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

[Bogdan-Andrei Iancu]

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
>>
>>
>>
>>
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>>
>>     
>
>