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A must-have, practical primer on true "carrier class" VoIP--how to use cutting-edge signaling schemes, quality of service (QoS) techniques, and existing protocols to deliver Ma Bell-quality service. This is where telecom managers, engineers, and network managers can go for an easy-to-grasp explantion of TIPHON--the solution to problems with H.323 and IETF specs--to ensure relliability of signaling over IP. Delves into resource reservation schemes that can provide very high QoS.
Leading-edge VoIP technologies, tools, and standards Efficiently deliver voice, data, and multimedia content over today's always-on broadband networks with guidance from this fully updated resource. Carrier-Grade VoIP, Third Edition, shows how to set up and administer a highly reliable unified communications platform using the latest tools. Find out how to choose from the complete spectrum of codecs, enable new HD voice and video services, handle security, and maintain optimal QoS. This comprehensive guide offers start-to-finish details on carrier-grade VoIP network design, troubleshooting, and interconnection. Coverage includes: HD voice services Internet, IP, and VoIP standards Speech-coding techniques H.323 and multimedia conferencing SIP messages and architecture The SS7 protocol suite Interconnecting VoIP networks QoS policies and enforcement Security and privacy issues VoIP network design
"A system administrator's guide to VoIP technologies"--Cover.
Includes new coverage on the advances in signaling protocols,second-generation switching and the development of non-switchedalternatives, and the implementation lessons learned. Contains in-depth coverage of network architectures used tosupport VoIP, performance and voice quality considerations,compression and integration methods for IP tranmissions.
More and more businesses today have their receive phone service through Internet instead of local phone company lines. Many businesses are also using their internal local and wide-area network infrastructure to replace legacy enterprise telephone networks. This migration to a single network carrying voice and data is called convergence, and it's revolutionizing the world of telecommunications by slashing costs and empowering users. The technology of families driving this convergence is called VoIP, or Voice over IP. VoIP has advanced Internet-based telephony to a viable solution, piquing the interest of companies small and large. The primary reason for migrating to VoIP is cost, as it equalizes the costs of long distance calls, local calls, and e-mails to fractions of a penny per use. But the real enterprise turn-on is how VoIP empowersbusinesses to mold and customize telecom and datacom solutions using a single, cohesive networking platform. These business drivers are so compelling that legacy telephony is going the way of the dinosaur, yielding to Voice over IP as the dominant enterprise communications paradigm. Developed from real-world experience by a senior developer, O'Reilly's Switching to VoIP provides solutions for the most common VoIP migration challenges. So if you're a network professional who is migrating from a traditional telephony system to a modern, feature-rich network, this book is a must-have. You'lldiscover the strengths and weaknesses of circuit-switched and packet-switched networks, how VoIP systems impact network infrastructure, as well as solutions for common challenges involved with IP voice migrations. Among the challenges discussed and projects presented: building a softPBX configuring IP phones ensuring quality of service scalability standards-compliance topological considerations coordinating a complete system ?switchover? migrating applications like voicemail and directoryservices retro-interfacing to traditional telephony supporting mobile users security and survivability dealing with the challenges of NAT To help you grasp the core principles at work, Switching to VoIP uses a combination of strategy and hands-on how-to that introduce VoIP routers and media gateways, various makes of IP telephone equipment, legacy analog phones, IPTables and Linux firewalls, and the Asterisk open source PBX software by Digium.You'll learn how to build an IP-based or legacy-compatible phone system and voicemail system complete with e-mail integration while becoming familiar with VoIP protocols and devices. Switching to VoIP remains vendor-neutral and advocates standards, not brands. Some of the standards explored include: SIP H.323, SCCP, and IAX Voice codecs 802.3af Type of Service, IP precedence, DiffServ, and RSVP 802.1a/b/g WLAN If VoIP has your attention, like so many others, then Switching to VoIP will help you build your own system, install it, and begin making calls. It's the only thing left between you and a modern telecom network.
"This book is like a good tour guide.It doesn't just describe the major attractions; you share in the history, spirit, language, and culture of the place." --Henning Schulzrinne, Professor, Columbia University Since its birth in 1996, Session Initiation Protocol (SIP) has grown up. As a richer, much more robust technology, SIP today is fully capable of supporting the communication systems that power our twenty-first century work and life. This second edition handbook has been revamped to cover the newest standards, services, and products. You'll find the latest on SIP usage beyond VoIP, including Presence, instant messaging (IM), mobility, and emergency services, as well as peer-to-peer SIP applications, quality-of-service, and security issues--everything you need to build and deploy today's SIP services. This book will help you * Work with SIP in Presence and event-based communications * Handle SIP-based application-level mobility issues * Develop applications to facilitate communications access for users with disabilities * Set up Internet-based emergency services * Explore how peer-to-peer SIP systems may change VoIP * Understand the critical importance of Internet transparency * Identify relevant standards and specifications * Handle potential quality-of-service and security problems
This book presents a review of the latest advances in speech and video compression, computer networking protocols, the assessment and monitoring of VoIP quality, and next generation network architectures for multimedia services. The book also concludes with three case studies, each presenting easy-to-follow step-by-step instructions together with challenging hands-on exercises. Features: provides illustrative worked examples and end-of-chapter problems; examines speech and video compression techniques, together with speech and video compression standards; describes the media transport protocols RTP and RTCP, as well as the VoIP signalling protocols SIP and SDP; discusses the concepts of VoIP quality of service and quality of experience; reviews next-generation networks based on the IP multimedia subsystem and mobile VoIP; presents case studies on building a VoIP system based on Asterisk, setting up a mobile VoIP system based on Open IMS and Android mobile, and analysing VoIP protocols and quality.
Now that virtually every leading telecommunications service provider has committed to delivering IP-based telephony services, communications professionals face the enormous challenge of implementation. This hands-on guide brings together today's best known answers and solutions for delivering VolP services with the quality customers deman. No other book covers the combined issues of protocol signaling, media transport methodology, reference topological considerations and voice quality testing in service offerings. No matter what your role in delivering Voice Over IP (VoIP) services, IP Telephony delivers the specifics you need to speed deployment, improve reliability, ensure quality, and simplify troubleshooting. Precise, thorough, and based firmly in the real-world, it is simply indispensable. The accompanying CD-ROM contains an extensive library of IP telephony-related RFCs, pertinent white papers and application notes that include real-world IP Telephony measurement examples.
This guide shows programmers and administrators how to implement, program and administer VOIP systems using open source tools instead of more expensive options.