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

[Johansson Olle E]

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

/O
-------------- next part --------------
A non-text attachment was scrubbed...
Name: smime.p7s
Type: application/pkcs7-signature
Size: 2207 bytes
Desc: not available
Url : http://lists.opensips.org/pipermail/users/attachments/20090210/b129f8ec/attachment.bin