[OpenSIPS-Users] Opensips 1.10 NAT

[Rodrigo Ferreira]

Forcing the traffic through RTPPROXY worked, but why isnt working the
nat_uac_test?

Kinda weird



Atenciosamente.
Eng.° Rodrigo Ferreira
ITIL v3 Certified

<http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901>


2013/10/4 Mike Tesliuk <mike at ultra.net.br>

> well, probably you softphone/ip phone, is using some kind of stun or other
> kind of nat features, so, nothing come to be detected, this can happen, so,
> if you will be ever using nat, you can force the rtpproxy without nat
> detection, this will solve your problem, if you read the documentation (
> http://www.opensips.org/html/docs/modules/1.9.x/nathelper.html#id292854 )
> you can see that this test are made over rfc1918 or different ip address
> from via and signalling
>
> The problem is probably the fact that when the call is stablished, the
> media cannot traverse, you have the correct ip information on sdp but the
> router does not permit the session to be opened, so, do a test forcing the
> use of rtpproxy without the nat detection, just force all trafic throught
> rtpproxy
>
>
> 2013/10/4 Rodrigo Ferreira <rsferreira08 at gmail.com>
>
>> I did that Mike ..
>>
>> my "nat_uac_client" isnt passing in any verification ...
>>
>> I did this ..
>>
>>         if ( nat_uac_test("1") ) xlog("UAC TEST = 1");
>>
>>         if ( nat_uac_test("2") ) xlog("UAC TEST = 2");
>>
>>         if ( nat_uac_test("4") ) xlog("UAC TEST = 4");
>>
>>         if ( nat_uac_test("8") ) xlog("UAC TEST = 8");
>>
>>         if ( nat_uac_test("16") ) xlog("UAC TEST = 16");
>>
>>         if ( nat_uac_test("32") ) xlog("UAC TEST = 32");
>>
>>         if ( nat_uac_test("64") ) xlog("UAC TEST = 64");
>>
>> in the beginning of the script, to see what is happening to my NAT, and i
>> got nothing.
>>
>>
>>
>> Atenciosamente.
>> Eng.° Rodrigo Ferreira
>> ITIL v3 Certified
>>
>> <http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901>
>>
>>
>> 2013/10/4 Mike Tesliuk <mike at ultra.net.br>
>>
>>> That howto is just a sample (with a lot of comments) to better
>>> understand of nat configuration (over my understand offcourse), so, you can
>>> check and compare with your configuration to identify about something
>>> missing
>>>
>>>
>>>
>>>
>>> 2013/10/4 Rodrigo Ferreira <rsferreira08 at gmail.com>
>>>
>>>> Yes I did Mike,
>>>>
>>>> and my SIP messages are ok.
>>>>
>>>> I will take a look at your tutorial.
>>>>
>>>> tks
>>>>
>>>>
>>>>
>>>> Atenciosamente.
>>>> Eng.° Rodrigo Ferreira
>>>> ITIL v3 Certified
>>>>
>>>> <http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901>
>>>>
>>>>
>>>> 2013/10/3 Mike Tesliuk <mike at ultra.net.br>
>>>>
>>>>> Did you try to made some debug rodrigo ? maybe some rule is missing on
>>>>> your route script
>>>>>
>>>>> i made a tutorial over version 1.9 that you can check
>>>>>
>>>>> [portugues]
>>>>> http://opensips.com.br/wiki/index.php?title=Opensips_NAT_Script_com_RTPproxy
>>>>> [english]
>>>>> http://opensips.com.br/wiki/index.php?title=Oopensips_Nat_script_with_RTPPROXY_-_English
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> 2013/10/3 Rodrigo Ferreira <rsferreira08 at gmail.com>
>>>>>
>>>>>>  Hi guys,
>>>>>>
>>>>>> After a long time without using Opensips (almost a year) I tried to
>>>>>> install the opensips 1.10 and everything went well BUT when I make a call,
>>>>>> there's no audio, I know that is something because of NAT, but I have the
>>>>>> nathelper and rtpproxy configuration on my opensips.cfg.
>>>>>>
>>>>>> There's anything else that I could take a look at?
>>>>>>
>>>>>> Thanks
>>>>>>
>>>>>>
>>>>>> Atenciosamente.
>>>>>> Eng.° Rodrigo Ferreira
>>>>>>  ITIL v3 Certified
>>>>>>
>>>>>> <http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901>
>>>>>>
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