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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.<br>
<br>
Sherwood McGowan wrote:
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<div><span class="562314000-01102005"><font face="Arial" size="2">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.</font></span></div>
<div><span class="562314000-01102005"></span> </div>
<div><span class="562314000-01102005"><font face="Arial" size="2">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:</font></span></div>
<div><span class="562314000-01102005"></span> </div>
<div><span class="562314000-01102005"><font face="Arial" size="2">1.
Can Asterisk be told to send the RTP stream for incoming and outgoing
always on the same set of ports?</font></span></div>
<div><span class="562314000-01102005"><font face="Arial" size="2">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)</font></span></div>
<div><span class="562314000-01102005"></span> </div>
<div><span class="562314000-01102005"><font face="Arial" size="2">Thank
you all in advance for your thoughts and comments. I apologize in
advance if I missed something that was publicly available.</font></span></div>
<div><span class="562314000-01102005"></span> </div>
<div><font face="Arial" size="2">Sherwood McGowan</font></div>
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