WebRTC 简介及服务搭建、测试

WebRTC主要解决如下三个问题

1、获取音频和视频。 可以访问输入设备，如话筒、（网络）摄像机（头），可以从上述设备获取媒体数据流。目前需要硬件设备支持VP8/VP9，如果设备仅仅支持H.264/H.265（HEVC：High Efficiency Video Coding）,则无法使用WebRTC进行编解码传输存储。

2、音频和视频交互通讯。 通过互联网，连接其它WebRTC终端，并且实时发送这些音视频流。

3、任意数据的交互通讯。 通过互联网，连接其它WebRTC终端，实时发送其它任意应用数据。

针对上述三个问题，WebRTC上使用的三种重要的JavaScript API

• MediaStream（aka getUserMedia)访问媒体
• RTCPeerConnection
• RTCDataChannel

简介：

1. Web Real Time Communication

2. What’s WebRTC?
WebRTC is a new front in the long war for an open and unencumbered web. — Brendan Eich , inventor of JavaScript

3. What’s WebRTC?
Web Real-Time Communication (WebRTC) is an upcoming standard that aims to enable real-time communication among Web browsers in a peer-to-peer fashion.
WebRTC project (opensource) aims to allow browsers to natively support interactive peer to peer communications and real time data collaboration.
Provide state of art audio/video communication stack in your browser.

4. Earlier Efforts?
Many web services already use RTC, but need downloads, native apps or plugins. These includes Skype, Facebook (uses Skype) and Google Hangouts (uses Google Talk plugin). Downloading, installing and updating plugins can be complex, error prone and annoying. Plugins can be difficult to deploy, debug, troubleshoot, test and maintain—and may require licensing and integration with complex, expensive technology. 4? Integrating RTC technology with existing content, data and

5. What does it change? No licenses or other fees. Integration via simple, standardized Web APIs. No Proprietary plugins. No Security issues. No downloads, no installation. Just surf to the right address!

6. Aims of WebRTC?
State of art audio/video communication stack in your browser. Seamless person-to-person communication. Specification to achieve inter-operability among Web browsers. Interoperability with legacy systems. Low cost and highly efficient communication solution

7. Architecture

8. Architecture? At startup, browsers do not know each other. JavaScript mediates the setup process through server. Media flows through the shortest possible path for latency.

9. How peers connect?

10. WebRTC Media Engine

11. WebRTC App. Need TO? Get streaming audio, video or other data. Get network information such as IP address and port, and exchange this with other WebRTC clients (known as peers ). Coordinate signaling communication to report errors and initiate or close sessions. Exchange information about media and client capability, such as resolution and codecs. 11? Communicate streaming audio, video or data.

12. Signaling? Mechanism to coordinate communication and to send control messages. Signaling methods and protocols are not specified by WebRTC but by application developer.
Signaling is used to exchange three types of information :
Session control messages : to initialize or close communication and report errors.
Network configuration : whats my computers IP address and port?

13. Signaling
The original idea to exchange Session Description information was in the form of Session Description Protocol (SDP) "blobs".
This approach had several shortcomings some of which would be difficult to address.
IETF is standardizing the JavaScript Session Establishment Protocol (JSEP).
JSEP provides the interface an application needs to deal with the negotiated local and remote session descriptions.
The JSEP approach leaves the responsibility for driving the signaling state machine entirely to the application.

14. WebRTC API
getUserMedia (MediaStream)
RTCPeerConnection
RTCDataChannel

15. WebRTC API Stack View

16. getUserMedia
A MediaStream is an abstract representation of an actual data stream of audio or video. Serves as a handle for managing actions on the media stream. A MediaStream can be extended to represent a stream that either comes from (remote stream) or is sent to (local stream) a remote node. A LocalMediaStream represents a media stream from a local media-capture device (such as a webcam or microphone).

