Operation Manual Voice Overview (Voice Volume) Table of Contents

Size: px
Start display at page:

Download "Operation Manual Voice Overview (Voice Volume) Table of Contents"

Transcription

1 Operation Manual Voice Over (Voice Volume) Table of Contents Table of Contents Chapter 1 Voice Over Introduction to VoIP VoIP System Basic VoIP Call Flow VoIP Features Voice Function Configuration Configuration Procedure Voice Subscriber Line Voice Entity Voice Protocols Dial Plan Command View i

2 Operation Manual Voice Over (Voice Volume) Chapter 1 Voice Over Chapter 1 Voice Over 1.1 Introduction to VoIP VoIP makes it possible that voice services such as plain old telephone service (POTS) can be carried over the IP network. VoIP is implemented via voice packets. In VoIP, the voice gateway segments the voice signal into frames and stores them in voice packets to transmit. Currently, interworking between PSTN and Internet is implemented via VoIP gateways. Meanwhile, the PC-to-telephone, telephone-to-pc, and telephone-to-telephone technologies are mature and the call quality has been improved greatly. Therefore, VoIP can completely meet the commercial requirements. H.323 and session initiation protocol (SIP) are two common protocols used in VoIP. For details about H.323 and SIP, refer to section Voice Protocols VoIP System For POTS, all functions from the call originator to the call receiver are implemented by the public switched telephone network (PSTN). VoIP is different from POTS. Gateway IP Gateway PSTN PSTN Telephone GK server /SIP server Telephone Figure 1-1 VoIP system In Figure 1-1, the VoIP gateway provides interfaces for communication between the IP network and PSTN/integrated services digital network (ISDN), users connect to the originating VoIP gateway through PSTN, the originating VoIP gateway converts analog signals into digital signals and compresses them into voice packets that can be transmitted over the IP network, and the IP network transmits the voice packets to the terminating VoIP gateway, which reduces the voice packets to recognizable analog signals and transmits them to the receiver. This is a complete telephone-to-telephone communication process. In practice, a gatekeeper (GK) server or SIP server may be applied in the VoIP system to implement the functions of routing and access control. 1-1

3 Operation Manual Voice Over (Voice Volume) Chapter 1 Voice Over Basic VoIP Call Flow The following describes a basic VoIP call flow: 1) A user picks up a telephone and the modular voice card detects the user s off-hook action in real time 2) The modular voice card transmits the off-hook signal to the VoIP signal processing module on the VoIP gateway. 3) The VoIP signal processing module generates a dial tone. 4) The user hears the dial tone played by the session application and begins dialing before the dial tone expires. 5) The session application collects the digits dialed by the user. 6) The session application compares the collected digits with the match template in real time during digit collection. 7) After finding a match template for the called number, the originating VoIP gateway maps the number to the terminating VoIP gateway. 8) The originating VoIP gateway initiates a VoIP call to the terminating VoIP gateway over the IP network and establishes a logical channel for the call to send and receive voice data. 9) The terminating VoIP gateway receives the call from the IP network and seeks the destination telephone according to the match template. If the call is to be processed by a private branch exchange (PBX), the terminating VoIP gateway passes the call via PSTN signaling to the PBX for processing until the destination telephone is connected. When the calling party or the called party hangs up, the conversation ends. Note: During call connection, the calling party and called party negotiate the encoding/decoding method for the call and voice data is transferred through real time protocol (RTP) The RTP voice channel is used to transfer prompt signals during call connection and other signals suitable for in-band transmission across the IP network. When either party hangs up, the session application will end the conversation VoIP Features Silence compression The voice traffic to be transmitted can be reduced by automatically detecting the time ranges of silence in a conversation and stopping generating voice traffic within these time ranges. 1-2

4 Operation Manual Voice Over (Voice Volume) Chapter 1 Voice Over Comfortable noise Silent gaps during a call can be filled by comfortable background noise. QoS As the voice service is highly time-sensitive, the priority transmission of voice packets must be guaranteed. Some measures, such as priority queuing (PQ), custom queuing (CQ), weighted fair queuing (WFQ), class-based queuing (CBQ), and real time protocol (RTP) can be adopted on the sender side for this purpose. To ensure an adequate bandwidth for voice transmission, you can adopt the committed access rate (CAR) mechanism to implement traffic classification and policing. Fax over IP On basis of VoIP, the fax over IP (FoIP) system is responsible for setup of fax channels and the receiving and sending of fax data. FoIP implementation involves modulation and demodulation, fax protocol processing, and IP channel maintenance. One-stage dialing and two-stage dialing One-stage dialing and two-stage dialing can well fit in with the situation where there are differences when various PBXs transmit called numbers to the VoIP gateway. If a PBX sends the called number to the VoIP gateway, the VoIP adopts the one-stage dialing to connect the calling user. If the PBX does not send the called number to the VoIP gateway, the VoIP gateway adopts the two-stage dialing and plays the prompt tone, instructing the calling user to enter information such as called number. Automatic busy tone detection Different PBXs are likely to play different busy tones with different frequency spectra. Therefore, it is hard to recognize a busy tone feature according to a fixed threshold. With the smart busy tone identification technology, the VoIP gateway can sample, calculate, and analyze the busy tone played by the PBX to obtain a set of parameters reflecting a busy tone feature to the greatest extent. The busy tone detection can be implemented by configuring these parameters on interfaces. 1.2 Voice Function Configuration As the voice functions and objects vary, the voice function configuration includes four parts: voice subscriber line configuration, voice entity configuration, voice protocol configuration, and dial plan configuration, as shown in Figure 1-2. Each part implements a type of function. The functions implemented by all these parts are related. 1-3

