diff --git a/README.md b/README.md
index 9463c9bd..b349ea1b 100644
--- a/README.md
+++ b/README.md
@@ -30,9 +30,8 @@ $ cordova plugin add cordova-plugin-iosrtc
 
 ## Building
 
-This plugin needs [Swift 2.0](https://developer.apple.com/swift/) support so some steps are needed to get your project working with it. These steps are explained in the [Building](docs/Building.md) documentation, please check it.
-
-**IMPORTANT:** The currently bundled libWebRTC does not contain bitcode, so you'll need to disable it in your Xcode project settings.
+* [Building](docs/Building.md): Guidelines for building a Cordova iOS application including the *cordova-plugin-iosrtc* plugin.
+* [Building `libwebrtc`](docs/BuildingLibWebRTC.md): Guidelines for building Google's *libwebrtc* with modifications needed by the *cordova-plugin-iosrtc* plugin.
 
 
 ## Usage
diff --git a/docs/Building.md b/docs/Building.md
index 829e6e9b..bfe0cbce 100644
--- a/docs/Building.md
+++ b/docs/Building.md
@@ -1,10 +1,19 @@
 # Building
 
+
+### Requirements
+
+* This plugin needs [Swift 2.0](https://developer.apple.com/swift/) so Xcode >= 7.0 is required.
+* The currently bundled *libwebrtc* does not contain bitcode, so you need to disable it in your Xcode project settings.
+
+
+### Building Steps
+
 An iOS Cordova application including the *cordova-plugin-iosrtc* plugin can be built using the [cordova-cli](https://cordova.apache.org/docs/en/edge/guide_cli_index.md.html#The%20Command-Line%20Interface) or Xcode.
 
-The plugin provides a ["hook"](../extra/hooks/iosrtc-swift-support.js) to automate required modifications in both *cordova-cli* and Xcode generated projects. These steps must be followed:
+The plugin provides a ["hook"](../extra/hooks/iosrtc-swift-support.js) to automate required modifications in both *cordova-cli* and Xcode generated projects.
 
-* Put the hook script under the "hooks/" folder of your Cordova project (or wherever you prefer) and give it execute permission:
+* Put the hook script under the "hooks/" folder of your Cordova project (or wherever you prefer) and give it execution permission:
 ```bash
 $ chmod +x hooks/iosrtc-swift-support.js
 ```
@@ -31,7 +40,7 @@ $ cordova build ios
 ```
 
 
-## Bridging Header
+#### Bridging Header
 
 The plugin is coded in Swift language but it makes use of the Cordova Objective-C headers and the [Google's WebRTC Objective-C wrapper](https://chromium.googlesource.com/external/webrtc/+/master/talk/app/webrtc/objc/) so a [Bridging Header](https://developer.apple.com/library/prerelease/ios/documentation/Swift/Conceptual/BuildingCocoaApps/MixandMatch.html) is required. When building (using the provided hook above) our [Bridging Header](../src/cordova-plugin-iosrtc-Bridging-Header.h) file is automatically added into your Cordova project.
 
@@ -50,12 +59,12 @@ It may happen that your Cordova application uses more than a single plugin coded
 And then set `Unified-Bridging-Header.h` as the value of the "Objective-C Bridging Header" build setting in your Xcode project. For more information check this [issue](https://github.com/eface2face/cordova-plugin-iosrtc/issues/9).
 
 
-## Xcode
+#### Xcode
 
 If you still prefer to do it manually open it with Xcode and follow these steps:
 
-- Set "iOS Deployment Target" to `7.0` or higher within your project settings.
-- Set "Deployment Target" to `7.0` or higher within the project target settings.
-- Within the project "Build Settings" add an entry to the "Runpath Search Paths" setting with value `@executable_path/Frameworks`.
-- Within the project "Build Settings" set "Objective-C Bridging Header" to `PROJECT_NAME/Plugins/cordova-plugin-iosrtc/cordova-plugin-iosrtc-Bridging-Header.h` (read more about the "Bridging Header" above).
-- Within the project "Build Settings" set "Enable Bitcode" to "No" (this will be addressed soon).
+* Set "iOS Deployment Target" to `7.0` or higher within your project settings.
+* Set "Deployment Target" to `7.0` or higher within the project target settings.
+* Within the project "Build Settings" add an entry to the "Runpath Search Paths" setting with value `@executable_path/Frameworks`.
+* Within the project "Build Settings" set "Objective-C Bridging Header" to `PROJECT_NAME/Plugins/cordova-plugin-iosrtc/cordova-plugin-iosrtc-Bridging-Header.h` (read more about the "Bridging Header" above).
