[Asterisk-Users] Asterisk and RTP streams

[Michael D Schelin]

Sherwood, I have never known the RTP audio to be on only one port in 
sip.  I believe it's always on 2.  The one way audio is always a 
nat/firewall problem in sip.
 
Sherwood McGowan wrote:

> 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)
>  
> Thank you all in advance for your thoughts and comments. I apologize 
> in advance if I missed something that was publicly available.
>  
> Sherwood McGowan
>
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