Hi Daniel

Thanks for your help on this , changing direction attr on rtpengine_offer fixed my issues

Best Regards

Gerry Kernan

From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Daniel-Constantin Mierla
Sent: 19 April 2017 10:28
To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org>
Subject: Re: [SR-Users] Kamailio rtpengine sdp

Hello,

you have to instruct rtpengine to do bridging between the two network interfaces.

Can you show the INVITE and the 200ok with all SDPs, both sides (incoming and outgoing to/from kamailio),  for the case you get audio problem? Then we can confirm if the SDP has been updated properly for bridging.

Cheers,
Daniel

On 18.04.17 17:23, gerry kernan wrote:

Hi

Thanks in advance if anyone can point me in the correct direction .

I have kamailio running in front of some asterisk VM’s.  SIP endpoint register to their asterisk PBX via Kamailio dispatcher module. I’m running rtpengine with a Wan and private interface to bridge audio stream between these endpoints on the WAN to asterisk PBX running on LAN IP behind Kamailio.

Calls from ext to ext work fine audio both directions , calls outbound to PSTN via SIP trunk to SIP provider via trunk on asterisk work fine audio both directions. But incoming calls via SIP provider I only get audio on stream from asterisk registered ext to external caller , no audio from external caller to the asterisk ext.

I reckon I have something wrong in my Kamailio.cfg . if I register an ext direct to asterisk I get audio both ways on incoming calls. And rtp logs from rtpenegine show it as trying to send the rtp to the private address of the sip endpoint rather that its WAN address.

I think my mistake in somewhere in the cfg below.

Do I need to handle invites from the backend asterisk servers different that invites from sip endpoints?

Gerry Kernan

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