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

[Bogdan-Andrei Iancu]

Hi Stefano,

OK - perfect :)

Regards,
Bogdan

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