Voice over IP Fundamentals: Second Edition (Cisco Press Fundamentals Series) / Edition 2

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Overview

A systematic approach to understanding the basics of voice over IP

  • Understand the basics of enterprise and public telephony networking, IP networking, and how voice is transported over IP networks
  • Learn the various caveats of converging voice and data networks
  • Examine the basic VoIP signaling protocols (H.323, MGCP/H.248, SIP) and primary legacy voice signaling protocols (ISDN, C7/SS7)
  • Explore how VoIP can run the same applications as the existing telephony system but in a more cost-efficient and scalable manner
  • Delve into such VoIP topics as jitter, latency, packet loss, codecs, QoS tools, and security

Voice over IP (VoIP) has become an important factor in network communications, promising lower operational costs, greater flexibility, and a variety of enhanced applications. To help you understand VoIP networks, Voice over IP Fundamentals provides a thorough introduction to the basics of VoIP.

Voice over IP Fundamentals explains how a basic IP telephony infrastructure is built and works today, major concepts concerning voice and data networking, and transmission of voice over data networks. You’ll learn how voice is signaled through legacy telephone networks, how IP signaling protocols are used to interoperate with current telephony systems, and how to ensure good voice quality using quality of service (QoS).

Even though Voice over IP Fundamentals is written for anyone seeking to understand how to use IP to transport voice, its target audience comprises both voice and data networking professionals. In the past, professionals working in voice and data networking did not have to understand each other’s roles. However, in this world of time-division multiplexing (TDM) and IP convergence, it is important to understand how these technologies work together. Voice over IP Fundamentals explains all the details so that voice experts can understand data networking and data experts can understand voice networking.

The second edition of this best-selling book includes new chapters on the importance of billing and mediation in a VoIP network, security, and the common types of threats inherent when packet voice environments, public switched telephone networks (PSTN), and VoIP interoperate. It also explains enterprise and service-provider applications and services.


Provides a through introduction to this new technology to help experts in both the data and telephone industries plan for the new networks. Covers various benefits and applications of VoIP and how to ensure good voice quality in your network.

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Editorial Reviews

From Barnes & Noble
The Barnes & Noble Review
Once, deploying new telecom services was a highly-specialized, complex, painful process best left to a few carrier-class switch providers and a few large service providers. Suddenly, thanks to deregulation, there are thousands of service providers. And thanks to VoIP, service providers can leverage the power and ubiquity of the Internet to deploy virtually any imaginable service — faster, and at lower cost. It's a revolution — and with Voice Over Ip Fundamentals from Cisco Press, you can get in on it.

Written for both data and voice networking professionals — even those without lengthy experience — the book covers all you'll need to know to get started. The authors start by demystifying the existing Public Switched Telephone Network (PSTN), including key elements such as SS7 — and showing why it isn't meeting the needs of a data-centric world. Next, they introduce VoIP, explaining how it can run the same applications as the PSTN, but more cost-effectively and with greater scalability. You'll review the functional components of a VoIP solution, and how they fit together. There are also candid discussions of the serious challenges associated with implementing VoIP, including jitter, packet loss, latency, and quality of service.

Next, the authors focus on the key protocols associated with VoIP, including H.323 for videoconferencing, SIP, gateway control protocols, and how Cisco's Virtual Switch Controller (VSC) wraps them all together. You walk through setting up calls, tearing down calls, and offering services — the bread-and-butter ofanytelephony network.

Some important VoIP standards are still in flux, and this book certainly takes a Cisco-centric view of the world. But those are two small caveats for an excellent introduction to VoIP that will serve most technical professionals extremely well.


bncom editor

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Product Details

  • ISBN-13: 9781587052576
  • Publisher: Cisco Press
  • Publication date: 8/10/2006
  • Series: Fundamentals Series
  • Edition description: REV
  • Edition number: 2
  • Pages: 432
  • Sales rank: 703,166
  • Product dimensions: 7.30 (w) x 9.00 (h) x 0.90 (d)

Meet the Author

Jonathan Davidson, CCIE No. 2560, is the Director of SP Solution Engineering in Integrated Network Systems Engineering. He has co-authored Voice over IP Fundamentals and edited Deploying Cisco Voice over IP. He has been with Cisco for 10 years in post-sales support, marketing, and engineering divisions.

James Peters is the Director of Product Marketing in the Carrier Core and Multiservice Business Unit at Cisco Systems. He co-authored the first edition of Voice over IP Fundamentals and is currently authoring a book on multiservice networking. James has more than 20 years experience in building, designing Internet-based voice and data networks, and product development.

