[Users] two invites, different session description => broken

[Bogdan-Andrei Iancu]

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 at gmail.com 
> <mailto:petersprc at 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. 
>
>
>------------------------------------------------------------------------
>
>_______________________________________________
>Users mailing list
>Users at openser.org
>http://openser.org/cgi-bin/mailman/listinfo/users
>  
>