10 feb 2009 kl. 12.25 skrev Iñaki Baz Castillo:

2009/2/10  <[email protected]>:
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

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