+* Within the project "Build Settings" set "Enable Bitcode" to "No".
diff --git a/docs/BuildingLibWebRTC.md b/docs/BuildingLibWebRTC.md
new file mode 100644
index 00000000..af7775ae
--- /dev/null
+++ b/docs/BuildingLibWebRTC.md
@@ -0,0 +1,71 @@
+# Build *libwebrtc* for cordova-plugin-iosrtc
+
+
+### Get the *libwebrtc* source code
+
+There are several ways to get and build the library. We choose a project that performs this task nicely:
+
+* Clone the [webrtc-build-scripts](https://github.com/pristineio/webrtc-build-scripts) project and enter into it:
+```bash
+$ git clone https://github.com/pristineio/webrtc-build-scripts
+$ cd webrtc-build-scripts
+```
+
+* Follow the project steps and fetch the *libwebrtc* source code:
+```bash
+$ source ios/build.sh
+$ WEBRTC_RELEASE=true
+$ get_webrtc
+```
+
+* Set a specific *libwebrtc* version:
+```bash
+$ update2Revision 10800
+```
+
+
+### Apply `libwebrtc-objc-iosrtc.patch`
+
+Google's *libwebrtc* Objective-C code lacks of `addtrack` and `removetrack` events in its `RTCMediaStream` class, making imposible to emulate them at JavaScript level. Thus a patch is needed to enable these W3C WebRTC features.
+
+**eFace2Face** created such a patch and sent it to the Google's *WebRTC* project, but [it was rejected](https://webrtc-codereview.appspot.com/50109004/) by arguing that the future of the WebRTC API is "track based" rather than "stream based".
+
+Since we live in the present we need those two events to properly implement the W3C WebRTC API in *cordova-plugin-iosrtc*, and hence the [patch](../extra/libwebrtc-objc-iosrtc.patch) is provided in the source code.
+
+* Apply the patch provided at `extra/libwebrtc-objc-iosrtc.patch` into the Objective-C source code of *libwebrtc*:
+```bash
+$ cd ios/webrtc/src/talk/app/webrtc/objc
+$ patch -p1 < PATH_TO_CORDOVA_PLUGIN_IOSRTC/extra/libwebrtc-objc-iosrtc.patch
+```
+
+
+### Remove iOS native H264 encoder/decoder
+
+(see issue #92 for details, help welcome!)
+
+Edit `ios/webrtc/src/webrtc/modules/video_coding/codecs/h264/include/h264.h` by removing the line 21:
+```c
+#define WEBRTC_VIDEO_TOOLBOX_SUPPORTED 1  // <-- remove this line
+```
+
+
+### Build *libwebrtc*
+
+* Go back to the `webrtc-build-scripts` root folder and build *libwebrtc*:
+```bash
+$ cd ../../../../../../../
+$ build_webrtc
+```
+
+* Copy the *libwebrtc* Objective-C headers to *cordova-plugin-iosrtc*:
+```bash
+$ rm -f PATH_TO_CORDOVA_PLUGIN_IOSRTC/src/webrtc-headers/*
+$ cp ios/webrtc/src/talk/app/webrtc/objc/public/* PATH_TO_CORDOVA_PLUGIN_IOSRTC/src/webrtc-headers/
+```
+
+* Copy the *libwebrtc* static library to *cordova-plugin-iosrtc*:
+```bash
+$ rm -f PATH_TO_CORDOVA_PLUGIN_IOSRTC/lib/*
+$ cp ios/webrtc/libWebRTC-LATEST-Universal-Release.a PATH_TO_CORDOVA_PLUGIN_IOSRTC/lib/
+```
+
diff --git a/docs/index.md b/docs/index.md
index 31f56aab..2b7b84a7 100644
--- a/docs/index.md
+++ b/docs/index.md
@@ -2,6 +2,7 @@
 
 * [`iosrtc` API](iosrtc.md): JavaScript API Documentation.
 * [Building](Building.md): Guidelines for building a Cordova iOS application including the *cordova-plugin-iosrtc* plugin.
+* [Building `libwebrtc`](BuildingLibWebRTC.md): Guidelines for building Google's *libwebrtc* with modifications needed by the *cordova-plugin-iosrtc* plugin.
 * [`<video>` CSS](videoCSS.md): Supported CSS properties.
