CHANGELOG.md

0.7.0

• Updated to WebRTC M96.
• Added target property to event objects.
• Support for multiple sinks per audio source.
• Fixed RTCRtpSender::GetParameters.

0.6.1

• Reduce size of npm package.

0.6.0

• Updated to WebRTC M95.
• Added support for Apple silicon.
• Builds with H.264 enabled.
• Use wrtc.s3.amazonaws.com as the distribution URL.

Forked to @cubicleai/wrtc

vNext (0.5.0)

• Move Javascript sources to Typescript
• Use Visual Studio 2022 for Windows builds
• Add cmake project() directive
• Add VS code C++ configuration
• Set up webrtc.astrocdn.com for future binary builds

Forked to @astronautlabs/webrtc

The previous changelog entries correspond to releases of the wrtc package from which @astronautlabs/webrtc was forked.

0.4.6

New Features

• Added target property to RTCPeerConnection events (thanks, @CharlesRA).
• Support for additional APIs in lib/browser.js (thanks, @piranna).
• Added a naïve version of getDeviceMedia that delegates to getUserMedia (thanks, @piranna).

Bug Fixes

• Although Node 14 support was confirmed in v0.4.5, it was not included in the "engines" property of package.json (thanks, @farnabaz).
• Potential fix for a crash mentioned in #637 (thanks, @thedracle).

0.4.5

New Features

• Updated to WebRTC M81.
• Added support for Node 14.
• Added rollback support.

Bug Fixes

• RTCPeerConnection no longer raises "icegatheringstatechange" when the RTCPeerConnection is closed. Thanks, @arlolra. (#625)

Breaking Changes

• With the update from M79 to M81, the dtx, ptime, and codecPayloadType parameters to RTCRtpEncodingParameters no longer take affect. They've also been removed from the WebRTC 1.0 specification (see here). Although this is technically a SemVer-breaking change, few users of this library are depending on the removed functionality, and I prefer not to increment the version number at this time.

0.4.4

New Features

• addTrack now supports multiple MediaStream arguments (#548). Additionally, MediaStreams can now be constructed with arbitrary IDs. For more information, see below. Thanks, @csheely and @sgodin.
• setStreams now supports multiple MediaStream arguments.

MediaStream

MediaStreams in node-webrtc can be constructed with arbitrary IDs. For example, the following MediaStream, stream, has its ID set to "foo".

const stream = new MediaStream({ id: "foo" });
stream.id === "foo"; // true

0.4.3

New Features

• Updated to WebRTC M79.
• Added support for Node 13.
• Added support for a number of new standard APIs (see below).

RTCPeerConnection

• Added support for restartIce.
• Added support for "icecandidateerror" events.

RTCDtlsTransport

• Added support for getRemoteCertificates.

RTCRtpSender, RTCRtpReceiver & RTCRtpTransceiver

• Added support for getCapabilities to RTCRtpSender and RTCRtpReceiver.
• Added support for setParameters to RTCRtpSender.
• Added support for setStreams to RTCRtpSender (at this time, up to one MediaStream argument is supported).
• Added support for sendEncodings to RTCRtpTransceiverInit.
• Added support for setCodecPreferences to RTCRtpTransceiver.

Bug Fixes

• Fixed a bug where VideoFrame timestamps reported via RTCP were incorrect (#566). Thanks, @lonocvb.
• Fixed a bug where, in some cases, Ninja builds failed on macOS (#582). Thanks, @taylorhoward92.
• Fixed bugs related to N-API usage in recent version of Node 12 and 13.

0.4.2

Bug Fixes

• Fix image stride issue at certain resolutions. (#536)

0.4.1

Bug Fixes

• Fix memory leak when receiving strings over RTCDataChannel. (#528)

0.4.0

node-webrtc is now implemented using N-API, which is ABI stable across Node releases. This means we can now ship fewer binaries while supporting a potentially greater number of Node releases. As of 0.4.0, node-webrtc targets N-API version 3.

New Features

• Updated to WebRTC M74.
• Added support for Node 12.
• Added support for Electron 4 and 5.
• Added initial RTCIceTransport support (see below).
• Added initial RTCSctpTransport support (see below).
• Expanded RTCIceCandidate support (see below).