5 Operation Manual Voice Over (Voice Volume) Chapter 1 Voice Over Voice function configuration Voice subscriber line Voice entity Voice protocol Dial plan Figure 1-2 Voice function configuration Configuration Procedure Figure 1-3 shows the voice function configuration procedure. For details, see Table 1-1. Start a link connection Is the link available? No Yes Is number substitution necessary? No Yes number substitution for dial plans voice entity voice subscriber line number application for dial plans voice protocol Troubleshoot No Is the call established? Yes End Figure 1-3 Voice function configuration procedure 1-4

6 Operation Manual Voice Over (Voice Volume) Chapter 1 Voice Over Table 1-1 Description of the voice function configuration procedure Step Operation Reference Connect the physical devices according to the network diagram. links and routes and ensure that the links and routes are available. Determine whether number substitution is necessary. If yes, configure number substitution for the dial plan. If no, proceed with the steps below. Access Volume and IP Routing Volume Dial Plan in Voice Volume. 4 POTS entity and VoIP entity VoIP in Voice Volume related voice subscriber lines for voice entities The physical characteristics of voice subscriber lines are usually set to the defaults. number substitution for the dial plan adopted in the network diagram. the following voice protocols according to the service and networking environment. H.323 protocol SIP protocol Fax protocol Check whether the network requirements can be met. If yes, the configuration is completed. If no, check and remove the fault and perform re-configuration. VoIP and E1&T1 in Voice Volume Dial Plan in Voice Volume H.323, SIP, and FoIP in Voice Volume Voice Subscriber Line Voice subscriber line configuration is to implement the functions of the voice subscriber line. Voice subscriber lines, which are connected to telephone network devices such as analog telephone and PBX, implement all physical layer functions between VoIP gateways and PSTN devices. These functions include power supply to analog telephones, off-hook state detection, ringing signal generation, receiving & sending of analog or digital voice calls, and receiving & sending of dialed digits for call routing. For the voice subscriber line configuration, refer to VoIP and E1&T1 in Voice Volume. The router provides the following voice subscriber lines: 1-5

7 Operation Manual Voice Over (Voice Volume) Chapter 1 Voice Over Foreign exchange station (FXS) analog voice subscriber line, namely, FXS interface. FXS interfaces are usually connected foreign exchange office (FXO) subscriber line terminals, for example, an ordinary analog telephone, to provide ringing current, ringing voltage, and dial tone. FXO analog voice subscriber line, namely, FXO interface or 2-port loop trunk interface. FXO interfaces are usually connected to analog telephone interfaces of PSTN central offices (PBXs). Ear & mouth or receive & transmit (E&M) analog voice subscriber line, namely, E&M interface. E&M interfaces support analog E&M signaling and divide each voice connection into trunk circuit side and signaling unit side (similar to the relationship between DCE and DTE). PBXs send signals to routers via M lines and receive signals from routers via E lines. Digital E1/T1 voice subscriber line, namely, timeslot (TS) group created on an E1/T1 interface. After TS groups and signaling types (for example, R2 signaling, digital E&M signaling, or digital LGS signaling) are configured on E1 voice interface cards, the system will automatically generate the corresponding voice subscriber lines for the TS groups. E1/T1 interfaces support R2, DSS1, QSIG, and digital E&M signaling Voice Entity Voice entity configuration is to implement the functions of the voice entity. For the configuration of the voice entity, refer to VoIP in Voice Volume. There are two kinds of voice entities: plain old telephone service (POTS) entity and VoIP entity. The POTS entity corresponds to the local telephone or PSTN. POTS entity configuration is to associate a voice subscriber line on the VoIP gateway with a local telephone. The POTS entity configuration implements the binding between telephone numbers and voice subscriber lines. The VoIP entity relates a call entity with a routing policy. Compared with the POTS entity, the VoIP entity corresponds to the IP network. VoIP configuration implements the binding between telephone numbers and destination addresses (IP addresses or server addresses) Voice Protocols The VoIP gateway can transfer voice or fax over the IP network by using different protocols. The basic voice protocols that the router supports are H.323 and session initiation protocol (SIP), and the fax protocol is T.38. 1) H.323 H.323 is a standard protocol established by ITU-T. The H.323 protocol stack, implemented at the application layer, mainly describes terminals, devices, and services used for multimedia communication without QoS guarantee over the IP network. An 1-6

