RingCentral WebPhone Library

The RingCentral WebPhone Library includes a JavaScript WebRTC library and a WebRTC phone demo app.

Version 1.0.0 release notes

https://gist.github.com/tylerlong/72b51a72cc16206850c4cdfa36c6793a

Prerequisites

• You will need an active RingCentral account. Don't have an account? Get your Free RingCentral Developer Account Now!
• App type should be either :
  • Browser-Based
  • Server/Web

Browser Compatibility

Currently, we officially support Google Chrome browser. Official support for Firefox and Safari browsers are coming soon.

Network Requirements

Please visit Network Requirement links below

1. Network Requirements and Recommendations | RingCentral Office : https://support.ringcentral.com/s/article/9233?language=en_US
2. Network Requirements and Recommendations - Resources : https://support.ringcentral.com/s/article/Network-Requirements-and-Recommendations-Resources?language=en_US

Demo application

Here is a demo application based on React. It is a work-in-progress project: https://github.com/tylerlong/ringcentral-web-phone-demo

Table of Contents

1. Installation
2. Usage
3. Configuring your RingCentral app
4. Include Library And HTML Elements
5. Application
6. Demo
7. API
8. Initiating The Call
9. Accepting Incoming Call
10. DTMF
11. Hold Unhold
12. Mute Unmute
13. Park
14. Flip
15. Transfer
16. Warm Transfer
17. Forward
18. Start/Stop Recording
19. Barge/Whisper

Installation

yarn add ringcentral-web-phone

If you are not using NPM:

1. Download SIP.JS: https://github.com/onsip/SIP.js/releases/tag/0.20.0
2. Download WebPhone SDK: https://github.com/ringcentral/ringcentral-web-phone/releases/latest
3. Download audio files:
   1. https://cdn.rawgit.com/ringcentral/ringcentral-web-phone/master/demo/audio/incoming.ogg
   2. https://cdn.rawgit.com/ringcentral/ringcentral-web-phone/master/demo/audio/outgoing.ogg

Usage

Configuring your RingCentral app

Ensure your app has the following properties set. If these are not set, the error specified will be returned.

| App Property | Value | Error if not set | | ------------- | --------------- | ----------------------------------------------------- | | Permissions | VoIP Calling | Specific application permission required | | Platform type | Browser-based | Client edition is not compatible with current Brand |

Since WebRTC enables dialing out, you need to have a DIGITAL LINE attached to an extension to use this capability. You can configure this in Online Web Portal for Production and Sandbox accounts. More information on Digital Lines and their configuration is available in the following RingCentral Knowledge Base article topics:

1. Digital Line Overview (KB 5862)
2. Adding a Digital Line (KB 3136). A limited number of Digital Lines are free with each sandbox account which can be configured with the free RingCentral for Desktop softphone.
3. Reassigning an Existing Digital Line (KB 3748)

These permissions be configured for your app in the RingCentral Developer Portal. Fill this Registration Form to get access to WebRTC permissions. Please contact [email protected] to request these permissions.

Include Library And HTML Elements

<video id="remoteVideo" hidden="hidden"></video>
<video id="localVideo" hidden="hidden" muted="muted"></video>

<script src=".../sip.js" type="text/javascript"></script>
<script src=".../ringcentral-web-phone.js" type="text/javascript"></script>

Application

For this example you will also need to have RingCentral JS SDK installed.

Configure the web-phone

var clientId = '...';
var clientSecret = '...';
var appName = '...';
var appVersion = '...';

var sdk = new RingCentral.SDK({
  clientId: clientId,
  clientSecret: clientSecret,
  appName: appName,
  appVersion: appVersion,
  server: RingCentral.SDK.server.production, // or .sandbox
});

var remoteVideoElement = document.getElementById('remoteVideo');
var localVideoElement = document.getElementById('localVideo');

var platform = sdk.platform();

platform
  .login({
    jwt: '...',
  })
  .then(function (loginResponse) {
    return platform
      .post('/client-info/sip-provision', {
        sipInfo: [{ transport: 'WSS' }],
      })
      .then(function (res) {
        // Doing nested then because we need loginResponse in a simple way

        return new RingCentral.WebPhone(res.json(), {
          // optional
          clientId: clientId,
          appName: appName,
          appVersion: appVersion,
          uuid: loginResponse.json().endpoint_id,
          logLevel: 1, // error 0, warn 1, log: 2, debug: 3
          audioHelper: {
            enabled: true, // enables audio feedback when web phone is ringing or making a call
            incoming: 'path-to-audio/incoming.ogg', // path to audio file for incoming call
            outgoing: 'path-to-audio/outgoing.ogg', // path to aduotfile for outgoing call
          },
          media: {
            remote: remoteVideoElement,
            local: localVideoElement,
          },
          //to enable QoS Analytics Feature
          enableQos: true,
        });
      });
  })
  .then(function (webPhone) {
    // YOUR CODE HERE
  })
  .catch(function (e) {
    console.error(e.stack);
  });

