WebRTC

General notes

Note

added in version 2016.04

Note

Current WebRTC implementation (since XiVO Freya 2020.18) does not require any of these manual configuration steps Users

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.

Former WebRTC implementation might require following configuration steps:

• configure Asterisk HTTP server,
• and create user with one line configured for WebRTC. To have user with both SIP and WebRTC line is not supported.

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:

Fallback Configuration

When the user is not connected to its WebRTC line, or disconnect from the assistant, you can route the call to a default number as for example the user mobile number. Update the fail option on the No Answer user tab configuration, and add an extension to the appropriate context.

Experimental video call feature

Important

Removed since 2021.08

Manual configuration of user with WebRTC line

For the records

• WebRTC manual configuration

Updating LineConfig Sip response

Since Helios 17, we added a new field sipPort inside LineConfig Sip response to avoid DNS requests

Before:

 "msgType": "LineConfig",
 "ctiMessage": {
    "hasDevice": false,
    "id": "1",
    "isUa": false,
    "mobileApp": true,
    "name": "l1ze3zd8",
    "number": "4000",
    "password": "e2xyubvn",
    "sipProxyName": "default",
    "vendor": null,
    "webRtc": true,
    "xivoIp": "192.168.56.3"
}

After:

 "msgType": "LineConfig",
 "ctiMessage": {
    "hasDevice": false,
    "id": "1",
    "isUa": false,
    "mobileApp": true,
    "name": "l1ze3zd8",
    "number": "4000",
    "password": "e2xyubvn",
    "sipProxyName": "default",
    "sipPort": "5060", <== New Field
    "vendor": null,
    "webRtc": true,
    "xivoIp": "192.168.56.3"
}