Voice over IP Fundamentals / Edition 2 available in Paperback
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Voice over IP Fundamentals / Edition 2
- ISBN-10:
- 1587052571
- ISBN-13:
- 9781587052576
- Pub. Date:
- 08/10/2006
- Publisher:
- Cisco Press
- ISBN-10:
- 1587052571
- ISBN-13:
- 9781587052576
- Pub. Date:
- 08/10/2006
- Publisher:
- Cisco Press
![Voice over IP Fundamentals / Edition 2](http://vs-images.bn-web.com/static/redesign/srcs/images/grey-box.png?v11.12.2_temp1)
Voice over IP Fundamentals / Edition 2
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Overview
A systematic approach to understanding the basics of voice over IP
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.
Product Details
ISBN-13: | 9781587052576 |
---|---|
Publisher: | Cisco Press |
Publication date: | 08/10/2006 |
Series: | Fundamentals Series |
Edition description: | REV |
Pages: | 432 |
Product dimensions: | 7.30(w) x 9.00(h) x 1.00(d) |
About 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.
Read an Excerpt
Chapter 4: Signaling System 7
Congestion ControlMTP2 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.
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
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...
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