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While learning at your own pace, you'll discover how to approach site concept creation and bring your vision from the drawing board through to full operability. Related News Wiley - HTML5 Digital Classroom Comments (0) Information Please Login or create a free account to leave comments. Imagine a world where your phone, TV and computer could all communicate on a common platform. Open the URL displayed at the bottom of the page in a new tab or, better still, on a different computer. Alternatively, jump straight into our WebRTC codelab: a step-by-step guide that explains how to build a complete video chat app, including a simple signaling server. One of the last major challenges for the web is to enable human communication via voice and video: Real Time Communication, RTC for short.
Historically, RTC has been corporate and complex, requiring expensive audio and video technologies to be licensed or developed in house. Gmail video chat became popular in 2008, and in 2011 Google introduced Hangouts, which use the Google Talk service (as does Gmail).
WebRTC has now implemented open standards for real-time, plugin-free video, audio and data communication. Plugins can be difficult to deploy, debug, troubleshoot, test and maintain—and may require licensing and integration with complex, expensive technology. The guiding principles of the WebRTC project are that its APIs should be open source, free, standardized, built into web browsers and more efficient than existing technologies. RTCPeerConnection is in Chrome (on desktop and for Android), Opera (on desktop and in the latest Android Beta) and in Firefox. Get network information such as IP addresses and ports, and exchange this with other WebRTC clients (known as peers) to enable connection, even through NATs and firewalls.
RTCPeerConnection: audio or video calling, with facilities for encryption and bandwidth management. Likewise on pages using HTTPS: permission only has to be granted once for for getUserMedia() (in Chrome at least). Also, Chrome will deprecate HTTP access for getUserMedia() at the end of 2015 due to it being classified as a Powerful feature. The intention is eventually to enable a MediaStream for any streaming data source, not just a camera or microphone. Webcam Toy is a photobooth app that uses WebGL to add weird and wonderful effects to photos which can be shared or saved locally. One gotcha: getUserMedia constraints set in one browser tab affect constraints for all tabs opened subsequently.
WebRTC uses RTCPeerConnection to communicate streaming data between browsers (aka peers), but also needs a mechanism to coordinate communication and to send control messages, a process known as signaling.
Instead, WebRTC app developers can choose whatever messaging protocol they prefer, such as SIP or XMPP, and any appropriate duplex (two-way) communication channel. Media capabilities: what codecs and resolutions can be handled by my browser and the browser it wants to communicate with?
The exchange of information via signaling must have completed successfully before peer-to-peer streaming can begin. Alice sends serialized candidate data to Bob, via whatever signaling channel they are using: WebSocket or some other mechanism.
When Bob gets a candidate message from Alice, he calls addIceCandidate, to add the candidate to the remote peer description.
WebRTC clients (known as peers, aka Alice and Bob) also need to ascertain and exchange local and remote audio and video media information, such as resolution and codec capabilities.
In the callback, Alice sets the local description using setLocalDescription() and then sends this session description to Bob via their signaling channel. Bob sets the description Alice sent him as the remote description using setRemoteDescription().
Bob runs the RTCPeerConnection createAnswer() method, passing it the remote description he got from Alice, so a local session can be generated that is compatible with hers. When Alice gets Bob's session description, she sets that as the remote description with setRemoteDescription.