Bug Fixes

• Avoid crashing when createDataChannel fails. (#508)

Breaking Changes

• Dropped support for Node 6.
• Installing from NPM only downloads pre-built binaries. If you wish to build from source, install from the source repository. (#200)
• Unified Plan is now the default. For Plan B behavior, set the sdpSemantics RTCConfiguration property or the SDP_SEMANTICS environment variable to "plan-b".

RTCIceTransport

RTCDtlsTransport now exposes RTCIceTransport under the iceTransport property.

The following attributes are supported:

• role
• component
• state
• gatheringState

The following events are supported:

• "statechange"
• "gatheringstatechange"

RTCSctpTransport

RTCPeerConnection now exposes RTCSctpTransport under the sctp property.

The following attributes are supported:

• transport
• state

The following attributes are partially supported:

• maxMessageSize (always null)
• maxChannels (always null)

The "statechange" event is also supported.

RTCIceCandidate

RTCIceCandidates now include the following attributes:

• foundation
• component
• priority
• address
• protocol
• port
• type
• tcpType
• relatedAddress
• relatedPort
• usernameFragment

0.3.7

New Features

RTCDtlsTransport

RTCRtpSender and RTCRtpReceiver now provide access to RTCDtlsTransport via the transport property (initially null). Currently, RTCDtlsTransport only supports the state property, the "statechange" event, and the "error" event.

Miscellaneous

• Updated to WebRTC M73.
• Added maxPacketLifeTime getter to RTCDataChannel (#492).
• Added negotiated getter to RTCDataChannel.

Bug Fixes

• Fixed addIceCandidate queueing behavior (#498).

0.3.6

New Features

Programmatic Audio

This release of node-webrtc adds non-standard, programmatic audio APIs in the form of RTCAudioSource and RTCAudioSink. These APIs are similar to the previously added RTCVideoSource and RTCVideoSink APIs. With these APIs, you can

• Pass audio samples to RTCAudioSource via the onData method. Then use the RTCAudioSource's createTrack method to create a local audio MediaStreamTrack.
• Construct an RTCAudioSink from a local or remote audio MediaStreamTrack. The RTCAudioSink will emit a "data" event every time audio samples are received. When you're finished, stop the RTCAudioSink by calling stop.

Because these APIs are non-standard, they are exposed via a nonstandard property on node-webrtc's exports object. For example,

const { RTCAudioSource, RTCAudioSink } = require("wrtc").nonstandard;

const source = new RTCAudioSource();
const track = source.createTrack();
const sink = new RTCAudioSink(track);

const sampleRate = 8000;
const samples = new Int16Array(sampleRate / 100); // 10 ms of 16-bit mono audio

const data = {
  samples,
  sampleRate,
};

const interval = setInterval(() => {
  // Update audioData in some way before sending.
  source.onData(data);
});

sink.ondata = (data) => {
  // Do something with the received audio samples.
};

setTimeout(() => {
  clearInterval(interval);
  track.stop();
  sink.stop();
}, 10000);

RTCAudioSource

[constructor]
interface RTCAudioSource {
  MediaStreamTrack createTrack();
  void onData(RTCAudioData data);
};

dictionary RTCAudioData {
  required Int16Array samples;
  required unsigned short sampleRate;
  octet bitsPerSample = 16;
  octet channelCount = 1;
  unsigned short numberOfFrames;
};

• Calling createTrack will return a local audio MediaStreamTrack whose source is the RTCAudioSource.
• Calling onData with RTCAudioData pushes a new audio samples to every non-stopped local audio MediaStreamTrack created with createTrack.
• RTCAudioData should represent 10 ms worth of 16-bit audio samples.

RTCAudioSink

[constructor(MediaStreamTrack track)]
interface RTCAudioSink {
  void stop();
  readonly attribute boolean stopped;
  attribute EventHandler ondata;
};

• RTCAudioSink's constructor accepts a local or remote audio MediaStreamTrack.
• As long as neither the RTCAudioSink nor the RTCAudioSink's MediaStreamTrack are stopped, the RTCAudioSink will raise a "data" event any time RTCAudioData is received.
• The "data" event has all the properties of RTCAudioData.
• RTCAudioSink must be stopped by calling stop.

RTCVideoFrame rotation

The RTCVideoFrame raised in RTCVideoSink's "frame" event now includes a property, rotation, which indicates rotation of the RTCVideoFrame. Possible values are 0, 90, 180, and 270.

EventListener handleEvent

EventListener instances now support handleEvent.