8 Operation Manual Voice Over (Voice Volume) Chapter 1 Voice Over H.323 network usually consists of VoIP gateway, gatekeeper (optional), multipoint control unit (MCU), and terminals. According to the ITU-T specifications, the gatekeeper (GK) should provide H.323 terminals, gateway, or MCU in LANs or WANs with the following functions: Address translation Access permission Bandwidth control and management Area management and security check Call control signaling and call management Routing control and accounting The GK not only controls the call service, but also functions as the central control point within its management area. The GK implements the control function by exchanging information with the VoIP gateway. If there is any GK, the router will be under the control of the GK. To implement the control function of the GK, you need to perform related configurations on the router. For detailed configurations, refer to H.323 in Voice Volume. 2) SIP SIP is the core protocol of the IETF multimedia data and control architecture and is used for signaling control and communication with a softswitch platform in the IP network. A SIP network consists of user agent (namely, SIP endpoint), proxy server, registration server, location server, and redirect server. Here, the proxy server, registration server, location server, and redirect server are only functional entities. In practice, multiple functional entities may be integrated into one physical entity. In a complete SIP system, all SIP endpoints serve as user agents and should register with the registration server to inform of their locations, session capabilities, and call policies. The registration server sends the registration information to the location server for storage. SIP endpoints can use the proxy server to set up calls. SIP endpoints send signaling messages to the proxy server and then the proxy server forwards them to the next hop. In this process, multiple proxy servers may be involved. Eventually, channels are established to transfer the upper layer voice service. Unlike the proxy server, the SIP redirect server will not forward the received session request messages, but inform the originating SIP endpoints of the addresses of the terminating SIP endpoints by returning reply messages. The originating SIP endpoints directly re-originate a session request message to the terminating SIP endpoints. The terminating SIP endpoints also directly return a reply message to the originating SIP endpoints. As a SIP endpoint, the voice router needs to exchange information with the servers to accomplish the functions such as registration. For detailed configurations, refer to SIP in Voice Volume. 1-7

9 Operation Manual Voice Over (Voice Volume) Chapter 1 Voice Over 3) Fax protocol Fax over IP (FoIP) complies with ITU-T T.30 and T.4 on PSTN and T.38 on the IP network. T.30 defines the procedures necessary for document transmission between facsimile terminals on PSTN. It gives detailed descriptions and stipulations on the communication process, signal format, control signaling, and error correction of Group 3 facsimile terminals on the general switched telephone network. T.4 is a standard protocol used for document transmission between Group 3 facsimile terminals. It standardizes image coding, signaling modulation, rate, transmission time, error correction, and document transmission of Group 3 facsimile terminals. T.38 describes the technical features necessary to transfer facsimile document in realtime between Group 3 facsimile terminals over the Internet or other networks by using IP protocols. It gives descriptions and stipulations on communication mode, message format, error correction, and part of communication processes. Before applying the fax service, you need to configure the technical protocols and physical characteristics. For detailed configurations, refer to FoIP in Voice Volume Dial Plan Dial plan configuration is to provide diversified number management functions. Dial plan configuration involves number substitution and number application. Number substitution means establishing some substitution rules and applying them to calling and called numbers. Number substitution includes number substitution rules and binding of number substitution rules. Number application means matching numbers, controlling the sending of numbers, and selecting voice entities according to match templates. Number application includes number match policy, rules in the match order for voice entity selection, maximum-call-connection set, and number sending mode. The dial plan configuration directly affects the selection of voice entity and the final call connection. The dial plan configuration involves global configuration, voice subscriber line configuration, and voice entity configuration. You can select one or more configurations for a dial plan. The global configuration acts on calls of the whole VoIP gateway, the voice entity configuration on those of the voice entity, and the voice subscriber line configuration on those of the voice subscriber line. For detailed configurations, refer to Dial Plan in Voice Volume Command View The voice subscriber line configuration, voice entity configuration, voice protocol configuration, and dial plan configuration are implemented via command lines. Command s are the command line interfaces of the voice router. Most voice 1-8

10 Operation Manual Voice Over (Voice Volume) Chapter 1 Voice Over functions are implemented in corresponding. The command s of the voice router are arranged in a hierarchical structure. You can enter different function s under system, and sub-function (s) under function s. Figure 1-4 shows the command structure of the voice router. System Ethernet interface E1/T1 interface Loopback interface Dial interface Voice subscriber line Voice dial program CAS Voice entity Voice Voice GK client Voice number substitution Voice SIP client Figure 1-4 Hierarchical command structure of the voice router Table 1-2 Basic functions of voice command s View name Function Prompt Command to enter Command to System system parameters [system] Log in to the system logout. Disconnect telnet connection with the VoIP gateway. Ethernet interface Ethernet interface parameters [system-ethe rnet1/1] Key in interface ethernet 1/1 in any Return to system Loopback interface loopback interface parameters [system-loop Back1] Key in interface loopback 1 in any Return to system E1/T1 interface Create a TS group [system-e1 1/0] Key in controller e1 1/0 in system Return to system CAS signaling parameters [system-cas 1/0:5] Key in cas in E1/T1 interface Return to E1/T1 interface Voice global voice parameters e] Key in voice-setup in system Return to system 1-9

