[OpenSIPS-Users] [Fwd: Openser with Audiocodes]
bogdan at voice-system.ro
Mon Oct 13 10:59:29 CEST 2008
Try to check out the IP addresses in SDP (INVITE + 200OK) to see if the
RTP is correctly routed (via mediaproxy).
Stefano Palleschi wrote:
> 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!
> 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.
>> 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
>>> 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.
>>> Users mailing list
>>> Users at lists.opensips.org
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