[Asterisk-Users] Why, why, why???

[Brian Wright]

Noah Miller wrote:

> It looks like it should work, but I don't use grandstream phones.  Has 
> anybody else had this problem?  Have you tried the latest version of 
> the Grandstream firmware - I know older versions had a number of 
> problems.
>
We have Grandstream SIP phones with the latest firmware versions and 
have also have this problem.  It appears to be something to do with RTP, 
I believe.  I don't know exactly what (simply because I don't know much 
about RTP as yet), but the packets don't seem to reach the Grandstream 
from the other phone.  The phones appear to work correctly when located 
on the same LAN segment.  But, when one is placed behind a NAT router, 
the dynamic changes and one-way audio seems to happen frequently.  I've 
tried mucking with some of the Asterisk configuration features in 
sip.conf to keep the phones from connecting directly to each other  as 
well as altering the RTP allocation ports without success.

> Thanks,
> Noah
>
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-- 
Brian Wright <bwright at boomerang.com>