Demo

$ git clone https://github.com/ringcentral/ringcentral-web-phone.git
$ cd ringcentral-web-phone
$ yarn install
$ yarn serve

1. Open http://localhost:8080 in the browser (port may change if 8080 will be already used by other app)
2. If your Application is of the Scope Server/Web Browser-Based Then you would need to add http://localhost:8080/callback.html as the OAuth Redirect URI for the application in Developer Portal
3. Add your RC credentials and click on Register
4. For making outbound calls, enter phone number and click on Call
5. For receiving incoming calls, Click on Accept button when window pops up (will be visible when there is an incoming call)

If there's any connection problems to Sandbox environment, you may need to switch to the Production environment.

WebRTC works with issues when served from file system directly to browser (e.g. file:// protocol), so you will need a local HTTP server (comes with this package).

Online demo is hosted at https://ringcentral.github.io/ringcentral-web-phone

** NOTE : If you are using the online demo, please add https://ringcentral.github.io/ringcentral-web-phone/callback.html to the app's OAuth Redirect URI

API

Except for some RingCentral-specific features the API is 100% the same as SIP.JS: https://github.com/onsip/SIP.js/releases/tag/0.20.0: most of the time you will be working with RC-flavored UserAgent and Session objects of SIP.JS.

We encourage you to take a look at Guides section, especially Make A Call and Receive A Call articles.

Constructor

var webPhone = new RingCentral.WebPhone(provisionData, options);

• Provision Data — the JSON returned from /client-info/sip-provision API endpoint
• Options — object with various configuration options that adjust WebPhone behavior
  • clientId — your application key
  • appName — your application short code name
  • appVersion — your application version
  • uuid — manually provide the unique identifier of WebPhone instance (should persist between page reloads)
  • logLevel — controls verboseness in browser console
    • 0 — Errors only (good for production)
    • 1 — Errors & warnings
    • 2 — Errors, warnings, logs
    • 3 — Everything including debug information (good for development)
  • audioHelper — audio feedback when web phone is ringing or making a call
    • enabled — turns feedback on and off
    • incoming — path to incoming.ogg, audio file for incoming call
    • outgoing — path to outgoing.ogg, audio file for outgoing call
  • onSession — this callback will be fired each time User Agent starts working with session (incoming or outgoing)
  • enableQos:true — will enable quality of service for webRTC calls , you can view the voice quality of calls in analytics portal

Attaching Media Streams

For futher information, refer SIP.js guide to attach media

Initiating The Call

var session = webPhone.userAgent.invite('PHONE_NUMBER', {
  fromNumber: 'PHONE_NUMBER', // Optional, Company Number will be used as default
  homeCountryId: '1', // Optional, the value of
});

Accepting Incoming Call

webPhone.userAgent.on('invite', function(session){
    session.accept().then(...);
});

DTMF

Callee will be put on hold and the another person can join into the call by dialing the extension number announced within the call.

session.dtmf('DTMF_DIGITS').then(...);

Hold Unhold

Callee will be put on hold and the another person can join into the call by dialing the extension number announced within the call.

session.hold().then(...);
session.unhold().then(...);

Mute Unmute

Callee will be put on mute or unmute

session.mute();
session.unmute();

Park

Callee will be put on hold and the another person can join into the call by dialing the extension number announced within the call.

session.park().then(...);

Flip

Caller can filp calls to different devices logged in through the same credentials.

session.flip('TARGET_NUMBER').then(...);

Transfer

session.transfer('TARGET_NUMBER').then(...);

Warm Transfer

If an agent has an active call with a customer and needs to transfer this call to a supervisor, then agent puts existing call on hold, makes a call to a supervisor and when ready performs a warm transfer. Customer will be connected to supervisor and the call between customer and agent will be disconnected.

Warm transfer puts current line on hold (if not done yet) then takes an existing line from arguments and makes transfer.

