How separate RTP traffice and sip signalling with asterisk?

Can I setup an asterisk server as only sip siganlling and another asterisk as only rtp media server process media?

Not really. You can configure Asterisk to reinvite so media doesn’t flow directly, but it’s not designed or written to allow components to run separately.

Thanks for your reply.
Server A: sip signalling
Server B: media server.
On Server A: sip.conf I set media_address=
and all sip trunk and sip device directmedia=no.
It’s no media on both led. From CLI it can not set soure rtp.
When I comment media_address para on server A. Call is normally.
On server B what paras is needed to config?

Configure for what? The media_address option changes the IP address that is placed into SDP. It does not provide functionality to split Asterisk up and have one do signaling and the other media without signaling, that doesn’t exist.

Is it mean that asterisk can not act like only sip signalling or rtp server separately?

It can not. It only does both, with the option of allowing media to flow directly between two SIP legs.

1 Like

Thank you very much :slightly_smiling_face:

If you need to be able to split signaling and rtp take a look at kamailio/opensips. Just realize they are both sipproxies and not b2bua.

1 Like

On kamailio, I configured successfully. Many thanks

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.