1 Curso de Telefonía IP para el MTC Sesión 1 Introducción Mg. Antonio Ocampo Zúñiga
2 Conceptos Generales
3 VoIP Essentials Family of technologies Carries voice calls over an IP network VoIP services convert traditional TDM analog voice streams into a digital signal Call from: Computer IP Phone Traditional (POTS) phone
4 Business Case for VoIP Cost savings Flexibility Advanced features: Advanced call routing Unified messaging Integrated information systems Voice security Telephony application services
5 Components of a VoIP Network Application Server Multipoint Control Unit PSTN IP Backbone PBX Call Agent IP Phone Router or Gateway Router or Gateway Router or Gateway IP Phone Videoconference Station
6 Basic Components of a Traditional Telephony Network Edge Devices Tie Trunks CO CO Tie Trunks PBX Switch Switch PBX CO Trunks CO Trunks San Jose Local Loops Local Loops Boston PSTN
7 Signaling Protocols Protocol H.323 MGCP SIP SCCP or Skinny Description ITU standard protocol for interactive conferencing; evolved from H.320 ISDN standard; flexible, complex IETF standard for PSTN gateway control; thin device control IETF protocol for interactive and noninteractive conferencing; simpler, but less mature, than H.323 Cisco proprietary protocol used between Cisco Unified Communications Manager and Cisco VoIP phones
8 H.323 Approved in 1996 by the ITU-T. Peer-to-peer protocol where end devices initiate sessions. Widely used with gateways, gatekeepers, or clients, especially video terminals
9 MGCP (Media Gateway Control Protocol) IETF RFC 2705 developed in Client/server protocol that allows a callcontrol device to take control of a specific port on a gateway. For an MGCP interaction to take place with Communications Manager, you have to make sure that the OS is compatible with Communications Manager version. The PRI backhaul and BRI backhauling concepts are the most powerful concepts to MGCP
10 SIP (Session Initiation Protocol) IETF RFC 2543 (1999), RFC 3261 (2002), and RFC 3665 (2003). Based on the logic of the World Wide Web. Widely used with gateways and proxy servers within service provider networks. Peer-to-peer protocol where end devices (user agents) initiate sessions. ASCII text-based for easy implementation and debugging.
11 SCCP (Skinny Call Control Protocol) Cisco proprietary terminal control protocol. Stimulus protocol: For every event, the end device sends a message to the Cisco Unified Communications Manager. Can be used to control gateway FXS ports. Proprietary nature allows quick additions and changes.
12 Comparing Signaling Protocols H.323 suite: Peer-to-peer protocol Gateway configuration necessary because gateway must maintain dial plan and route pattern. PSTN H.323 Q.921 Q.931
13 Comparing Signaling Protocols (Cont.) MGCP: Works in a client/server architecture Simplified configuration Communications Manager maintains the dial plan PSTN MGCP Q.921 Q.931
14 Comparing Signaling Protocols (Cont.) SIP: Peer-to-peer protocol. Gateway configuration is necessary because the gateway must maintain a dial plan and route pattern. PSTN SIP Q.921 Q.931
15 Comparing Signaling Protocols (Cont.) SCCP Works in a client/server architecture. Simplified configuration. Communications Manager maintains a dial plan and route patterns. PSTN SCCP Endpoint SCCP
16 VoIP Service Considerations Latency Jitter Bandwidth Packet loss Reliability Security
17 Media Transmission Protocols Real-Time Transport Protocol: Delivers the actual audio and video streams over networks Real-Time Transport Control Protocol: Provides out-of-band control information for an RTP flow
18 Media Transmission Protocols crtp: Compresses IP/UDP/RTP headers on low-speed serial links SRTP: Provides encryption, message authentication and integrity, and replay protection to the RTP data
19 Real-Time Transport Protocol GateKeeper GW1 GW2 Provides end-to-end network functions and delivery services for delay-sensitive, real-time data, such as voice and video Runs on top of UDP RTP Stream Works well with queuing to prioritize voice traffic over other traffic
20 Real-Time Transport Protocol GateKeeper GW1 GW2 RTP Stream Services include: Payload-type identification Sequence numbering Time stamping Delivery monitoring
21 Real-Time Transport Control Protocol Define in RFCs 1889, 3550 Provides out-of-band control information for a RTP flow Used for QoS reporting Monitors the quality of the data distribution and provides control information
22 Real-Time Transport Control Protocol Provides feedback on current network conditions Allows hosts involved in an RTP session to exchange information about monitoring and controlling the session Provides a separate flow from RTP for UDP transport use
23 Compressed RTP S0/0 GW1 crtp on Slow-Speed Serial Links RTP Stream S0/0 GW2 RFCs RFC 2508, Compressing IP/UDP/RTP Headers for Low-Speed Serial Links RFC 2509, IP Header Compression over PPP
24 Compressed RTP S0/0 GW1 crtp on Slow-Speed Serial Links RTP Stream S0/0 GW2 Enhanced CRTP RFC 3545, Enhanced Compressed RTP (CRTP) for Links with High Delay, Packet Loss and Reordering Compresses 40-byte header to approximately 2 to 4 bytes
26 Summary VoIP is the family of technologies that allow IP networks to be used for voice applications, such as telephony, voice instant messaging, and teleconferencing. VoIP uses H.323, MGCP, SIP, and SCCP call signaling and call control protocols.
