[OpenSIPS-Users] [Fwd: Openser with Audiocodes]

[Stefano Palleschi]

Hi Bogdan,
thanks for your reply.

Yes, with Asterisk I use mediaproxy also, and when UA is behind nat the 
rtp packets flow through Openser (obviously).
The only one difference between two scenarios is that when using 
Asterisk the MGC there isn't.
With Asterisk I have only one server (Asterisk) that allow SIP signaling 
and termination.
In Audiocodes scenario I have two servers interested, MGC for SIP 
signaling and Audiocodes Mediant 3000 for termination.
In my openser.cfg I have only changed  the Asterisk IP address with the 
MGC IP address in the rewritehostport() function.
Do I have to add anything else? ... I think not!

Regards,
Stefano.



Bogdan-Andrei Iancu ha scritto:
> Hi Stefano,
>
> When using Asterisk, do you also use mediaproxy? If no, maybe Asterisk 
> is automatically doing COMEDIA (direction=active in SDP) and the 
> Audiocodes  not.
>
> Regards,
> Bogdan
>
>
>
> Stefano Palleschi wrote:
>> Hi all,
>> I'm trying to use openser with Audiocodes 3000 as pstn gateway.
>> This is my scenario:
>>
>> UA----------->  openser-------> MGC-------->Audiocodes-------> PSTN.
>>
>> When I use Asterisk as PSTN gateway I haven't  any problem for rtp 
>> traffic, even when UA is behind nat.
>> Using Audiocodes I noticed that the rtp traffic doesn't flow from 
>> Audiocodes to Openser (or viceversa), but the rtp flow  bypasses 
>> openser.
>> This cause problems when UA is behind nat because mediaproxy doesn't 
>> fix nat.
>> All my outbound calls are redirect to MGC, and in my route section 
>> the Audiocodes's IP address doesn't compare.
>> My questions are:
>> is this an Audiocodes problem? .... or I can adjust openser 
>> configuration for fix that?
>>
>> Thanks for your attention.
>> Regards,
>> Stefano.
>>
>>
>>
>>
>>
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>>   
>
>
>