WebRTC服务器:不全是点对点

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视频地址

Types of WebRTC servers

• Signaling 信令 Always needed
• NAT Traversal 内网穿透　Needed for production
• Media 多媒体 Depends on the app
• Gateway 网关 Depends on the app

Signaling

WebRTC is peer-to-peer for media, but WebRTC is not peer-to-peer for Signaling.

Session Description Protocol (SDP)

WebRTC使用会话描述协议（SDP）来协商媒体参数

样例展示

Server

简单的 Socket.io server:

• 只有32行
• 没有逻辑，它只转发 messages

webRTC github地址
sending webRTC via avian carrier, which means a bird, a pigeon, a passenger pigeon.

var socketIO = requrie('socket.io');
var server = require('http').createServer().listen(7000, '0.0.0.0');
var io = socketIO.listen(server);

// Super simple server:
// * On room only.
// * We expect two people max.
// * No error handling.

io.sockets.on('connection', function (client) {
   console.log('new connection: ' + client.id);
   
   client.on('offer', function (details) {
      client.broadcast.emit('offer', details);
      console.log('offer:' + JSON.stringify(details));     
  });
 
  client.on('answer', function (details) {
     client.broadcast.emit('answer', details);
     console.log('answer: ' + JSON.stringify(details));
  });

 client.on('candidate', function (details) {
    client.broadcast.emit('candidate', details);
    console.log('candidate: ' + JSON.stringify(details));
 });

// Here starts evertying!
// The first connection doesn't send anything (no other clients)
// Second connection emits the message to start the SDP negotation
client.broadcast.emit('createoffer', {});
});

有个叫fippo 的黑客，有一些很炫的东东

Signaling considerations outside of webRTC

• user authentication
• Security & access controls
• mobile environments. 程序不能太耗电，信令通信不能太繁琐，保持通话状态，保持网络唤醒。想办法推送通知服务消息。
• 通话人数的扩增。

如何 获取/使用 一台 signaling server

• Ask your CPaas provider all include signaling eg: Twilio TokBox
• Run your own write your own in node.js Matrix.org EasyRTC SimpleWebRTC
• Use a messaging service eg: Firebase PubNub Pusher GCM, Google Cloud Messager

NAT Traversal

NAT Traversal firewalls.

信令需要通过Signaling Server 进行传输. 　WebRTC media 是不需要通过Signaling Server 进行forward传输的。

Interactive Connectivity Establishment(ICE)

A protocol for establishing a peer-to-peer media connection between peers behind NAT and firewall devices.

Two types of ICE servers

• STUN
  session Traversal Utilities for NAT
• TURN
  Traversal using Relay NAT a relay server

STUN 告诉你，你的公网IP地址是多少。

Comparing STUN vs. TURN

STUN 和 TURN 的使用情况统计, 大约24%的情况需要用到TURN.

example iceServers object

ICE candidate protocols

• UDP
• TCP
• TLS over TCP
  

Types of TURN

ICE Candidate Types

• host local address
• srflx address from STUN server
• relay addre of TURN server

Where to test TURN

• https://test.webrtc.org
• https://webrtc.github.io/samples/src/content/peerconnection/trickle-ice/

Distribute geographically for improved performance & redundancy

可以在不同的地区建立　TURN server 云服务

media servers

reasons you might need a media server

• Multi-party
• Recording
• Heavy media manipulation
• broadcasting

The Mesh Problem

参与会话的 Peer 需要给其他所有的Peer 发送数据，这带来了CPU和带宽的消耗。特别是参与的Peer数量增加的时候。

参与 call 的 Peer数量是 n. 那么每个Ｃient的 Streams/client　的数量是 2(n-1), 总计n*n-n

Mulipoint Control Unit　多方控制单元

MCU 的优点和缺点

MCU Issue: Choosing the Layout
https://freeswitch.org/confluence/display/FRESSWITCH/FreeSWITCH+1.6+Video

Selective Forwarding Unit (SFU)

有Ｎ个 Peer(client) 参与 Call, 每一个Peer的stream链接数是Ｎ，总共有n*n个链接.

SFU with Simulcast

Simulcast

How to get/use media server

• Ask your CPaaS provider only some support media servers
• Open source
  • Jitsi Video Bridge
  • Kurento
  • Janus
  • Licode
  • Freeswitch
• Commercial software/hardware

WebRTC Gateways

not for browser-to-browser,it’s browser to something else

Gateways adapt signaling & media
需要做一些视频编码转换，信令协议转换。因为WebRTC 并没有直接定义信令协议。他只要求你需要使用一种信令协议，通过　offer-answer mechanism. 如何将　SDP 从一端传送到另外一端. 无论你使用哪一种 signaling mechanism. 你需要一种方法，将它转换成通信行业里面的标准协议，比如SIP.

WebRTC Server Considerations

源视频地址