[OpenSIPS-Users] One way audio with AudioCodes Mediant 2000 and NAT

[Bogdan-Andrei Iancu]

Hi Julian,

You can still handle the NAT wih COMEDIA even for T.38, but you have to 
handle the re-INVITE also . In your scenario, who is generating the 
re-INVITE?

Regards,
Bogdan

Julian Yap wrote:
> The full story is that I was looking to get T.38 working behind NAT.
>
> Unfortunately, no matter what I tried, it wouldn't work behind NAT.  I
> had the initial INVITE (G.711) working fine but when there was the
> T.38 re-INVITE, the RTP media would connect up fine but just wouldn't
> negotiate properly with T.38.  Very strange as it worked fine with the
> UA behind a public IP.
>
> In the end, I implemented RTPProxy and T.38 works fine behind NAT.
>
> - Julian
>
> On Sun, Feb 15, 2009 at 1:25 AM, Bogdan-Andrei Iancu
> <bogdan at voice-system.ro> wrote:
>   
>> Hi Julian,
>>
>> That is cool - in this way you save a lot of bandwidth and processing power
>> with media relaying.
>>
>> Regards,
>> Bogdan
>>
>> Julian Yap wrote:
>>     
>>> Hi all,
>>>
>>> I eventually played around with the Audiocodes box and enabled some
>>> settings so it worked with Comedia support.
>>>
>>> Thanks,
>>> Julian
>>>
>>>
>>> On 2/10/09, Bogdan-Andrei Iancu <bogdan at voice-system.ro> wrote:
>>>
>>>       
>>>> HI Julian,
>>>>
>>>> If it has, you can actually force it by adding "direction=active" into
>>>> SDP as indication. See "fix_nated_sdp("1") :
>>>>
>>>>  http://www.opensips.org/html/docs/modules/1.4.x/nathelper.html#id270439
>>>>
>>>> Regards,
>>>> Bogdan
>>>>
>>>> Julian Yap wrote:
>>>>
>>>>         
>>>>> Thanks all. I'll check to see if the AudioCodes gateway does have
>>>>> comedia support.
>>>>>
>>>>> That clarifies some half baked NAT/RTP knowledge in my head.
>>>>>
>>>>> - Julian
>>>>>
>>>>>
>>>>> On 2/10/09, Bogdan-Andrei Iancu <bogdan at voice-system.ro> wrote:
>>>>>
>>>>>
>>>>>           
>>>>>> Hi Olle,
>>>>>>
>>>>>> Johansson Olle E wrote:
>>>>>>
>>>>>>
>>>>>>             
>>>>>>> 10 feb 2009 kl. 12.25 skrev Iñaki Baz Castillo:
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>>>> 2009/2/10  <julianokyap at gmail.com>:
>>>>>>>>
>>>>>>>>
>>>>>>>>                 
>>>>>>>>>> You don't know if RtpProxy should be running, does it mean you are
>>>>>>>>>> trying to use it or not? I don't want to spend time inspecting what
>>>>>>>>>> you want to do by reading your config, sorry.
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>                     
>>>>>>>>> Yeah, I'm trying not to run RTPProxy. After more testing, I'm
>>>>>>>>> thinking I may
>>>>>>>>> need to.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>                   
>>>>>>>> You cannot decide if you need RtpProxy or not based on testing, it's
>>>>>>>> pure theory:
>>>>>>>>
>>>>>>>> A RTP proxy is NOT needed when (assuming the proxy has in the public
>>>>>>>> internet):
>>>>>>>>
>>>>>>>> - Both caller and callee have public IP or use STUN.
>>>>>>>> - Both caller and callee are in the *SAME* private LAN.
>>>>>>>> - The caller is in a private LAN and the callee has public IP and
>>>>>>>> supports Comedia mode (typical in some media servers and gateways).
>>>>>>>> - The callee is in a private LAN and the caller has public IP and
>>>>>>>> supports Comedia mode.
>>>>>>>>
>>>>>>>>
>>>>>>>> A RTP proxy is needed when:
>>>>>>>>
>>>>>>>> - Caller is in private LAN (with no STUN) and callee in public
>>>>>>>> internet (and not supporting Comedia).
>>>>>>>> - Caller and callee are in different private LAN's with no STUN.
>>>>>>>>
>>>>>>>>
>>>>>>>>                 
>>>>>>> I would like to add that it's the device that can't receive audio that
>>>>>>> needs the RTP proxy to get incoming audio.
>>>>>>>
>>>>>>> If both devices are on private IP's, there's going to be two
>>>>>>> RTP proxys involved if they're on different SIP networks.
>>>>>>>
>>>>>>> Each SIP service needs an RTP proxy for supporting their
>>>>>>> local users.
>>>>>>>
>>>>>>> To simplify:
>>>>>>>
>>>>>>> - If my user is on a private IP and sends an INVITE, add RTP proxy
>>>>>>> handling to the INVITE
>>>>>>>
>>>>>>> - If my user receives a call and sends a 200 OK, add RTP proxy
>>>>>>> handling to the 200 OK
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>> This logic is simple but not efficient....Theoretically, if a call has
>>>>>> already a leg in public net, there is not need for a media relay for
>>>>>> traversing the nat.
>>>>>>
>>>>>> The only requirement is that all the devices to support symmetric media
>>>>>> (comedia support).
>>>>>>
>>>>>> So, after the caller proxy forced RTPproxy, the callee should not do
>>>>>> the
>>>>>> same because the SDP already have a public IP, the nat traversal works
>>>>>> even if the callee is behind a nat.
>>>>>>
>>>>>> Regards,
>>>>>> Bogdan
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Users mailing list
>>>>>> Users at lists.opensips.org
>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>
>>>>>>
>>>>>>
>>>>>>             
>>>>>           
>>>>         
>>>       
>>     
>
>