Das Session Initiation Protocol (SIP) ist eine neue Internet-Technologie, die einen entscheidenden Durchbruch bei der Übermittlung nahtloser Multimedia-Dienste über das Internet ermöglichen wird. Momentan werden die ersten echten SIP-Netzwerke in Betrieb genommen. Dieses Buch zeigt, wie Sie die Leistungsfähigkeit von SIP maximieren können. Eine unentbehrliche Informationsquelle für Telekommunikationsingenieure und Webentwickler, die multimediale IP-Telefoniedienste aufbauen wollen!
Das Session Initiation Protocol (SIP) ist eine neue Internet-Technologie, die einen entscheidenden Durchbruch bei der Übermittlung nahtloser Multimedia-Dienste über das Internet ermöglichen wird. Momentan werden die ersten echten SIP-Netzwerke in Betrieb genommen. Dieses Buch zeigt, wie Sie die Leistungsfähigkeit von SIP maximieren können. Eine unentbehrliche Informationsquelle für Telekommunikationsingenieure und Webentwickler, die multimediale IP-Telefoniedienste aufbauen wollen!
Dr. Henry Sinnreich (Richardson, TX) is Chief Technology Officer at Pulver.com, a leading media company for VoIP and Internet communication services. Dr. Sinnreich has held engineering and executive positions at MCI where he was an MCI fellow and has been involved in Internet and multimedia services for more than 12 years, including the development of the flagship MCI Advantage service based on SIP. Henry Sinnreich is also a contributor to IETF standards for Internet communications in such areas as SIP telephony devices and using RTP extensions for voice quality monitoring. He was awarded the title Pioneer for VoIP in 2000 at the VON Europe conference. Henry Sinnreich has been a cofounder and board member of the International SIP Forum based in Stockholm. He is a frequent speaker and is known as the leading evangelist, worldwide, for SIP based VoIP, presence, IM, multimedia, and integration of applications with communications. Dr. Sinnreich is also a guest lecturer at the Engineering School of the Southern Methodist University in Dallas, TX. Alan B. Johnston (St. Louis, MO) is a Consulting Member of Technical Staff at Avaya, Inc. He has coauthored the core Internet SIP standard RFC 3261 and four other SIP related RFCs. He is the co-chair of the IETF Centralized Conferencing Working Group and is on the board of directors of the International SIP Forum. His current areas of interest include peer-to-peer SIP and security. Dr. Johnston is a frequent speaker and lecturer on SIP and contributor to various publications, and is an adjunct professor at Washington University in St. Louis, MO.
Inhaltsangabe
Foreword. Acknowledgments. Introduction. Chapter 1 Introduction. Chapter 2 Internet Communications Enabled by SIP. Chapter 3 Architectural Principles of the Internet. Chapter 4 DNS and ENUM. Chapter 5 Real-Time Internet Multimedia. Chapter 6 SIP Overview. Chapter 7 SIP Service Creation. Chapter 8 User Preferences. Chapter 9 SIP Security. Chapter 10 NAT and Firewall Traversal. Chapter 11 SIP Telephony. Chapter 12 Voicemail and Universal Messaging. Chapter 13 Presence and Instant Messaging. Chapter 14 SIP Conferencing. Chapter 15 SIP Application Level Mobility. Chapter 16 Emergency and Preemption Communication Services. Chapter 17 Accessibility for the Disabled. Chapter 18 Quality of Service for Real-Time Internet Communications. Chapter 19 SIP Component Services. Chapter 20 Peer-to-Peer SIP. Chapter 21 Conclusions and Future Directions. Index.