27 Summary Signaling protocol models range from peerto-peer, client server, and stimulus protocol. Configuring voice in a data network requires network services with low delay, minimal jitter, and minimal packet loss. The actual voice conversations are transported across the transmission media using RTP and other RTP related protocols.
28 Voice Gateways
29 Understanding Gateways A gateway connects IP communication networks to analog devices, to the PSTN, or to a PBX Specifically, its role is the following: Convert IP telephony packets into analog or digital signals Connect an IP telephony network to analog or digital trunks or to individual analog stations Two gateway signaling types: Analog Digital
30 Gateways Support these gateway protocols: H.323 MGCP SIP SCCP Provide advanced gateway functionality DTMF relay Supplementary services
31 Gateways Work with redundant Communication Manager Enable call survivability Provide QSIG support. Provide fax or modem services, or both
32 Deploying Gateways Lima Unified Communications Manager Cluster IP WAN Arequipa MGCP LI-GW PSTN Unified Communications Manager Express H.323 AQ-GW Trujillo SIP TR-GW SIP Proxy Server
33 Gateway Hardware Platforms Modern enterprise models: Cisco 2800 Series Routers Cisco 3800 Series Routers Cisco Catalyst 6500 Series
34 Gateway Hardware Platforms (Cont.) Well-known and widely used older enterprise models: Cisco 1751-V Router EOS: 03/2007 EOL: 03/2012 Cisco 1760-V Router EOS: 03/2007 EOL: 03/2012 Cisco 2600XM Series Routers EOS: 03/2007 EOL: 03/2012 Cisco 3600 Series Platforms EOS: 12/2004 EOL: 12/2008 Cisco 3700 Series Routers EOS: 03/2007 EOL: 03/2012
35 Gateway Hardware Platforms (Cont.) Special voice gateways: Cisco VG224 and VG248 Gateways Cisco AS5300 and AS5400 Series Gateways Cisco AS5850 Gateway Cisco 827-4V Router EOS: 05/2005 EOL: 05/2010 Cisco ATA 186 Cisco 7200 Series Routers
36 Gateway Hardware Platforms (Cont.) H.323 Cisco Unified Communications Manager MGCP Cisco 827-4V Router Yes No No No Cisco 2800 Series Routers Yes Yes Yes Yes Cisco 3800 Series Routers Yes Yes Yes Yes Cisco 1751-V and 1760-V Routers Yes Yes No Yes 1 Cisco 2600XM Series Router Yes Yes No No 3 Cisco 3600 Series Platforms Yes Yes No No 3 Cisco 3700 Series Routers Yes Yes No No 3 Cisco VG224 Gateway Yes 2 Yes 2 No Yes Cisco VG248 Gateway No No No Yes Cisco AS53XX and AS5400 and AS5850 Cisco Gateways SIP SCCP Yes No No No Communication Media Module Yes Yes Yes Yes GW Module WS-X6608-x1 and FXS Module WS-X6624 No Yes No Yes Cisco ATA 180 Series Yes 2 Yes 2 No Yes 2 Cisco 7200 Series Routers Yes No No No 1 Conferencing and transcoding only 2 FXS only 3 DSP farm
37 IP Telephony Deployment Models Applications Communications Manager Cluster PSTN Communications Manager Cluster Applications IP WAN Branch Headquarters Single-site deployment Multisite WAN with centralized call processing
38 IP Telephony Deployment Models Applications Communications Manager Cluster PSTN Communications Manager Cluster Applications IP WAN Branch Headquarters Multisite WAN with distributed call processing Clustering over the IP WAN
39 Single-Site Deployment Communications Manager servers, applications, and DSP resources at same physical location IP WAN (if one) used for data traffic only Communications Manager Cluster PSTN SIP or SCCP WAN
40 Single-Site Deployment PSTN used for all external calls Supports approximately 30,000 IP phones per cluster Communications Manager Cluster PSTN SIP or SCCP WAN
41 Design Guidelines Provide a highly available, fault-tolerant infrastructure. Understand the current calling patterns within the enterprise. Use the G.711 codec for all endpoints; DSP resources can be allocated to other functions, such as conferencing and MTP.
