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

Stefano Palleschi stefano.palleschi at okcom.it
Mon Oct 13 11:30:15 CEST 2008


Hi Bogdan,
thanks again for your reply.
I was going to answer you to your previous reply.
Luckily there isn't a configuration problem but all depends to the sip 
client used.
Using X-lite 3.0 (free version) the rtp traffic doesn't flow , but with 
Linksys SPA 2102 I don't have any problem.
I'm going to try X-lite without nat for check out  if the problem 
disappears.

Thanks again.
Regards,
Stefano


Bogdan-Andrei Iancu ha scritto:
> Hi Stefano,
>
> Try to check out the IP addresses in SDP (INVITE + 200OK) to see if 
> the RTP is correctly routed (via mediaproxy).
>
> Regards,
> Bogdan
>
>
> 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!
>>
>> 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.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>   
>>>
>>>
>>>
>>
>>
>
>
>




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