Manoj Bhatia is a Business Development Manager for Partner Programs at IP Communications Business Unit (IPCBU) for Cisco Systems, Inc. He was among the first to start the software development for SIP technology on Cisco VoIP gateways and IOS-based routers. His past projects include technical marketing for VoIP products such as media gateways, call agents, and SIP-based residential voice solutions. Prior to Cisco, Manoj worked in Nortel Networks and Summa Four (now Cisco) and has 14+ years of experience in telephony protocols such as SS7, call control, and VoIP technologies.

Satish Kalidindi is a Software Engineer with Cisco Systems. He has more than six years experience working on development and deployment of VoIP technologies. He has been involved with various products, including IOS gateways and Cisco CallManager. More recently he has been involved with security features on CCM. He is a graduate of Purdue University with an M.S in Engineering.

Sudipto Mukherjee is a Software Development Engineer with Cisco Systems. He has product development and deployment experience for a variety of telecommunication devices for wireline, wireless, and VoIP networks. More recently at Cisco he has been working on SIP gateway development. Sudipto has a Bachelors of Engineering degree in Electronics Communication engineering from GS Institute of Technology, Indore and a Masters degree in Electronics Design and technology from Indian Institute of Science, Bangalore.

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Read an Excerpt


Chapter 4: Signaling System 7

Congestion Control

MTP2 monitors the level of messages queued in buffers (both output and retransmission) and alerts SNM in case of congestion.

Onset of congestion messages are sent to SNM when the threshold value for the buffers is exceeded. The SNM process considers all destinations across the link to be congested.

Now consider congestion from the signaling endpoint and STP perspective:

  • Signaling endpoints (SSP, SCP) receive congestion information from MTP2 onset of congestion indications. Excessive higher-layer messages can cause congestion over signal endpoint (SSP and SCP) links. In this case, SNM sends status messages to applications indicating which DPCs are affected. The application should reduce outgoing messages for a period of time. SNM continues to send the congestion status message until MTP2 receives the end of congestion indication. At this point, SNM stops sending the status messages, and after the timeout period, user applications resume normal activity.
  • If the STP SNM process receives an onset of congestion alert concerning a particular link, it considers that the route to its adjacent node is congested. When messages are received for the affected node, the STP SNM process sends a Transfer Controlled (TFC) message to the SNM of the transmitting endpoint. The STP indicates the affected node in the TFC message. This enables the signaling endpoint to choose an alternate route to the affected node. When the SNM process receives the end of congestion indication, it stops sending the status indications to the transmitting endpoint.
Rerouting

The SNM rerouting process reroutes traffic around an affected node without causing congestion or losing messages. STPs use this process when the route to a specific endpoint is unavailable. SNM uses the Transfer Prohibited (TFP) message to advise all directly connected nodes of the lost route to the specific endpoint. This enables the other STPs to choose an alternate route to the affected node. When the links are restored, Transfer Allowed (TFA) messages alert the directly connected nodes that normal routing procedures can resume.

Changeover and Changeback

You use changeover procedures when signaling links become unavailable and messages need to be diverted over alternate links. You use changeback procedures when the signaling links become available and normal routing needs to be re-established. Changeover and changeback procedures require SNM actions from both signaling points to maintain sequence and minimize loss.

You initiate the changeover procedure using the changeover order (COO) message between the signaling points. The COO message indicates the affected link in the SLC field of the MSU. The SMH function does not select the signaling link identified in the SLC field as the outgoing link. SMH selects an alternate route to reach the adjacent signaling point.

When the receiving point receives the COO message, it selects an alternate route and sends a changeover acknowledgment (COA) to the transmitting signaling point. The COO and COA messages contain the FSNs of the last message accepted on the unavailable link. Both signaling points retrieve the messages in the output buffers of the unavailable link and move these messages to the output of the alternate link. At this point, all waiting messages are sent in sequence and without loss, completing the changeover procedure.

You use the changeback procedure when the affected link becomes available. Either signaling point can initiate changeback procedures. SNM advises the SMH process that the messages destined for the alternate link should be stored in the changeback buffer (CBB) instead. The changeback declaration (CBD) is then sent to the adjacent signaling point identifying that the link is now available. The receiving signaling point responds with a changeback acknowledgment (CBA). When the signaling point receives the CBA, SNM advises SMH to send the buffered messages out the primary link and resume normal routing procedures.

SCCP

The SCCP provides network services on top of MTP3: The combination of those two layers is called the Network Service Part (NSP) of SS7. TCAP typically uses SCCP services to access databases in the SS7 network. As illustrated in Figure 4-8, the SCCP provides service interfaces to TCAP and ISUP. SCCP routing services enable the STP to perform Global Title Translation (GTT) by determining the DPC and subsystem number of the destination database.

The following SCCP features are covered in the next few sections:

  • Connection-Oriented Services
  • Connectionless Services and Messages
  • SCCP Management Functions
Connection-Oriented Services

SCCP supports connection-oriented services for TCAP and ISUP, however none of these services is used today. As such, this section does not cover SCCP connection-oriented capabilities, messages, or services.