11 Operation Manual Voice Over (Voice Volume) Chapter 1 Voice Over View name Function Prompt Command to enter Command to Voice dial program dial plan e-dial] Key in dial-program in voice Return to voice Voice subscriber line voice subscriber line e-line3/0] Key in subscriber-line 3/0 in voice Return to voice Voice entity voice entity e-dial-entity1] Key in entity 1 pots or entity 1 voip in voice dial program Return to voice dial program Voice GK client H.323 e-gk] Key in gk-client in voice Return to voice Voice SIP client SIP e-sip] Key in sip in voice Return to voice 1-10

Peer-to-Peer SIP Mode with FXS and FXO Gateways

Peer-to-Peer SIP Mode with FXS and FXO Gateways 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

More information

Understanding Voice over IP

Understanding Voice over IP Introduction Understanding Voice over IP For years, many different data networking protocols have existed, but now, data communications has firmly found its home in the form of IP, the Internet Protocol.

More information

Troubleshooting Voice Over IP with WireShark

Troubleshooting Voice Over IP with WireShark 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

More information

Introduction to VoIP Technology

Introduction to VoIP Technology Lesson 1 Abstract Introduction to VoIP Technology 2012. 01. 06. This first lesson of contains the basic knowledge about the terms and processes concerning the Voice over IP technology. The main goal of

More information

Gateways and Their Roles

Gateways and Their Roles 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

More information

Need for Signaling and Call Control

Need for Signaling and Call Control Need for Signaling and Call Control VoIP Signaling In a traditional voice network, call establishment, progress, and termination are managed by interpreting and propagating signals. Transporting voice

More information

CVOICE Exam Topics Cisco Voice over IP Exam #642-432 7/14/2005

CVOICE Exam Topics Cisco Voice over IP Exam #642-432 7/14/2005 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

More information

Network Connection Considerations for Microsoft Response Point 1.0 Service Pack 2

Network Connection Considerations for Microsoft Response Point 1.0 Service Pack 2 Network Connection Considerations for Microsoft Response Point 1.0 Service Pack 2 Updated: February 2009 Microsoft Response Point is a small-business phone solution that is designed to be easy to use and

More information

Overview of Voice Over Internet Protocol

Overview of Voice Over Internet Protocol 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

More information

2- Technical Training (9 weeks) 3- Applied Project (3 weeks) 4- On Job Training (OJT) (4 weeks)

2- Technical Training (9 weeks) 3- Applied Project (3 weeks) 4- On Job Training (OJT) (4 weeks) 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

More information

Configuring the Sonus SBC 2000 with Cisco Unified Call Manager 10.5 for Verizon Deployment

Configuring the Sonus SBC 2000 with Cisco Unified Call Manager 10.5 for Verizon Deployment Configuring the Sonus SBC 2000 with Cisco Unified Call Manager 10.5 for Verizon Deployment Application Notes Rev 1.0 P/N 550-06690 Last Updated: October 26, 2015 Revision History Revision Date Revised

More information

Mediatrix 3000 with Asterisk June 22, 2011

Mediatrix 3000 with Asterisk June 22, 2011 Mediatrix 3000 with Asterisk June 22, 2011 Proprietary 2011 Media5 Corporation Table of Contents Introduction... 3 Network Topology... 3 Equipment Detail... 3 Configuration of the Fax Extension... 4 Configuration

More information

VoIP QoS. Version 1.0. September 4, 2006. AdvancedVoIP.com. sales@advancedvoip.com support@advancedvoip.com. Phone: +1 213 341 1431

VoIP QoS. Version 1.0. September 4, 2006. AdvancedVoIP.com. sales@advancedvoip.com support@advancedvoip.com. Phone: +1 213 341 1431 VoIP QoS Version 1.0 September 4, 2006 AdvancedVoIP.com sales@advancedvoip.com support@advancedvoip.com Phone: +1 213 341 1431 Copyright AdvancedVoIP.com, 1999-2006. All Rights Reserved. No part of this

More information

VoIP H.323 Series. VoIP Gatways: VoIP 422/404/440/800 VoIP Routers: VoIP 404R/440R/200R/110R. Quick Setup Guide

VoIP H.323 Series. VoIP Gatways: VoIP 422/404/440/800 VoIP Routers: VoIP 404R/440R/200R/110R. Quick Setup Guide VoIP H.323 Series VoIP Gatways: VoIP 422/404/440/800 VoIP Routers: VoIP 404R/440R/200R/110R Quick Setup Guide Important Information In this guide, you will learn how to setup our VoIP gateway and Routers

More information

Curso de Telefonía IP para el MTC. Sesión 1 Introducción. Mg. Antonio Ocampo Zúñiga

