RTP Stream for internal calls with Hosted FREEPBX


I’m going to deploy about 15 SIP phones in a company with a hosted PBX (Freepbx in cloud).

Is there a way that internal calls (between the SIP Phones) can be processed without transiting though the cloud PBX ?

Exemple : The SIP phones are is the same Local Network.

For internal calls, Instead of : Phone 1 <RTP> Cloud PBX <RTP> Phone 2

I want : Phone 1 <RTP> Phone 2

So the RTP stream will stay in LAN and not use internet bandwith for internal calls.

Thank you

Technically yes, but I’ve never done it. You would enable re-invite in Asterisk so that the media goes direct to endpoints instead of thru the PBX. Unless you are using satellite or dial-up internet, it is not worth the effort for a small number of concurrent calls.

Thank you. I will try and give you a feedback.

I’ve to turn “YES” in both ?

  • Chan_sip Settings “Reinvite Behavior” to YES
  • Advanced Settings “SIP canrenivite” to YES


Good luck - for the price of a “server” that could do this in the local network, you’re going to be hard pressed to make this work in a cost-effective time frame.

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In fact it is not possible to use canreinvite or directmedia options because there is NAT on extensions so Freepbx only see the public IP (WAN) for an extension and not the LAN IP.

A solution is to set up a VPN between the customer routeur and Freepbx so Freepbx would have a IP in the LAN.

Too complicated. I will put a local serveur or upgrade the WAN bandwith.