42 Design Guidelines Use H.323, SIP, SRST, and MGCP gateways for the PSTN. Implement the recommended network infrastructure for high availability, connectivity options for phones, QoS mechanisms, and security.
43 Multisite WAN with Centralized Call Processing Unified Communications Manager at central site; applications and DSP resources centralized or distributed IP WAN carries voice traffic and call control signaling SRSTcapable SRSTcapable SIP or SCCP PSTN IP WAN Communications Manager Cluster SIP or SCCP SIP or SCCP
44 Multisite WAN with Centralized Call Processing Supports approximately 30,000 IP phones per cluster Call admission control (limit number of calls per site) SRST for remote branches SRSTcapable SRSTcapable SIP or SCCP PSTN IP WAN Communications Manager Cluster SIP or SCCP SIP or SCCP
45 Design Guidelines Minimize delay between Communications Manager and remote locations to reduce voice cut-through delays. Use the locations mechanism in Communications Manager to provide call admission control into and out of remote branches. The number of IP phones and line appearances supported in SRST mode at each remote site depends on the branch router platform.
46 Design Guidelines At the remote sites, use SRST, Communications Manager Express in SRST mode, SIP SRST, and MGCP gateway fallback to ensure call-processing survivability in the event of a WAN failure. Use HSRP to provide backup gateways and gatekeepers.
47 Multisite WAN with Distributed Call Processing Communications Manager and applications located at each site IP WAN carries intercluster call control signaling SIP or SCCP SIP or SCCP PSTN Call Manager Cluster GK IP WAN Call Manager Cluster Gatekeeper SIP or SCCP
48 Multisite WAN with Distributed Call Processing Scales to hundreds of sites Transparent use of the PSTN if the IP WAN is unavailable SIP or SCCP SIP or SCCP PSTN GK IP WAN Call Manager Cluster Gatekeeper SIP or SCCP Call Manager Cluster
49 Design Guidelines Use HSRP gatekeeper pairs, gatekeeper clustering, and alternate gatekeeper support for resiliency. Size the gateway and gatekeeper platforms appropriately per the SRND. Deploy a single WAN codec.
50 Design Guidelines Gatekeeper networks scale to hundreds of sites. Provide adequate redundancy for the SIP proxies. Ensure that the SIP proxies have the capacity for the call rate and number of calls required in the network.
51 Clustering over the IP WAN Publisher or TFTP server <40 ms Round-Trip Delay IP WAN SIP or SCCP QoS-Enabled Bandwidth Applications and Communications Managers of the same cluster distributed over the IP WAN IP WAN carries intracluster server communication and signaling SIP or SCCP
52 Clustering over the IP WAN Publisher or TFTP server <40 ms Round-Trip Delay IP WAN SIP or SCCP QoS-Enabled Bandwidth Limited number of sites SIP or SCCP
53 WAN Considerations 40-ms maximum RTT between any two Communications Manager servers in the cluster Use QoS to minimize jitter for the IP Precedence 3 ICCS traffic. Design network to provide sufficient prioritized bandwidth for all ICCS traffic, especially the priority ICCS traffic.
