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WebRTC

( Duration: 5 Days )

The WebRTC training course will help you master the application and all its fundamentals. You will learn the architecture of WebRTC, concepts of signalling channels using Web Sockets, understand the WebRTC APIs, implement WebRTC security and much more. This course is great for software professionals who want to stay up to date with all the latest happenings in WebRTC.

By attending WebRTC workshop, delegates will learn:

The WebRTC architecture and how it is revolutionizing modern web applications
The VoIP architecture
WebRTC APIs, their architecture and uses
How to implement security in WebRTC applications

JavaScript and SIP/HTTP/Web programming experience.
A knowledge of Real Time communications is ideal, as is a background with RT signalling using SIP.
Experience with Node.js is helpful but not required.

This WebRTC class is intended for:

Software Developers, Web Application Developers, Voip Technicians and Programmers.

COURSE AGENDA

1

Introduction to WebRTC

What is WebRTC
Next evolution of RT communications
PSTN/SS7 – VoIP/SIP – WebRTC/Any Signaling
Real Time Communications natively from a web browser
WebRTC current vendor support

Chrome
Firefox
Opera
Status of other browsers and devices

Standards Process

IETF standards track overview
Overview of RFCs and emerging internet drafts
W3C API standards overview

Understanding the legacy of RT Communications
PSTN
VoIP/SIP
WebRTC
PSTN Architecture

PSTN Equipment

Central Office Switches
POTS phones
PBX
Analog and Digital Trunks (T-1, OC-192 Fiber, etc)
Conference call media transcoding/mux
SS&Signaling infrastructure

Isolation of the media plane and the signaling plane
Signaling with SS7
Media transport with Digital TDM trunks

VoIP Architecture

VoIP Equipment

SIP Server / Soft switch
SIP Endpoints (IP phones, etc)
VoIP-PSTN Gateways
Media Control Unit for Transcoding/conferencing
Firewalls
Session Border Controllers - SBC

Isolation of the media plane and the signaling plane
Signaling Options

Many early signaling protocols

H.323. IAX2, MGCP, SIP

Consolidation to SIP in Carrier and Enterprise networks

Session Initiation Protocol - SIP
Use of Session Description Protocol - SDP
SDP/SIP/UDP/IP/Layer 2
Implementation of Secure SIP - SIPS

Media Transport

Encoding of Audio and Video using codecs
Transport of Audio and Video using Real Time Protocol – RTP
Encryption with SRTP
Codec/RTP/UDP/IP/layer 2

NAT Traversal Solutions developed for VoIP explained

STUN
TURN
ICE

WebRTC Architecture

WebRTC Equipment

Endpoint: Browser, embedded device (Google Glass, etc)
SIP or other Signaling Server(s)
Media Control Unit for Transcoding/conferencing
WebRTC Gateways – SIP/VoIP – PSTN – Others not yet defined

Signaling
Media
Data

Firewalls
WebRTC SBC (not yet defined)

Isolation of the Media plane and the Signaling plane
Signaling plane “agnostic”
Four Tasks

Media Stream Acquisition
Signaling
Media Transport
“Arbitrary” Data Transport

ustyme
PIPE: File Transfer/Sharing on Facebook
Fluke Connect
Capiche
Be My Eyes
Rap Battles

2

WebRTC APIs

WebRTC API Flowchart
Four Primary Requirements:

Acquire Audio and Video from device inputs
Establish connection between peers

Exchange Offer/Answer
createOffer()
createAnswer()
setLocalDescription()
setRemoteDescription()

Communicating audio and video
Communicating arbitrary data

Four primary APIs
getUserMedia

MediaStreams

RTCPeerConnection

makes connections and communicates media streams

RTCSessionDescription

Represents the parameters of a session

RTCDataChannel

Send and receive “arbitrary” data

getUserMedia -MediaStreams

gUM
Represent a stream of synchronized media
Can contain more than one audio and/or video MediaStream Tracks
Obtain a MediaStream with navigator.getUserMedia()
javascript examples
navigator.getUserMedia(constraints, successCallback, errorCallback)
simpl.info/gum
Processing video and audio streams
Examples

gUM + Canvas
idevelop.github.com/ascii-camera
gUMscreencapture
Build hangouts using WebRTC

Constraints
Dynamic constraints through API
getUserMedia constraint of audio true only
webaudiodemos.appspot.com/Vocoder/index.html

RTCPeerConnection API

Abstracts away complexity
Sets up everything

Echo cancelation
noise reduction
codecs
etc

setting up peer to peer pathway through NATs and Firewalls

RTCSessionDescription

SDP format
candidate options
codecs, ports, RTP configuration, etc.

