ONsip has some tips for handling re-INVITEs with rtpproxy:
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@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.