Brekeke 3.4.1 Beta & WebRTC

1. Brekeke Product Name and Version: Brekeke SIP Server 3.4.1 Beta Advanced Edition

2. Java version: 1.8

3. OS type and the version: Windows Server 2008 SP2

4. UA (phone), gateway or other hardware/software involved: None. Softphone WebRTC using JsSIP.

5. Your problem / question :

Hello,
I am looking for more information about the WebRTC feature of Brekeke. I have a working WebRTC phone targetting Asterisk PBX, I would like to target Brekeke now.
Any help appreciated !
Sébastien

Brekeke SIP Server v3.4 supports WebSocket.
You can make a call between WebSocket clients (such as JsSIP) over Brekeke SIP Server v3.4.


Brekeke PBX ver 3.5 will support both WebRTC and WebSocket. (ver3.5 is not related yet.)
You can make a call between JsSIP and non-websocket clients (such as Asterisk) over Brekeke PBX v3.5.

Hello,

I have now successfully configured Brekeke SIP Advanced so I can make an audio call between 2 WebRTC clients (JsSIP) (Activate WebSockets and TLS following the Wiki's instructions).

Could you please help me to answer the following questions ?
1- Will Brekeke Advanced support WebRTC Video call ?
2- How to configure the Dial Plan so a SIP client can call a WebSocket-WebRTC client ?

Thank you for your support,

Sebastien

> 1- Will Brekeke Advanced support WebRTC Video call ?

Yes.
If both caller and callee (UAC and UAS) support WbRTC, you can use any media such as audio and video between clients.

For JsSIP, enable "video" at the JsSIP's GUI.


> 2- How to configure the Dial Plan so a SIP client can call a WebSocket-WebRTC client ?

What kind of DialPlan do you need?

If a callee client is registered in Brekeke SIP Server, you don't need any DialPlan rules to make a call to the callee.