VOIP and SIP Training Contents

8/13/2019 VOIP and SIP Training Contents

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VOIP and SIP Training contents

Duration 3 days

VoIP Training Content

Overview of VoIP

Voice and data convergence Components of a VoIP system Standards employed in current VoIP solutions The role of Voice Processing The speech encoding process Sampling, Quantization, Coding, Framing Silence suppression Voice coding and compression standards

Adaptive encoding techniques Coding fax signals Voice codecs: G.711, G.722, G.721, G.723, G.726, G.723 Assessing voice quality Mean Opinion Scores (MOS) Detecting flaws in transmitted voice Employing MOS ratings for codecs and real networks Assessing Voice Quality Measurable components What to test and measure P.800 / P.861 recommendations PESQ

Operating Voice over IP

The issues when operating Voice over IP Delay, Talker overlap, Echo Jitter, Packet loss Out of Order Delivery The role of Voice Processing and DSP Real-time Transport Protocol (RTP) The role of RTP RTP header in detail

RTP payload types Real-time Transport Control Protocol (RTCP) Conclusions

Introduction to Voice over IP signalling

Overview of signalling in PSTN networks Overview of private network signalling



The major architectures and standards for Voice over IP ITU H.323 IETF SIP MGCP and Megaco/H.248 Cisco SCCP (Skinny) VOIP in the enterprise VOIP in PSTN Emulation Service (PES)

ITU / IETF Megaco / H.248

MGCP and Megaco The Media Gateway Reference Architecture End-to-End call setup IETF Megaco Megaco Terminations and Contexts Megaco Commands Megaco Packages

Megaco IP phone Media Gateway

Overview of IP QoS

Classifying IP traffic Review of the IPv4 Datagram format IPv4 Service (TOS) field Precedence bits DTR(C) bits Characteristics of RTP media flows Classifying packets in IPv6 networks

The need for QoS IP Differentiated Services (Diff-Serv) Queuing and Scheduling mechanisms First-In First-Out (FIFO), Strict priority scheduling, Fair Queuing, Weighted-Fair Queuing

(WFQ), Class-Based Queuing, Hierarchical Class Based Queuing (CBQ) Coping with packet loss Controlling admission Employing Random Early Detection

Employing traffic shaping IEEE 802.1p/Q Operating IP over ATM networks Overview of MPLS



SIP Training Contents

Module 1 SIP Overview

Introduction

SIP design requirements The development of SIP

SIP and VOIP SIP Vs. H.323 and H.248/MEGACO What SIP doeso SIP and Next Generation Networks, NGNo SIP and mobility

Why we would deploy SIP

The role of SIP within:o Mobileo Provider VOIPo NGN solutionso Unified Communicationso Multimediao SIP based contact centres and using SIP for contact centre hostingo SIP Trunkingo Mini case studies

SIP Architecture and Components

SIP User Agents SIP Registrar SIP Proxies SIP Location server SIP Redirect Server SIP Back to Back User Agent (B2BUA) SIP PBX

Module 2 Overview of SIP Operation

The SIP User Agent client and server

The SIP URI SIP Methods and Responses SIP message exchangeo SIP Messages involved setting up a simple SIP callo INVITE methodo 100 Tryingo 180 Ringing



o 200 OKo ACKo BYE method

Overview of other widely used SIP messageso SUBSCRIBEo OPTIONSo NOTIFYo REINVITEo PUBLISHo INFOo PRACK

SDP Description and Role

Role of SDP Structure of SDP SDP operation

SIP Registration and Location Servers

Role of the SIP Registration Server

SIP Registration Method SIP Registration/location to provide roaming/mobility

SIP Proxy

SIP Stateful Proxy SIP Call Stateful Proxy SIP Stateless Proxy SIP Stateful/stateless proxies

SIP URI and DNS

Mapping E.164 dialled digits to SIP using DNS/ENUM SIP and ENUM

The SIP Redirect Server

SIP REDIRECT methods Using redirection to route calls Using redirection to implement mobility SIP redirection, proxies, registrars, and location services to provide mobility and roaming

Problems of SIP NAT Traversal

Common solutions to SIP NAT Traversal

Simple Traversal of UDP through NAT's, STUN



Traversal using Relay NAT, TURN Universal Plug and Play, UPnP Tunnelling/VPN Session Border Controller, SBC

SIP In the Cloud

SIP Gateways SIP integration with the PSTN/ISDN SIPI and SS7 ISUP SIP and H.248/Megaco SIP and H.225/H245/H.323 SIP and MPLS/QOS SIP Trunking Hosted SIP PBX Hosted SIP Conferencing Hosted SIP based contact centre

Potential threats to SIP

Registration Hijacking

Impersonating a Server Tampering with Message Bodies Tearing Down Sessions Denial of Service and Amplification

Securing SIP

Transport and Network Layer Security SIPS URI Scheme SIP Method Authentication Registration Interdomain Requests Peer-to-Peer Requests DoS Protection HTTP Digest S/MIME TLS

Module 3 SIP Solutions and Capacity Planning

Design considerations when deploying SIP solutions

Feature requirementso Matching customer requirement to cloud offeringo Identifying the Customer Premises Equipment (CPE) requirementso Server implementation



Security and resilience of SIP servers Security to the Cloud Interconnection and communication with other severs such as DNS, and

RADIUS/DIAMETER

Connecting to the Cloud

SIP based VOIP Key Performance Indicators (KPI’s) Sizing VOIP voice channel capacity Impact of VOIP on data applications Translating Erlangs and Grade of Service (GOS) into VOIP channel capacity Impact of VOIP QOS on Grade of Service, GOS Concept of Connection Admissions Control, CAC/VCAC SIP and QOSo Possible QOS signalling within SIPo Enforcing the GOS from the SIP servers

Determining the bandwidth requirements for SIP signalling

Documents

VOIP and SIP Training Contents