 
 
diff --git a/extra/libwebrtc-objc-iosrtc.patch b/extra/libwebrtc-objc-iosrtc.patch
new file mode 100644
index 00000000..d084da2d
--- /dev/null
+++ b/extra/libwebrtc-objc-iosrtc.patch
@@ -0,0 +1,288 @@
+diff -Naur objc/RTCMediaStream+Internal.h objc.IOSRTC/RTCMediaStream+Internal.h
+--- objc/RTCMediaStream+Internal.h	2015-11-26 12:13:34.000000000 +0100
++++ objc.IOSRTC/RTCMediaStream+Internal.h	2015-11-26 13:13:48.000000000 +0100
+@@ -37,4 +37,12 @@
+ - (id)initWithMediaStream:
+         (rtc::scoped_refptr<webrtc::MediaStreamInterface>)mediaStream;
+ 
++- (void)update;
++- (BOOL)hasNativeAudioTrack:(RTCAudioTrack *)objcTrack;
++- (BOOL)hasNativeVideoTrack:(RTCVideoTrack *)objcTrack;
++- (BOOL)hasObjcAudioTrack:
++    (rtc::scoped_refptr<webrtc::AudioTrackInterface>)nativeTrack;
++- (BOOL)hasObjcVideoTrack:
++    (rtc::scoped_refptr<webrtc::VideoTrackInterface>)nativeTrack;
++
+ @end
+diff -Naur objc/RTCMediaStream.mm objc.IOSRTC/RTCMediaStream.mm
+--- objc/RTCMediaStream.mm	2015-11-26 12:13:34.000000000 +0100
++++ objc.IOSRTC/RTCMediaStream.mm	2015-11-26 13:21:23.000000000 +0100
+@@ -31,16 +31,31 @@
+ 
+ #import "RTCMediaStream+Internal.h"
+ 
+-#import "RTCAudioTrack+Internal.h"
+ #import "RTCMediaStreamTrack+Internal.h"
++#import "RTCAudioTrack+Internal.h"
+ #import "RTCVideoTrack+Internal.h"
+ 
+ #include "talk/app/webrtc/mediastreaminterface.h"
+ 
++namespace webrtc {
++  class RTCMediaStreamObserver : public ObserverInterface {
++    public:
++    RTCMediaStreamObserver(RTCMediaStream* stream) { _stream = stream; }
++
++    void OnChanged() override {
++      [_stream update];
++    }
++
++   private:
++    __weak RTCMediaStream* _stream;
++  };
++}
++
+ @implementation RTCMediaStream {
+   NSMutableArray* _audioTracks;
+   NSMutableArray* _videoTracks;
+   rtc::scoped_refptr<webrtc::MediaStreamInterface> _mediaStream;
++  rtc::scoped_ptr<webrtc::RTCMediaStreamObserver> _observer;
+ }
+ 
+ - (NSString*)description {
+@@ -64,7 +79,6 @@
+ 
+ - (BOOL)addAudioTrack:(RTCAudioTrack*)track {
+   if (self.mediaStream->AddTrack(track.audioTrack)) {
+-    [_audioTracks addObject:track];
+     return YES;
+   }
+   return NO;
+@@ -72,7 +86,6 @@
+ 
+ - (BOOL)addVideoTrack:(RTCVideoTrack*)track {
+   if (self.mediaStream->AddTrack(track.nativeVideoTrack)) {
+-    [_videoTracks addObject:track];
+     return YES;
+   }
+   return NO;
+@@ -83,7 +96,6 @@
+   NSAssert(index != NSNotFound,
+            @"|removeAudioTrack| called on unexpected RTCAudioTrack");
+   if (index != NSNotFound && self.mediaStream->RemoveTrack(track.audioTrack)) {
+-    [_audioTracks removeObjectAtIndex:index];
+     return YES;
+   }
+   return NO;
+@@ -93,9 +105,7 @@
+   NSUInteger index = [_videoTracks indexOfObjectIdenticalTo:track];
+   NSAssert(index != NSNotFound,
+            @"|removeAudioTrack| called on unexpected RTCVideoTrack");
+-  if (index != NSNotFound &&
+-      self.mediaStream->RemoveTrack(track.nativeVideoTrack)) {
+-    [_videoTracks removeObjectAtIndex:index];
++  if (index != NSNotFound && self.mediaStream->RemoveTrack(track.nativeVideoTrack)) {
+     return YES;
+   }
+   return NO;
+@@ -119,6 +129,8 @@
+     _audioTracks = [NSMutableArray arrayWithCapacity:audio_tracks.size()];
+     _videoTracks = [NSMutableArray arrayWithCapacity:video_tracks.size()];
+     _mediaStream = mediaStream;
++    _observer.reset(new webrtc::RTCMediaStreamObserver(self));
++    _mediaStream->RegisterObserver(_observer.get());
+ 
+     for (size_t i = 0; i < audio_tracks.size(); ++i) {
+       rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
+@@ -139,8 +151,157 @@
+   return self;
+ }
+ 
++- (void)dealloc {
++  _mediaStream->UnregisterObserver(_observer.get());
++}
++
+ - (rtc::scoped_refptr<webrtc::MediaStreamInterface>)mediaStream {
+   return _mediaStream;
+ }
+ 
++- (void)update {
++  size_t i;
++  webrtc::AudioTrackVector native_audio_tracks = _mediaStream->GetAudioTracks();
++  webrtc::VideoTrackVector native_video_tracks = _mediaStream->GetVideoTracks();
++  std::vector<size_t> removedAudioTrackIndexes;
++  std::vector<size_t> removedVideoTrackIndexes;
++
++  // Detect audio tracks removal.