17. getUserMedia
The MediaStream represents synchronized streams of media. For example, a stream taken from camera and microphone input has synchronized video and audio tracks. The getUserMedia() method takes three parameters: A constraints object. A success callback which, if called, is passed a LocalMediaStream. A failure callback which, if called, is passed an error object. In Chrome, the URL.createObjectURL () method converts a LocalMediaStream to a Blob URL which can be set as the src of a

18. getUserMedia
<video id="sourcevid" autoplay>
</video>
<script> var video = document.getElementById('sourcevid');
navigator.getUserMedia(video, success, error);
function success(stream)
{
video.src =window.URL.createObjectURL(stream);
}
</script>

19. RTCPeerConnection
RTCPeerConnection is the WebRTC API that handles stable and efficient communication of streaming data between peers . Communications are coordinated via a signaling channel provided by scripting code in the page via the Web server — for instance, using XMLHttpRequest or WebSocket.

20. RTCPeerConnectionIn the real world, WebRTC needs servers, so the followingcan happen:? Users discover each other and exchange real world details such as names. WebRTC client applications (peers) exchange network information. Peers exchange data about media such as video format and resolution.

21. RTCPeerConnection

22. NAT Traversal
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. Initially, ICE tries to connect peers directly , with the lowest possible latency, via UDP. In this process, STUN servers have a single task: to enable a peer behind a NAT to find out its public address and port.
If UDP fails,
ICE tries TCP: first HTTP, then HTTPS.
If direct connection fails—in particular, because of enterprise NAT traversal and firewalls—ICE uses an intermediary (relay) TURN server.
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.
The expression finding candidates refers to the process of

25. NAT Traversal

26. RTCDataChannel
As well as audio and video, WebRTC supports real-time communication for other types of data.
The RTCDataChannel API will enable peer-to-peer exchange of arbitrary data, with low latency and high throughput.
The API has several features to make the most of RTCPeerConnection and enable powerful and flexible peer-to-peer communication.

27. RTCDataChannel
Stream Control Transmission Protocol (SCTP) encapsulated in DTLS is used to handle DataChannel Data.
DataChannel API is bidirectional, which means that each DataChannel bundles an incoming and an outgoing SCTP stream.
Encapsulating "SCTP over DTLS over ICE over UDP" provides a NAT traversal solution together with confidentiality, source authentication, and integrity protected transfers.

28. SecurityThere are several ways a real-time communication applicationor plugin might compromise security.
For example: Unencrypted media or data might be intercepted en route between browsers, or between a browser and a server.
An application might record and distribute video or audio without the user knowing.
Malware or viruses might be installed alongside an apparently innocuous plugin or application.

29. Security
WebRTC has several features to avoid these problems: WebRTC implementations use secure protocols such as DTLS and SRTP.
Encryption is mandatory for all WebRTC components, including signaling mechanisms.
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.

30. Current Limitations
Cloud Infrastructure – A server is required by WebRTC to complete four tasks: User discovery, Signalling and NAT/firewall traversal.
Native Applications – WebRTC enables real-time communication between web browsers. It is not a software development kit that can be used in native iOS or Android applications or in native desktop applications.
Multiparty Conferencing – WebRTC is peer-to-peer by nature which allows WebRTC to be extremely scalable, but it is very inefficient when setting up communications between more than two end users.
Recording – WebRTC does not support recording as of now.

31. Conclusion
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.
WebRTC will have great impact on open web and interoperable browser technologies including the existing enterprise solutions.

References
Salvatore Loreto, Simon Pietro Romano (2012) ‘Real-Time Communications in the Web’ - IEEE paper October, 2012
IETF.org
WebRTC book by Alan B. Johnston and Daniel C. Burnett : webrtcbook.com
Video of Justin Ubertis WebRTC session at Google I/O, 27 June 2012.
webrtc.org
Google Developers Google Talk documentation, which gives more information about NAT traversal, STUN, relay servers and candidate gathering.
WebPlatform.org (http://docs.webplatform.org/wiki/concepts/internet_and_web/web

Demo:simpl.info/dc

部署调试WebRTC的步骤详细记录如下：

参考：

https://www.youtube.com/watch?v=p2HzZkd2A40

https://www.slideshare.net/dhamavijays/webrtc-17414738

g.co/webrtc

https://www.youtube.com/watch?v=p2HzZkd2A40

http://transcriptvids.com/v2/p2HzZkd2A40.html