BandwidthRTC Node SDK Documentation

Initialize the BandwidthRTC Node SDK

importBandwidthRtcfrom"bandwidth-rtc";constbandwidthRtc=newBandwidthRtc();

API Methods

connect

• Params:
  • authParams: the device token to connect with
  • options: optional SDK settings (can be omitted)
    • websocketUrl: override the default Bandwidth RTC connection url (this should not generally be needed)
• Description: connect device to the Bandwidth RTC platform

awaitbandwidthRtc.connect({deviceToken: deviceToken,});

publish

• Params:
  • input: the input to publish; this can be an instance of:
    • mediaStream: An already existing media stream such as a screen share
      • Type: MediaStream
• Return:
  • rtcStream: a media stream with the supplied media stream constraints
    • Type: RtcStream
• Description: publish media

Publish with default settings:

let rtcStream: RtcStream=awaitbandwidthRtc.publish();

Publish audio only

Publish with customized constraints

constmediaConstraints: MediaStreamConstraints={audio: {autoGainControl: true,channelCount: 1,deviceId: "default",echoCancellation: true,latency: 0.01,noiseSuppression: true,sampleRate: 48000,sampleSize: 16,}};letmicrophoneStream=awaitnavigator.mediaDevices.getUserMedia(mediaConstraints);letsourceNode=audioContext.createMediaStreamSource(microphoneStream);
let rtcStream: RtcStream=awaitbandwidthRtc.publish(mediaConstraints);

Publish with existing media stream

letscreenShare=awaitnavigator.mediaDevices.getDisplayMedia({video: true,});
let rtcStream: RtcStream=awaitbandwidthRtc.publish(screenShare);

Please see the following resources for more information on MediaStreamConstraints and MediaTrackConstraints that can be specified here:

• https://developer.mozilla.org/en-US/docs/Web/API/MediaStreamConstraints
• https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackConstraints

disconnect

• Description: disconnect device from the Bandwidth RTC platform

DTMF

• Description: send DTMF tones via the browser's native RTCDTMFSender (RFC 4733). Tones are forwarded as telephone-event RTP packets by the Bandwidth gateway.
• Params:
  • tone: the digits to send, as a string, chosen from the set of valid DTMF characters [0-9,*,#,A-D,,]
  • streamId (optional): the stream to send the tone on; defaults to all published streams
  • duration (optional): tone duration in milliseconds, between 40 and 6000 (default: 100)
  • interToneGap (optional): gap between tones in milliseconds, minimum 30 (default: 70)

bandwidthRtc.sendDtmf("3");bandwidthRtc.sendDtmf("313,3211*#");bandwidthRtc.sendDtmf("5",undefined,200,100);// 200ms tone, 100ms gap

Event Listeners

onStreamAvailable

• Description: called when a media stream is available to attach to the UI. This will be called for every independent stream presented to the Participant, meaning that if a new remote Participant is added to a subscribed Session a new stream will be made available to the browser, and will need to be presented to the UI for consumption by the user.

bandwidthRtc.onStreamAvailable((event)=>{console.log(`A stream is available with streamId=${event.mediaStream.id}, its media types are ${event.mediaTypes} and the stream itself is ${event.mediaStream}`,);});

onStreamUnavailable

• Description: called when a media stream is now unavailable and should be removed from the UI

bandwidthRtc.onStreamUnavailable((event)=>{console.log(`The stream with streamId=${event.mediaStream.id} is now unavailable and should be removed from the UI because the media is likely to freeze imminently.`,);});

Notes

• Pinning rpc-websockets library to version 7.10.0 so that the transpiling and compilation of the RPC Client works.