++  for (i = 0; i < [_audioTracks count]; i++) {
++    RTCAudioTrack* objcTrack = [_audioTracks objectAtIndex:i];
++    NSNumber* index;
++
++    if (![self hasNativeAudioTrack:objcTrack]) {
++      removedAudioTrackIndexes.push_back(i);
++    }
++  }
++
++  // Detect video tracks removal.
++  for (i = 0; i < [_videoTracks count]; i++) {
++    RTCVideoTrack* objcTrack = [_videoTracks objectAtIndex:i];
++    NSNumber* index;
++
++    if (![self hasNativeVideoTrack:objcTrack]) {
++      removedVideoTrackIndexes.push_back(i);
++    }
++  }
++
++  // Remove old audio tracks and notify the delegate.
++  for (std::vector<size_t>::iterator it = removedAudioTrackIndexes.begin();
++    it != removedAudioTrackIndexes.end(); ++it) {
++    RTCAudioTrack* objcTrack = [_audioTracks objectAtIndex:*it];
++
++    // Remove the track from the ObjC container.
++    [_audioTracks removeObjectAtIndex:*it];
++
++    // Notify the delegate.
++    [_delegate OnRemoveAudioTrack:self track:objcTrack];
++  }
++
++  // Remove old video tracks and notify the delegate.
++  for (std::vector<size_t>::iterator it = removedVideoTrackIndexes.begin();
++    it != removedVideoTrackIndexes.end(); ++it) {
++    RTCVideoTrack* objcTrack = [_videoTracks objectAtIndex:*it];
++
++    // Remove the track from the ObjC container.
++    [_videoTracks removeObjectAtIndex:*it];
++
++    // Notify the delegate.
++    [_delegate OnRemoveVideoTrack:self track:objcTrack];
++  }
++
++  // Detect audio tracks addition and notify the delegate.
++  for (i = 0; i < native_audio_tracks.size(); ++i) {
++    rtc::scoped_refptr<webrtc::AudioTrackInterface> nativeTrack =
++      native_audio_tracks[i];
++
++    if (![self hasObjcAudioTrack:nativeTrack]) {
++      // Create the RTCAudioTrack instance and add it to the ObjC container.
++      RTCAudioTrack* objcTrack =
++        [[RTCAudioTrack alloc] initWithMediaTrack:nativeTrack];
++      [_audioTracks addObject:objcTrack];
++
++      // Notify the delegate.
++      [_delegate OnAddAudioTrack:self track:objcTrack];
++    }
++  }
++
++  // Detect video tracks addition and notify the delegate.
++  for (i = 0; i < native_video_tracks.size(); ++i) {
++    rtc::scoped_refptr<webrtc::VideoTrackInterface> nativeTrack =
++      native_video_tracks[i];
++
++    if (![self hasObjcVideoTrack:nativeTrack]) {
++      // Create the RTCVideoTrack instance and add it to the ObjC container.
++      RTCVideoTrack* objcTrack =
++        [[RTCVideoTrack alloc] initWithMediaTrack:nativeTrack];
++      [_videoTracks addObject:objcTrack];
++
++      // Notify the delegate.
++      [_delegate OnAddVideoTrack:self track:objcTrack];
++    }
++  }
++
++  unsigned long num_native_audios = native_audio_tracks.size();
++  unsigned long num_native_videos = native_video_tracks.size();
++  unsigned long num_objc_audios = (unsigned long)[_audioTracks count];
++  unsigned long num_objc_videos = (unsigned long)[_videoTracks count];
++
++  NSAssert(num_native_audios == num_objc_audios,
++    @"ObjC audio tracks does not match number of native audio tracks");
++
++  NSAssert(num_native_videos == num_objc_videos,
++    @"ObjC video tracks does not match number of native video tracks");
++}
++
++- (BOOL)hasNativeAudioTrack:(RTCAudioTrack *)objcTrack {
++  webrtc::AudioTrackVector audio_tracks = _mediaStream->GetAudioTracks();
++
++  for (size_t i = 0; i < audio_tracks.size(); ++i) {
++    rtc::scoped_refptr<webrtc::AudioTrackInterface> track = audio_tracks[i];
++
++    if (track->id().compare(objcTrack.audioTrack->id()) == 0) {
++      return YES;
++    }
++  }
++  return NO;
++}
++
++- (BOOL)hasNativeVideoTrack:(RTCVideoTrack *)objcTrack {
++  webrtc::VideoTrackVector video_tracks = _mediaStream->GetVideoTracks();
++
++  for (size_t i = 0; i < video_tracks.size(); ++i) {