Connectionless Services and Messages

SCCP provides the transport layer for the connectionless services of TCAP (discussed in the section entitled "Transaction Capabilities Applications Part [TCAP]"). TCAP-based services include 800, 888, 900, calling card, and mobile applications. Together, SCCP and MTP3 transfer non-circuit based messages used in these services. The SCCP also enables the STP to perform GTT on behalf of the end office exchange. The end office exchange views the 800 number as a functional address or, in other words, as a global title address. Because global title addresses are not routed, the SCCP in the end office exchange routes query messages to its home STP.

In this section, connectionless services are based on end office exchanges querying a database to obtain the routing number for an 800 number. The following is an example of how this works in the network.

Together, SCCP and MTP3 transport TCAP 800-based queries to centralized databases. The connectionless messages passed between the SCCP and MTP are called Unitdata Messages (UDTs) and Unitdata Service Messages (UDTSs).

The SCUP sends a UDT to transfer subsystem information, and it sends a UDT to perform the GTT function. UDTs also are used to query and receive responses from databases. Table 4-2 lists parameters used in the UDT message...

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Table of Contents

Introduction

Part I PSTN

Chapter 1 Overview of the PSTN and Comparisons to Voice over IP

The Beginning of the PSTN

Understanding PSTN Basics

Analog and Digital Signaling

Digital Voice Signals

Local Loops, Trunks, and Interswitch Communication

PSTN Signaling

PSTN Services and Applications

PSTN Numbering Plans

Drivers Behind the Convergence Between Voice and Data Networking

Drawbacks to the PSTN

Packet Telephony Network Drivers

Standards-Based Packet Infrastructure Layer

Open Call-Control Layer

VoIP Call-Control Protocols

Open Service Application Layer

New PSTN Network Infrastructure Model

Summary

Chapter 2 Enterprise Telephony Today

Similarities Between PSTN and ET

Differences Between PSTN and ET

Signaling Treatment

Advanced Features

Common ET and PSTN Interworking

ET Networks Provided by PSTN

Private ET Networks

Summary

Chapter 3 Basic Telephony Signaling

Signaling Overview

Analog and Digital Signaling

Direct Current Signalin8

In-Band and Out-of-Band Signaling

Loop-Start and Ground-Start Signaling

CAS and CCS

E&M Signaling

Type I

Type II

Type III

Type IV

Type V

CAS

Bell System MF Signaling

CCITT No. 5 Signaling

R1

R2

ISDN

ISDN Service5

ISDN Access Interface6

ISDN L2 and L3 Protocols

Basic ISDN Call

QSIG

QSIG Service4

QSIG Architecture and Reference Points

QSIG Protocol Stac5

QSIG Basic Call Setup and Teardown Example

DPNSS

Summary

Chapter 4 Signaling System 7

SS7 Network Architecture

Signaling Elements

Signaling Links

SS7 Protocol Overview

Physical Layer—MTP L1

Data Layer—MTP L2

Network Layer—MTP3

SCCP

TUP

ISUP

TCAP

SS7 Examples

Basic Call Setup and Teardown Example

800 Database Query Example

List of SS7 Specifications

Summary

Chapter 5 PSTN Services

Plain Old Telephone Service

Custom Calling Features

CLASS Features

Voice Mail

Business Services

Virtual Private Voice Networks

Centrex Services

Call Center Services

Service Provider Services

Database Service

Operator Services

Summary

Part II Voice over IP Technology

Chapter 6 IP Tutorial

OSI Reference Model

The Application Layer

The Presentation Layer

The Session Layer

The Transport Layer

The Network Layer

The Data Link Layer

The Physical Layer

Internet Protocol

Data Link Layer Addresses

IP Addressing

Routing Protocols

Distance-Vector Routing

Link-State Routing

BGP

IS-IS

OSPF

IGRP

EIGRP

RIP

IP Transport Mechanisms

TCP

UDP

Summary

References

Chapter 7 VoIP: An In-Depth Analysis

Delay/Latency

Propagation Delay

Handling Delay

Queuing Delay

Jitter

Pulse Code Modulation

What Is PCM?

