jssip_client

Flowroute JsSIP Client

Flowroute SIP over WebSocket and WebRTC JavaScript client.

It's actually is a facade for WebRTC, DOM and JsSIP APIs to easy development of Flowroute applications on frontend.

Online demo

https://demo.webrtc.flowroute.com/fr.html?cache_invalidate=xxx

Installation for usage

NPM

Available on Node Package Manager: npm install --save jssip_client

CDN

For a static script installation. Soon.

Setting up core development environment

You'll only require node (+10.x.y) (if you haven't installed it yet, use nvm). Then after clonning this repository, change directory to its root and run once:

npm install

And after this, you probably will enjoy to see a demo app with hot reloading on your browser:

npm start

This demo HTML code is pretty simple and you can use it to further developing a frontend application on any lib/framework (React, jQuery, Angular, Vue etc.) or even vanilla JavaScript.

With the UI debugging console, you can see many of actions being dispatched by the user agent and its calls, everything that is available to you.

Creating releases

For package maintainers:

./release.sh x.y.z

Where "x.y.z" is actually the version you want to release (like ./release.sh 0.0.1).

This script will change the version in package.json, build production artifacts into releases/ directory and and commit them. After this, you will want to git push these changes and, for Node maintainers, also run npm publish command.

Examples of features

First, create an instance of Flowroute client. You can pass some optional parameters to turn on the browser console SIP debugging and handle every client actions. All available param keys are available on constructor docstring (so if you use a mainstrain editor/IDE, it'll even show them for you with descriptions). Remember that these parameters are optional and you can make a call work without them, they'll be assigned to default ones.

const flowrouteClient = new FlowrouteClient({
  debug: true,
  onUserAgentAction: console.warn,
  intervalOfQualityReport: 5000,
  pointOfPresence: 'us-west-or',
});

Note that the example above is passing a param of QoS report. That's because during this interval of milliseconds, a SIP MESSAGE will be sent with WebRTC stats about the call RTC connection.

There's also a selected PoP, because many Flowroute PoPs are equiped with WebSocket/WebRTC gateways. The preferred PoP and its related edge strategy can be set using Flowroute APIs and the same ones are supported by this client, is therefore possible for a customer to control where the media should be relayed to ensure we can avoid unnecessary media relay.

Some other common parameters are callerId and password. Always check the docstring for the full list.

Having the client ready, you can start a connection with the signaling server and invoke the SIP REGISTER:

flowrouteClient.start();

After receiving the { type: 'registered' } action on onUserAgentAction callback, you're free to make calls. And it's simple as:

flowrouteClient.call({
  to: '',
  onCallAction: console.warn,
});

As always, parameters are optionals (although you can consider this to a required one, another approach is using flowrouteClient.setDID method before this and here omitting to). If you want, a audioConstraints parameter is available so you can pick, for example, a selected audio output device id.

You can easily control player volume, from 0 (mutting it) to 100 (loudest):

flowrouteClient.setOutputVolume(100);

During a call, you can get its instance and make actions with JsSIP.RTCSession API, like sending DTMFs:

flowrouteClient.getActiveCall().sendDTMF('4');
flowrouteClient.getActiveCall().sendDTMF('2');

Technical debts so far

Feel free to contribute on this:

• answering calls
• unit tests
• polyfill for IE
• easy to download and use released script versions