0.3.5

New Features

Programmatic Video

This release of node-webrtc adds non-standard, programmatic video APIs in the form of RTCVideoSource and RTCVideoSink. With these APIs, you can

• Pass I420 frames to RTCVideoSource via the onFrame method. Then use RTCVideoSource's createTrack method to create a local video MediaStreamTrack.
• Construct an RTCVideoSink from a local or remote video MediaStreamTrack. The RTCVideoSink will emit a "frame" event every time an I420 frame is received. When you're finished, stop the RTCVideoSink by calling stop.

Because these APIs are non-standard, they are exposed via a nonstandard property on node-webrtc's exports object. For example,

const { RTCVideoSource, RTCVideoSink } = require("wrtc").nonstandard;

const source = new RTCVideoSource();
const track = source.createTrack();
const sink = new RTCVideoSink(track);

const width = 320;
const height = 240;
const data = new Uint8ClampedArray(width * height * 1.5);
const frame = { width, height, data };

const interval = setInterval(() => {
  // Update the frame in some way before sending.
  source.onFrame(frame);
});

sink.onframe = ({ frame }) => {
  // Do something with the received frame.
};

setTimeout(() => {
  clearInterval(interval);
  track.stop();
  sink.stop();
}, 10000);

This release also adds bindings to some libyuv functions for handling I420 frames. These can be useful when converting to and from RGBA.

RTCVideoSource

[constructor(optional RTCVideoSourceInit init)]
interface RTCVideoSource {
  readonly attribute boolean isScreencast;
  readonly attribute boolean? needsDenoising;
  MediaStreamTrack createTrack();
  void onFrame(RTCVideoFrame frame);
};

dictionary RTCVideoSourceInit {
  boolean isScreencast = false;
  boolean needsDenoising;
};

dictionary RTCVideoFrame {
  required unsigned long width;
  required unsigned long height;
  required Uint8ClampedArray data;
};

• Calling createTrack will return a local video MediaStreamTrack whose source is the RTCVideoSource.
• Calling onFrame with an RTCVideoFrame pushes a new video frame to every non-stopped local video MediaStreamTrack created with createTrack.
• An RTCVideoFrame represents an I420 frame.

RTCVideoSink

[constructor(MediaStreamTrack track)]
interface RTCVideoSink {
  void stop();
  readonly attribute boolean stopped;
  attribute EventHandler onframe;
};

• RTCVideoSink's constructor accepts a local or remote video MediaStreamTrack.
• As long as neither the RTCVideoSink nor the RTCVideoSink's MediaStreamTrack are stopped, the RTCVideoSink will raise a "frame" event any time an RTCVideoFrame is received.
• The "frame" event has a property, frame, of type RTCVideoFrame.
• RTCVideoSink must be stopped by calling stop.

i420ToRgba and rgbaToI420

These two functions are bindings to libyuv that provide conversions between I420 and RGBA frames. WebRTC expects I420, whereas APIs like the Canvas API expect RGBA, so these functions are useful for converting between. For example,

const { i420ToRgba, rgbaToI420 } = require("wrtc").nonstandard;

const width = 640;
const height = 480;
const i420Data = new Uint8ClampedArray(width * height * 1.5);
const rgbaData = new Uint8ClampedArray(width * height * 4);
const i420Frame = { width, height, data: i420Data };
const rgbaFrame = { width, height, data: rgbaData };

i420ToRgba(i420Frame, rgbaFrame);
rgbaToI420(rgbaFrame, i420Frame);

MediaStreamTrack

• Added support for setting MediaStreamTrack's enabled property (#475).

0.3.4

New Features

• Updated to WebRTC M71.
• Relay remote audio MediaStreamTracks on Windows (0.1.5 initially introduced this feature for Linux and macOS; now, Windows supports it, too).
• Added support for pkg (#404).

Bug Fixes

• Calling certain methods, like addTrack, removeTrack, etc., with objects that were not instances of MediaStreamTrack, RTCRtpSender, etc., could lead to segfaults. This was because we did not properly validate objects before attempting to unwrap them. (#448)

0.3.3

New Features

• Experimental support for armv7l and arm64. Binaries built for these architectures have been tested with QEMU but not on real devices. Please test them out. If you install node-webrtc directly on an ARM device, node-pre-gyp should pull the correct binaries automatically. Otherwise, you may need to set the TARGET_ARCH environment variable to "arm" (armv7l) or "arm64". For example,

  TARGET_ARCH=arm64 npm install
  

• Set DEBUG=true to install debug binaries (Linux- and macOS-only). For example,

  DEBUG=true npm install
  

0.3.2

New Features

• Support for Node 11 on Windows.

