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VoIP and unified communications : internet telephony and the future voice network /

Translates technical jargon into practical business communications solutions This book takes readers from traditional voice, fax, video, and data services delivered via separate platforms to a single, unified platform delivering all of these services seamlessly via the Internet. With its clear, jarg...

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Detalles Bibliográficos
Clasificación:Libro Electrónico
Autor principal: Flanagan, William A.
Formato: Electrónico eBook
Idioma:Inglés
Publicado: Hoboken, N.J. : Wiley, ©2012.
Temas:
Acceso en línea:Texto completo (Requiere registro previo con correo institucional)
Tabla de Contenidos:
  • VoIP and Unified Communications: Internet Telephony and the Future Voice Network; Contents; Preface; Acknowledgments; 1: IP Technology Disrupts Voice Telephony; 1.1: Introduction to the Public Switched Telephone Network; 1.2: The Digital PSTN; 1.3: The Packet Revolution in Telephony; 1.3.1: Summary of Packet Switching; 1.3.2: Link Capacity: TDM versus Packets; 1.3.3: VoIP and The Cloud
  • IN SHORT: Reading Network Drawings; 2: Traditional Telephones Still Set Expectations; 2.1: Availability: How the Bell System Ensured Service; 2.2: Call Completion.
  • 2.3: Sound Quality: Encoding for Recognizable Voices2.4: Low Latency; 2.5: Call Setup Delays; 2.6: Impairments Controlled: Echo, Singing, Distortion, Noise; 3: From Circuits to Packets; 3.1: Data and Signaling Preceded Voice; 3.1.1: X.25 Packet Data Service; 3.1.2: SS7: PSTN Signaling on Packets; 3.1.3: ISDN; 3.2: Putting Voice into Packets; 3.2.1: Voice Encoding; 3.2.2: Dicing and Splicing Voice Streams; 3.2.3: The Latency Budget; 4: Packet Transmission and Switching; 4.1: The Physical Layer: Transmission; IN SHORT: The Endian Wars; 4.2: Data Link Protocols; 4.3: IP, the Network Protocol.
  • 4.4: Layer 4 Transport Protocols4.4.1: Transmission Control Protocol; 4.4.2: User Datagram Protocol; 4.4.3: Stream Control Transmission Protocol; 4.5: Higher Layer Processes; 4.5.1: RTP; 4.5.2: RTCP; 4.5.3: Multiplexing RTP and RTCP on One UDP Port; 4.5.4: RTP Mixers and Translators; 4.5.5: Layered Encoding; 4.5.6: Profiles for Audio and Video Conferences; 4.5.7: Security via Encryption; IN SHORT: Public Key Infrastructure (PKI); 4.6: Saving Bandwidth; 4.6.1: Voice Compression; 4.6.2: Header Compression; 4.6.3: Silence Suppression, VAD; 4.6.4: Sub-Packet Multiplexing.
  • 4.6.5: Protocol and Codec Selection4.7: Differences: Circuit versus Packet Switched; 4.7.1: Power to the Desktop Phone; 4.7.2: Phone as Computer and Computer as Phone; 4.7.3: Length of a Phone Line; 4.7.4: Scaling to Large Size; 4.7.5: Software Ownership and Licenses; 5: VoIP Signaling and Call Processing; 5.1: What Packet Voice and UC Systems Share; 5.2: Session Initiation Protocol (SIP); 5.2.1: SIP Architecture; 5.2.2: SIP Messages; 5.2.3: SIP Header Fields and Behaviors; 5.3: Session Description Protocol; IN SHORT: ABNF; 5.4: Media Gateway Control Protocol; 5.4.1: MGW Functions.
  • 5.4.2: MGW Connection Model5.4.3: Megaco Procedures; 5.4.4: Megaco Details; 5.4.5: Signaling Conversion; 5.4.6: Voice Transcoding; 5.5: H.323; 5.5.1: H.323 Architecture; 5.5.2: Gatekeeper; 5.5.3: Gateway; 5.5.4: Terminal; 5.5.5: Multipoint Control Unit; 5.5.6: Call Procedures; 5.6: Directory Services; 5.6.1: Domain Name Service (DNS); 5.6.2: ENUM; 6: VoIP and Unified Communications Define the Future; 6.1: Voice as Before, with Additions; 6.2: Legacy Services to Keep and Improve with VoIP; 6.2.1: Flexible Call Routing and 800 Numbers; 6.2.2: Call on Hold; 6.2.3: Call Transfer.
  • 6.2.4: Call Forwarding.