54 WAN Considerations The general rule of thumb for bandwidth is to over-provision and undersubscribe. QoS-enabled bandwidth must be engineered into the network infrastructure.
55 Summary Gateways connect IP communications networks to traditional telephony networks. There are several types of voice gateways that can be used to meet all kinds of customer needs, from small enterprises to large service provider networks.
56 Summary Telephony deployment models are singlesite, multisite with centralized call processing, multisite with distributed call processing, and clustering over the IP WAN. In the single-site deployment model, the Communications Manager applications and the DSP resources are at the same physical location; the PSTN handles all external calls.
57 Summary The multisite centralized model has a single call-processing agent, applications and DSP resources are centralized or distributed, and the IP WAN carries voice traffic and call control signaling between sites. The multisite distributed model has multiple independent sites, each with a callprocessing agent, and the IP WAN carries voice traffic but not call control signaling between sites.
58 Summary Clustering over an IP WAN provides central administration, a unified dial plan, feature extension to all offices, and support for more remote phones during failover, but places strict delay and bandwidth requirements on the WAN.
CHAPTER 2 Sections in this chapter address the following topics: Single Site, page 2-1 Multisite Implementation with Distributed Call Processing, page 2-3 Design Considerations for Section 508 Conformance,
Encapsulating Voice in IP Packets Major VoIP Protocols This topic defines the major VoIP protocols and matches them with the seven layers of the OSI model. Major VoIP Protocols 15 The major VoIP protocols
Curso de Telefonía IP para el MTC Sesión 2 Requerimientos principales Mg. Antonio Ocampo Zúñiga Factors Affecting Audio Clarity Fidelity: Audio accuracy or quality Echo: Usually due to impedance mismatch
Gateways and Their Roles Understanding Gateways This topic describes the role of voice gateways and their application when connecting VoIP to traditional PSTN and telephony equipment. Analog vs. Digital
Requirements of Voice in an IP Internetwork Real-Time Voice in a Best-Effort IP Internetwork This topic lists problems associated with implementation of real-time voice traffic in a best-effort IP internetwork.
Voice over IP Basics for IT Technicians White Paper Executive summary The IP phone is coming or has arrived on desk near you. The IP phone is not a PC, but does have a number of hardware and software elements
Cisco CME Features and Functionality Supported Protocols and Integration Options This topic describes the supported protocols and integration options of Cisco CME. Supported Protocols and Integration FAX
IMPLEMENTING CISCO VOICE COMMUNICATIONS AND QOS Volume 1 Course Introduction Overview Learner Skills and Knowledge Course Goal and Course Flow Additional References Cisco Glossary of Terms Your Training
CVOICE Exam Topics Cisco Voice over IP Exam #642-432 7/14/2005 From the Cisco CVOICE 642-432 Exam Topics Voice over IP Technologies Describe the similarities and differences between PSTN and VoIP including
Survivable Remote Site Telephony Version 4.1 As the enterprise extends its IP telephony deployments from central sites to remote offices, a critical factor in achieving a successful deployment is the capability
Cisco Catalyst 6500 Series and Cisco 7600 Series Communication Media Module Product Overview Cisco Unified Communications is a comprehensive IP communications system of voice, video, data, and mobility
Voice over IP (VoIP) Basics for IT Technicians VoIP brings a new environment to the network technician that requires expanded knowledge and tools to deploy and troubleshoot IP phones. This paper provides
Remote Site Telephony Version 4.1 As the enterprise extends its IP telephony deployments from central sites to remote offices, one of the critical factors in achieving a successful deployment is the ability
VoIP Bandwidth Considerations - design decisions When calculating the bandwidth requirements for a VoIP implementation the two main protocols are: a signalling protocol such as SIP, H.323, SCCP, IAX or
An Introduction to VoIP Protocols www.netqos.com Voice over IP (VoIP) offers the vision of a converged network carrying multiple types of traffic (voice, video, and data, to name a few). To carry out this
Gen2 Ventures White Paper Clearing the Way for VoIP An Alternative to Expensive WAN Upgrades Executive Overview Enterprises have traditionally maintained separate networks for their voice and data traffic.
Survivable Remote Site Telephony Version 7.0 Communications solutions unify voice, video, data, and mobile applications on fixed and mobile networks, enabling easy collaboration every time from any workspace.
