diff --git a/webrtc/MediaStream.d.ts b/webrtc/MediaStream.d.ts
index a7bebb857..c4a836d61 100644
--- a/webrtc/MediaStream.d.ts
+++ b/webrtc/MediaStream.d.ts
@@ -1,204 +1,204 @@
-// Type definitions for WebRTC
-// Project: http://dev.w3.org/2011/webrtc/
-// Definitions by: Ken Smith <https://github.com/smithkl42/>
-// Definitions: https://github.com/borisyankov/DefinitelyTyped
-
-// Taken from http://dev.w3.org/2011/webrtc/editor/getusermedia.html
-// version: W3C Editor's Draft 29 June 2015
-
-/// <reference path="../es6-promise/es6-promise.d.ts" />
-
-interface ConstrainBooleanParameters {
-    exact?: boolean;
-    ideal?: boolean;
-}
-
-interface NumberRange {
-    max?: number;
-    min?: number;
-}
-
-interface ConstrainNumberRange extends NumberRange {
-    exact?: number;
-    ideal?: number;
-}
-
-interface ConstrainStringParameters {
-    exact?: string | string[];
-    ideal?: string | string[];
-}
-
-interface MediaStreamConstraints {
-    video?: boolean | MediaTrackConstraints;
-    audio?: boolean | MediaTrackConstraints;
-}
-
-declare module W3C {
-    type LongRange = NumberRange;
-    type DoubleRange = NumberRange;
-    type ConstrainBoolean = boolean | ConstrainBooleanParameters;
-    type ConstrainNumber = number | ConstrainNumberRange;
-    type ConstrainLong = ConstrainNumber;
-    type ConstrainDouble = ConstrainNumber;
-    type ConstrainString = string | string[] | ConstrainStringParameters;
-}
-
-interface MediaTrackConstraints extends MediaTrackConstraintSet {
-    advanced?: MediaTrackConstraintSet[];
-}
-
-interface MediaTrackConstraintSet {
-    width?: W3C.ConstrainLong;
-    height?: W3C.ConstrainLong;
-    aspectRatio?: W3C.ConstrainDouble;
-    frameRate?: W3C.ConstrainDouble;
-    facingMode?: W3C.ConstrainString;
-    volume?: W3C.ConstrainDouble;
-    sampleRate?: W3C.ConstrainLong;
-    sampleSize?: W3C.ConstrainLong;
-    echoCancellation?: W3C.ConstrainBoolean;
-    latency?: W3C.ConstrainDouble;
-    deviceId?: W3C.ConstrainString;
-    groupId?: W3C.ConstrainString;
-}
-
-interface MediaTrackSupportedConstraints {
-    width?: boolean;
-    height?: boolean;
-    aspectRatio?: boolean;
-    frameRate?: boolean;
-    facingMode?: boolean;
-    volume?: boolean;
-    sampleRate?: boolean;
-    sampleSize?: boolean;
-    echoCancellation?: boolean;
-    latency?: boolean;
-    deviceId?: boolean;
-    groupId?: boolean;
-}
-
-interface MediaStream extends EventTarget {
-    id: string;
-    active: boolean;
-
-    onactive: EventListener;
-    oninactive: EventListener;
-    onaddtrack: (event: MediaStreamTrackEvent) => any;
-    onremovetrack: (event: MediaStreamTrackEvent) => any;
-
-    clone(): MediaStream;
-    stop(): void;
-
-    getAudioTracks(): MediaStreamTrack[];
-    getVideoTracks(): MediaStreamTrack[];
-    getTracks(): MediaStreamTrack[];
-
-    getTrackById(trackId: string): MediaStreamTrack;
-
-    addTrack(track: MediaStreamTrack): void;
-    removeTrack(track: MediaStreamTrack): void;
-}
-
-interface MediaStreamTrackEvent extends Event {
-    track: MediaStreamTrack;
-}
-
-declare enum MediaStreamTrackState {
-	"live",
-    "ended"
-}
-
-interface MediaStreamTrack extends EventTarget {
-    id: string;
-    kind: string;
-    label: string;
-    enabled: boolean;
-    muted: boolean;
-    remote: boolean;
-    readyState: MediaStreamTrackState;
-
-    onmute: EventListener;
-    onunmute: EventListener;
-    onended: EventListener;
-    onoverconstrained: EventListener;
-
-    clone(): MediaStreamTrack;
-
-    stop(): void;
-
-    getCapabilities(): MediaTrackCapabilities;
-    getConstraints(): MediaTrackConstraints;
-    getSettings(): MediaTrackSettings;
-    applyConstraints(constraints: MediaTrackConstraints): Promise<void>;
-}
-
-interface MediaTrackCapabilities {
-    width: number | W3C.LongRange;
-    height: number | W3C.LongRange;
-    aspectRatio: number | W3C.DoubleRange;
-    frameRate: number | W3C.DoubleRange;
-    facingMode: string;
-    volume: number | W3C.DoubleRange;
-    sampleRate: number | W3C.LongRange;
-    sampleSize: number | W3C.LongRange;
-    echoCancellation: boolean[];
-    latency: number | W3C.DoubleRange;
-    deviceId: string;
-    groupId: string;
-}
-
-interface MediaTrackSettings {
-    width: number;
-    height: number;
-    aspectRatio: number;
-    frameRate: number;
-    facingMode: string;
-    volume: number;
-    sampleRate: number;
-    sampleSize: number;
-    echoCancellation: boolean;
-    latency: number;
-    deviceId: string;
-    groupId: string;
-}
-
-interface MediaStreamError {
-    name: string;
-    message: string;
-    constraintName: string;
-}
-
-interface NavigatorGetUserMedia {
-    (constraints: MediaStreamConstraints,
-     successCallback: (stream: MediaStream) => void,
-     errorCallback: (error: MediaStreamError) => void): void;
-}
-
-// to use with adapter.js, see: https://github.com/webrtc/adapter
-declare var getUserMedia: NavigatorGetUserMedia;
-
-interface Navigator {
-    getUserMedia: NavigatorGetUserMedia;
-
-    webkitGetUserMedia: NavigatorGetUserMedia;
-
-    mozGetUserMedia: NavigatorGetUserMedia;
-
-    msGetUserMedia: NavigatorGetUserMedia;
-
-    mediaDevices: MediaDevices;
-}
-
-interface MediaDevices {
-    getSupportedConstraints(): MediaTrackSupportedConstraints;
-