Curso de Telefonía IP para el MTC. Sesión 1 Introducción. Mg. Antonio Ocampo Zúñiga Curso de Telefonía IP para el MTC Sesión 1 Introducción Mg. Antonio Ocampo Zúñiga Conceptos Generales VoIP Essentials Family of technologies Carries voice calls over an IP network VoIP services convert

More information

IP Implementation in Private Branch Exchanges From 9:30 a.m until 4:30 p.m (7 hrs./day) 5 days / week

IP Implementation in Private Branch Exchanges From 9:30 a.m until 4:30 p.m (7 hrs./day) 5 days / week Course Title: No. of Hours: IP Implementation in Private Branch Exchanges From 9:30 a.m until 4:30 p.m (7 hrs./day) 5 days / week 1 Course Duration: 3 Months (12weeks) No. Of Hours: 7 Hrs./Day- 5 days/week.

More information

PATTON TECH NOTES What are FXS and FXO?

PATTON TECH NOTES What are FXS and FXO? Introduction The question What is the difference between FXS and FXO? is frequently asked by those deploying Patton Voice-over-Internet Protocol (VoIP) SmartNode solutions. Foreign exchange Subscriber

More information

ehealth and VoIP Overview

ehealth and VoIP Overview 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.

More information

Version 0.1 June 2010. Xerox WorkCentre 7120 Fax over Internet Protocol (FoIP)

Version 0.1 June 2010. Xerox WorkCentre 7120 Fax over Internet Protocol (FoIP) Version 0.1 June 2010 Xerox WorkCentre 7120 Fax over Internet Protocol (FoIP) Thank you for choosing the Xerox WorkCentre 7120. Table of Contents Introduction.........................................

More information

Configuration Notes 0217

Configuration Notes 0217 PBX Remote Line Extension using Mediatrix 1104 and 1204 Introduction... 2 Application Scenario... 2 Running the Unit Manager Network (UMN) Software... 3 Configuring the Mediatrix 1104... 6 Configuring

More information

Let's take a look at another example, which is based on the following diagram:

Let's take a look at another example, which is based on the following diagram: Chapter 3 - Voice Dial Peers In order to understand the concept of dial peers, it is important to understand call legs. A voice call over a packet network is segmented into discrete call legs. A call leg

More information

GW400 VoIP Gateway. User s Guide

GW400 VoIP Gateway. User s Guide GW400 VoIP Gateway User s Guide P/N: 956YD30001 Copyright 2006. All Rights Reserved. Document Version: 1.0 All trademarks and trade names are the properties of their respective owners. i Table of Contents

More information

IP Telephony v1.0 Scope and Sequence. Cisco Networking Academy Program

IP Telephony v1.0 Scope and Sequence. Cisco Networking Academy Program 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

More information

Voice Traffic over SIP Trunks

Voice Traffic over SIP Trunks 61210916L1-29.1D July 2008 Configuration Guide This configuration guide explains the concepts behind transmitting Voice over Internet Protocol (VoIP) over Session Initiation Protocol (SIP) trunks, using

More information

Application Note - IP Trunking

Application Note - IP Trunking Application Note - IP Trunking End-to-End Configuration for IP Trunking This document gives you a detailed description of how to configure IP Trunking in a Tenor VoIP system. The following topics are included

More information

Direct IP Calls. Quick IP Call Mode

Direct IP Calls. Quick IP Call Mode Unicorn3112 Tips Direct IP Calls...1 Quick IP Call Mode...1 PSTN Pass Through...2 VoIP-to-PSTN Calls...2 PSTN-to-VoIP Calls...3 Route Calls to PSTN...4 Forward Calls to PSTN...4 Forward Calls to VoIP...4

More information

AP200 VoIP Gateway Series Design Features & Concept. 2002. 3.5 AddPac R&D Center

AP200 VoIP Gateway Series Design Features & Concept. 2002. 3.5 AddPac R&D Center 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

More information

Hands on VoIP. Content. Tel +44 (0) 845 057 0176 enquiries@protelsolutions.co.uk. Introduction

Hands on VoIP. Content. Tel +44 (0) 845 057 0176 enquiries@protelsolutions.co.uk. Introduction 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

More information

H.323 / SIP VoIP Gateway VIP GW. Quick Installation Guide

H.323 / SIP VoIP Gateway VIP GW. Quick Installation Guide H.323 / SIP VoIP Gateway VIP GW Quick Installation Guide Overview This quick installation guide describes the objectives; organization and basic installation of the PLANET VIP-281/VIP-480/VIP-880/VIP-1680/VIP-2480

More information

Indepth Voice over IP and SIP Networking Course

Indepth Voice over IP and SIP Networking Course 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.

More information

Implementing Cisco Voice Communications and QoS

Implementing Cisco Voice Communications and QoS Implementing Cisco Voice Communications and QoS Course CVOICE v8.0; 5 Days, Instructor-led Course Description Implementing Cisco Voice Communications and QoS (CVOICE) v8.0 teaches learners about voice

More information

Configuration Guide for connecting the Eircom Advantage 4800/1500/1200 PBXs to the Eircom SIP Voice platform.

