Carrier Grade Voice Over IP, 2/e (IE-Paperback)

Daniel Collins

  • 出版商: McGraw-Hill
  • 出版日期: 2002-09-16
  • 售價: $1,060
  • 貴賓價: 9.5$1,007
  • 語言: 英文
  • 頁數: 522
  • ISBN: 0071231552
  • ISBN-13: 9780071231558

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Description:

In 2002 voice over IP will constitute more than 25% of all long distance voice calls, according to Network World. That’s more than a 30% ramp-up from 2001. The emergence of SIP, MPLS and new quality of service tools is making carrier grade voice over IP a service reality, and a potentially huge margin booster and revenue driver for service providers. The first edition of Carrier Grade Voice over IP played a roll in VoIP growth, in less than year becoming an essential tool for carriers working to provide high quality IP telephony. This new edition vastly updates the SIP chapter, details MPLS, and takes the explanations of the previous edition a step further in a final chapter that shows, step by step, how to design working VoIP networks.

 

Table of Contents:

CHAPTER 1: INTRODUCTION
What is Meant by Carrier-Grade?
What is Meant by VoIP?
A Little About IP
Why VoIP?
Why Carry Voice?
Why Use IP for Voice?
Lower Equipment Cost
Voice/Data Integration and Advanced Services
Potentially Lower Bandwidth Requirements
The Widespread Availability of IP
The VoIP Market
VoIP Challenges
Speech Quality
Managing Access and Prioritizing Traffic
Speech-Coding Techniques
Network Reliability and Scalability
Overview of the Following Chapters
CHAPTER 2: TRANSPORTING VOICE BY USING IP
Overview of the IP Protocol Suite
Internet Standards and the Standards Process
The Internet Society
The Internet Architecture Board (IAB)
The Internet Engineering Task Force (ETF)
The Internet Engineering Steering Group (ESG)
The Internet Assigned Numbers Authority (IANA)
The Internet Standards Process
The Internet Prototol (IP)
The IP Header
IP Routing
The Transmission Control Protocol (TCP)
The TCP Header
TCP Connections
The User Datagram Protocol (UDP)
Voice over UDP, not TCP
The Real-Time Transport Protocol (RTP)
RTP Payload Formats
The RTP Header
Mixers and Translators
The RTP Control Protocol (RTCP)
RTCP Sender Report (SR)
RTCP Receiver Report (RR)
RTCP Source Description Packet (SDES)
RTCP BYE Packet
Application-Defined RTCP Packet
Calculating Round-Trip Time
Calculating Jitter
Timing of RTCP Packets
IP Multicast
IP Version 6 (IPv6)
IPv6 Header
IPv6 Addresses
IPv6 Header Extensions
Interworking IPv4 and IPv6
CHAPTER 3: SPEECH-CODING TECHNIQUES
Voice Quality
A Little About Speech
Voice Sampling
Quantization
Types of Speech Coders
G.711
Adaptive Differential PCM (ADPCM)
Analysis-by-Synthesis (AbS) Codecs
G.728 Low-Delay CELP (LD-CELP)
G.723.1 Algebraic Code-Excited Linear Prediction (ACELP)
G.729
Selecting Codecs
Cascaded Codecs
Tones, Signals, and Dual-Tone Multifrequency (DTMF) Digits
CHAPTER 4: H.323
The H.323 Architecture
Overview of H.323 Signaling
Overview of H.323 Protocols
H.323 Addressing
Codecs
RAS Signaling
Gatekeeper Discovery
Endpoint Registration and Registration Cancellation
Endpoint Location
Admission
Bandwidth Change
Status
Disengage
Resource Availability
Service Control
Request in Progress
Call Signaling
Setup
Call-Proceeding
Alerting
Progress
Connect
Release Complete
Facility
Interaction Between Call Signaling and H.245 Control Signaling
Call Scenarios
Basic Call Without Gatekeepers
A Basic Call with Gatekeepers and Direct Endpoint Call Signaling
A Basic Call with Gatekeeper/Direct Routed Call Signaling
A Basic Call with Gatekeeper-Routed Call Signaling
Optional Called-Endpoint Signaling
H.245 Control Signaling