-    getUserMedia(constraints: MediaStreamConstraints): Promise<MediaStream>;
-    enumerateDevices(): Promise<MediaDeviceInfo[]>;
-}
-
-interface MediaDeviceInfo {
-    label: string;
-    id: string;
-    kind: string;
-    facing: string;
-}
+// Type definitions for WebRTC
+// Project: http://dev.w3.org/2011/webrtc/
+// Definitions by: Ken Smith <https://github.com/smithkl42/>
+// Definitions: https://github.com/borisyankov/DefinitelyTyped
+
+// Taken from http://dev.w3.org/2011/webrtc/editor/getusermedia.html
+// version: W3C Editor's Draft 29 June 2015
+
+/// <reference path="../es6-promise/es6-promise.d.ts" />
+
+interface ConstrainBooleanParameters {
+    exact?: boolean;
+    ideal?: boolean;
+}
+
+interface NumberRange {
+    max?: number;
+    min?: number;
+}
+
+interface ConstrainNumberRange extends NumberRange {
+    exact?: number;
+    ideal?: number;
+}
+
+interface ConstrainStringParameters {
+    exact?: string | string[];
+    ideal?: string | string[];
+}
+
+interface MediaStreamConstraints {
+    video?: boolean | MediaTrackConstraints;
+    audio?: boolean | MediaTrackConstraints;
+}
+
+declare module W3C {
+    type LongRange = NumberRange;
+    type DoubleRange = NumberRange;
+    type ConstrainBoolean = boolean | ConstrainBooleanParameters;
+    type ConstrainNumber = number | ConstrainNumberRange;
+    type ConstrainLong = ConstrainNumber;
+    type ConstrainDouble = ConstrainNumber;
+    type ConstrainString = string | string[] | ConstrainStringParameters;
+}
+
+interface MediaTrackConstraints extends MediaTrackConstraintSet {
+    advanced?: MediaTrackConstraintSet[];
+}
+
+interface MediaTrackConstraintSet {
+    width?: W3C.ConstrainLong;
+    height?: W3C.ConstrainLong;
+    aspectRatio?: W3C.ConstrainDouble;
+    frameRate?: W3C.ConstrainDouble;
+    facingMode?: W3C.ConstrainString;
+    volume?: W3C.ConstrainDouble;
+    sampleRate?: W3C.ConstrainLong;
+    sampleSize?: W3C.ConstrainLong;
+    echoCancellation?: W3C.ConstrainBoolean;
+    latency?: W3C.ConstrainDouble;
+    deviceId?: W3C.ConstrainString;
+    groupId?: W3C.ConstrainString;
+}
+
+interface MediaTrackSupportedConstraints {
+    width?: boolean;
+    height?: boolean;
+    aspectRatio?: boolean;
+    frameRate?: boolean;
+    facingMode?: boolean;
+    volume?: boolean;
+    sampleRate?: boolean;
+    sampleSize?: boolean;
+    echoCancellation?: boolean;
+    latency?: boolean;
+    deviceId?: boolean;
+    groupId?: boolean;
+}
+
+interface MediaStream extends EventTarget {
+    id: string;
+    active: boolean;
+
+    onactive: EventListener;
+    oninactive: EventListener;
+    onaddtrack: (event: MediaStreamTrackEvent) => any;
+    onremovetrack: (event: MediaStreamTrackEvent) => any;
+
+    clone(): MediaStream;
+    stop(): void;
+
+    getAudioTracks(): MediaStreamTrack[];
+    getVideoTracks(): MediaStreamTrack[];
+    getTracks(): MediaStreamTrack[];
+
+    getTrackById(trackId: string): MediaStreamTrack;
+
+    addTrack(track: MediaStreamTrack): void;
+    removeTrack(track: MediaStreamTrack): void;
+}
+
+interface MediaStreamTrackEvent extends Event {
+    track: MediaStreamTrack;
+}
+
+declare enum MediaStreamTrackState {
+	"live",
+    "ended"
+}
+
+interface MediaStreamTrack extends EventTarget {
+    id: string;
+    kind: string;
+    label: string;
+    enabled: boolean;
+    muted: boolean;
+    remote: boolean;
+    readyState: MediaStreamTrackState;
+
+    onmute: EventListener;
+    onunmute: EventListener;
+    onended: EventListener;
+    onoverconstrained: EventListener;
+
+    clone(): MediaStreamTrack;
+
+    stop(): void;
+
+    getCapabilities(): MediaTrackCapabilities;
+    getConstraints(): MediaTrackConstraints;
+    getSettings(): MediaTrackSettings;
+    applyConstraints(constraints: MediaTrackConstraints): Promise<void>;
+}
+
+interface MediaTrackCapabilities {
+    width: number | W3C.LongRange;
+    height: number | W3C.LongRange;
+    aspectRatio: number | W3C.DoubleRange;
+    frameRate: number | W3C.DoubleRange;
+    facingMode: string;
+    volume: number | W3C.DoubleRange;
+    sampleRate: number | W3C.LongRange;
+    sampleSize: number | W3C.LongRange;
+    echoCancellation: boolean[];
+    latency: number | W3C.DoubleRange;
+    deviceId: string;
+    groupId: string;
+}
+
+interface MediaTrackSettings {
+    width: number;
+    height: number;
+    aspectRatio: number;
+    frameRate: number;
+    facingMode: string;
+    volume: number;
+    sampleRate: number;
+    sampleSize: number;
+    echoCancellation: boolean;
+    latency: number;
+    deviceId: string;
+    groupId: string;
+}
+
+interface MediaStreamError {
+    name: string;
+    message: string;
+    constraintName: string;
+}
+
+interface NavigatorGetUserMedia {
+    (constraints: MediaStreamConstraints,
+     successCallback: (stream: MediaStream) => void,
+     errorCallback: (error: MediaStreamError) => void): void;
+}
+
+// to use with adapter.js, see: https://github.com/webrtc/adapter
+declare var getUserMedia: NavigatorGetUserMedia;
+
+interface Navigator {
+    getUserMedia: NavigatorGetUserMedia;
+
+    webkitGetUserMedia: NavigatorGetUserMedia;
+
+    mozGetUserMedia: NavigatorGetUserMedia;
+
+    msGetUserMedia: NavigatorGetUserMedia;
+
+    mediaDevices: MediaDevices;
+}
+
+interface MediaDevices {
