Why this SIP course?
Session Initiation Protocol (SIP) is the signalling protocol that forms the basis of future Voice over IP (VoIP) and converged networks, both fixed and mobile. This two day course enables attendees to:
- Gain an understanding of the architecture, components and functions of the SIP suite.
- Learn SIP and its operation in detail.
- Understand SIP’s strengths, reliability, performance, separation, extensibility, modularity and simplicity.
- Understand the role of SIP within the wider context of signalling, content and communications protocols and in Fixed and Mobile Convergence (FMC).
The course takes a modular approach to covering the Planning, Structure and Operation of Session Initiation Protocol (SIP). The course also covers the integration of SIP with other network technologies such as mobile and SS7.
SIP Training Course Objectives
On completing this course, you will be able to:
- 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.
Who should attend this SIP course
- 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.
SIP Course Pre-requisites
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.
Hands-on Practicals
This SIP training course features extensive hands-on practical working with:
- SIP Servers
- SIP Clients
- Hard SIP IP phones
- SIP softphones
- SIP-enabled PBXs
- SIP protocol capture and decoding using Wireshark
- SIP Troubleshooting
and more.
Lifetime Post-Course Support
After completing this SIP training course, delegates receive lifetime post-training support from LEVER Technology Group, to help them apply the technologies and skills they have learned with us, to provide career-long support, and to ensure they are better equipped for their future roles in IT and networking.
SIP Training Course Content
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