0.3.1

This release adds a number of new features and brings us closer to spec-compliance, thanks to the tests at web-platform-tests/wpt.

New Features

getUserMedia

This release adds limited getUserMedia support. You can create audio and video MediaStreamTracks; however, the resulting MediaStreamTracks do not capture any media. You can add these MediaStreamTracks to an RTCPeerConnection; however, no media will be transmitted. You can confirm by checking bytesSent and bytesReceived in getStats.

const { getUserMedia } = require("wrtc");

getUserMedia({
  audio: true,
  video: true,
}).then((stream) => {
  stream.getTracks().forEach((track) => stop());
});

Although we will parse and validate some members of the MediaStreamConstraints and related dictionaries, we do not use their values at this time.

getStats

This release adds limited standards-compliant getStats support. Previous node-webrtc releases exposed the legacy, callback-based getStats API. This release preserves that API but adds the Promise-based API. Neither the MediaStreamTrack selector argument nor the RTCRtpSender- and RTCRtpReceiver-level getStats APIs are implemented at this time.

// Legacy API
pc.getStats((response) => {
  /* ... */
}, console.error);

// Standards-compliant API
pc.getStats().then((report) => {
  /* ... */
}, console.error);

Unified Plan and sdpSemantics

This release adds support for RTCRtpTransceivers and Unified Plan SDP via

• A non-standard RTCConfiguration option, sdpSemantics, and
• An environment variable, SDP_SEMANTICS.

Construct an RTCPeerConnection with sdpSemantics set to "unified-plan" or launch your application with SDP_SEMANTICS=unified-plan to enable RTCRtpTransceiver support; otherwise, "plan-b" is the default.

const { RTCPeerConnection } = require("wrtc");

const pc = new RTCPeerConnection({
  sdpSemantics: "unified-plan", // default is "plan-b"
});

SDP_SEMANTICS=unified-plan node app.js

RTCRtpTransceiver

You can use RTCRtpTransceivers and related APIs when Unified Plan is enabled. This includes the following RTCPeerConnection methods

• addTransceiver
• getTransceivers

and the following RTCTrackEvent properties

• transceiver

The following RTCRtpTransceiver methods are supported

• stop

as well as the following RTCRtpTransceiver properties

• mid
• sender
• receiver
• stopped
• direction
• currentDirection

setCodecPreferences is not yet implemented. When calling addTransceiver, only the following RTCRtpTransceiverInit dictionary members are supported

• direction
• streams

const assert = require("assert");
const { MediaStream, RTCPeerConnection, RTCRtpTransceiver } = require("wrtc");

const pc = new RTCPeerConnection({
  sdpSemantics: "unified-plan",
});

const t1 = pc.addTransceiver("audio", {
  direction: "recvonly",
});

const t2 = pc.addTransceiver(t1.receiver.track, {
  direction: "sendonly",
  streams: [new MediaStream()],
});

MediaStreamTrack

Added limited support for the muted property (it always returns false).

Miscellaneous

• APIs that should throw DOMExceptions, such as addTrack, will use domexception to construct those DOMExceptions, if installed.

Bug Fixes

• Calling addTrack twice with the same MediaStreamTrack should throw an InvalidAccessError (#442).
• MediaStream's getTrackById did not work for video MediaStreamTracks.
• MediaStream's clone method did not clone MediaStreamTracks.
• MediaStreamTrack's readyState was not updated when stop was called.

0.3.0

New Features

• Support for Node 11. Binaries are available for Linux and macOS. Windows binaries will become available in a subsequent release once AppVeyor gains support for Node 11.
• Updated to WebRTC M70. This release no longer uses mayeut/libwebrtc; instead, WebRTC is built from source.

Breaking Changes

• Dropped support for Node 9
• Minimum CMake version bumped to 3.12
• Minimum GCC version bumped to 5.4
• Minimum Microsoft Visual Studio version bumped to 2017

Bug Fixes

• Updating to WebRTC M70 fixes an RTCDataChannel-related interop bug with recent Firefox releases (#444).

0.2.2

Bug Fixes

• Destroy AudioDeviceModule on the worker thread.

0.2.1

Bug Fixes

• Fixed an AudioDeviceModule memory and thread leak (#429).
• Fixed an issue where closing an RTCPeerConnection would raise "open" events on any RTCDataChannels whose readyState was "connecting" (#436).