+    getSupportedConstraints(): MediaTrackSupportedConstraints;
+
+    getUserMedia(constraints: MediaStreamConstraints): Promise<MediaStream>;
+    enumerateDevices(): Promise<MediaDeviceInfo[]>;
+}
+
+interface MediaDeviceInfo {
+    label: string;
+    id: string;
+    kind: string;
+    facing: string;
+}
diff --git a/webrtc/RTCPeerConnection-tests.ts b/webrtc/RTCPeerConnection-tests.ts
index b85c61806..1ed79bf2c 100644
--- a/webrtc/RTCPeerConnection-tests.ts
+++ b/webrtc/RTCPeerConnection-tests.ts
@@ -1,90 +1,90 @@
-/// <reference path="MediaStream.d.ts" />
-/// <reference path="RTCPeerConnection.d.ts" />
-
-var config: RTCConfiguration =
-    { iceServers: [{ urls: "stun.l.google.com:19302" }] };
-var constraints: RTCMediaConstraints =
-    { mandatory: { offerToReceiveAudio: true, offerToReceiveVideo: true } };
-
-var peerConnection: RTCPeerConnection =
-    new RTCPeerConnection(config, constraints);
-
-navigator.getUserMedia({ audio: true, video: true },
-  stream => {
-    peerConnection.addStream(stream);
-  },
-  error => {
-    console.log('Error message: ' + error.message);
-    console.log('Error name: ' + error.name);
-  });
-
-peerConnection.onaddstream = ev => console.log(ev.type);
-peerConnection.ondatachannel = ev => console.log(ev.type);
-peerConnection.oniceconnectionstatechange = ev => console.log(ev.type);
-peerConnection.onnegotiationneeded = ev => console.log(ev.type);
-peerConnection.onopen = ev => console.log(ev.type);
-peerConnection.onicecandidate = ev => console.log(ev.type);
-peerConnection.onremovestream = ev => console.log(ev.type);
-peerConnection.onstatechange = ev => console.log(ev.type);
-
-peerConnection.createOffer(
-  offer => {
-    peerConnection.setLocalDescription(offer,
-      () => console.log("set local description"),
-      error => console.log("Error setting local description: " + error));
-  },
-  error => console.log("Error creating offer: " + error));
-
-var type: string = RTCSdpType[RTCSdpType.offer];
-var offer: RTCSessionDescriptionInit = { type: type, sdp: "some sdp" };
-var sessionDescription = new RTCSessionDescription(offer);
-
-peerConnection.setRemoteDescription(sessionDescription, () => {
-  peerConnection.createAnswer(
-    answer => {
-      peerConnection.setLocalDescription(answer,
-        () => console.log('Set local description'),
-      error => console.log(
-          "Error setting local description from created answer: " + error +
-          "; answer.sdp=" + answer.sdp));
-    },
-  error => console.log("Error creating answer: " + error));
-},
-error => console.log('Error setting remote description: ' + error +
-    "; offer.sdp=" + offer.sdp));
-
-var webkitSessionDescription = new webkitRTCSessionDescription(offer);
-
-peerConnection.setRemoteDescription(webkitSessionDescription, () => {
-  peerConnection.createAnswer(
-    answer => {
-      peerConnection.setLocalDescription(answer,
-        () => console.log('Set local description'),
-      error => console.log(
-         "Error setting local description from created answer: " + error +
-         "; answer.sdp=" + answer.sdp));
-    },
-  error => console.log("Error creating answer: " + error));
-},
-error => console.log('Error setting remote description: ' + error +
-    "; offer.sdp=" + offer.sdp));
-
-var mozSessionDescription = new mozRTCSessionDescription(offer);
-
-peerConnection.setRemoteDescription(mozSessionDescription, () => {
-  peerConnection.createAnswer(
-    answer => {
-      peerConnection.setLocalDescription(answer,
-        () => console.log('Set local description'),
-      error => console.log(
-         "Error setting local description from created answer: " + error +
-         "; answer.sdp=" + answer.sdp));
-    },
-  error => console.log("Error creating answer: " + error));
-},
-error => console.log('Error setting remote description: ' + error +
-    "; offer.sdp=" + offer.sdp));
-
-var wkPeerConnection: webkitRTCPeerConnection =
-    new webkitRTCPeerConnection(config, constraints);
-
+/// <reference path="MediaStream.d.ts" />
+/// <reference path="RTCPeerConnection.d.ts" />
+
+var config: RTCConfiguration =
+    { iceServers: [{ urls: "stun.l.google.com:19302" }] };
+var constraints: RTCMediaConstraints =
+    { mandatory: { offerToReceiveAudio: true, offerToReceiveVideo: true } };
+
+var peerConnection: RTCPeerConnection =
+    new RTCPeerConnection(config, constraints);
+
+navigator.getUserMedia({ audio: true, video: true },
+  stream => {
+    peerConnection.addStream(stream);
+  },
+  error => {
+    console.log('Error message: ' + error.message);
+    console.log('Error name: ' + error.name);
+  });
+
+peerConnection.onaddstream = ev => console.log(ev.type);
+peerConnection.ondatachannel = ev => console.log(ev.type);
+peerConnection.oniceconnectionstatechange = ev => console.log(ev.type);
+peerConnection.onnegotiationneeded = ev => console.log(ev.type);
+peerConnection.onopen = ev => console.log(ev.type);
+peerConnection.onicecandidate = ev => console.log(ev.type);
+peerConnection.onremovestream = ev => console.log(ev.type);
+peerConnection.onstatechange = ev => console.log(ev.type);
+
+peerConnection.createOffer(
+  offer => {
+    peerConnection.setLocalDescription(offer,
+      () => console.log("set local description"),
+      error => console.log("Error setting local description: " + error));
