[OpenSIPS-Users] Opensips 1.10 NAT

[Mike Tesliuk]

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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>>>>> Users mailing list
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>>>>>
>>>>>
>>>>
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