Installation Procedures

Purpose
List the Installation tasks for the Genesys WebRTC Service.

Note: As mentioned below, you must create a WebRTC Gateway application object before you can run and test the Genesys WebRTC Service.

Tasks	Procedures
1. Create a WebRTC Gateway application object in Genesys Administrator or Genesys Administrator Extensions	If you are using version 8.1.3 or later of Configuration Server, import the WebRTC_Gateway_850.apd template. If you are using an earlier version of Configuration Server, import the WebRTC_Gateway_850_generic.apd template.
2. Install the Genesys WebRTC Service	Complete one of the following procedures, according to your operating system: Installing the Genesys WebRTC Service on Windows Installing the Genesys WebRTC Service on Linux
3. Configure the necessary options in the rsmp section of your WebRTC Gateway application object	Review the Security Considerations and Platform configuration sections and set the required options.

Validate the Installation

• After the WebRTC gateway has been installed, configured and started, you can run a basic gateway health check by going to the following URL in a browser:
  

  http[s]://<WebRTC_Gateway_FQDN>:<HTTP_Port>/ping

  You should get a response with the text "up".
  If you want to test this locally from the gateway machine on Linux, you can run:
  

  wget --no-check-certificate -q -O - http[s]://localhost:<HTTP_Port>/ping

• After the WebRTC JavaScript API (JSAPI) has also been installed on a Web server, you can test the WebRTC operation using the demo1 application that comes with the JSAPI, by making a loop-back call or a call with another demo instance or a SIP endpoint. For running demo1, go to the following URL:
  

  https://<WebServer_FQDN>/webrtc/demo/demo1/audio_video.html

  Here it is assumed that the JSAPI files are installed in the Web server under sub-directory "webrtc".
  Follow the steps in the page using the information there to make a test call and perform some operations with the call, such as hold and resume.
  The audio_video.html may need to be modified to set the gateway URI (http[s]://<address>:<port>) and possibly other JSAPI configuration parameters, especially STUN or TURN server information, although most of these parameters can be input during STEP 2 of the demo. You may also want to add the configured DN values to be used in STEP 4 (for registering with the SIP Server configured in the gateway) and STEP 5 (for calling a DN, which another endpoint is registered with) in the HTML page itself for easy use.
  You would need to have SSL certificates installed for the gateway (as described in the Platform Configuration section) as well as the Web server; or for testing purposes, you could manually allow the security exceptions in the browser for the Web server and the gateway. The caller app/demo1 can be registered with the gateway anonymously using the ID "Anon", and can make a loop-back call by calling the ID "self" to test the signaling and media connections with the gateway. If you want to make a call to another endpoint, however, you will need to have a SIP Server configured in the gateway, and have the caller endpoint register with the SIP Server using an extension DN, which the caller can use to make the call via the gateway.