+  },
+  error => console.log("Error creating offer: " + error));
+
+var type: string = RTCSdpType[RTCSdpType.offer];
+var offer: RTCSessionDescriptionInit = { type: type, sdp: "some sdp" };
+var sessionDescription = new RTCSessionDescription(offer);
+
+peerConnection.setRemoteDescription(sessionDescription, () => {
+  peerConnection.createAnswer(
+    answer => {
+      peerConnection.setLocalDescription(answer,
+        () => console.log('Set local description'),
+      error => console.log(
+          "Error setting local description from created answer: " + error +
+          "; answer.sdp=" + answer.sdp));
+    },
+  error => console.log("Error creating answer: " + error));
+},
+error => console.log('Error setting remote description: ' + error +
+    "; offer.sdp=" + offer.sdp));
+
+var webkitSessionDescription = new webkitRTCSessionDescription(offer);
+
+peerConnection.setRemoteDescription(webkitSessionDescription, () => {
+  peerConnection.createAnswer(
+    answer => {
+      peerConnection.setLocalDescription(answer,
+        () => console.log('Set local description'),
+      error => console.log(
+         "Error setting local description from created answer: " + error +
+         "; answer.sdp=" + answer.sdp));
+    },
+  error => console.log("Error creating answer: " + error));
+},
+error => console.log('Error setting remote description: ' + error +
+    "; offer.sdp=" + offer.sdp));
+
+var mozSessionDescription = new mozRTCSessionDescription(offer);
+
+peerConnection.setRemoteDescription(mozSessionDescription, () => {
+  peerConnection.createAnswer(
+    answer => {
+      peerConnection.setLocalDescription(answer,
+        () => console.log('Set local description'),
+      error => console.log(
+         "Error setting local description from created answer: " + error +
+         "; answer.sdp=" + answer.sdp));
+    },
+  error => console.log("Error creating answer: " + error));
+},
+error => console.log('Error setting remote description: ' + error +
+    "; offer.sdp=" + offer.sdp));
+
+var wkPeerConnection: webkitRTCPeerConnection =
+    new webkitRTCPeerConnection(config, constraints);
+
diff --git a/webrtc/RTCPeerConnection.d.ts b/webrtc/RTCPeerConnection.d.ts
index 661b9685b..0594f84c3 100644
--- a/webrtc/RTCPeerConnection.d.ts
+++ b/webrtc/RTCPeerConnection.d.ts
@@ -1,363 +1,363 @@
-// Type definitions for WebRTC
-// Project: http://dev.w3.org/2011/webrtc/
-// Definitions by: Ken Smith <https://github.com/smithkl42/>
-// Definitions: https://github.com/borisyankov/DefinitelyTyped
-//
-// Definitions taken from http://dev.w3.org/2011/webrtc/editor/webrtc.html
-//
-// For example code see:
-//   https://code.google.com/p/webrtc/source/browse/stable/samples/js/apprtc/js/main.js
-//
-// For a generic implementation see that deals with browser differences, see:
-//   https://code.google.com/p/webrtc/source/browse/stable/samples/js/base/adapter.js
-
-/// <reference path='MediaStream.d.ts' />
-
-// TODO(1): Get Typescript to have string-enum types as WebRtc is full of string
-// enums.
-// https://typescript.codeplex.com/discussions/549207
-
-// TODO(2): get Typescript to have union types as WebRtc uses them.
-// https://typescript.codeplex.com/workitem/1364
-
-interface RTCConfiguration {
-  iceServers: RTCIceServer[];
-}
-declare var RTCConfiguration: {
-  prototype: RTCConfiguration;
-  new (): RTCConfiguration;
-};
-
-interface RTCIceServer {
-  urls: string;
-  credential?: string;
-}
-declare var RTCIceServer: {
-  prototype: RTCIceServer;
-  new (): RTCIceServer;
-};
-
-// moz (Firefox) specific prefixes.
-interface mozRTCPeerConnection extends RTCPeerConnection {
-}
-declare var mozRTCPeerConnection: {
-  prototype: mozRTCPeerConnection;
-  new (settings: RTCPeerConnectionConfig,
-       constraints?:RTCMediaConstraints): mozRTCPeerConnection;
-};
-// webkit (Chrome) specific prefixes.
-interface webkitRTCPeerConnection extends RTCPeerConnection {
-}
-declare var webkitRTCPeerConnection: {
-  prototype: webkitRTCPeerConnection;
-  new (settings: RTCPeerConnectionConfig,
-       constraints?:RTCMediaConstraints): webkitRTCPeerConnection;
-};
-
-// For Chrome, look at the code here:
-// https://code.google.com/p/chromium/codesearch#chromium/src/third_party/libjingle/source/talk/app/webrtc/webrtcsession.cc&sq=package:chromium&dr=C&l=63
-interface RTCOptionalMediaConstraint {
-  // When true, will use DTLS/SCTP data channels
-  DtlsSrtpKeyAgreement?: boolean;
-  // When true will use Rtp-based data channels (depreicated)
-  RtpDataChannels?: boolean;
-}
-
-// ks 12/20/12 - There's more here that doesn't seem to be documented very well yet.
-// http://www.w3.org/TR/2013/WD-webrtc-20130910/
-interface RTCMediaConstraints {
-  mandatory?: RTCMediaOfferConstraints;
-  optional?: RTCOptionalMediaConstraint[]
-}
-
-interface RTCMediaOfferConstraints {
-  offerToReceiveAudio: boolean;
-  offerToReceiveVideo: boolean;
-}
-
-interface RTCSessionDescriptionInit {
-  type: string;  // RTCSdpType; See TODO(1)
-  sdp: string;
-}
-
-interface RTCSessionDescription {
-  type?: string;  // RTCSdpType; See TODO(1)
-  sdp?: string;
-}
-declare var RTCSessionDescription: {
-  prototype: RTCSessionDescription;
-  new (descriptionInitDict?: RTCSessionDescriptionInit): RTCSessionDescription;
-  // TODO: Add serializer.
