[OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway

Mundkowsky, Robert rmundkowsky at ets.org
Wed Jun 28 09:28:34 EDT 2017

Curious, why would you want to use OpenSIPS and Kamailio?

There both SIP proxies.


From: Users [mailto:users-bounces at lists.opensips.org] On Behalf Of Alex Megalokonomos
Sent: Wednesday, June 28, 2017 5:47 AM
To: users at lists.opensips.org
Subject: [OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway


We have the following scenario: our office call center is an Alcatel OmniPCX Office setup.

This handles most of our needs and also provides 4 SIP extensions.

These are provided by what appears to be a Kamailio SIP server v 3.2.2 (no webrtc or websockets support)

What we would like to do is set up an OpenSIPS instance to handle WebRTC and proxy everything to this Kamailio SIP server.

The idea is to allow a web client (using sip js or something similar) to register / make / receive calls as one of the Kamailio extensions.

I think half of the configuration is this : http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.opensips.org%2FDocumentation%2FTutorials-WebSocket-2-1&data=02%7C01%7Crmundkowsky%40ets.org%7Cbc057a0583854c1f54eb08d4be0ae688%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636342401422676698&sdata=Y%2BsdaA4hevNYQyWx0zkB4rVvG5glervN54dVx1XArNo%3D&reserved=0>

which I've already completed and indeed, clients can register to opensips and chat/make calls over websockets between them.

How do I go about proxying registrations/invites/etc to the kamailio server instead?

best regards


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