Handle Warm Transfer scenario (Attended Transfer usecase) :

Steps:

1. Put the current session on Hold as shown in the demo code
2. Initiate a new session (Start new call)
3. a. Once new call is answered , Complete the transfer , or terminate new session. b. If you want to switch to original call, switch the session context and Unhold the session

$modal.find('.transfer-form button.warm').on('click', function (e) {
  session.hold().then(function () {
    console.log('Placing the call on hold, initiating attended transfer');
    var newSession = session.userAgent.invite($transfer.val().trim());
    newSession.once('established', function () {
      console.log('New call initated. Click Complete to complete the transfer');
      $modal.find('.transfer-form button.complete').on('click', function (e) {
        session
          .warmTransfer(newSession)
          .then(function () {
            console.log('Warm transfer completed');
          })
          .catch(function (e) {
            console.error('Transfer failed', e.stack || e);
          });
      });
    });
  });
});

Forward

session.forward('TARGET_NUMBER').then(...);

Start/Stop Recording

session.startRecord().then(...);
session.stopRecord().then(...);

Barge/Whisper

Not yet implemented. Could be done by dialing *83. The account should be enabled for barge/whisper access through system admin.

Upgrade Procedure from 0.8.x to 0.9.0

• Migration Doc

Upgrade Procedure from v0.4.X to 0.8.9

• SDK now only supports only Unified SDP plan. You can find more information about this here: https://chromestatus.com/feature/5723303167655936

• SDK now only supports "require" as rtcp-mux policy. We no more support "negotiate". You can find more information about this here: https://www.juandebravo.com/2017/02/15/rtcp-mux-in-webrtc/

• SDK now handles SIP Re-Invites, which helps in handling one-way audio issues / reconnecting media due to network reconnections.

• SDK constructor now allows to add custom UA Configuration parameters like sessionDescriptionHandlerFactory , sessionDescriptionHandlerFactoryOptions

• SDK now handles rendering HTML Media Elements. Pass remoteVideo and localVideo elements via SDK constructor

• SDK also offers to addTrack() to handle remoteVideo and localVideo elements outside the constructor too

• For FireFox browser support

  • Client application needs to detect if the browser is firefox.
  • Client application needs to set custom UA configuration option 'options.enableMidLinesInSDP' to true for browser >= FF v63 for hold functionality to work
  • QoS feature is not supported on FireFox due to browser related bugs. Please set the custom UA configuration option options.enableQos to false

• SDK can now detect AudioInputLevel if the microphone device is not present or the input volume is set to 0. Added event listner no-input-volume for the same

• SDK can now detect AudioOutputLevel if the headset/speaker device is not configured correctly or the output volume is set to 0. Added event listner no-output-volume for the same

• You can now enable logging for AudioInputLevel, AudioOutputLevel and Media Reports by setting the custom UA configuration option options.enableMediaReportLogging to true. This will help in providing more information on one-way audio issues if there are any

Initialization

Before:

webPhone = new RingCentral.WebPhone(data, {
  clientId: localStorage.getItem('webPhoneClientId'),
  audioHelper: {
    enabled: true,
  },
  logLevel: parseInt(logLevel, 10),
  appName: 'WebPhoneDemo',
  appVersion: '1.0.0',
});

After:

var remoteVideoElement = document.getElementById('remoteVideo');
var localVideoElement = document.getElementById('localVideo');
webPhone = new RingCentral.WebPhone(data, {
  clientId: localStorage.getItem('webPhoneClientId'),
  audioHelper: {
    enabled: true,
  },
  logLevel: parseInt(logLevel, 10),
  appName: 'WebPhoneDemo',
  appVersion: '1.0.0',
  media: {
    remote: remoteVideoElement,
    local: localVideoElement,
  },
  //to enable QoS Analytics Feature
  enableQos: true,
  //to enable media stats logging
  enableMediaReportLogging: true,
});

Accept Invites:

Before:

var acceptOptions = {
            media: {
                render: {
                    remote: document.getElementById('remoteVideo'),
                    local: document.getElementById('localVideo')
                }
            }
      };
...
...
session.accept(acceptOptions).then(function() {
...
});;

After:

session.accept().then(function() {
...
})

Send Invite:

Before:

var session = webPhone.userAgent.invite(number, {
  media: {
    render: {
      remote: document.getElementById('remoteVideo'),
      local: document.getElementById('localVideo'),
    },
  },
  fromNumber: username,
  homeCountryId: homeCountryId,
});