-  // See: http://dev.w3.org/2011/webrtc/editor/webrtc.html#idl-def-RTCSdpType)
-};
-
-interface webkitRTCSessionDescription extends RTCSessionDescription{
-  type?: string;
-  sdp?: string;
-}
-declare var webkitRTCSessionDescription: {
-  prototype: webkitRTCSessionDescription;
-  new (descriptionInitDict?: RTCSessionDescriptionInit): webkitRTCSessionDescription;
-};
-
-interface mozRTCSessionDescription extends RTCSessionDescription{
-  type?: string;
-  sdp?: string;
-}
-declare var mozRTCSessionDescription: {
-  prototype: mozRTCSessionDescription;
-  new (descriptionInitDict?: RTCSessionDescriptionInit): mozRTCSessionDescription;
-};
-
-
-
-interface RTCDataChannelInit {
-  ordered             ?: boolean; // messages must be sent in-order.
-  maxPacketLifeTime   ?: number;  // unsigned short
-  maxRetransmits      ?: number;  // unsigned short
-  protocol            ?: string;  // default = ''
-  negotiated          ?: boolean; // default = false;
-  id                  ?: number;  // unsigned short
-}
-
-// TODO(1)
-declare enum RTCSdpType {
-  // http://dev.w3.org/2011/webrtc/editor/webrtc.html#rtcsdptype
-  'offer',
-  'pranswer',
-  'answer'
-}
-
-interface RTCMessageEvent {
-  // http://dev.w3.org/2011/webrtc/editor/webrtc.html#event-datachannel-message
-  // At present, this can be an: ArrayBuffer, a string, or a Blob.
-  // See TODO(2)
-  data: any;
-}
-
-// TODO(1)
-declare enum RTCDataChannelState {
-  // http://dev.w3.org/2011/webrtc/editor/webrtc.html#idl-def-RTCDataChannelState
-  'connecting',
-  'open',
-  'closing',
-  'closed'
-}
-
-interface RTCDataChannel extends EventTarget {
-  label: string;
-  reliable: boolean;
-  readyState: string; // RTCDataChannelState; see TODO(1)
-  bufferedAmount: number;
-  binaryType: string;
-
-  onopen: (event: Event) => void;
-  onerror: (event: Event) => void;
-  onclose: (event: Event) => void;
-  onmessage: (event: RTCMessageEvent) => void;
-
-  close(): void;
-
-  send(data: string): void ;
-  send(data: ArrayBuffer): void;
-  send(data: ArrayBufferView): void;
-  send(data: Blob): void;
-}
-declare var RTCDataChannel: {
-  prototype: RTCDataChannel;
-  new (): RTCDataChannel;
-};
-
-interface RTCDataChannelEvent extends Event {
-  channel: RTCDataChannel;
-}
-declare var RTCDataChannelEvent: {
-  prototype: RTCDataChannelEvent;
-  new (eventInitDict: RTCDataChannelEventInit): RTCDataChannelEvent;
-};
-
-interface RTCIceCandidateEvent extends Event {
-  candidate: RTCIceCandidate;
-}
-
-interface RTCMediaStreamEvent extends Event {
-  stream: MediaStream;
-}
-
-interface EventInit {
-}
-
-interface RTCDataChannelEventInit extends EventInit {
-  channel: RTCDataChannel;
-}
-
-interface RTCVoidCallback {
-  (): void;
-}
-interface RTCSessionDescriptionCallback {
-  (sdp: RTCSessionDescription): void;
-}
-interface RTCPeerConnectionErrorCallback {
-  (errorInformation: DOMError): void;
-}
-
-// TODO(1)
-declare enum RTCIceGatheringState {
-  // http://dev.w3.org/2011/webrtc/editor/webrtc.html#rtcicegatheringstate-enum
-  'new',
-  'gathering',
-  'complete'
-}
-
-// TODO(1)
-declare enum RTCIceConnectionState {
-  // http://dev.w3.org/2011/webrtc/editor/webrtc.html#idl-def-RTCIceConnectionState
-  'new',
-  'checking',
-  'connected',
-  'completed',
-  'failed',
-  'disconnected',
-  'closed'
-}
-
-// TODO(1)
-declare enum RTCSignalingState {
-  // http://dev.w3.org/2011/webrtc/editor/webrtc.html#idl-def-RTCSignalingState
-  'stable',
-  'have-local-offer',
-  'have-remote-offer',
-  'have-local-pranswer',
-  'have-remote-pranswer',
-  'closed'
-}
-
-// This is based on the current implementation of WebRtc in Chrome; the spec is
-// a little unclear on this.
