30 Nov
2006
30 Nov
'06
10:42 a.m.
Hi John,
actually if you use Asterisk, there is no need for using RTPproxy as
Asterisk is able to cope with nated rtp by itslef (using Comedia).
regards,
bogdan
John Peters wrote:
ONsip has some tips for handling re-INVITEs with
rtpproxy:
http://siprouter.onsip.org/doc/gettingstarted/ch08s02.html#rtp_loose_route
<http://siprouter.onsip.org/doc/gettingstarted/ch08s02.html#rtp_loose_route>
Advises to use force_rtp_proxy(l) on reinvites.
On 11/29/06, *John Peters* <petersprc(a)gmail.com
<mailto:petersprc@gmail.com>> wrote:
Not sure why that's happening. Probably setting canreinvite=no on
the asterisk side will eliminate the re-INVITEs as a temporary
solution, but still would like to know what is happening...
wrote:
Sometimes, a calls b and b hears a, and a hears b
for a second
but a second
INVITE comes to phone B that causes it to
redirect rtp to be
point to point.
Sometimes there is no audio.
Sometimes, everything works fine.
At one point, rtp from a was going to asterisk,
but asterisk was
not sending
the rtp on to b, and b was trying to send traffic
point to point.
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