RTCDataChannel

Exchanging Arbitrary data
Same API as WebSockets
Ultra low latency
simpl.info/dc

Overview of additional APIs of interest
Resources

chrome://webrtc-internals
adapter.js
Code Samples
github.com/GoogleChrome/webrtc
apprtc.appspot.com

3

WebRTC Architecture

Isolation of the Media plane and the Signaling plane
Concept of “Arbitrary data” channel
Signaling plane “agnostic”
Four Tasks

Media Stream Acquisition
Signaling
Media Transport
“Arbitrary” Data Transport

Media Streams Acquisition

internal devices (webcam, mic, etc)
Media Transport
Inherently peer-to-peer
Supported codecs defined, including G.711, Opus, V8
secure transmission options
Requires Secure RTP or DTLS for transport
What to secure?
DTLS implementation Examples

Signaling

Signaling Agnostic: You Decide
"BYOS"; Bring You Own Signaling!
JavaScript Establishment Protocol JSEP
Need to exchange Session Description – SDP
Requirement to exchange SDP media descriptions in offer/answer
Alternative proposals

Object RTC (ORTC API) – Not standardized

Understanding the role of Session Description Protocol RFC4566

Why SDP?
Unified SDP for WebRTC
Limitation of SDP in WebRTC architecture
number of m lines
conference calling
traditional media mixer conference server
Single RTP audio stream
Cannot adjust, mute, enhance individual streams
challenge of media relay server
SSRC values in single RTP stream
need for server logic to re-invite with new multi-m line SDP

Using Session Description Protocol

codec selection
peer to peer routing
encryption and authentication
Bandwidth management
Offer / Answer Exchange
createOffer
RFC 3264 offer
Complete set of capabilities
gotOffer
createAnswer
SDP answer
Codec, RTP, RTCP options plus ICE candidates
Negotiated set of capabilities
gotAnswer
gotRemoteStream
RTCSessionDescription object
SDP failure requirement for DOMError object

Application specific signaling
Isolation of browser from signaling
Possible Signaling approaches:
SIP over WebSockets
RFC 7118: WebSocket Protocol as a Transport for the Session Initiation Protocol (SIP)
Signaling with Node and Socket.io
Socket.io uses WebSocket with fallbacks
jSIP
sip-js
QoffeeSIP
sipML5
Call Control API
GSMA/OMA extending RCS-s API to include WebRTC support
XMPP/Jingle

Interoperability with XMPP clients
XMPP server with XMPP WebSocket transport

XHR

no special signaling infrastructure
web server with proprietary protocol

JSON
Proprietary signaling

Connecting and Sending Data. Peer-to-peer Data API
RTCDataChannel

sending application data over peer to peer connection
Ultra low latency requirements
gaming communications
Signaling considerations
MediaStreamSignaling options plus
Inside Data Channel
latency, NAT traversal, privacy considerations
WebSocket or web server with proprietary implementation for Data Channel establishment
API similarity to WebSockets
Latency considerations
Reliability or unreliable considerations
When does unreliable make sense?
RTCPeerConnection
ondatachannel
receiveChannel
createDataChannel
sendChannel
simpl.info/dc

4

IPv4 & NAT

IPv4 vs IPv6 addressing
Impact of Firewalls
Web Proxy Server considerations
How NATs function

How NATs cause Peer to Peer applications to break
Impact on SIP message
Impact on SDP message

Nat Traversal solutions built into access devices

How these solutions can break your "solution"!

IPv6 Address space solution for NAT (hint: Don’t NAT!)
Public vs Private IPv4 addresses
IETF NAT Traversal Architecture
ICE – Interactive Connectivity Establishment

Architecture for STUN and TURN
Path determination

STUN, TURN, and ICE support
STUN

Session Traversal Utilities for NAT
Punching hole in the NAT
Contacting the STUN server to determine Public IP and Port information
Relatively Free to connect
Firewall considerations

TURN

Fallback for STUN
Relay Server in the Cloud
Cost and delay considerations
Bandwidth and Server requirements
Availability

Examples

stun.l.google.com:19302
rfc5766-turn-server
deploying in Amazon cloud
restund

Quality of Service

IETF support for Q0S in TCP/IP
Integration of DiffServ with Ethernet (LAN) and MPLS(WAN)
Use of DSCP
Support for DiffServ in WebRTC using DSCP
Practical considerations when considering DiffSer

LAN/WAN architecture
IT support
Carrier support
Internet connectivity or VPN

Security

Defining WebRTC security
WebRTC Security Architecture
Brower Permissions for accessing webcam, audio, etc
WebRTC Threat Model and Attacks
Signaling Threats and Attacks
API Attacks
Implementing DTLS

DTLS SRTP Key Agreement
Client / Server

Implementing SRTP

Secure Profile of RTP –SRTP
Encryption and Authentication
Key Exchange and Key Management
SDP Security Descriptions

WebRTC impacts on browser security
using HTTPS for signaling

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