30 Nov
2006
30 Nov
'06
9:42 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@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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