Start for free ApiRTC Training ApiRTC Training Start for free Make the most of ApiRTC!

ApiRTC Training

Goals and presentation

Acquire the fundamentals of WebRTC architecture, protocols and associated software technologies. Understand WebRTC by theoretical analyzes of varied use-cases, and experimental work on platform. Get an introduction to its potential in term of new services.

 

WebRTC is a project initiated by the W3C and the IETF, whose objective is to achieve real-time mutimedia communications between web browsers. By introducing a native convergence of synchronous/asynchronous communication services, landline/ mobile, voice/data… This new approach made the web the natural support and definitively any professional or public communication service.

Participants and requirements

  • Computer and/or network engineers, developers of telecommunications services, web developers, technical managers.
  • Knowledge of network protocols TCP/IP,HTTP,HTML et JavaScript languages are required to take better advantage of training.

Program

Introduction

 

  • WebRTC origins and history
  • WebRTC Standardization
  • Use-cases and deployments

Technical description of WebRTC

 

WebRTC support (browsers, devices, mobile …)

 

WebRTC APIs

  • MediaStream – getUserMedia
    • API explanation with Web sample
    • Media devices acceptation: HTTPS
    • Media constraints usage (Video resolution, QoS…)
    • Media devices selection
    • Screensharing
  • Peerconnection
    • Datachannel

 

SDP – Session description

  • Peer-to-Peer call sample with ApiRTC: Media routing on a local network
  • Network capture analyses and SDP exchange
  • offer / answer/ candidates

 

DTLS – SRTP : Protocoles

DTLS – SCTP : Protocoles

Encryptions

  • Keys exchanges description s et analyse des paramètres dans les offres SDP

 

Audio, video and DTMF

Audio and video codecs in WebRTC – OPUS audio codec

 

Network address Traversal – Media communication with Firewall

  • STUN, TURN, ICE: protocols description
  • SDP description (offer / answer/ candidates)
  • TrickleIce mechanism

 

WebRTC issues and how ApiRTC will help in your deployment

  • Signalling
  • API evolutions
  • Browser interoperability
  • Restrictive firewall traversal

Technical description of ApiRTC

 

This part of the training includes practical work on platform for use-case analysis and development of a service.

 

Browser compatibility

ApiRTC platform architecture description (SaaS – On-premise deployment)

ApiRTC Library architecture description

Starting with ApiRTC

  • apiKey / apiCCId
  • events management
  • Version management
  • Log management

ApiRTC Dashboard description

Offline features (UserAgent: Whiteboard…)

Signalling

  • ApiRTC connexion (Long-polling and WebSocket)
  • Disconnection management / retry
  • Presence group management / userData

 

Calls management (tutorial 3)

  • Audio / video call
  • Error management
  • Call end management
  • Safari
  • Hangup
  • Mute
  • Accept/Refuse call – (tutorial 4)
  • Media Device selection – (tutorial 8)
  • Resolution configuration – (tutorial 9)
  • Recording calls – (tutorial 7)
  • External stream usage: Stream tutorial
  • Disconnection management / retry during calls
    • Call hand over with/without network change
  • Call Forking

 

Media routing

  • ApiRTC media routing mode sample
  • Restrictive firewall traversal

 

Conference calls – (tutorial 12 + record tutorial 15)

  • Media routing with ApiRTC in conference: P2P / SFU
  • Quality of service (QoS) adaptation mechanisms

 

Chat with history management: Chat sample

Group Chat

 

SendData usage:  JSON data transfer

 

File transfer (tutorial 14 + DataChannel sample)

Datachannel to server

 

Whiteboard (tutorial 13)

  • Disconnection management
  • Whiteboard hand over with/without network change

 

Video pointer sharing

 

Call establishment with PSTN

  • SIP Trunk interconnection for call and conference access

 

Security aspects on ApiRTC

  • Media flows – DTLS/SRTP
  • Datachannel flows – DTLS/SCTP
  • Users authentication

 

WebRTC monitoring tool:

  • Browser tools : Chrome / Firefox
    • WebRTC QoS Statistics display: bandwidth, delay, packet lost, jitter…
  • Diagnostic tools :
    • Precall test
    • WebRTC test

Summary and conclusion

Duration

2 days-training
2 000€
Up to 5 participants

Person in charge

Frédéric LUART

CTO and co-founder of Apizee, Frederic has a large experience delivering VoIP systems for telecom carriers and is an established expert in WebRTC technology and WebRTC based multimedia applications development.