Data Sheet Telephony Version 3.4 As the enterprise extends its IP telephony deployments from central sites to remote offices, one of the critical factors in achieving a successful deployment is the ability
CHAPTER 2 Telephony Deployment odels Last revised on: February 13, 2008 This chapter describes the Telephony deployment models for Cisco Unified Callanager 4.2. For design guidance with earlier releases
Survivable Remote Site Telephony Version 4.2 Communications solutions unify voice, video, data, and mobile applications on fixed and mobile networks, delivering a media-rich collaboration experience across
Implementing Cisco IP Telephony & Video, Part 2 CIPTV2 v1.0; 5 Days; Instructor-led Course Description Implementing Cisco IP Telephony & Video, Part 2 (CIPTV2) v1.0 is a five-day course that prepares the
VoIP and IP Telephony @ IT Tralee email@example.com Presentation outline: Basic overview of IP telephony and technology Detailed overview of VoIP @ IT Tralee deployment How IPT has benefited
Enterprise Vo Terena 2000 ftp://ftpeng.cisco.com/sgai/t2000voip.pdf Silvano Gai Cisco Systems, USA Politecnico di Torino, IT firstname.lastname@example.org Terena 2000 1 Compass Motivation for Vo Voice over in the Enterprise
Survivable Remote Site Telephony Version 7.1 Communications Solutions unify voice, video, data, and mobile applications on fixed and mobile networks, enabling easy collaboration every time from any workspace.
VOICE SERVICES AND AVIATION DATA NETWORKS Anuj Bhatia, Anant Shah, Nagaraja Thanthry, and Ravi Pendse, Department of Electrical and Computer Engineering, Wichita State University, Wichita KS Abstract The
CHAPTER 1 Introducing Cisco Unified Communications Express Cisco Unified Communications Express is an award-winning communications solution that is provided with the Cisco Integrated Services Router portfolio.
Curso de Telefonía IP para el MTC Sesión 4-1 Tipos de llamadas Mg. Antonio Ocampo Zúñiga Call Types Local: Does not traverse the WAN or PSTN. On-net: Occurs between two telephones on the same data network.
Communications Transformations 2: Steps to Integrate SIP Trunk into the Enterprise The Changing Landscape IP-based unified communications is widely deployed in enterprise networks, both for internal calling
Overview of Voice Over Internet Protocol Purva R. Rajkotia, Samsung Electronics November 4,2004 Overview of Voice Over Internet Protocol Presentation Outline History of VoIP What is VoIP? Components of
Survivable Remote Site Telephony Version 8.0 Communications Solutions unify voice, video, data, and mobile applications on fixed and mobile networks, enabling easy collaboration every time from any workspace.
1 Voice over IP (VoIP) - Basic Voice over IP - Voice over IP (VoIP) is a digital form of transport for voice transmissions, replacing analog phone systems. The benefits of VoIP are considerable: Better
Voice over IP (VoIP) Professor Richard Harris School of Engineering and Advanced Technology (SEAT) Presentation Outline Basic IP phone set up The SIP protocol Computer Networks - 1/2 Learning Objectives
IP Telephony v1.0 Scope and Sequence Cisco Networking Academy Program Table of Content COURSE OVERVIEW...4 Course Description...4 Course Objectives...4 Target Audience...5 Prerequisites...5 Lab Requirements...5
Article VoIP Introduction Internet telephony refers to communications services voice, fax, SMS, and/or voice-messaging applications that are transported via the internet, rather than the public switched
SIP Trunking Christina Hattingh Darryl Sladden ATM Zakaria Swapan Cisco Press 800 East 96th Street Indianapolis, IN 46240 SIP Trunking Contents Introduction xix Part I: From TDM Trunking to SIP Trunking
Introduction This 4-day course offers a practical introduction to 'hands on' VoIP engineering. Voice over IP promises to reduce your telephony costs and provides unique opportunities for integrating voice
White Paper Local Session Controller: Cisco s Solution for the U.S. Department of Defense Network of the Future What You Will Learn The future of the Department of Defense s (DoD) networks focuses on the
Voice over IP Professor Richard Harris Presentation Outline Brief overview of VoIP and applications Challenges of VoIP IP Support for Voice Protocols used for VoIP (current views) RTP RTCP RSVP H.323 Semester
