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https://github.com/bandwidth/webrtc-browser

Bandwidth's WebRTC Browser SDK
https://github.com/bandwidth/webrtc-browser

bandwidth outage-risk sdk webrtc

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Bandwidth's WebRTC Browser SDK

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README

        

# Bandwidth WebRTC Browser SDK Documentation

## Initialize the Bandwidth WebRTC Browser SDK

```javascript
import BandwidthRtc from "@bandwidth/webrtc-browser";

const bandwidthRtc = new BandwidthRtc();
```

## 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

```javascript
await bandwidthRtc.connect({
deviceToken: deviceToken,
});
```

### publish

- Params:
- input: the input to publish; this can be an instance of:
- constraints: the media stream constraints such as audio, peerIdentity, video
- Type: MediaStreamConstraints
- 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:

```javascript
let rtcStream: RtcStream = await bandwidthRtc.publish();
```

#### Publish audio only

```javascript
const mediaConstraints: MediaStreamConstraints = {
audio: true,
video: false,
};
let rtcStream: RtcStream = await bandwidthRtc.publish(mediaConstraints);
```

#### Publish with customized constraints

```javascript
const mediaConstraints: MediaStreamConstraints = {
audio: {
autoGainControl: true,
channelCount: 1,
deviceId: "default",
echoCancellation: true,
latency: 0.01,
noiseSuppression: true,
sampleRate: 48000,
sampleSize: 16,
},
video: {
aspectRatio: 1.3333333333333333,
frameRate: 30,
width: { min: 640, ideal: 1280 },
height: { min: 480, ideal: 720 },
resizeMode: "none",
},
};
let rtcStream: RtcStream = await bandwidthRtc.publish(mediaConstraints);
```

#### Publish with existing media stream

```javascript
let screenShare = await navigator.mediaDevices.getDisplayMedia({
video: true,
});
let rtcStream: RtcStream = await bandwidthRtc.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 a set of VoIP-network-friendly DTMF tones. The tone amplitude and duration can not be controlled
- Params:
- tone: the digits to send, as a string, chosen from the set of valid DTMF characters [0-9,*,#,\,]
- streamId (optional): the stream to 'play' the tone on

```javascript
bandwidthRtc.sendDtmf("3");
bandwidthRtc.sendDtmf("313,3211*#");
```

## 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.

```javascript
bandwidthRtc.onStreamAvailable((event) => {
console.log(
`A stream is available with endpointId=${event.endpointId}, 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

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