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| [[Category:Concept|myPBX WebRTC Softwarephone]] | | [[Category:Concept|myPBX WebRTC Softwarephone]] |
| The myPBX toolkit is a set of Java Script files that can be used for integrating functionality of the innovaphone PBX into arbitrary web sites.
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| = Supported features =
| | Version 12r1 is not compatible with WebRTC any more due to the missing RTCP-Mux feature that is only included in 12r2 and up. |
| * Presence monitoring
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| * Outgoind WebRTC calls with audio, video and application sharing
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| = Requirements =
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| ;Web site:
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| * The needed JavaScript files of the libraries have to be included in the web site.
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| ;PBX:
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| * User object with password for login and a WebRTC device
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| * Valid STUN/TURN configuration
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| ;Licenses:
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| * myPBX license
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| * Video license (for video telephony)
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| * Application Sharing license (for viewing shared applications)
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| * WebRTC channel license (per WebRTC call)
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| ;Network:
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| * The PBX service (<code>https://xxx/PBX0/WEBSOCKET</code>) must be accessible from the public internet (NAT port forwarding or reverse proxy).
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| = Reference =
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| == innovaphone.pbxwebsocket.WebRtcEndpoint ==
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| This file contains a WebRTC endpoint implementation that can be used for adding WebRTC calls to a web page. For that the credentials of a user object on the PBX is needed. Visitors of the web page will use that user object for making phone calls.
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| === Includes ===
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| The following files have to be included in the html file in order to use the WebRTC endpoint.
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| * <code>innovaphone.common.crypto.js</code>
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| * <code>innovaphone.pbxwebsocket.Connection.js</code>
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| * <code>innovaphone.pbxwebsocket.ToneGenerator.js</code>
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| * <code>innovaphone.pbxwebsocket.WebRtcEndpoint.js</code>
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| === Constructor ===
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| ;new WebRtcEndpoint(url, username, password, device, physicalLocation, regContext, onLog, onCall):
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| :;url: The URL of the PBX websocket service (e.g. <code>ws://10.0.0.1/PBX0/WEBSOCKET/websocket</code>)
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| :;username: The username of the PBX user object.
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| :;password: The password of the PBX user object.
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| :;device: The device ID that shall be used for making calls.
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| :;physicalLocation: The physical location of the user (optional).
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| :;regContext: A numeric value that will be used for identifying the registration in the PBX (optional).
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| :;logFunction: A callback function that is called for logging debug info (optional).
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| :;onCall: A callback function that is called when calls are added, updated or removed. Applications that want to use call control have to specify this callback function. Applications that don't should give <code>null</code>. (optional)
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| === Methods ===
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| ;function close(): Closes the WebRTC endpoint and disconnects from the PBX.
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| ;function initCall(name, number, video, sharing): Starts a phone call. This is only possible if the application supplied an <code>onCall</code> callback to the constructor.
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| :;name: The URI to be called (optional, supply name or number).
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| :;number: The phone number to be called (optional, supply name or number).
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| :;video: Set to <code>true</code> for starting a video call (optional).
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| :;sharing: Set to <code>true</code> for starting an application sharing call (optional).
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| ;function clearCall(id): Terminates a call.
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| :;id: The ID of the call (optional). If no ID is supplied, all calls are terminated.
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| ;function attachVideo(local, remote): Attaches HTML video elements to the WebRTC endpoints. The endpoint will use them to playback video. Applications can both attach before or during a video call. Also attaching multiple video elements for a single call is possible. The application should mute the video elements (<code>muted="muted"</code>) in order to avoid playback of audio.
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| :;local: The HTML video element for playback of the local webcam image (optional, may be <code>null</code>).
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| :;remote: The HTML video element for playback of the remote video image (optional, may be <code>null</code>).
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| ;function detachVideo(local, remote): Detaches HTML video elements that have previously attached. This will stop the playback on the supplied elements.
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| :;local: The HTML video element to be detached (optional, may be <code>null</code>).
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| :;remote: The HTML video element to be detached (optional, may be <code>null</code>).
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| === Callbacks ===
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| ;function onLog(text): Should write the supplied text to the log of the application.
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| ;function onCall(event, call): Is called when the state of calls of this WebRTC endpoint changes.
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| :;event: A string that can be <code>added</code>, <code>updated</code> and <code>removed</code>.
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| :;call: The call object containing the current info about the call.
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| ::;id: The numeric id of the call.
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| ::;dir: The direction of the call (<code>in</code> or <code>out</code>).
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| ::;state:The state of the call (<code>idle</code>, <code>calling</code>, <code>incomplete</code>, <code>complete</code>, <code>alerting</code>, <code>connected</code>, <code>disconnecting</code>, <code>disconnected</code>, <code>parked</code>).
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| ::;hold: <code>true</code>, if the call is on hold (optional).
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| ::;name: The URI of the remote party (optional).
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| ::;number: The phone number of the remote party (optional).
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| ::;video: <code>true</code>, if video is active (optional).
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| ::;sharing: <code>true</code>, if application sharing is active (optional).
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| ::;cause: The cause code (optional).
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| === Example ===
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| <script type="text/javascript" src="innovaphone.common.crypto.js"></script>
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| <script type="text/javascript" src="innovaphone.pbxwebsocket.Connection.js"></script>
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| <script type="text/javascript" src="innovaphone.pbxwebsocket.ToneGenerator.js"></script>
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| <script type="text/javascript" src="innovaphone.pbxwebsocket.WebRtcEndpoint.js"></script>
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| <script type="text/javascript">
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| // dependencies
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| var WebRtcEndpoint = innovaphone.pbxwebsocket.WebRtc.Endpoint;
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|
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| var endpoint = null;
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| var config = {
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| url: "ws://192.168.0.1/PBX0/WEBSOCKET/websocket",
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| username: "user",
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| password: "password",
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| device: "user-webrtc",
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| physicalLocation: null,
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| regContext: "0"
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| };
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|
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| function logFunction(text) {
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| console.log("WebRTC demo: " + text);
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| }
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|
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| function onCall(event, call) {
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| console.log("Call " + event + ": " + JSON.stringify(call));
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| }
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|
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| function initCall(number, video) {
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| if (endpoint) endpoint.initCall(null, number, video, false);
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| }
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|
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| function clearAllCalls() {
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| if (endpoint) endpoint.clearCall();
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| }
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|
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| function close() {
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| if (endpoint) {
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| endpoint.detachVideo(document.getElementById("video-lcoal"), document.getElementById("video-remote"));
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| endpoint.close();
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| }
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| endpoint = null;
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| }
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| function start() {
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| if (endpoint) endpoint.close();
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| endpoint = new WebRtcEndpoint(config.url, config.username, config.password, config.device, config.physicalLocation, config.regContext, logFunction, onCall);
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| endpoint.attachVideo(document.getElementById("video-lcoal"), document.getElementById("video-remote"));
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| }
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| </script>
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Version 12r1 is not compatible with WebRTC any more due to the missing RTCP-Mux feature that is only included in 12r2 and up.