RTCSessionDescription objects are blobs that conform to the Session Description Protocol, SDP.
The acquisition and exchange of network and media information can be done simultaneously, but both processes must have completed before audio and video streaming between peers can begin. Once the signaling process has completed successfully, data can be streamed directly peer to peer, between the caller and callee—or if that fails, via an intermediary relay server (more about that below). RTCPeerConnection is the WebRTC component that handles stable and efficient communication of streaming data between peers. From a JavaScript perspective, the main thing to understand from this diagram is that RTCPeerConnection shields web developers from the myriad complexities that lurk beneath. In this example, pc1 represents the local peer (caller) and pc2 represents the remote peer (callee).
Create an offer and set it as the local description for pc1 and as the remote description for pc2. NAT traversal, peer-to-peer networking, and the requirements for building a server app for user discovery and signaling, are beyond the scope of this article.
The code is deliberately verbose in its logging: check the console to understand the order of events. This code initializes variables for the HTML video elements that will display video streams from the local camera (localVideo) and from the camera on the remote client (remoteVideo).
For signaling, this demo uses the Google App Engine Channel API, which enables messaging between JavaScript clients without polling. This causes video from the local camera to be displayed in the localVideo element, by creating an object (Blob) URL for the camera's data stream and then setting that URL as the src for the element. The underlying purpose is to set up a connection, using a STUN server, with onIceCandidate() as the callback (see above for an explanation of ICE, STUN and 'candidate').
The offer creation process here is similar to the no-signaling example above but, in addition, a message is sent to the remote peer, giving a serialized SessionDescription for the offer. XHR works fine for sending signaling messages from the client to the server, but some mechanism is needed for server-to-client messaging: this application uses the Google App Engine Channel API.
If the message is an answer from a peer (a response to an offer), RTCPeerConnection sets the remote SessionDescription and communication can begin.
Many existing WebRTC apps only demonstrate communication between web browsers, but gateway servers can enable a WebRTC app running on a browser to interact with devices such as telephones (aka PSTN) and with VOIP systems. As well as audio and video, WebRTC supports real-time communication for other types of data.
The RTCDataChannel API enables peer-to-peer exchange of arbitrary data, with low latency and high throughput. Communication occurs directly between browsers, so RTCDataChannel can be much faster than WebSocket even if a relay (TURN) server is required when 'hole punching' to cope with firewalls and NATs fails. There are several ways a real-time communication application or plugin might compromise security.
Unencrypted media or data might be intercepted en route between browsers, or between a browser and a server.
Malware or viruses might be installed alongside an apparently innocuous plugin or application. WebRTC is not a plugin: its components run in the browser sandbox and not in a separate process, components do not require separate installation, and are updated whenever the browser is updated. Camera and microphone access must be granted explicitly and, when the camera or microphone are running, this is clearly shown by the user interface. The APIs and standards of WebRTC can democratize and decentralize tools for content creation and communication—for telephony, gaming, video production, music making, news gathering and many other applications. We look forward to what JavaScript developers make of WebRTC as it becomes widely implemented.
Imagine it was easy to add video chat and peer-to-peer data sharing to your web application. Integrating RTC technology with existing content, data and services has been difficult and time consuming, particularly on the web. Google bought GIPS, a company which had developed many components required for RTC, such as codecs and echo cancellation techniques.

These includes Skype, Facebook (which uses Skype) and Google Hangouts (which use the Google Talk plugin). Take a look at the cross-browser demo at demo and Chris Wilson's amazing examples using getUserMedia as input for Web Audio. Often this actually just means that getUserMedia is supported, but not any of the other RTC components.
For example, a stream taken from camera and microphone input has synchronized video and audio tracks. This would enable streaming from disc, or from arbitrary data sources such as sensors or other inputs. Note that screen capture requires HTTPS and should only be used for development due to it being enabled via a command line flag as explaind in this discuss-webrtc post. Signaling methods and protocols are not specified by WebRTC: signaling is not part of the RTCPeerConnection API. Here's a code sample from the WebRTC W3C Working Draft, which shows the signaling process in action. The callback argument of this is passed an RTCSessionDescription: Alice's local session description. The createAnswer() callback is passed an RTCSessionDescription: Bob sets that as the local description and sends it to Alice. Suffice to say that the STUN protocol and its extension TURN are used by the ICE framework to enable RTCPeerConnection to cope with NAT traversal and other network vagaries.
This step-by-step guide explains how to build a complete video chat application, including a simple signaling server built with running on a Node server. The Channel API is used in this demo, but other methods (such as WebSocket) could be used instead. So—if a connection hasn't already been made, and a local stream is available, and a channel is ready for signaling, a connection is created and passed the local video stream. Handlers are then set for each of the RTCPeerConnection events: when a session is connecting or open, and when a remote stream is added or removed. The caller and callee have discovered each other and exchanged information about their capabilities, a call session is initiated, and real-time data communication can begin. The magnificent Cube Slam game uses the API to communicate game state: play a friend or play the bear!
Without it, we're limited in our ability to innovate and develop new ways for people to interact. Google open sourced the technologies developed by GIPS and engaged with relevant standards bodies at the IETF and W3C to ensure industry consensus. The intention is to implement support for other constraints such as aspect ratio, facing mode (front or back camera), frame rate, height and width, along with an applyConstraints() method. For screencast, code and more information, see the HTML5 Rocks update: Screensharing with WebRTC. The code assumes the existence of some signaling mechanism, created in the createSignalingChannel() method. In this process, STUN servers have a single task: to enable a peer behind a NAT to find out its public address and port.
In other words, ICE will first use STUN with UDP to directly connect peers and, if that fails, will fall back to a TURN relay server. Once that happens, started is set to true, so a connection won't be started more than once. In this case, there is only one video track and no audio, but it is easy to imagine use cases where there are more: for example, a chat application that gets streams from the front camera, rear camera, microphone, and a 'screenshared' application. The expression 'finding candidates' refers to the process of finding network interfaces and ports. This app uses adapter.js, a JavaScript shim, maintained Google with help from the WebRTC community, that abstracts away browser differences and spec changes.

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