A Sampling Example for Satellite Networks

Voice Compression

Voice Coding Standards

Mean Opinion Score

Perceptual Speech Quality Measurement

Echo

Packet Loss

Voice Activity Detection

Digital-to-Analog Conversion

Tandem Encoding

Transport Protocols

RTP

Reliable User Data Protocol

Dial-Plan Design

End Office Switch Call-Flow Versus IP Phone Call

Summary

References

Chapter 8 Quality of Service

QoS Network Toolkit

Edge Functions

Bandwidth Limitations

cRTP

Queuing

Packet Classification

Traffic Policing

Traffic Shaping

Edge QoS Wrap-Up

Backbone Networks

High-Speed Transport

Congestion Avoidance

Backbone QoS Wrap-Up

Rules of Thumb for QoS

Cisco Labs’ QoS Testing

Summary

Chapter 9 Billing and Mediation Services

Billing Basics

Authentication, Authorization, and Accounting (AAA)

RADIUS

Vendor-Specific Attributes (VSA)

Billing Formats

Case Study: Cisco SIP Proxy Server and Billing

RADIUS Server Accounting

Challenges for VoIP Networks

Mediation Services

Summary

Chapter 10 Voice Security

Security Requirements

Security Technologies

Shared-Key Approaches

Public-Key Cryptography

Protecting Voice Devices

Disabling Unused Ports/Services

HIPS

Protecting IP Network Infrastructure

Segmentation

Traffic Policing

802.1x Device Authentication

Layer 2 Tools

NIPS

Layer 3 Tools

Security Planning and Policies

Transitive Trust

VoIP Protocol-Specific Issues

Complexity Tradeoffs

NAT/Firewall Traversal

Password and Access Control

Summary

Part III IP Signaling Protocols

Chapter 11 H.323

H.323 Elements

Terminal

Gateway

Gatekeeper

The MCU and Elements

H.323 Proxy Server

H.323 Protocol Suite

RAS Signaling

Call Control Signaling (H.225)

Media Control and Transport (H.245 and RTP/RTCP)

H.323 Call-Flows

Summary

Chapter 12 SIP

SIP Overview

Functionality That SIP Provides

SIP Network Elements

Interaction with Other IETF Protocols

Message Flow in SIP Network

SIP Message Building Blocks

SIP Addressing

SIP Messages

SIP Transactions and Dialog

Transport Layer Protocols for SIP Signaling

Basic Operation of SIP

Proxy Server Example

Redirect Server Example

B2BUA Server Example

SIP Procedures for Registration and Routing

User Agent Discovering SIP Servers in a Network

SIP Registration and User Mobility

SIP Message Routing

Routing of Subsequent Requests Within a SIP Dialog

Signaling Forking at the Proxy

Enhanced Proxy Routing

SIP Extensions

SIP Extension Negotiation Mechanism: Require, Supported, Allow Headers

Caller and Callee Preferences

SIP Event Notification Framework: Subscription and Notifications

SUBSCRIBE and NOTIFY Methods

Monitoring Registration State Using the Subscription-Notification Framework

SIP REFER Request

Presence and Instant Messaging Overview

SIP Extensions for IM and Presence

Summary

Chapter 13 Gateway Control Protocols

MGCP Overview

MGCP Model

Endpoints

Connections

Calls

MGCP Commands and Messages

CreateConnection (CRCX)

ModifyConnection (MDCX)

DeleteConnection (DLCX)

NotificationRequest (RQNT)

Notification (NTFY)

AuditEndpoint (AUEP)

AuditConnection (AUCX)

RestartIn-Progress (RSIP)

EndpointConfiguration (EPCF)

MGCP Response Messages

MGCP Call Flows

Basic MGCP Call Flow

Trunking GW-to-Trunking GW Call Flow

Advanced MGCP Features

Events and Event Packages

Digit Maps

Embedded Notification Requests

Non-IP Bearer Networks

H.248/MEGACO

Summary

Part IV VoIP Applications and Services

Chapter 14 PSTN and VoIP Interworking

Cisco Packet Telephony

Packet Voice Network Overview

Network Elements

Residential Gateway

Network Interfaces

PGW2200 Architecture and Operations

PGW2200-Supported Protocols

Execution Environment

North American Numbering Plan

PGW2200 Implementation

Application Check-Pointing

MGC Node Manager

Accounting

PSTN Signaling Over IP

SCTP

IUA

Changing Landscape of PSTN-IP Interworking

Session Border Controller (SBC)

Summary

Chapter 15 Service Provider VoIP Applications and Services

The Service Provider Dilemma

Service Provider Applications and Benefits

Service Provider VoIP Deployment: Vonage

VoIP Operational Advantages

Service Provider Case Study: Prepaid Calling Card

BOWIE.net Multiservice Networks

Session Border Control: Value Addition

VoIP Peering: Top Priority for the Service Providers

Service Provider VoIP and Consumer Fixed Mobile Convergence

Summary

Chapter 16 Enterprise Voice over IP Applications and Services

Migrating to VoIP Architecture

Enterprise Voice Applications and Benefits

Advanced Enterprise Applications

Web-Based Collaboration and Conference

The Need for Presence Information

Presence-Aware Services

Wi-Fi–Enabled Phones

Better Voice Quality Using Wideband Codecs

Summary

1587052571 TOC 7/6/2006

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