Configuration Guide for connecting the Eircom Advantage 4800/1500/1200 PBXs to the Eircom SIP Voice platform. Configuration Guide for connecting the Eircom Advantage 4800/1500/1200 PBXs to the Eircom SIP Voice platform. 1 Contents Introduction.... 3 Installing the Applications Module... 4 Ordering a Licence for

More information

Configuring Voice over IP

Configuring Voice over IP CHAPTER 4 This chapter explains how to configure voice interfaces and ports, which convert telephone voice signals for transmission over an IP network. This chapter presents the following major topics:

More information

Voice over IP. Abdus Salam ICTP, February 2004 School on Digital Radio Communications for Research and Training in Developing Countries

Voice over IP. Abdus Salam ICTP, February 2004 School on Digital Radio Communications for Research and Training in Developing Countries 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)

More information

- Basic Voice over IP -

- Basic Voice over IP - 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

More information

640-460. IIUC Implementing Cisco IOS Unified Communications (IIUC) Version: Demo. Page <<1/9>>

640-460. IIUC Implementing Cisco IOS Unified Communications (IIUC) Version: Demo. Page <<1/9>> 640-460 IIUC Implementing Cisco IOS Unified Communications (IIUC) Version: Demo Page 1. You are CCNA VOICE associate in XXXX.com. You need configure a voice port that will allow the gateway to

More information

Packetized Telephony Networks

Packetized Telephony Networks 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.

More information

White paper. SIP An introduction

White paper. SIP An introduction White paper An introduction Table of contents 1 Introducing 3 2 How does it work? 3 3 Inside a normal call 4 4 DTMF sending commands in sip calls 6 5 Complex environments and higher security 6 6 Summary

More information

159.334 Computer Networks. Voice over IP (VoIP) Professor Richard Harris School of Engineering and Advanced Technology (SEAT)

159.334 Computer Networks. Voice over IP (VoIP) Professor Richard Harris School of Engineering and Advanced Technology (SEAT) 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

More information

Integrate VoIP with your existing network

Integrate VoIP with your existing network 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

More information

4. H.323 Components. VOIP, Version 1.6e T.O.P. BusinessInteractive GmbH Page 1 of 19

4. H.323 Components. VOIP, Version 1.6e T.O.P. BusinessInteractive GmbH Page 1 of 19 4. H.323 Components VOIP, Version 1.6e T.O.P. BusinessInteractive GmbH Page 1 of 19 4.1 H.323 Terminals (1/2)...3 4.1 H.323 Terminals (2/2)...4 4.1.1 The software IP phone (1/2)...5 4.1.1 The software

More information

Frequently Asked Questions about Integrated Access

Frequently Asked Questions about Integrated Access Frequently Asked Questions about Integrated Access Phone Service How are local, long distance, and international calls defined? Local access transport areas (LATAs) are geographical boundaries set by the

More information

Special-Purpose Connections

Special-Purpose Connections Special-Purpose Connections Connection Commands This topic identifies different special-purpose connection commands. Special-Purpose Connection Commands connection plar Associates a voice port directly

More information

Network Overview. Background Traditional PSTN Equipment CHAPTER

Network Overview. Background Traditional PSTN Equipment CHAPTER CHAPTER 1 Background Traditional PSTN Equipment Traditional telephone services are engineered and offered over the public switched telephone network (PSTN) via plain old telephone service (POTS) equipment

More information

LevelOne VOI-9000. H.323 VoIP Gatekeeper. User Manual

LevelOne VOI-9000. H.323 VoIP Gatekeeper. User Manual LevelOne VOI-9000 H.323 VoIP Gatekeeper User Manual Quick Guide Step 1: Broadband (ADSL/Cable Modem) Connections For VOI-9000 A. Connect VOI-9000 RJ45 LAN port to Router/ADSL as one of the following connections.

More information

Internet Telephony Terminology

Internet Telephony Terminology Internet Telephony Terminology Understanding the business phone system world can be a daunting task to a lay person who just wants a system that serves his or her business needs. The purpose of this paper

More information

Extend the Life of Your Legacy PBX while Benefiting from SIP Trunks. December 5, 2013

Extend the Life of Your Legacy PBX while Benefiting from SIP Trunks. December 5, 2013 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

More information

Implementing Cisco Voice Communications and QoS **Part of the CCNP Voice certification track**

Implementing Cisco Voice Communications and QoS **Part of the CCNP Voice certification track** Course: Duration: Price: $ 3,695.00 Learning Credits: 37 Certification: Implementing Cisco Voice Communications and QoS Implementing Cisco Voice Communications and QoS**Part of the CCNP Voice certification

More information

IMPLEMENTING CISCO VOICE COMMUNICATIONS AND QOS Volume 1

IMPLEMENTING CISCO VOICE COMMUNICATIONS AND QOS Volume 1 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