0.2.0

Breaking Changes

• Dropped support for Node 4, 5 and 7 (#408).

Bug Fixes

• Fixed a race when closing an RTCDataChannel (#358).
• Fixed memory leaks in createOffer, createAnswer, addIceCandidate, and getStats (#425).

0.1.6

Bug Fixes

• Fixed an issue with receiving multiple ArrayBuffers over an RTCDataChannel that could cause invalid memory accesses (#406).

0.1.5

New Features

• This release allows relaying remote MediaStreamTracks. This can be useful for test applications. Note: currently, Windows cannot relay audio MediaStreamTracks, only video.

MediaStream

This release adds support for MediaStream. Most MediaStream APIs are supported, excluding the "addtrack" and "removetrack" events. You can construct MediaStreams as follows:

const { MediaStream } = require("wrtc");

const stream1 = new MediaStream();
const stream2 = new MediaStream(stream1);

Assuming you already have some Array of MediaStreamTracks, tracks, you can also construct a MediaStream with

const stream3 = new MediaStream(tracks);

Or, if you have an existing MediaStream, you can clone it.

const stream4 = stream3.clone();

This release also adds support for the following methods

• getTracks
• getAudioTracks
• getVideoTracks
• getTrackById
• addTrack
• removeTrack

and the following attributes

• id
• active

RTCTrackEvent

This release adds support for the streams property.

RTCPeerConnection

This release adds support for addTrack, removeTrack, and getSenders. Although we don't yet provide a way to construct local MediaStreamTracks, you can relay remote MediaStreamTracks as follows:

pc.ontrack = ({ track, streams }) => {
  pc.addTrack(track, ...streams);
};

RTCRtpSender

This release adds support for the following methods

• getCapabilities (always throws for now)
• getParameters
• setParameters (always returns a rejected Promise for now)
• getStats (always returns a rejected Promise for now)
• replaceTrack

and the following attributes

• track
• transport (always returns null for now)
• rtcpTransport (always returns null for now)

0.1.4

New Features

• Added support for Node 10 (#402)

RTCPeerConnection

• Added support for getReceivers.
• Added partial support for the RTCTrackEvent. RTCPeerConnection will emit the RTCTrackEvent with two attributes: track and receiver. In the future, we will add support for the streams and transceiver attributes.

RTCRtpReceiver

This release adds partial support for RTCRtpReceiver, including methods

• getParameters
• getContributingSources (always returns an empty array for now)
• getSynchronizationSources (always returns an empty array for now)
• getStats (always returns a rejected Promise for now)

and attributes

• track
• transport (always returns null for now)
• rtcpTransport (always returns null for now)

RTCRtpReceiver also includes partial support for the static method getCapabilities; however, it always returns a rejected Promise.

MediaStreamTrack

This release adds partial support for remote MediaStreamTracks, including attributes

• enabled (read-only for now)
• id
• kind
• readyState

0.1.3

Bug Fixes

• Fixed memory leaks related to RTCPeerConnection events.

0.1.2

Bug Fixes

• Fixed memory leaks related to sending and receiving messages over RTCDataChannels (#205, #304, #319). There are some less severe leaks related to RTCPeerConnection events that remain. These will be addressed in a future release.

0.1.1

Bug Fixes

• Calling createDataChannel on a closed RTCPeerConnection no longer returns undefined; instead, it raises an InvalidStateError (#314, #382).
• Worked around WebRTC Issue 7585 on Linux by backporting the epoll-based PhysicalSocketServer from WebRTC M61 into node-webrtc. This allows many more concurrent RTCPeerConnections on Linux (for example, up to 3000 in my tests, not exceeding thread limits). (#362)

0.1.0

This project will begin to follow SemVer in preparation for a 1.0.0 release.

New Features

Besides updating to WebRTC M60 (using mayeut/libwebrtc), this release adds a number of features that bring node-webrtc closer to standards-compliance. We still have a ways to go, but we're now testing against w3c/web-platform-tests.