++    rtc::scoped_refptr<webrtc::VideoTrackInterface> track = video_tracks[i];
++
++    if (track->id().compare(objcTrack.nativeVideoTrack->id()) == 0) {
++      return YES;
++    }
++  }
++  return NO;
++}
++
++- (BOOL)hasObjcAudioTrack:
++    (rtc::scoped_refptr<webrtc::AudioTrackInterface>)nativeTrack {
++  for (size_t i = 0; i < [_audioTracks count]; i++) {
++    RTCAudioTrack* track = [_audioTracks objectAtIndex:i];
++
++    if (track.audioTrack->id().compare(nativeTrack->id()) == 0) {
++      return YES;
++    }
++  }
++  return NO;
++}
++
++- (BOOL)hasObjcVideoTrack:
++    (rtc::scoped_refptr<webrtc::VideoTrackInterface>)nativeTrack {
++  for (size_t i = 0; i < [_videoTracks count]; i++) {
++    RTCVideoTrack* track = [_videoTracks objectAtIndex:i];
++
++    if (track.nativeVideoTrack->id().compare(nativeTrack->id()) == 0) {
++      return YES;
++    }
++  }
++  return NO;
++}
++
+ @end
+diff -Naur objc/public/RTCMediaStream.h objc.IOSRTC/public/RTCMediaStream.h
+--- objc/public/RTCMediaStream.h	2015-11-26 12:13:34.000000000 +0100
++++ objc.IOSRTC/public/RTCMediaStream.h	2015-11-26 13:22:41.000000000 +0100
+@@ -29,6 +29,22 @@
+ 
+ @class RTCAudioTrack;
+ @class RTCVideoTrack;
++@class RTCMediaStream;
++@class RTCMediaStreamTrack;
++
++@protocol RTCMediaStreamDelegate<NSObject>
++
++- (void)OnAddAudioTrack:
++    (RTCMediaStream*)mediaStream track:(RTCMediaStreamTrack*)track;
++- (void)OnAddVideoTrack:
++    (RTCMediaStream*)mediaStream track:(RTCMediaStreamTrack*)track;
++- (void)OnRemoveAudioTrack:
++    (RTCMediaStream*)mediaStream track:(RTCMediaStreamTrack*)track;
++- (void)OnRemoveVideoTrack:
++    (RTCMediaStream*)mediaStream track:(RTCMediaStreamTrack*)track;
++
++@end
++
+ 
+ // RTCMediaStream is an ObjectiveC wrapper for MediaStreamInterface.
+ @interface RTCMediaStream : NSObject
+@@ -36,6 +52,7 @@
+ @property(nonatomic, strong, readonly) NSArray *audioTracks;
+ @property(nonatomic, strong, readonly) NSArray *videoTracks;
+ @property(nonatomic, strong, readonly) NSString *label;
++@property(nonatomic, weak) id<RTCMediaStreamDelegate> delegate;
+ 
+ - (BOOL)addAudioTrack:(RTCAudioTrack *)track;
+ - (BOOL)addVideoTrack:(RTCVideoTrack *)track;
diff --git a/lib/libWebRTC-LATEST-Universal-Release.a b/lib/libWebRTC-LATEST-Universal-Release.a
index 38947ce6..da2741b3 100644
Binary files a/lib/libWebRTC-LATEST-Universal-Release.a and b/lib/libWebRTC-LATEST-Universal-Release.a differ
diff --git a/package.json b/package.json
index e75f3cc7..a2bf5d48 100644
--- a/package.json
+++ b/package.json
@@ -1,6 +1,6 @@
 {
   "name": "cordova-plugin-iosrtc",
-  "version": "2.0.2",
+  "version": "2.1.0-pre",
   "description": "Cordova iOS plugin exposing the full WebRTC W3C JavaScript APIs",
   "author": "Iñaki Baz Castillo at eFace2Face, inc. (https://eface2face.com)",
   "license": "MIT",
diff --git a/plugin.xml b/plugin.xml
index fd28a606..bf35f822 100644
--- a/plugin.xml
+++ b/plugin.xml
@@ -1,7 +1,7 @@
 <?xml version="1.0" encoding="UTF-8"?>
 <plugin xmlns="http://apache.org/cordova/ns/plugins/1.0"
 		id="cordova-plugin-iosrtc"
-		version="2.0.2">
+		version="2.1.0-pre">
 
 	<name>iosrtc</name>
 	<description>Cordova iOS plugin exposing the full WebRTC W3C JavaScript APIs</description>
@@ -31,13 +31,16 @@
 		</config-file>
 
 		<!-- WebRTC ObjetiveC headers -->
+		<header-file src="src/webrtc-headers/RTCAVFoundationVideoSource.h" />
 		<header-file src="src/webrtc-headers/RTCAudioSource.h" />
 		<header-file src="src/webrtc-headers/RTCAudioTrack.h" />
 		<header-file src="src/webrtc-headers/RTCDataChannel.h" />
 		<header-file src="src/webrtc-headers/RTCEAGLVideoView.h" />
+		<header-file src="src/webrtc-headers/RTCFileLogger.h" />
 		<header-file src="src/webrtc-headers/RTCI420Frame.h" />
 		<header-file src="src/webrtc-headers/RTCICECandidate.h" />
