My scenario: asterisk with sip clients and the same asterisk registering as client to voip provider.
SIP CLIENTS ---------> ASTERISK ----------> VOIP PROVIDER
—> indicates “REGISTER”
I want sip clients to be able to call voip provider via asterisk extensions, but RTP traffic must not go via asterisk, the audio must go direct between sip client and voip provider.
Sip clients may be behind NAT, but voip provider never.
We know that sip protocol permits it, but is there a way to force it on asterisk?
More, is it possible only RTP go directly, but SIP maintain via asterisk, for CDR?
My motivation is simple: lower bandwidth on asterisk link and call latency.
Any hint or pointer is welcome.