H.245 Message Groupings
The Concept of Logical Channels
H.245 Procedures
Fast Connect Procedure
H.245 Message Encapsulation
Conference Calls
Pre-arranged Conference
Ad Hoc Conference
The Decomposed Gateway
CHAPTER 5: THE SESSION INITIATION PROTOCOL (SIP)
The Popularity of SIP
The SIP Architecture
SIP Network Entities
SIP Call Establishment
SIP Advantages over Other Signaling Protocols
Overview of SIP Messaging Syntax
SIP Requests
SIP Responses
SIP Addressing
Message Headers
Examples of SIP Message Sequences
Registration
Invitation
Termination of a Call
Redirect and Proxy Servers
Redirect Services
Proxy Servers
The Session Description Protocol (SDP)
The Structure of SDP
SDP Syntax
Usage of SDP with SIP
Negotiation of Media
SIP Extensions and Enhancements
The SIP INFO Method
SIP Event Notification
SIP for Instant Messaging
The SIP REFER Method
Reliability of Provisional Responses
The SIP UPDATE Method
Integration of SIP Signaling and Resource Management
Usage of SIP for Features and Services
Call Forwarding
Consultation Hold
Interworking
PSTN Interworking
Interworking with H.323
Summary
CHAPTER 6: MEDIA GATEWAY CONTROL AND THE SOFTSWITCH ARCHITECTURE
Separation of Media and Call Control
Softswitch Architecture
Requirements for Media Gateway Control
Protocols for Media Gateway Control
MGCP
The MGCP Model
MGCP Endpoints
MGCP Calls and Connections
Overview of MGCP Commands
Overview of MGCP Responses
Command and Response Details
Call Setup Using MGCP
MGCP Events, Signals, and Packages
Interworking Between MGCP and SIP
MEGACO.248
MEGACO Architecture
Overview of MEGACO Commands
Descriptors
Packages
MEGACO Command and Response Details
Call Setup Using MEGACO
Interworking Between MEGACO and SIP
CHAPTER 7: VoIP and SS7
The SS7 Protocol Suite
The Message Transfer Part (MTP)
ISDN User Part (ISUP) and Signaling Connection Control Part (SCCP)
SS7 Network Architecture
Signaling Points (SPs)
Signal Transfer Point (STP)
Service Control Point (SCP)
Message Signal Units (MSUs)
SS7 Addressing
ISUP
Performance Requirements for SS&
Sigtran
Sigtran Architecture
SCTP
M3UA Operation
M2UA Operation
M2PA Operation
Interworking SS7 and VoIP Architectures
Interworking Softswitch and SS7
Interworking H.323 and SS7
CHAPTER 8: QUALITY OF SERVICE (QoS)
The Need for QoS
End-to-End QoS
It's Not Just the Network
Overview of QoS Solutions
More Bandwidth
QoS Protocols and Architectures
QoS Policies
The Resource Reservation Protocol (RSVP)
RSVP Syntax
Establishing Reservations
Reservation Errors
Guaranteed Service
Controlled-Load Service
Removing Reservations and the Use of Soft State
DiffServ
The DiffServ Architecture
The Need for SLAs
Per-Hop Behavior (PHB)
Multiprotocol Label Switching (MPLS)
The MPLS Architecture
FEC and Label Formats
Actions at LSRs
MPLS Traffic Engineering
Label Distribution Protocols and Constraint-Based Routing
RSVP Traffic Engineering (RSVP-TE)
Combining QoS Solutions
CHAPTER 9: DESIGNING A VOICE OVER IP NETWORK
Design Criteria
Build-Ahead or Capacity Buffer
Fundamental Technology Assumptions
Network-Level Redundancy
Voice Coder/Decoder (Codec) Selection Issues
Blocking Probability
QoS Protocol Considerations and Layer 2 Protocol Choices
Product and Vendor Selection
Generic VoIP Product Requirements
Element Management
Traffic Forecasts
Voice Usae Forecast
Traffic Distribution Forecast
Node Locations and Bandwidth Requirements
MG Locations and PSTN Trunk Dimensioning
MSG, SG, and EMS Dimensioning and Placement
Calculating VoIP Bandwidth Requirements
Physical Connectivity
APPENDIX A: TABLE OF ERLANG B
APPENDIX B: VISUAL BASIC CODE FOR ERLANG CALCULATIONS
Glossary of Acronyms
References
Index