Configuring Campus Switches to Support Voice BCMSN Module 7 1 The Basics VoIP is a technology that digitizes sound, divides that sound into packets, and transmits those packets over an IP network. VoIP
Course 4: IP Telephony and VoIP Telecommunications Technical Curriculum Program 3: Voice Knowledge 6/9/2009 1 Telecommunications Technical Curriculum Program 1: General Industry Knowledge Course 1: General
ACD: Average Call Duration is the average duration of the calls routed bya a VoIP provider. It is a quality parameter given by the VoIP providers. API: An application programming interface (API) is a source
Understanding Voice over IP Protocols Cisco Systems Service Provider Solutions Engineering February, 2002 1 Topics to Discuss History of VoIP VoIP Early Adopters VoIP Standards and Standards Bodies VoIP
VoIP Architecture VoIP: Architectural Differences of SIP and MGCP/NCS Protocols and What It Means in Real World VoIP Service Marcin Godlewski Lead Engineer Scientific Atlanta, a Cisco Company Charles Moreman
CIPT1 Implementing Cisco Unified Communications IP Telephony Part 1 Volume 2 Version 6.0 Student Guide Editing, Production, and Web Services: 02-15-08 DISCLAIMER WARRANTY: THIS CONTENT IS BEING PROVIDED
Peer-to-Peer SIP Mode with FXS and FXO Gateways New Rock s SIP based VoIP gateways with FXS and FXO ports support peer-to-peer mode which has many applications in deploying enterprise multi-site telephone
Hands-On Course Description Voice over IP is being widely implemented both within companies and across the Internet. The key problems with IP voice services are maintaining the quality of the voice service
Integrate VoIP with your existing network As organisations increasingly recognise and require the benefits voice over Internet Protocol (VoIP) offers, they stop asking "Why?" and start asking "How?". A
Fax over IP Contents Introduction Why Fax over IP? How Real-time Fax over IP works Implementation with MessagePlus/Open Summary About this document This document describes how Fax over IP works in general
Voice over IP Abdus Salam ICTP, February 2004 School on Digital Radio Communications for Research and Training in Developing Countries Ermanno Pietrosemoli Latin American Networking School (Fundación EsLaRed)
EarthLink Business SIP Trunking ININ IC3 IP PBX Customer Configuration Guide Publication History First Release: Version 1.0 August 30, 2011 CHANGE HISTORY Version Date Change Details Changed By 1.0 8/30/2011
IP Telephony Technology IEEE ComSoc Meeting Corey Coffin, SE Cisco Systems 1 Agenda IP Convergence Call Manager Deployment Models Case Studies Summary 2 Toll Bypass and IP Telephony PBX PBX Router/GW IP
Introduction SIP is fast becoming the Voice over IP protocol of choice. During this 3-day course delegates will examine SIP technology and architecture and learn how a functioning VoIP service can be established.
IP Telephony Management How Cisco IT Manages Global IP Telephony A Cisco on Cisco Case Study: Inside Cisco IT 1 Overview Challenge Design, implement, and maintain a highly available, reliable, and resilient
5. DEPLOYMENT ISSUES Having described the fundamentals of VoIP and underlying IP infrastructure, let s address deployment issues. 5.1 LEGACY INTEGRATION In most cases, enterprises own legacy PBX systems,
Cisco Unified Communications Manager 7.0 Cisco Unified Communications Solutions unify voice, video, data, and mobile applications on fixed and mobile networks, enabling easy collaboration every time from
Packetized Telephony Networks Benefits of Packet Telephony Networks Traditionally, the potential savings on long-distance costs was the driving force behind the migration to converged voice and data networks.
Extend the Life of Your Legacy PBX while Benefiting from SIP Trunks December 5, 2013 Agenda About Sangoma VoIP Gateways Defined Sangoma Gateway Features Gateways Product Specifications Business Applications
ehealth and VoIP Overview Voice over IP (VoIP) configurations can be very complex. Your network could contain a variety of devices, applications, and configuration capabilities to support voice traffic.
