Sip sign but no RTP in the total asteriskS

Asterisk Asterisk Support
1

I have that:

ASTERISK_CLIENT1<–>
-----------------------------ASTERISK_SRV1 <–> PROXY_SIP <–> ASTERISK_SRV2
ASTERISK_CLIENT2<–>

I need that, if ASTERISK_CLIENT1 call ASTERISK_CLIENT2, sip signaling go to ASTERISK_SRV2
but RTP don’t go more than ASTERISK_SRV1

is it possible to do it?

thank you

francois

2

Assuming that you meet the other criteria for external bridging, the proxy and server 2 should be taken out of the loop, but it’s not clear whether or not one will still end up with a loopback on the server one interface.

What is unlikely to happen is that that loop back is moved deeper into server one, but it is possible that the two clients will be re-invited to send RTP directly to each other.

The easiest way is to actually try it.

3

the solution could be canreinvite=yes in the server2 ?