After:

var session = webPhone.userAgent.invite(number, {
  fromNumber: username,
  homeCountryId: homeCountryId,
});

Auto Answer incoming calls if Invites containing Alert-Info: Auto Answer header field:

For incoming calls
function onInvite(session) {
    if (session.request.headers['Alert-Info'][0].raw === 'Auto Answer') {
            session
                .accept()
                .then(function() {
                    onAccepted(session);
                })
                .catch(function(e) {
                    console.error('Accept failed', e.stack || e);
                });
    }
...
...
}

Compatibility Matrix

| Date | SDK | SIPJS | Chrome | Firefox | | -------- | --------- | ------ | --------------------------------------------- | ------------------------------------------------------------ | | Feb 2016 | 0.2.0 | 0.6.4 | not known may be v50-70 | :warning: NA | | Apr 2016 | 0.3.0 | 0.7.3 | not known may be v50-70 | :warning: NA | | Jun 2016 | 0.3.1 | 0.7.4 | not known may be v50-70 | :warning: NA | | Aug 2016 | 0.3.2 | 0.7.5 | 54 to 56 | :warning: NA | | Sep 2016 | 0.4.0-RC1 | 0.7.5 | 54 to 56 | :warning: NA | | Jan 2017 | 0.4.0 | 0.7.5 | 54 to 56 | :warning: NA | | Mar 2017 | 0.4.1 | 0.7.7 | 54 to 70, rtcp mux support, media API changes | :warning: Issues with Audio, SBC | | Aug 2017 | 0.4.2 | 0.7.7 | 61 to 70 | :warning: Issues with Audio, SBC | | Aug 2017 | 0.4.3 | 0.7.8 | 61 to 70 | :warning: Not Tested | | Sep 2017 | 0.4.4 | 0.7.8 | 62 to 70 | :warning: Issues with DTMF | | Nov 2017 | 0.4.5 | 0.7.8 | 64 to 70 | :warning: Issues with DTMF | | Jul 2018 | 0.5.0 | 0.10.0 | 68 to 70 | :warning: Issues with DTMF | | Nov 2018 | 0.6.0 | 0.11.3 | 68 to 70 | Regression tested for 62, 63 supported with custom modifiers | | Nov 2018 | 0.6.1 | 0.11.6 | 71+, explicit plan b SDP support | 62 to 64 | | Dec 2018 | 0.6.2 | 0.11.6 | 71+ | 62 to 65 | | Feb 2019 | 0.6.3 | 0.11.6 | 71+ | 62 to 65 , :warning: QoS feature not supported | | Apr 2019 | 0.7.0 | 0.13.5 | 71+ | 62 to 65 , :warning: QoS feature not supported | | May 2019 | 0.7.1 | 0.13.5 | 71+ | 62 to 65 , :warning: QoS feature not supported | | Jun 2019 | 0.7.2 | 0.13.5 | 71+ | 62 to 65 , :warning: QoS feature not supported | | Nov 2019 | 0.7.3 | 0.13.5 | 71+ | 62 to 65 , :warning: QoS feature not supported | | Nov 2019 | 0.7.5 | 0.13.5 | 71+ | 62 to 65 , :warning: QoS feature not supported | | Jan 2020 | 0.7.6 | 0.13.5 | 71+ | 62 to 65 , :warning: QoS feature not supported | | Jan 2020 | 0.7.7 | 0.13.5 | 71+ | 62 to 65 , :warning: QoS feature not supported | | Feb 2020 | 0.7.8 | 0.13.5 | 71+ | 62 to 65 , :warning: QoS feature not supported | | Mar 2020 | 0.8.0 | 0.13.5 | 71+ | 62 to 65 , :warning: QoS feature not supported | | May 2020 | 0.8.1 | 0.13.5 | 71+ | 62 to 65 , :warning: QoS feature not supported | | Jul 2020 | 0.8.2 | 0.13.5 | 71+ | 62 to 65 , :warning: QoS feature not supported | | Oct 2020 | 0.8.3 | 0.13.5 | 71+ | 62 to 65 , :warning: QoS feature not supported | | Dec 2020 | 0.8.4 | 0.13.5 | 71+ | 62 to 65 , :warning: QoS feature not supported | | Feb 2021 | 0.8.5 | 0.13.5 | 71+ | 62 to 65 , :warning: QoS feature not supported |