Course Title: Prerequisites: Training Program (5 months) IP Implementation in Private Branch Exchanges Must fresh graduates Communication/Electronics Engineers" 1- Soft Skills Training (4 weeks) 1. Communication
Implementing Cisco IP Telephony & Video, Part 1 CIPTV1 v1.0; 5 Days; Instructor-led Course Description Implementing Cisco IP Telephony & Video, Part 1 (CIPTV1) v1.0 is a five-day course that prepares the
KEY VOIP TERMS 1 ACD: Automatic Call Distribution is a system used to determine how incoming calls are routed. When the ACD system receives an incoming call it follows user-defined specifications as to
Cisco Multiservice IP-to-IP Gateway the Cisco IOS Session Border Controller Cisco Unified Communications is a comprehensive IP communications system of voice, video, data, and mobility products and applications.
CHAPTER 1 Introducing Cisco Hosted Unified Communications Services This chapter provides a high-level overview of the architecture and components of Cisco Hosted UCS, Release 7.1(a), describes applications
EarthLink Business SIP Trunking Cisco Call Manager and Cisco CUBE Customer Configuration Guide Publication History First Release: Version 2.0 April 20, 2012 CHANGE HISTORY Version Date Change Details Changed
Cisco Unified Video Conferencing Configuration This topic provides a reference configuration for Cisco Unified Video Conferencing within a Cisco Unified Communications deployment. The information is based
EarthLink Business SIP Trunking NEC SV8100 IP PBX Customer Configuration Guide Publication History First Release: Version 1.0 August 30, 2011 CHANGE HISTORY Version Date Change Details Changed By 1.0 8/30/2011
Voice over IP Fundamentals Duration: 5 Days Course Code: GK3277 Overview: The aim of this course is for delegates to gain essential data networking and Voice over IP (VoIP) knowledge in a single, week-long
CHAPTER 4 Call Control Protocols and IPv6 in IP Video Solutions Revised: March 30, 2012, Protocols provide a complete set of specifications and suite of standards for communications between devices, This
AP200 VoIP Gateway Series Design Features & Concept 2002. 3.5 AddPac R&D Center Contents Design Features Design Specifications AP200 Series QoS Features AP200 Series PSTN Backup Features AP200 Series Easy
VoIP and IP Telephony Reach Out and Ping Someone ISAC Spring School 2006 21 March 2006 Anthony Kava, Sr. Network Admin Pottawattamie County IT Definition VoIP Voice over Internet Protocol Voice Transport
EarthLink Business SIP Trunking NEC SV8300 IP PBX Customer Configuration Guide Publication History First Release: Version 1.0 May 18, 2012 CHANGE HISTORY Version Date Change Details Changed By 1.0 5/18/2012
Enterprise Edge Communications Manager Data Capabilities Data Module Objectives After the completion of this module you will be able to describe the following Data components of the Enterprise Edge Communications
VOICE over IP H.323 Advanced Computer Network SS2005 Presenter : Vu Thi Anh Nguyet 1 Outlines 1. Introduction 2. QoS in VoIP 3. H323 4. Signalling in VoIP 5. Conclusions 2 1. Introduction to VoIP Voice
Configuration Guide for Cisco Unified Communications Environments CA Unified Communications Monitor Version 3.7 This Documentation, which includes embedded help systems and electronically distributed materials,
SIP Trunking and Voice over IP Agenda What is SIP Trunking? SIP Signaling How is Voice encoded and transported? What are the Voice over IP Impairments? How is Voice Quality measured? VoIP Technology Confidential
Allstream Converged IP Telephony SIP Trunking Solution An Allstream White Paper 1 Table of contents Introduction 1 Traditional trunking: a quick overview 1 SIP trunking: a quick overview 1 Why SIP trunking?
SIP A Technology Deep Dive Anshu Prasad Product Line Manager, Mitel June 2010 Laith Zalzalah Director, Mitel NetSolutions What is SIP? Session Initiation Protocol (SIP) is a signaling protocol for establishing
Convergence Technologies Professional (CTP) Course 1: Data Networking The Data Networking course teaches you the fundamentals of networking. Through hands-on training, you will learn the vendor-independent
ETM System SIP Trunk Support Technical Discussion Release 6.0 A product brief from SecureLogix Corporation Rev C SIP Trunk Support in the ETM System v6.0 Introduction Today s voice networks are rife with