@twilio/webrtc

twilio-webrtc.js
================

 

twilio-webrtc.js contains the various WebRTC shims used by twilio-video.js.
It is not intended for general consumption.

Installation

``
npm install --save @twilio/webrtc
`

Exports

The following WebRTC API shims are available:

`javascript
const {
getStats,
getUserMedia,
MediaStream,
MediaStreamTrack,
RTCIceCandidate,
RTCPeerConnection,
RTCSessionDescription
} = require('@twilio/webrtc');
`

$3

getStats resolves with normalized WebRTC statistics for the active ICE
candidate pair and each MediaStreamTrack, local or remote, of a particular
RTCPeerConnection.

`javascript
/**
* Get the statistics for a given RTCPeerConnection.
* @param {RTCPeerConnection} peerConnection
* @returns {Promise}
*/
function getStats(peerConnection) {}
`

__NOTE__: StandardizedStatsResponse
normalizes the different formats of the stats returned by RTCPeerConnection#getStats in different
browsers. It does not conform to the W3C spec.

$3

getUserMedia accepts a MediaStreamConstraints object and resolves
with a MediaStream. By default, it requests both audio and video.

`javascript
/**
* Request media from the user.
* @param {MediaStreamConstraints} [constraints={audio: true, video: true}]
* @returns {Promise}
*/
function getUserMedia(constraints) {}
`

$3

RTCPeerConnection abstracts away some of the browser-specific implementations
of WebRTC, and implements some WebRTC features that are not present in some
browsers.

#### Chrome
* Adds rollback support, according to the workaround specified here.
* Adds "track" event support, as per the workaround in webrtc-adapter.
* Provides a workaround for the case where, when the SSRC of a MediaStreamTrack changes, the
browser treats this as a removal of the existing MediaStreamTrack and the addition of a new
MediaStreamTrack.
* Adds support for getting and setting maxPacketLifeTime on RTCDataChannels by
remapping the legacy property maxRetransmitTime to maxPacketLifeTime. See
this bug for
more information.
* Provides a workaround for this bug, where calling removeTrack
with an RTCRtpSender that is not created by the RTCPeerConnection in question throws an exception.

#### Firefox
* For new offers, adds support for calling setLocalDescription and setRemoteDescription in
have-local-offer and have-remote-offer signaling states respectively.
* Adds support for calling createOffer in signaling state have-local-offer.
* The above features are implemented using rollback to work around this bug.
* Provides a workaround for this bug, where the browser may
change the previously negotiated DTLS role in an answer, which breaks Chrome.
* Provides a workaround for this bug,
where the browser throws when RTCPeerConnection.prototype.peerIdentity is accessed.
* Works around Firefox Bug 1480277.

#### Safari
* Adds rollback support, according to the workaround specified here.
* Provides a workaround for the case where, when the SSRC of a MediaStreamTrack changes, the
browser treats this as a removal of the existing MediaStreamTrack and the addition of a new
MediaStreamTrack.
* Provides a workaround for this bug, where webrtc-adapter's shimmed
addTrack method does not return the RTCRtpSender associated with the added track.

$3

RTCSessionDescription abstracts away some of the browser-specific implementations
of WebRTC for Firefox and Safari, and works around this bug
in Chrome, where the native RTCSessionDescription constructor throws when its argument is
{ type: 'rollback'}.

$3

MediaStream, MediaStreamTrack, and RTCIceCandidate` abstracts away their
browser-prefixed counterparts for earlier browser versions.