[Asterisk-Users] Asterisk and RTP streams (just bumping)

[Olle E. Johansson]

Sherwood McGowan wrote:
> Bumping, just in case it got lost in the shuffle today... I think this is an
> important thing to be able to do.
> 
> Subject: [Asterisk-Users] Asterisk and RTP streams
> 	
> Guys, I've been poking around trying to find a good answer for this via
> voip-info, google, etc... Haven't found anything that helps, so maybe you
> mates could.
> 	 
> A lot of my customers are using Linksys UAs (router/ATA PAP2) and some using
> Sipura SPA-2002s. Every once in a while, the customer will get one-way
> audio. I've read that this is commonly caused by the outgoing RTP port not
> being the same as the incoming RTP port. A lot of other devices (I found
> info on forcing Xten to do it) can be forced to use the same port for both,
> but these devices don't have an option (that I've been able to find, even in
> the provisioning configs) to do this. So, my question is two-fold:
> 	 
> 1. Can Asterisk be told to send the RTP stream for incoming and outgoing
> always on the same set of ports?
> 2. Does anyone know something that I'm missing for the above mentioned
> devices? They're all the 2 line version of the ATA and/or router configs
> (wireless and wired)
> 	 
If you turn on nat=yes this will affect both SIP and RTP. Asterisk will
then send to the same address as we receive RTP from, this is called
symmetric RTP. THere's no way we can affect the address port range that
the device tell us to send to, but we can ignore that in the case there
is a NAT in between and send to whatever address the device sends audio
from.

The RTP port address we receive RTP on *from* the device is settable in
rtp.conf.

/Olle