 		<header-file src="src/webrtc-headers/RTCICEServer.h" />
+		<header-file src="src/webrtc-headers/RTCLogging.h" />
 		<header-file src="src/webrtc-headers/RTCMediaConstraints.h" />
 		<header-file src="src/webrtc-headers/RTCMediaSource.h" />
 		<header-file src="src/webrtc-headers/RTCMediaStream.h" />
@@ -48,6 +51,7 @@
 		<header-file src="src/webrtc-headers/RTCPeerConnection.h" />
 		<header-file src="src/webrtc-headers/RTCPeerConnectionDelegate.h" />
 		<header-file src="src/webrtc-headers/RTCPeerConnectionFactory.h" />
+		<header-file src="src/webrtc-headers/RTCPeerConnectionInterface.h" />
 		<header-file src="src/webrtc-headers/RTCSessionDescription.h" />
 		<header-file src="src/webrtc-headers/RTCSessionDescriptionDelegate.h" />
 		<header-file src="src/webrtc-headers/RTCStatsDelegate.h" />
diff --git a/src/webrtc-headers/RTCDataChannel.h b/src/webrtc-headers/RTCDataChannel.h
index 7c225802..24a46f65 100644
--- a/src/webrtc-headers/RTCDataChannel.h
+++ b/src/webrtc-headers/RTCDataChannel.h
@@ -82,6 +82,12 @@ typedef enum {
 - (void)channel:(RTCDataChannel*)channel
     didReceiveMessageWithBuffer:(RTCDataBuffer*)buffer;
 
+@optional
+
+// Called when the buffered amount has changed.
+- (void)channel:(RTCDataChannel*)channel
+    didChangeBufferedAmount:(NSUInteger)amount;
+
 @end
 
 // ObjectiveC wrapper for a DataChannel object.
diff --git a/src/webrtc-headers/RTCFileLogger.h b/src/webrtc-headers/RTCFileLogger.h
new file mode 100644
index 00000000..3900cb6f
--- /dev/null
+++ b/src/webrtc-headers/RTCFileLogger.h
@@ -0,0 +1,70 @@
+/*
+ * libjingle
+ * Copyright 2015 Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ *  1. Redistributions of source code must retain the above copyright notice,
+ *     this list of conditions and the following disclaimer.
+ *  2. Redistributions in binary form must reproduce the above copyright notice,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#import <Foundation/Foundation.h>
+
+// TODO(tkchin): Move this to a common location.
+#ifndef NS_DESIGNATED_INITIALIZER
+#define NS_DESIGNATED_INITIALIZER
+#endif
+
+typedef NS_ENUM(NSUInteger, RTCFileLoggerSeverity) {
+  kRTCFileLoggerSeverityVerbose,
+  kRTCFileLoggerSeverityInfo,
+  kRTCFileLoggerSeverityWarning,
+  kRTCFileLoggerSeverityError
+};
+
+// This class intercepts WebRTC logs and saves them to a file. The file size
+// will not exceed the given maximum bytesize. When the maximum bytesize is
+// reached logs from the beginning and the end are preserved while the middle
+// section is overwritten instead.
+// This class is not threadsafe.
+@interface RTCFileLogger : NSObject
+
+// The severity level to capture. The default is kRTCFileLoggerSeverityInfo.
+@property(nonatomic, assign) RTCFileLoggerSeverity severity;
+
+// Default constructor provides default settings for dir path and file size.
+- (instancetype)init;
+
+- (instancetype)initWithDirPath:(NSString *)dirPath
+                    maxFileSize:(NSUInteger)maxFileSize
+    NS_DESIGNATED_INITIALIZER;
+
+// Starts writing WebRTC logs to disk if not already started. Overwrites any
+// existing file(s).
+- (void)start;
+
+// Stops writing WebRTC logs to disk. This method is also called on dealloc.
+- (void)stop;
+
+// Returns the current contents of the logs, or nil if start has been called
+// without a stop.
+- (NSData *)logData;
+
+@end
diff --git a/src/webrtc-headers/RTCLogging.h b/src/webrtc-headers/RTCLogging.h
new file mode 100644
index 00000000..bcd160d9
--- /dev/null
+++ b/src/webrtc-headers/RTCLogging.h
@@ -0,0 +1,92 @@
+/*
+ * libjingle
+ * Copyright 2015 Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ *  1. Redistributions of source code must retain the above copyright notice,
+ *     this list of conditions and the following disclaimer.
