AG Tech Presentation WebRTC

WebRTCFUTURE OF THE WEB

@folio_3 www.folio3.com Copyright 2015

WebRTC

Web Real-Time Communication

free

open-source

high quality

Real Time Communications

right in the browser.


No plugins...!!!

Support browser-to-browser applications for:

voice calling

video chat

and P2P file sharing

without the need of either internal or external plugins.


Yes, you got it right..!!!

Simple JavaScript APIs and HTML5

No downloads required

No installing and updating plugins of any sort.


So, what's the big idea?

Flash Player is like patching an incomplete system.

With WebRTC there is no external dependency beyond the basic

web browser.

You know HTML and Javascript, right?

What about Actionscript/MXML???


Browser Support

Desktop: 17+

Mobile: 29+

Desktop: 18+

Mobile: 20+Desktop: 18+

Mobile: 24+


Quality WebRTC - High quality video codec

Flash Player - Outdated Sorenson codec

(which is difficult to interoperate with non-Flash products)

@folio_3 Copyright 2015www.folio3.com

But lets face it...

Many users are reluctant to change/update their browser (IE,

Safari,...)

Flash Player has shown more ubiquitous availability on user's

desktops and laptops than any specific web browser.


Cross-compatibility

In the past, incompatibility in HTML among browsers has been a

nightmare for web developers

(remember IE6...? Dark ages...? anyone...?)

Solution:

adapter.js

or similar shims


The main problem with Flash?!!?!

It's PROPRIETARY..!!!(the protocol for end-to-end media path is proprietary so interoperating with existing

VoIP gears is inefficient without buying server pieces from the plugin vendor.)


So What it takes for a browser to

support WebRTC?

Lets look at some boring jargons...

Media EngineThe technology that manages microphones, cameras, and

speakers and gets audio and video onto and off the internet.

Transport layer

(ISO-OSI) which is relatively simple, but hard to tweak for real-time

communications

CODECs

Audio/Video encoders and decoders (something that is usually acquired from third parties)


How about the network?

NATan Internet standard that enables a local-area network (LAN) to use one set of IP

addresses for internal traffic and a second set of addresses for external traffic.

STUN(Session Traversal Utilities for NAT) is a standardized set of methods and a network

protocol that allows an end host to discover its public IP address if it is located behind

a NAT.

TURNTraversal Using Relays around NAT (TURN) is a protocol that allows a client behind a

network address translator (NAT) or firewall to receive incoming data over TCP or UDP

connections.


ICEInteractive Connectivity Establishment (ICE) is a technique used in computer

networking involving network address translators (NATs) in Internet applications of

Voice over Internet Protocol (VoIP), peer-to-peer communications, video, instant

messaging and other interactive media. [Wikipedia]

Stream An abstraction referring to any flow of data from a source (or sender, producer) to a

single sink (or receiver, consumer). A stream usually flows through a channel of some

kind.

SDPThe Session Description Protocol (SDP) is a format for describing streaming media

initialization parameters.

SDP does not deliver media itself but is used for negotiation

between end points of media type, format, and all associated properties.


How all of this fits together?


Ready...steady...RT.........

peerConnection = new mozRTCPeerConnection();

peerConnection.onaddstream = pconaddstream;

navigator.mozGetUserMedia({video: true, audio: true}, mediaSuccess,

mediaFailure);


function setRemoteDesc(connectionString) {

var connectionObj = JSON.parse(connectionString);

var sessionObj = new mozRTCSessionDescription();

sessionObj.type = connectionObj.type;

sessionObj.sdp = connectionObj.sdp;

peerConnection.setRemoteDescription(sessionObj, doNothing,

callFailed);

}


WebRTC beyond peer-to-peer

3 types of architectures:

Mesh

Mixing

Routing


Mesh

Powerful clients

High network bandwidth

Best quality

Minimal Latency


Mixing

MCU (Multipoint Conferencing Unit)

Powerful server

Compromised Quality

High Latency


Routing

SFU (Selective Forwarding Unit)

Compression on clients

Scalable

Low Latency


Is it all upon me???

Muaz Khan

on GitHub


PetZam

PetZam uses WebRTC

using TokBox API

peer-to-peer video calling

for pet care


http://web.folio3.com/vetspert/




Future

Support on Safari and IE

Unified standard API

Lots of fun and interactivity

(screen sharing, video conferences, team collaboration, distance

learning, online gaming, real-time dashboards, ... ... ... ... ...)

www.folio3.com Copyright 2015@folio_3

AG Tech Presentation WebRTC

Technology

Transcript of AG Tech Presentation WebRTC