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

Julian Yap julianokyap at gmail.com
Tue Feb 10 19:10:21 CET 2009


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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