Dear all,
I am using an Asterisk since several months now. I have a question concerning the RTP-stream. What I recognized is, that the RTP-stream is running through the Asterisk and is forwarded to my media gateways.
Is it possible to tell the Asterisk to handle only the signalling? RTP should be seperatley streamed by my media gateways.
If possible, how can I configure this?
best regards
If you’re using sip and * 1.4 check in sip.conf the usage of the parameters canreinvite and directrtpsetup.
Cheers.
Marco Bruni
Thank you for your quick reply.
I use version 1.2.23. Are the parameters the same?
I already tried with canreinvite but without success. I guess the parameter must be enabled on both sides? On my Asterisk and on the device where i receive the traffic from?
Best regards
Felix Tröger
Sorry, I don’t know.
Cheers.
Marco Bruni
I am wondering why v1.4 would make a difference. canreinvite existed before 1.4 so it should work. i have heard issues about NAT being a problem, such as so forth.
[quote=“lix”]Thank you for your quick reply.
I use version 1.2.23. Are the parameters the same?
I already tried with canreinvite but without success. I guess the parameter must be enabled on both sides? On my Asterisk and on the device where i receive the traffic from?
Best regards
Felix Tröger[/quote]