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

[Stefano Palleschi]

Hi Bogdan,
I've done several tests with tcpdump and wireshark.
In SDP negotiation I didn't find  any difference and mediaproxy  worked 
fine in both situations.
I found  only one difference when I was using X-lite. In the wireshark 
output file there is a icmp message with info: " Destination unreachable 
(port unreachable)"  exchanged  from Openser to Audiocodes and from 
Openser to UAC public IP address.
I adjusted X-lite configuration (domain:listen port, use rport,...) for 
this situation, now all works fine.
Thanks for your help.

Regards,
Stefano



Bogdan-Andrei Iancu ha scritto:
> 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
>>>>>>
>>>>>>   
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