[Asterisk-Users] RTP not thru asterisk

[Rich Adamson]

> I want to make sure that RTP is not going thru my asterisk.
> 
> I read you should avoid in the dial commands:
> m   music while ringing
> t,T transfer calls from caller and called party
> 
> What else do I need to take care?
> 
> remote phone ===> registered to local asterisk ===>  calling remote gateway
> 
> should have the RTP     remote phone ===(RTP)==> calling remote gateway

That should be doable "if" the remote phone and remote gateway are
truly accessible to each other on registered IP addresses (and not 
behind nat boxes or firewalls).

If there is a nat box or firewall involved with either device, you
might be able to get it to function, but it can be very difficult
without knowing exactly how the nat boxes function and exactly how
the sip phones operate. Part of that understanding is knowing exactly
which udp ports the phones & gateway use for the negotiated rtp
session. (E.g., Xten uses udp ports in the 8000 range, cisco's in the
16384-32766 range, asterisk in the 10000-20000 range, etc. There
is no industry or rfc standard.)

In addition, the phones and gateway will likely need some sort of 
stun support or static parameters to define the true external ip address. 
Not all phones support that.

If a nat box is present, the _only_ way for you to diagnose rtp port
negotiation problems is to use a packet sniffer at the phone and/or
gateway locations as the two will attempt to negotiate a set of rtp
ports on their own.  Don't bother posting "it don't work" to the list
without such traces as absolutely no one is going to be able to help.

Also keep in mind that not all nat boxes operate the way that you 
think they should. Some will actually do port mapping and the OEM
doesn't necessarily document that in their spec sheets.