-// http://dev.w3.org/2011/webrtc/editor/webrtc.html#idl-def-RTCStatsReport
-interface RTCStatsReport {
-  stat(id: string): string;
-}
-
-interface RTCStatsCallback {
-  (report: RTCStatsReport): void;
-}
-
-interface RTCPeerConnection {
-  createOffer(successCallback: RTCSessionDescriptionCallback,
-              failureCallback?: RTCPeerConnectionErrorCallback,
-              constraints?: RTCMediaConstraints): void;
-  createAnswer(successCallback: RTCSessionDescriptionCallback,
-               failureCallback?: RTCPeerConnectionErrorCallback,
-               constraints?: RTCMediaConstraints): void;
-  setLocalDescription(description: RTCSessionDescription,
-                      successCallback?: RTCVoidCallback,
-                      failureCallback?: RTCPeerConnectionErrorCallback): void;
-  localDescription: RTCSessionDescription;
-  setRemoteDescription(description: RTCSessionDescription,
-                        successCallback?: RTCVoidCallback,
-                        failureCallback?: RTCPeerConnectionErrorCallback): void;
-  remoteDescription: RTCSessionDescription;
-  signalingState: string; // RTCSignalingState; see TODO(1)
-  updateIce(configuration?: RTCConfiguration,
-            constraints?: RTCMediaConstraints): void;
-  addIceCandidate(candidate:RTCIceCandidate,
-                  successCallback:() => void,
-                  failureCallback:RTCPeerConnectionErrorCallback): void;
-  iceGatheringState: string;  // RTCIceGatheringState; see TODO(1)
-  iceConnectionState: string;  // RTCIceConnectionState; see TODO(1)
-  getLocalStreams(): MediaStream[];
-  getRemoteStreams(): MediaStream[];
-  createDataChannel(label?: string,
-                    dataChannelDict?: RTCDataChannelInit): RTCDataChannel;
-  ondatachannel: (event: Event) => void;
-  addStream(stream: MediaStream, constraints?: RTCMediaConstraints): void;
-  removeStream(stream: MediaStream): void;
-  close(): void;
-  onnegotiationneeded: (event: Event) => void;
-  onconnecting: (event: Event) => void;
-  onopen: (event: Event) => void;
-  onaddstream: (event: RTCMediaStreamEvent) => void;
-  onremovestream: (event: RTCMediaStreamEvent) => void;
-  onstatechange: (event: Event) => void;
-  oniceconnectionstatechange: (event: Event) => void;
-  onicecandidate: (event: RTCIceCandidateEvent) => void;
-  onidentityresult: (event: Event) => void;
-  onsignalingstatechange: (event: Event) => void;
-  getStats: (successCallback: RTCStatsCallback,
-             failureCallback: RTCPeerConnectionErrorCallback) => void;
-}
-declare var RTCPeerConnection: {
-  prototype: RTCPeerConnection;
-  new (configuration: RTCConfiguration,
-       constraints?: RTCMediaConstraints): RTCPeerConnection;
-};
-
-interface RTCIceCandidate {
-  candidate?: string;
-  sdpMid?: string;
-  sdpMLineIndex?: number;
-}
-declare var RTCIceCandidate: {
-  prototype: RTCIceCandidate;
-  new (candidateInitDict?: RTCIceCandidate): RTCIceCandidate;
-};
-
-interface webkitRTCIceCandidate extends RTCIceCandidate {
-  candidate?: string;
-  sdpMid?: string;
-  sdpMLineIndex?: number;
-}
-declare var webkitRTCIceCandidate: {
-  prototype: webkitRTCIceCandidate;
-  new (candidateInitDict?: webkitRTCIceCandidate): webkitRTCIceCandidate;
-};
-
-interface mozRTCIceCandidate extends RTCIceCandidate {
-  candidate?: string;
-  sdpMid?: string;
-  sdpMLineIndex?: number;
-}
-declare var mozRTCIceCandidate: {
-  prototype: mozRTCIceCandidate;
-  new (candidateInitDict?: mozRTCIceCandidate): mozRTCIceCandidate;
-};
-
-interface RTCIceCandidateInit {
-  candidate: string;
-  sdpMid: string;
-  sdpMLineIndex: number;
-}
-declare var RTCIceCandidateInit:{
-  prototype: RTCIceCandidateInit;
-  new (): RTCIceCandidateInit;
-};
-
-interface PeerConnectionIceEvent {
-  peer: RTCPeerConnection;
-  candidate: RTCIceCandidate;
-}
-declare var PeerConnectionIceEvent: {
-  prototype: PeerConnectionIceEvent;
-  new (): PeerConnectionIceEvent;
-};
-
-interface RTCPeerConnectionConfig {
-  iceServers: RTCIceServer[];
-}
-declare var RTCPeerConnectionConfig: {
-  prototype: RTCPeerConnectionConfig;
-  new (): RTCPeerConnectionConfig;
-};
-
-interface Window{
-  RTCPeerConnection: RTCPeerConnection;
-  webkitRTCPeerConnection: webkitRTCPeerConnection;
-  mozRTCPeerConnection: mozRTCPeerConnection;
-  RTCSessionDescription: RTCSessionDescription;
-  webkitRTCSessionDescription: webkitRTCSessionDescription;
-  mozRTCSessionDescription: mozRTCSessionDescription;
-  RTCIceCandidate: RTCIceCandidate;
-  webkitRTCIceCandidate: webkitRTCIceCandidate;
-  mozRTCIceCandidate: mozRTCIceCandidate;
-}
+// Type definitions for WebRTC
+// Project: http://dev.w3.org/2011/webrtc/
+// Definitions by: Ken Smith <https://github.com/smithkl42/>
+// Definitions: https://github.com/borisyankov/DefinitelyTyped
+//
+// Definitions taken from http://dev.w3.org/2011/webrtc/editor/webrtc.html
+//
+// For example code see:
+//   https://code.google.com/p/webrtc/source/browse/stable/samples/js/apprtc/js/main.js
+//
+// For a generic implementation see that deals with browser differences, see:
+//   https://code.google.com/p/webrtc/source/browse/stable/samples/js/base/adapter.js
+
+/// <reference path='MediaStream.d.ts' />
+
+// TODO(1): Get Typescript to have string-enum types as WebRtc is full of string
+// enums.
+// https://typescript.codeplex.com/discussions/549207
+
+// TODO(2): get Typescript to have union types as WebRtc uses them.
+// https://typescript.codeplex.com/workitem/1364
+
+interface RTCConfiguration {
+  iceServers: RTCIceServer[];
+}
+declare var RTCConfiguration: {
+  prototype: RTCConfiguration;
+  new (): RTCConfiguration;
+};
+
+interface RTCIceServer {
+  urls: string;
+  credential?: string;
+}
+declare var RTCIceServer: {
+  prototype: RTCIceServer;
+  new (): RTCIceServer;
+};
+
+// moz (Firefox) specific prefixes.
