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* <[EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]>> 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.
------------------------------------------------------------------------
_______________________________________________
Users mailing list
[email protected]
http://openser.org/cgi-bin/mailman/listinfo/users
_______________________________________________
Users mailing list
[email protected]
http://openser.org/cgi-bin/mailman/listinfo/users
