General notes


added in version 2016.04

XiVO comes with a WebRTC lines support, you can use in with XiVO UC Assistant and Desktop Assistant. Before starting, please check the WebRTC Environment.

Current WebRTC implementation requires following configuration steps:

Configuration of XiVO PBX for WebRTC


Security warning: when enabling WebRTC you need to ensure that you do it securely:

  1. by securing the access to the ARI,
  2. and by securing (e.g. via an external firewall) the access to the asterisk HTTP server (which listens on port 5039).

First, secure ARI connection (this step is very important otherwise your XiVO PBX won’t be secure) :

  • Generate a password (e.g. with pwgen -s 16)

  • Edit file /etc/asterisk/ari.conf

  • replace:

    password = Nasheow8Eag



Then, open asterisk HTTP server to accept outside websocket connections:

  • Edit file /etc/asterisk/http.conf

  • replace:




By default, asterisk HTTP server has a limit of 100 websocket connections. You can change this limit in the /etc/asterisk/http.conf file:


Restart XiVO PBX services to apply the new settings:

xivo-service restart

Configuration of user with WebRTC line

  1. Create user
  2. Add line to user without any device
  3. Edit the line created and, in the Advanced tab, add webrtc=yes options:

Manual configuration of user with WebRTC line

For the records