+interface mozRTCPeerConnection extends RTCPeerConnection {
+}
+declare var mozRTCPeerConnection: {
+  prototype: mozRTCPeerConnection;
+  new (settings: RTCPeerConnectionConfig,
+       constraints?:RTCMediaConstraints): mozRTCPeerConnection;
+};
+// webkit (Chrome) specific prefixes.
+interface webkitRTCPeerConnection extends RTCPeerConnection {
+}
+declare var webkitRTCPeerConnection: {
+  prototype: webkitRTCPeerConnection;
+  new (settings: RTCPeerConnectionConfig,
+       constraints?:RTCMediaConstraints): webkitRTCPeerConnection;
+};
+
+// For Chrome, look at the code here:
+// https://code.google.com/p/chromium/codesearch#chromium/src/third_party/libjingle/source/talk/app/webrtc/webrtcsession.cc&sq=package:chromium&dr=C&l=63
+interface RTCOptionalMediaConstraint {
+  // When true, will use DTLS/SCTP data channels
+  DtlsSrtpKeyAgreement?: boolean;
+  // When true will use Rtp-based data channels (depreicated)
+  RtpDataChannels?: boolean;
+}
+
+// ks 12/20/12 - There's more here that doesn't seem to be documented very well yet.
+// http://www.w3.org/TR/2013/WD-webrtc-20130910/
+interface RTCMediaConstraints {
+  mandatory?: RTCMediaOfferConstraints;
+  optional?: RTCOptionalMediaConstraint[]
+}
+
+interface RTCMediaOfferConstraints {
+  offerToReceiveAudio: boolean;
+  offerToReceiveVideo: boolean;
+}
+
+interface RTCSessionDescriptionInit {
+  type: string;  // RTCSdpType; See TODO(1)
+  sdp: string;
+}
+
+interface RTCSessionDescription {
+  type?: string;  // RTCSdpType; See TODO(1)
+  sdp?: string;
+}
+declare var RTCSessionDescription: {
+  prototype: RTCSessionDescription;
+  new (descriptionInitDict?: RTCSessionDescriptionInit): RTCSessionDescription;
+  // TODO: Add serializer.
+  // See: http://dev.w3.org/2011/webrtc/editor/webrtc.html#idl-def-RTCSdpType)
+};
+
+interface webkitRTCSessionDescription extends RTCSessionDescription{
+  type?: string;
+  sdp?: string;
+}
+declare var webkitRTCSessionDescription: {
+  prototype: webkitRTCSessionDescription;
+  new (descriptionInitDict?: RTCSessionDescriptionInit): webkitRTCSessionDescription;
+};
+
+interface mozRTCSessionDescription extends RTCSessionDescription{
+  type?: string;
+  sdp?: string;
+}
+declare var mozRTCSessionDescription: {
+  prototype: mozRTCSessionDescription;
+  new (descriptionInitDict?: RTCSessionDescriptionInit): mozRTCSessionDescription;
+};
+
+
+
+interface RTCDataChannelInit {
+  ordered             ?: boolean; // messages must be sent in-order.
+  maxPacketLifeTime   ?: number;  // unsigned short
+  maxRetransmits      ?: number;  // unsigned short
+  protocol            ?: string;  // default = ''
+  negotiated          ?: boolean; // default = false;
+  id                  ?: number;  // unsigned short
+}
+
+// TODO(1)
+declare enum RTCSdpType {
+  // http://dev.w3.org/2011/webrtc/editor/webrtc.html#rtcsdptype
+  'offer',
+  'pranswer',
+  'answer'
+}
+
+interface RTCMessageEvent {
+  // http://dev.w3.org/2011/webrtc/editor/webrtc.html#event-datachannel-message
+  // At present, this can be an: ArrayBuffer, a string, or a Blob.
+  // See TODO(2)
+  data: any;
+}
+
+// TODO(1)
+declare enum RTCDataChannelState {
+  // http://dev.w3.org/2011/webrtc/editor/webrtc.html#idl-def-RTCDataChannelState
+  'connecting',
+  'open',
+  'closing',
+  'closed'
+}
+
+interface RTCDataChannel extends EventTarget {
+  label: string;
+  reliable: boolean;
+  readyState: string; // RTCDataChannelState; see TODO(1)
+  bufferedAmount: number;
+  binaryType: string;
+
+  onopen: (event: Event) => void;
+  onerror: (event: Event) => void;
+  onclose: (event: Event) => void;
+  onmessage: (event: RTCMessageEvent) => void;
+
+  close(): void;
+
+  send(data: string): void ;
+  send(data: ArrayBuffer): void;
+  send(data: ArrayBufferView): void;
+  send(data: Blob): void;
+}
+declare var RTCDataChannel: {
+  prototype: RTCDataChannel;
+  new (): RTCDataChannel;
+};
+
+interface RTCDataChannelEvent extends Event {
+  channel: RTCDataChannel;
+}
+declare var RTCDataChannelEvent: {
+  prototype: RTCDataChannelEvent;
+  new (eventInitDict: RTCDataChannelEventInit): RTCDataChannelEvent;
+};
+
+interface RTCIceCandidateEvent extends Event {
+  candidate: RTCIceCandidate;
+}
+
+interface RTCMediaStreamEvent extends Event {
+  stream: MediaStream;
+}
+
+interface EventInit {
+}
+
+interface RTCDataChannelEventInit extends EventInit {
+  channel: RTCDataChannel;
+}
+
+interface RTCVoidCallback {
+  (): void;
+}
+interface RTCSessionDescriptionCallback {
+  (sdp: RTCSessionDescription): void;
+}
+interface RTCPeerConnectionErrorCallback {
+  (errorInformation: DOMError): void;
+}
+
+// TODO(1)
+declare enum RTCIceGatheringState {
+  // http://dev.w3.org/2011/webrtc/editor/webrtc.html#rtcicegatheringstate-enum
+  'new',
+  'gathering',
+  'complete'
+}
+
+// TODO(1)
+declare enum RTCIceConnectionState {
+  // http://dev.w3.org/2011/webrtc/editor/webrtc.html#idl-def-RTCIceConnectionState
+  'new',
+  'checking',
+  'connected',
+  'completed',
+  'failed',
+  'disconnected',
+  'closed'
+}
+
+// TODO(1)
+declare enum RTCSignalingState {
+  // http://dev.w3.org/2011/webrtc/editor/webrtc.html#idl-def-RTCSignalingState
+  'stable',
+  'have-local-offer',
+  'have-remote-offer',
+  'have-local-pranswer',
+  'have-remote-pranswer',
+  'closed'
+}
+
+// This is based on the current implementation of WebRtc in Chrome; the spec is
+// a little unclear on this.