More information

Implementing Cisco IOS Telephony and Unified Communications Express (IITUCX)

Implementing Cisco IOS Telephony and Unified Communications Express (IITUCX) Implementing Cisco IOS Telephony and Unified Communications Express (IITUCX) Course Objectives Explain the benefits and components of a Cisco Unified Communications system Describe how traditional telephony

More information

Getting Started KX-TDA5480

Getting Started KX-TDA5480 4-Channel VoIP Gateway Card Getting Started KX-TDA5480 Model KX-TDA0484 Thank you for purchasing the Panasonic 4-Channel VoIP Gateway Card, KX-TDA5480/KX-TDA0484. Please read this manual carefully before

More information

Voice over IP Basics for IT Technicians

Voice over IP Basics for IT Technicians 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

More information

Contents Introduction Why Fax over IP? How Real-time Fax over IP works Implementation with MessagePlus/Open Summary. About this document

Contents Introduction Why Fax over IP? How Real-time Fax over IP works Implementation with MessagePlus/Open Summary. About this document 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

More information

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.

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. 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

More information

6.40A AudioCodes Mediant 800 MSBG

6.40A AudioCodes Mediant 800 MSBG AudioCodes Mediant 800 MSBG Page 1 of 66 6.40A AudioCodes Mediant 800 MSBG 1. Important Notes Check the SIP 3 rd Party Validation Website for current validation status. The SIP 3 rd party Validation Website

More information

Internet Telephony PBX System

Internet Telephony PBX System Internet Telephony PBX System T1/E1 Gateway With IP PBX Application Copyright PLANET Technology Corporation. All rights reserved. Case 35: With IP PBX Application Head Office E1 PABX interconnect with

More information

IMPLEMENTING CISCO IOS TELEPHONY AND UNIFIED COMMUNICATIONS EXPRESS (IITUCX)

IMPLEMENTING CISCO IOS TELEPHONY AND UNIFIED COMMUNICATIONS EXPRESS (IITUCX) Temario IMPLEMENTING CISCO IOS TELEPHONY AND UNIFIED COMMUNICATIONS EXPRESS (IITUCX) This course is designed to be the primary training for Cisco Unified Communications Manager Express and Cisco Unity

More information

Implementing Cisco IOS Telephony and Unified Communications Express (IITUCX)

Implementing Cisco IOS Telephony and Unified Communications Express (IITUCX) Implementing Cisco IOS Telephony and Unified Communications Express (IITUCX) Who should attend The primary audience for this course is as follows: Network administrators Network engineers Systems engineers

More information

(Refer Slide Time: 6:17)

(Refer Slide Time: 6:17) Digital Video and Picture Communication Prof. S. Sengupta Department of Electronics and Communication Engineering Indian Institute of Technology, Kharagpur Lecture - 39 Video Conferencing: SIP Protocol

More information

IP PBX. SD Card Slot. FXO Ports. PBX WAN port. FXO Ports LED, RED means online

IP PBX. SD Card Slot. FXO Ports. PBX WAN port. FXO Ports LED, RED means online 1 IP PBX SD Card Slot FXO Ports PBX LAN port PBX WAN port FXO Ports LED, RED means online 2 Connect the IP PBX to Your LAN Internet PSTN Router Ethernet Switch FXO Ports 3 Access the PBX s WEB GUI The

More information

EdgeMarc 4508T4/4508T4W Converged Networking Router

EdgeMarc 4508T4/4508T4W Converged Networking Router Introduction The EdgeMarc 4508T4W combines multiple voice and data features into a single, easy to use converged networking router. It includes models that have up to 4 T1 WAN interfaces or a single Ethernet

More information

802.1p An IEEE standard for providing QoS using three bits (defined in 802.1q) to allow switches to reorder packets based on priority level.

802.1p An IEEE standard for providing QoS using three bits (defined in 802.1q) to allow switches to reorder packets based on priority level. Glossary and Terms 802.1p An IEEE standard for providing QoS using three bits (defined in 802.1q) to allow switches to reorder packets based on priority level. 802.1q An IEEE standard for providing virtual

More information

Configuration Notes 283

Configuration Notes 283 Mediatrix 4400 Digital Gateway VoIP Trunking with a Legacy PBX June 21, 2011 Proprietary 2011 Media5 Corporation Table of Contents Table of Contents... 2 Introduction... 3 Mediatrix 4400 Digital Gateway

More information

640-460 - Implementing Cisco IOS Unified Communications (IIUC)

640-460 - Implementing Cisco IOS Unified Communications (IIUC) 640-460 - Implementing Cisco IOS Unified Communications (IIUC) Course Introduction Course Introduction Module 1 - Cisco Unified Communications System Introduction Cisco Unified Communications System Introduction

More information

SIP Trunking and Voice over IP

SIP Trunking and Voice over IP 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

More information

Integration of GSM Module with PC Mother Board (GSM Trunking) WHITE/Technical PAPER. Author: Srinivasa Rao Bommana (srinivasrao.bommana@wipro.

