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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