Hi Olle,
even if the devices register with OpenSER and you use Asterisk as B2BUA behind the proxy, it still works if: 1) handle SIP signalling NAT mangle in OpenSER 2) disable the can-reinvites option in Asterisk. 3) do not use rtpproxy.
From Asterisk perspective, do you see any issues in this scenario?
thanks and regards, bogdan
Olle E Johansson wrote:
30 nov 2006 kl. 10.42 skrev 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).
Depends on the setup really, Bogdan.
If your devices are registering with OpenSER, you need RTP proxy. If they're registering with Asterisk and calling through Asterisk, asterisk can handle media and NAT.
If you have NATs, you should really disable can-reinvites since you don't want ASterisk to set up media stream that will fail.
/O
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 topoint.
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