Overview

Session Initiation Protocol (SIP) is the signalling protocol that forms the basis of modern Voice over IP (VoIP) and converged networks, both fixed and mobile. This comprehensive two day course enables attendees to understand the architecture, components and functions of the SIP suite, to learn SIP and its operation in detail. Attendees will also learn about the role of SIP in Fixed and Mobile Convergence (FMC), and the integration of SIP with other network technologies including 5G mobile networks and SS7.

Duration

2 days

Audience

  • People responsible for evaluating, planning, installing, configuring, administrating or supporting SIP products and services.
  • Level 2, 3 and 4 support staff responsible for diagnosing and troubleshooting issues with SIP servers, PBXs, media gateways and Session Border Controllers.
  • Personnel wanting to move into the field of modern SIP-based voice communications or media streaming systems.

Prerequisites

It is essential that course participants have good understanding of IP, DNS, IP Multicasting/IGMP, TCP/UDP, legacy TDM-based voice telephony, VoIP principles, RTP/RTCP, H.323 and IP QoS principles. The alternative course 328 combines those topics together with the content of this SIP Fundamentals course.

Objectives

Discuss and evaluate SIP products, services and SIP network deployments with customers or suppliers.
Plan SIP solutions, including capacity planning and system sizing.
Configure SIP phones, applications, SIP converters and more.
Understand the SIP protocol interactions for call/session signalling.
Capture and analyse SIP signalling traffic using a network protocol analyser.

Content Headings

SIP Overview
Introduction
SIP design requirements
The development of SIP
SIP and VOIP
SIP Vs. H.323 and H.248/MEGACO
What SIP does
– SIP and Next Generation Networks, NGN
– SIP and mobility
Why we would deploy SIP
The role of SIP within:
– Mobile
– Provider VOIP
– NGN solutions
– Unified Communications
– Multimedia
– SIP based contact centres and using SIP for contact centre hosting
– SIP Trunking
– 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

Overview of SIP Operation
The SIP User Agent client and server
The SIP URI
SIP Methods and Responses
SIP message exchange
– SIP Messages involved setting up a simple SIP call
– INVITE method
– 100 Trying
– 180 Ringing
– 200 OK
– ACK
– BYE method
Overview of other widely used SIP messages
– SUBSCRIBE
– OPTIONS
– NOTIFY
– REINVITE
– PUBLISH
– INFO
– 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

SIP Solutions and Capacity Planning
Design considerations when deploying SIP solutions
Feature requirements
– Matching customer requirement to cloud offering
– Identifying the Customer Premises Equipment (CPE) requirements
– 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 (KPIs)
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 QOS
– Possible QOS signalling within SIP
– Enforcing the GOS from the SIP servers
Determining the bandwidth requirements for SIP signalling