RTCConfiguration

RTCPeerConnection's constructor now accepts the following standard properties:

• bundlePolicy
• iceCandidatePoolSize
• iceServers (no support for OAuth yet)
• iceTransportPolicy
• rtcpMuxPolicy

RTCConfiguration also accepts a non-standard property, portRange. This property constrains the port range used by the RTCPeerConnection's ICE transports. For example,

const { RTCPeerConnection } = require("wrtc");

const pc = new RTCPeerConnection({
  portRange: {
    min: 10000, // defaults to 0
    max: 20000, // defaults to 65535
  },
});

RTCPeerConnection

RTCPeerConnection now supports two new methods:

• getConfiguration
• setConfiguration

RTCPeerConnection now supports the following properties:

• canTrickleIceCandidates (always returns null for now)
• connectionState (derived from iceConnectionState)
• currentLocalDescription
• currentRemoteDescription
• pendingLocalDescription
• pendingRemoteDescription

RTCPeerConnection now supports the following events:

• "connectionstatechange"
• "negotiationneeded"

RTCOfferOptions and RTCAnswerOptions

RTCPeerConnection's createOffer method now accepts RTCOfferOptions, and RTCPeerConnection's createAnswer method now accepts RTCAnswerOptions. RTCOfferOptions supports

• iceRestart
• offerToReceiveAudio
• offerToReceiveVideo

Both RTCOfferOptions and RTCAnswerOptions support voiceActivityDetection.

RTCDataChannelInit

RTCPeerConnection's createDataChannel method now accepts

• id
• maxPacketLifeTime
• maxRetransmits
• negotiated
• ordered
• protocol

RTCDataChannel

RTCDataChannel supports the following properties:

• id
• maxRetransmits
• ordered
• priority (always returns "high")
• protocol

RTCDataChannel's send method now supports sending Blobs provided by jsdom; however, there is no support for receiving Blobs.

Top-level Exports

Added top-level exports for

• RTCDataChannel
• RTCDataChannelEvent
• RTCPeerConnectionIceEvent

Bug Fixes

• Fixed a failed assertion when closing RTCPeerConnection's or RTCDataChannel's event loop (#376).
• Moved AudioDeviceModule construction to the worker thread. This fixes a thread checker assertion raised by debug builds of libwebrtc.
• Copy StatsReports on the signaling thread. This fixes a thread checker assertion raised by debug builds of libwebrtc.

Breaking Changes

• Dropped support for "moz"- and "webkit"-prefixed WebRTC APIs in lib/browser.js. This means that, when bundling JavaScript that depends on node-webrtc for the browser, the resulting bundle will no longer depend on these APIs (#369).
• Dropped support for the now non-standard url attribute in RTCIceServer.
• Dropped support for the RTCIceConnectionStates, RTCIceGatheringStates, and RTCSignalingStates properties on the RTCPeerConnection prototype. These were an implementation detail that some libraries used for detecting node-webrtc.
• Dropped support for the RTCDataStates and BinaryTypes properties on the RTCDataChannel prototype. This, too, was an implementation detail.

0.0.67

Bug Fixes

• ObjectWrap instances accessed in an event loop (like PeerConnection and DataChannel) were getting freed before the event loop completed, which caused segfaults. Now, we call Ref in the ObjectWrap instances' constructors and Unref in their event loops.
• Fixed another source of segfaults, where, if a DataChannel's PeerConnectionFactory was freed, accessing the underlying DataChannelInterface would try to use Threads which had been freed.

0.0.66

Bug Fixes

• Fixed a CPU regression introduced in 0.0.63. We now share a single PeerConnectionFactoryInterface across PeerConnectionInterfaces, and we now use a "dummy" AudioDeviceModule instead of FakeAudioDeviceModule.

0.0.65

New Features

• Added support for sending Buffers (#103)

Bug Fixes

• Sending an ArrayBufferView over an RTCDataChannel did not take into account the ArrayBufferView's offset or length properties. This resulted in sending the entire backing ArrayBuffer instead of just the data in the ArrayBufferView.
• unzip-stream 0.2.2 breaks compatibility with Node 4 and 5. This release pins to unzip-stream 0.2.1.

Breaking Changes

• Building from source requires CMake 3.1 or newer

0.0.64

Bug Fixes

• We no longer Externalize ArrayBuffers. This fixes an error when sending ArrayBuffers mutliple times (#262 and #264) and a memory leak (#304).
• Fixed RTCDataChannel-related segfaults by checking for nullptr (#236 and #325)

0.0.63

New Features

• Support for Node 9
• Updated to WebRTC M57 (using libwebrtc)

Breaking Changes

• Minimum Mac OS X version bumped to 10.9
• Minimum Microsoft Visual Studio version bumped to 2015