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

[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.


>> > Basically, I'm getting one way audio :(
>>
>>
>> Inspect, by yourself, the SDP arriving to the UAS in INVITE and the
>> SDP arrives to the UAC in the 200 OK. Do they contain a media address
>> reachable from each UA?
>
> I'll check this out.

It's really easy:
a) In the caller check the media address in the 200 OK SDP. Can you
ping that IP from the caller?
b) In the callee check the media address in the INVITE SDP. Can you
ping that IP from the callee?

If a) or b) are "no", then you get one-way-audio.
If both are "no", then you get no audio at all.


-- 
Iñaki Baz Castillo
<ibc at aliax.net>