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.
|1. Create a WebRTC Gateway application object in Genesys Administrator or Genesys Administrator Extensions||
|2. Install the Genesys WebRTC Service||Complete one of the following procedures, according to your operating system:|
|3. Configure the necessary options in the rsmp section of your WebRTC Gateway application object|
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:
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
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.