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* <[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.
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