+// http://dev.w3.org/2011/webrtc/editor/webrtc.html#idl-def-RTCStatsReport
+interface RTCStatsReport {
+  stat(id: string): string;
+}
+
+interface RTCStatsCallback {
+  (report: RTCStatsReport): void;
+}
+
+interface RTCPeerConnection {
+  createOffer(successCallback: RTCSessionDescriptionCallback,
+              failureCallback?: RTCPeerConnectionErrorCallback,
+              constraints?: RTCMediaConstraints): void;
+  createAnswer(successCallback: RTCSessionDescriptionCallback,
+               failureCallback?: RTCPeerConnectionErrorCallback,
+               constraints?: RTCMediaConstraints): void;
+  setLocalDescription(description: RTCSessionDescription,
+                      successCallback?: RTCVoidCallback,
+                      failureCallback?: RTCPeerConnectionErrorCallback): void;
+  localDescription: RTCSessionDescription;
+  setRemoteDescription(description: RTCSessionDescription,
+                        successCallback?: RTCVoidCallback,
+                        failureCallback?: RTCPeerConnectionErrorCallback): void;
+  remoteDescription: RTCSessionDescription;
+  signalingState: string; // RTCSignalingState; see TODO(1)
+  updateIce(configuration?: RTCConfiguration,
+            constraints?: RTCMediaConstraints): void;
+  addIceCandidate(candidate:RTCIceCandidate,
+                  successCallback:() => void,
+                  failureCallback:RTCPeerConnectionErrorCallback): void;
+  iceGatheringState: string;  // RTCIceGatheringState; see TODO(1)
+  iceConnectionState: string;  // RTCIceConnectionState; see TODO(1)
+  getLocalStreams(): MediaStream[];
+  getRemoteStreams(): MediaStream[];
+  createDataChannel(label?: string,
+                    dataChannelDict?: RTCDataChannelInit): RTCDataChannel;
+  ondatachannel: (event: Event) => void;
+  addStream(stream: MediaStream, constraints?: RTCMediaConstraints): void;
+  removeStream(stream: MediaStream): void;
+  close(): void;
+  onnegotiationneeded: (event: Event) => void;
+  onconnecting: (event: Event) => void;
+  onopen: (event: Event) => void;
+  onaddstream: (event: RTCMediaStreamEvent) => void;
+  onremovestream: (event: RTCMediaStreamEvent) => void;
+  onstatechange: (event: Event) => void;
+  oniceconnectionstatechange: (event: Event) => void;
+  onicecandidate: (event: RTCIceCandidateEvent) => void;
+  onidentityresult: (event: Event) => void;
+  onsignalingstatechange: (event: Event) => void;
+  getStats: (successCallback: RTCStatsCallback,
+             failureCallback: RTCPeerConnectionErrorCallback) => void;
+}
+declare var RTCPeerConnection: {
+  prototype: RTCPeerConnection;
+  new (configuration: RTCConfiguration,
+       constraints?: RTCMediaConstraints): RTCPeerConnection;
+};
+
+interface RTCIceCandidate {
+  candidate?: string;
+  sdpMid?: string;
+  sdpMLineIndex?: number;
+}
+declare var RTCIceCandidate: {
+  prototype: RTCIceCandidate;
+  new (candidateInitDict?: RTCIceCandidate): RTCIceCandidate;
+};
+
+interface webkitRTCIceCandidate extends RTCIceCandidate {
+  candidate?: string;
+  sdpMid?: string;
+  sdpMLineIndex?: number;
+}
+declare var webkitRTCIceCandidate: {
+  prototype: webkitRTCIceCandidate;
+  new (candidateInitDict?: webkitRTCIceCandidate): webkitRTCIceCandidate;
+};
+
+interface mozRTCIceCandidate extends RTCIceCandidate {
+  candidate?: string;
+  sdpMid?: string;
+  sdpMLineIndex?: number;
+}
+declare var mozRTCIceCandidate: {
+  prototype: mozRTCIceCandidate;
+  new (candidateInitDict?: mozRTCIceCandidate): mozRTCIceCandidate;
+};
+
+interface RTCIceCandidateInit {
+  candidate: string;
+  sdpMid: string;
+  sdpMLineIndex: number;
+}
+declare var RTCIceCandidateInit:{
+  prototype: RTCIceCandidateInit;
+  new (): RTCIceCandidateInit;
+};
+
+interface PeerConnectionIceEvent {
+  peer: RTCPeerConnection;
+  candidate: RTCIceCandidate;
+}
+declare var PeerConnectionIceEvent: {
+  prototype: PeerConnectionIceEvent;
+  new (): PeerConnectionIceEvent;
+};
+
+interface RTCPeerConnectionConfig {
+  iceServers: RTCIceServer[];
+}
+declare var RTCPeerConnectionConfig: {
+  prototype: RTCPeerConnectionConfig;
+  new (): RTCPeerConnectionConfig;
+};
+
+interface Window{
+  RTCPeerConnection: RTCPeerConnection;
+  webkitRTCPeerConnection: webkitRTCPeerConnection;
+  mozRTCPeerConnection: mozRTCPeerConnection;
+  RTCSessionDescription: RTCSessionDescription;
+  webkitRTCSessionDescription: webkitRTCSessionDescription;
+  mozRTCSessionDescription: mozRTCSessionDescription;
+  RTCIceCandidate: RTCIceCandidate;
+  webkitRTCIceCandidate: webkitRTCIceCandidate;
+  mozRTCIceCandidate: mozRTCIceCandidate;
+}