+ *  2. Redistributions in binary form must reproduce the above copyright notice,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#import <Foundation/Foundation.h>
+
+// Subset of rtc::LoggingSeverity.
+typedef NS_ENUM(NSInteger, RTCLoggingSeverity) {
+  kRTCLoggingSeverityVerbose,
+  kRTCLoggingSeverityInfo,
+  kRTCLoggingSeverityWarning,
+  kRTCLoggingSeverityError,
+};
+
+#if defined(__cplusplus)
+extern "C" void RTCLogEx(RTCLoggingSeverity severity, NSString* logString);
+extern "C" void RTCSetMinDebugLogLevel(RTCLoggingSeverity severity);
+extern "C" NSString* RTCFileName(const char* filePath);
+#else
+
+// Wrapper for C++ LOG(sev) macros.
+// Logs the log string to the webrtc logstream for the given severity.
+extern void RTCLogEx(RTCLoggingSeverity severity, NSString* logString);
+
+// Wrapper for rtc::LogMessage::LogToDebug.
+// Sets the minimum severity to be logged to console.
+extern void RTCSetMinDebugLogLevel(RTCLoggingSeverity severity);
+
+// Returns the filename with the path prefix removed.
+extern NSString* RTCFileName(const char* filePath);
+
+#endif
+
+// Some convenience macros.
+
+#define RTCLogString(format, ...)                    \
+  [NSString stringWithFormat:@"(%@:%d %s): " format, \
+      RTCFileName(__FILE__),                         \
+      __LINE__,                                      \
+      __FUNCTION__,                                  \
+      ##__VA_ARGS__]
+
+#define RTCLogFormat(severity, format, ...)                    \
+  do {                                                         \
+    NSString *logString = RTCLogString(format, ##__VA_ARGS__); \
+    RTCLogEx(severity, logString);                             \
+  } while (false)
+
+#define RTCLogVerbose(format, ...)                                \
+  RTCLogFormat(kRTCLoggingSeverityVerbose, format, ##__VA_ARGS__) \
+
+#define RTCLogInfo(format, ...)                                   \
+  RTCLogFormat(kRTCLoggingSeverityInfo, format, ##__VA_ARGS__)    \
+
+#define RTCLogWarning(format, ...)                                \
+  RTCLogFormat(kRTCLoggingSeverityWarning, format, ##__VA_ARGS__) \
+
+#define RTCLogError(format, ...)                                  \
+  RTCLogFormat(kRTCLoggingSeverityError, format, ##__VA_ARGS__)   \
+
+#if !defined(NDEBUG)
+#define RTCLogDebug(format, ...) RTCLogInfo(format, ##__VA_ARGS__)
+#else
+#define RTCLogDebug(format, ...) \
+  do {                           \
+  } while (false)
+#endif
+
+#define RTCLog(format, ...) RTCLogInfo(format, ##__VA_ARGS__)
diff --git a/src/webrtc-headers/RTCPeerConnection.h b/src/webrtc-headers/RTCPeerConnection.h
index 7177fd6c..a13ed3ec 100644
--- a/src/webrtc-headers/RTCPeerConnection.h
+++ b/src/webrtc-headers/RTCPeerConnection.h
@@ -27,6 +27,7 @@
 
 #import "RTCPeerConnectionDelegate.h"
 
+@class RTCConfiguration;
 @class RTCDataChannel;
 @class RTCDataChannelInit;
 @class RTCICECandidate;
@@ -97,10 +98,12 @@
     setRemoteDescriptionWithDelegate:(id<RTCSessionDescriptionDelegate>)delegate
                   sessionDescription:(RTCSessionDescription *)sdp;
 
-// Restarts or updates the ICE Agent process of gathering local candidates
-// and pinging remote candidates.
-- (BOOL)updateICEServers:(NSArray *)servers
-             constraints:(RTCMediaConstraints *)constraints;
+// Sets the PeerConnection's global configuration to |configuration|.
+// Any changes to STUN/TURN servers or ICE candidate policy will affect the
+// next gathering phase, and cause the next call to createOffer to generate
+// new ICE credentials. Note that the BUNDLE and RTCP-multiplexing policies
+// cannot be changed with this method.
+- (BOOL)setConfiguration:(RTCConfiguration *)configuration;
 
 // Provides a remote candidate to the ICE Agent.
