Browser SDK V2

    Plivo BrowserSDK allows you to make and receive calls using Plivo applications directly from any web browser that supports webRTC. Using our SDK you can create applications like Click to Call, Conferencing Apps and even Webphones. Currently we support Google Chrome 55 and above and Firefox 51 and above.

    Try out the demo here:

    Get started with examples

    Including the javascript

    Include the plivo javascript file as shown below

    <script type="text/javascript" src="https://cdn.plivo.com/sdk/browser/v2.0.24/plivo.min.js"/>

    Initializing the plivoBrowserSdk Object

    The plivoBrowserSdk object needs to be initialized.

    var options = {
        "debug":"DEBUG",
        "permOnClick":true,
        "audioConstraints":{"optional":[{"googAutoGainControl":false},{"googEchoCancellation":false}]},
        "enableTracking":true
    };
    var plivoBrowserSdk = new window.Plivo(options);
    

    Event registration

    Pass function references to the events produced from the sdk. This is where your UI manipulation should handle all the different call flows.

    plivoBrowserSdk.client.on('onWebrtcNotSupported', onWebrtcNotSupported);
    plivoBrowserSdk.client.on('onLogin', onLogin);
    plivoBrowserSdk.client.on('onLogout', onLogout);
    plivoBrowserSdk.client.on('onLoginFailed', onLoginFailed);
    plivoBrowserSdk.client.on('onCallRemoteRinging', onCallRemoteRinging);
    plivoBrowserSdk.client.on('onIncomingCallCanceled', onIncomingCallCanceled);
    plivoBrowserSdk.client.on('onCallFailed', onCallFailed);
    plivoBrowserSdk.client.on('onCallAnswered', onCallAnswered);
    plivoBrowserSdk.client.on('onCallTerminated', onCallTerminated);
    plivoBrowserSdk.client.on('onCalling', onCalling);
    plivoBrowserSdk.client.on('onIncomingCall', onIncomingCall);
    plivoBrowserSdk.client.on('onMediaPermission', onMediaPermission);
    plivoBrowserSdk.client.on('mediaMetrics',mediaMetrics);
    plivoWebSdk.client.on('onConnectionChange',onConnectionChange);
    

    Registering using your Plivo Endpoint

    Register using your Plivo Endpoint credentials

    var username = 'johndoe12345';
    var pass = 'XXXXXXXX';
    plivoBrowserSdk.client.login(username, password);

    Making a call

    Making a call to any number/sip endpoint. Do remember that the application attached to the registered endpoint should have an answer url that will ensure that the correct <Dial> element is returned

    var dest = “jane1234@phone.plivo.com”;
    var extraHeaders = {'X-PH-Test1': 'test1', 'X-PH-Test2': 'test2'};
    plivoBrowserSdk.client.call(dest, extraHeaders);

    Accepting a call

    This is how an incoming call should be answered once the onIncomingCall event is received

    plivoBrowserSdk.client.answer();

    Sending DTMF

    The snippet below shows how you can send a DTMF tone when in a phone call

    plivoBrowserSdk.client.sendDtmf("1");

    Configuration

    The following are the configuration parameters:

    AttributeDescriptionAllowed ValuesDefault Value
    debugEnable debug message in JS logOFF, ERROR, WARN, INFO, DEBUG, ALLINFO
    permOnClickSet to true if you want to ask for mic permission before call. Otherwise it will be asked on page load.true/falsefalse
    audioConstraintsAudio constraints object that will be passed to webRTC getUserMedia().Audio constraints values are browser specific. The following values can be set to true or false: googEchoCancellation, googAutoGainControl, googNoiseSuppression, googHighpassFilter{}
    enableTrackingSet to true if you want to get mediaMetrics events and enable call quality tracking.true/falsetrue
    dscpSet to true if you want to enable QoS in voice traffic. Differentiated Services field in the packet headers for all WebRTC traffic.true/falsetrue
    preDetectOwaDetect one way audio before answering/sending the call.true/falsefalse
    clientRegionInitialisation options to set and route calls to specific MediaServer POPs["usa_west","usa_east","australia","europe","asia","south_america"]

    Methods

    These are the methods supported on plivoBrowserSdk.client

    </table>

    These are the methods supported on plivoBrowserSdk.client.audio (Supported only on Chrome)

