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

[Julian Yap]

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
>>>
>>>
>>>
>>>
>>> _______________________________________________
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>>>
>>>
>>
>>
>
>