 - (BOOL)addICECandidate:(RTCICECandidate *)candidate;
diff --git a/src/webrtc-headers/RTCPeerConnectionFactory.h b/src/webrtc-headers/RTCPeerConnectionFactory.h
index f0b2e3ad..e1e69b4b 100644
--- a/src/webrtc-headers/RTCPeerConnectionFactory.h
+++ b/src/webrtc-headers/RTCPeerConnectionFactory.h
@@ -28,6 +28,7 @@
 #import <Foundation/Foundation.h>
 
 @class RTCAudioTrack;
+@class RTCConfiguration;
 @class RTCMediaConstraints;
 @class RTCMediaStream;
 @class RTCPeerConnection;
@@ -44,7 +45,7 @@
 + (void)initializeSSL;
 + (void)deinitializeSSL;
 
-// Create an RTCPeerConnection object.   RTCPeerConnectionFactory will create
+// Create an RTCPeerConnection object. RTCPeerConnectionFactory will create
 // required libjingle threads, socket and network manager factory classes for
 // networking.
 - (RTCPeerConnection *)
@@ -52,6 +53,11 @@
                      constraints:(RTCMediaConstraints *)constraints
                         delegate:(id<RTCPeerConnectionDelegate>)delegate;
 
+// Creates a peer connection using the default port allocator factory and identity service.
+- (RTCPeerConnection *)peerConnectionWithConfiguration:(RTCConfiguration *)configuration
+                                           constraints:(RTCMediaConstraints *)constraints
+                                              delegate:(id<RTCPeerConnectionDelegate>)delegate;
+
 // Create an RTCMediaStream named |label|.
 - (RTCMediaStream *)mediaStreamWithLabel:(NSString *)label;
 
diff --git a/src/webrtc-headers/RTCPeerConnectionInterface.h b/src/webrtc-headers/RTCPeerConnectionInterface.h
new file mode 100644
index 00000000..b0cc72b5
--- /dev/null
+++ b/src/webrtc-headers/RTCPeerConnectionInterface.h
@@ -0,0 +1,75 @@
+/*
+ * libjingle
+ * Copyright 2015 Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ *  1. Redistributions of source code must retain the above copyright notice,
+ *     this list of conditions and the following disclaimer.
+ *  2. Redistributions in binary form must reproduce the above copyright notice,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+// See talk/app/webrtc/peerconnectioninterface.h.
+
+#import <Foundation/Foundation.h>
+
+typedef NS_ENUM(NSInteger, RTCIceTransportsType) {
+  kRTCIceTransportsTypeNone,
+  kRTCIceTransportsTypeRelay,
+  kRTCIceTransportsTypeNoHost,
+  kRTCIceTransportsTypeAll,
+};
+
+// https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-08#section-4.1.1
+typedef NS_ENUM(NSInteger, RTCBundlePolicy) {
+  kRTCBundlePolicyBalanced,
+  kRTCBundlePolicyMaxBundle,
+  kRTCBundlePolicyMaxCompat,
+};
+
+// https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-09#section-4.1.1
+typedef NS_ENUM(NSInteger, RTCRtcpMuxPolicy) {
+  kRTCRtcpMuxPolicyNegotiate,
+  kRTCRtcpMuxPolicyRequire,
+};
+
+typedef NS_ENUM(NSInteger, RTCTcpCandidatePolicy) {
+  kRTCTcpCandidatePolicyEnabled,
+  kRTCTcpCandidatePolicyDisabled,
+};
+
+// Configuration object used for creating a peer connection.
+@interface RTCConfiguration : NSObject
+
+@property(nonatomic, assign) RTCIceTransportsType iceTransportsType;
+@property(nonatomic, copy) NSArray *iceServers;
+@property(nonatomic, assign) RTCBundlePolicy bundlePolicy;
+@property(nonatomic, assign) RTCRtcpMuxPolicy rtcpMuxPolicy;
+@property(nonatomic, assign) RTCTcpCandidatePolicy tcpCandidatePolicy;
+@property(nonatomic, assign) int audioJitterBufferMaxPackets;
+@property(nonatomic, assign) int iceConnectionReceivingTimeout;
+
+- (instancetype)initWithIceTransportsType:(RTCIceTransportsType)iceTransportsType
+                             bundlePolicy:(RTCBundlePolicy)bundlePolicy
+                            rtcpMuxPolicy:(RTCRtcpMuxPolicy)rtcpMuxPolicy
+                       tcpCandidatePolicy:(RTCTcpCandidatePolicy)tcpCandidatePolicy
+              audioJitterBufferMaxPackets:(int)audioJitterBufferMaxPackets
+            iceConnectionReceivingTimeout:(int)iceConnectionReceivingTimeout;
+
+@end
diff --git a/src/webrtc-headers/RTCTypes.h b/src/webrtc-headers/RTCTypes.h
index 946148af..99ac192c 100644
--- a/src/webrtc-headers/RTCTypes.h
+++ b/src/webrtc-headers/RTCTypes.h
@@ -36,6 +36,7 @@ typedef enum {
   RTCICEConnectionFailed,
   RTCICEConnectionDisconnected,
   RTCICEConnectionClosed,
+  RTCICEConnectionMax,
 } RTCICEConnectionState;
 
 // RTCICEGatheringState the states in webrtc::ICEGatheringState.