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

[Bogdan-Andrei Iancu]

Hi Stefano,

have you spotted what is the difference in the SDP negotiation.

Regards,
Bogdan

Stefano Palleschi wrote:
> 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
>>>>>
>>>>>   
>>>>
>>>>
>>>>
>>>
>>>
>>
>>
>>
>
>