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Voice Over IP Fundamentals als Buch

Voice Over IP Fundamentals

Updated. Sprache: Englisch.
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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 d … weiterlesen

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Voice Over IP Fundamentals als Buch


Titel: Voice Over IP Fundamentals
Autor/en: Jonathan Davidson, James F. Peters, Manoj Bhatia

ISBN: 1587052571
EAN: 9781587052576
Sprache: Englisch.

Mai 2006 - kartoniert - 394 Seiten


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


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


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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