    MethodDescription
    login(username, password)Registering a Plivo endpoint
    logout()Logging out from the registered endpoint
    call(number, extraHeaders)Call a number or sip address. 'number' takes a String value and 'extraHeaders' takes a JSON object.
    Example of 'extraHeaders':
    {'X-PH-Test1': 'test1', 'X-PH-Test2': 'test2'}
    answer()Answer an incoming call
    hangup()Hangup an ongoing call
    reject()Reject an incoming call
    sendDtmf(digit)Send the digits as dtmf
    'digit' can be any of the following one character strings: "1", "2", "3", "4", "5", "6", "7", "8", "9", "0", "*", "#"
    mute()Mute the mic
    unmute()Unmute the mic
    setRingTone(url)Set the ringtone that plays when calling. The default tone here will be used in case the URL is not specified. </tr>
    setRingToneBack(url)Set the ringtone that plays when being called. The default tone here will be used in case the URL is not specified.
    setConnectTone(boolean)Dial beep will play till we get alert response from network. Setting false will not play beep tone. DEFAULT is True , connect tone is enabled . </tr>
    setDebug(debug)Set the log level of the sdk. Allowed values are OFF, ERROR, WARN, INFO, DEBUG, ALL
    getPeerConnection()Returns an RTCPeerConnection object
    sendQualityFeedback(score, comment)Send call quality feedback at the end of a call. 'score' can take an integer value between 1 to 5. Comment must be one of the following values: ['bad-audio', 'call-dropped', 'wrong-callerid', 'post-dial-delay', 'dtmf-not-captured', 'audio-latency', 'unsolicited-call', 'one-way-audio', 'no-audio', 'never-connected']. This feedback would be sent only if 'enableTracking' flag is set to true.
    getCallUUID()Returns a String call UUID if a call is active, else returns null
    getLastCallUUID()Returns last call UUID, Useful in the cases if you want to send feedback for last call
    webRTC()Returns true if webRTC is supported and false if there is no webrtc support
    versionIt is a variable that returns current version of the Plivo SDK
    isLoggedInIt is a variable that returns 'true' if the user is logged in and 'false' otherwise
    MethodDescription
    availableDevices(filter)This is a promise based callback that returns all available devices. The filter parameter is optional and takes a String value. Pass 'input' as filter by Input audio devices,'output' as filter by Output audio devices, null to get all audio devices
    revealAudioDevices(arg)This is a promise based callback that will force and ask for allowing permission before returning all available devices. arg parameter is optional and takes a string. Passing 'returnStream' as arg will return local stream as well in Promise success

    Objects in Audio Device API plivoBrowserSdk.client.audio are as follows:

    ObjectMethods and ParametersDescription
    microphoneDevicesset(deviceID)
    get()
    reset()
    set method will set the audioDevice ID as default Microphone device. The deviceID parameter is mandatory and takes a String value.
    get method will return the Microphone device ID which is set already.
    reset method will remove any Microphone device ID which is already set.
    speakerDevicesset(deviceId)
    get()
    reset()
    media(source)
    set method will set the audioDevice ID as default Speaker device for DTMF and remote audio. The deviceID parameter is mandatory and takes a String value.
    get method will return the Speaker device ID which is set already.
    reset method will remove any Speaker device ID which is already set.
    mediawill take 'dtmf' or 'ringback' as source parameter and will return the corresponding HTML audio element. This parameter is mandatory.
    ringtoneDevicesset(deviceId)
    get()
    reset()
    media()
    set method will set the audioDevice ID as default Ringtone device for DTMF and remote audio. The deviceID parameter is mandatory and takes a String value.
    get method will return the Ringtone device ID which is set already.
    reset method will remove any Ringtone device ID which is already set.
    media method will return Ringtone HTML audio element.

    Events

    plivoBrowserSdk object receives events that can be listened to.

    EventDescription
    onLoginOccurs when a login is successful
    onLoginFailed(cause)Occurs when a login has failed.cause returns the login failure reason
    onLogoutOccurs when a logout is successful
    onCallingOccurs when a call is initiated
    onCallRemoteRingingOccurs when the remote end starts ringing during an outbound call
    onCallAnsweredOccurs when the an outbound or an inbound call is answered
    onCallTerminatedOccurs when the an outbound or an inbound call has ended
    onIncomingCall(callerID, extraHeaders)Occurs when there is an incoming call. callerID provides the callerID and extraHeaders return the X-Headers from Plivo
    onIncomingCallCanceledOccurs when an incoming call is cancelled by the caller
    onCallFailed(cause)Occurs when an outbound or an inbound call fails. cause returns the reason for call failing
    onMediaPermission(event)Occurs when media permission has been granted. event returns the stream access status. The success event returns {'status':'success','stream':true}. On failure the event returns {'status':'failure','error':errorName}
    onWebrtcNotSupportedOccurs when browser does not support web rtc
    mediaMetricsWorks only for Chrome. Raises the following events:
    high_jitter: when the jitter is higher than 30 ms for 3 out of last 5 samples.
    high_rtt: When the RTT is higher than 400 ms for 3 out of last 5 samples.
    high_packetloss: When the packet loss is > 10% for OPUS and loss > 20% PCMU.
    low_mos: When sampled mos is < 3 for 3 out of last 5 samples, no_microphone_access When we detect one way audio (<80 bytes sent in 3 seconds).
    no_audio_received : When the user is not able to hear the callee or When audio level is stable for last 3 samples.
    ice_timeout : Alert if ICE gathering takes more than 2 sec either for outgoing call invite or incoming call answer.
    All the above will have `level`:`warning`, its `group` : `network/audio`, related params will have its own `value` and state `active`:`true/false`.
    Example: {active : true, desc : "local_audio", group : "audio", level : "warning", type : "no_audio_received", value : 0}
    onAudioDeviceChange(deviceObj)Occurs when there is a change in USB audio device, Device added or removed. This event will emit an object with two properties "change" and "device". change will have values "added" or "removed". Device provides device specific properties
    onConnectionChangeThis event is generated when the state of Plivo's WebSocket connection changes. For example,
    when the WebSocket is disconnected due to internet issues.

    On WebSocket disconnect {
    'state': 'disconnected',
    'eventCode': <code>,
    'eventReason': <reason>
    }


    On WebSocket reconnect {
    'state': 'connected'
    }


    Common WebSocket event codes ( RFC 6455 )
    1006: indicates that the connection was closed abnormally, e.g., without sending or
    receiving a Close control frame. For example, internet disconnection.
    1009: indicates that an endpoint is terminating the connection
    because it has received a message that is too big for it to
    process.
    1011: indicates that a server is terminating the connection because
    it encountered an unexpected condition that prevented it from
    fulfilling the request.

    Examples

    Checkout our github repo for examples