Integration of GSM Module with PC Mother Board (GSM Trunking) WHITE/Technical PAPER. Author: Srinivasa Rao Bommana (srinivasrao.bommana@wipro. (GSM Trunking) WHITE/Technical PAPER Author: Srinivasa Rao Bommana (srinivasrao.bommana@wipro.com) Table of Contents 1. ABSTRACT... 3 2. INTRODUCTION... 3 3. PROPOSED SYSTEM... 4 4. SOLUTION DESCRIPTION...

More information

Getting Started. 16-Channel VoIP Gateway Card. Model No. KX-TDA0490

Getting Started. 16-Channel VoIP Gateway Card. Model No. KX-TDA0490 16-Channel VoIP Gateway Card Getting Started Model No. KX-TDA0490 Thank you for purchasing a Panasonic 16-Channel VoIP Gateway Card. Please read this manual carefully before using this product and save

More information

Toll-bypass Long Distance Calling... 1. What Is VOIP?... 2. Immediate Cost Savings... 3. Applications... 3. Business Quality Voice...

Toll-bypass Long Distance Calling... 1. What Is VOIP?... 2. Immediate Cost Savings... 3. Applications... 3. Business Quality Voice... telephony internet access remote access modems Content Toll-bypass Long Distance Calling... 1 What Is VOIP?... 2 That Was Then... This is Now... Immediate Cost Savings... 3 Applications... 3 Office-to-office

More information

IPNext 50 NGN IP-PBX High-performance Next Generation IP-PBX Solution

IPNext 50 NGN IP-PBX High-performance Next Generation IP-PBX Solution IPNext 50 NGN IP-PBX High-performance Next Generation IP-PBX Solution IP-PBX Features www.addpac.com AddPac Technology 2008, Sales and Marketing Contents IP-PBX Features Smart Multimedia Manager VoIP Gateway

More information

Agilent Technologies Performing Pre-VoIP Network Assessments. Application Note 1402

Agilent Technologies Performing Pre-VoIP Network Assessments. Application Note 1402 Agilent Technologies Performing Pre-VoIP Network Assessments Application Note 1402 Issues with VoIP Network Performance Voice is more than just an IP network application. It is a fundamental business and

More information

IP Telephony Deployment Models

IP Telephony Deployment Models 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,

More information

Configuring a Mediatrix 500 / 600 Enterprise SIP Trunk SBC June 28, 2011

Configuring a Mediatrix 500 / 600 Enterprise SIP Trunk SBC June 28, 2011 Configuring a Mediatrix 500 / 600 Enterprise SIP Trunk SBC June 28, 2011 Proprietary 2011 Media5 Corporation Table of Contents Introduction... 3 Solution Overview... 3 Network Topology... 4 Network Configuration...

More information

CVOICE - Cisco Voice Over IP

CVOICE - Cisco Voice Over IP CVOICE - Cisco Voice Over IP Table of Contents Introduction Audience At Course Completion Prerequisites Applicable Products Program Contents Course Outline Introduction This five-day course covers the

More information

Voice over IP. Presentation Outline. Objectives

Voice over IP. Presentation Outline. Objectives 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

More information

EarthLink Business SIP Trunking. Toshiba IPedge Customer Configuration Guide

EarthLink Business SIP Trunking. Toshiba IPedge Customer Configuration Guide EarthLink Business SIP Trunking Toshiba IPedge 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

More information

Encapsulating Voice in IP Packets

Encapsulating Voice in IP Packets 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

More information

Three Network Technologies

Three Network Technologies Three Network Technologies Network The largest worldwide computer network, specialized for voice ing technique: Circuit-switching Internet The global public information infrastructure for data ing technique:

More information

Voice over IP (VoIP) Basics for IT Technicians

Voice over IP (VoIP) Basics for IT Technicians 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

More information

NetVanta 7100 Exercise Service Provider SIP Trunk

NetVanta 7100 Exercise Service Provider SIP Trunk NetVanta 7100 Exercise Service Provider SIP Trunk PSTN NetVanta 7100 FXS 0/1 x2001 SIP Eth 0/0 x2004 SIP Server 172.23.102.87 Hosted by x2003 www.voxitas.com In this exercise, you will create a SIP trunk

More information

Call Routing through Analog Voice Ports and Issues at the Analog Voice Ports Connection Points

Call Routing through Analog Voice Ports and Issues at the Analog Voice Ports Connection Points International Journal of Information & Computation Technology. ISSN 0974-2239 Volume 4, Number 14 (2014), pp. 1373-1378 International Research Publications House http://www. irphouse.com Call Routing through

More information

Table of Contents. Cisco Mapping Outbound VoIP Calls to Specific Digital Voice Ports

Table of Contents. Cisco Mapping Outbound VoIP Calls to Specific Digital Voice Ports Table of Contents Mapping Outbound VoIP Calls to Specific Digital Voice Ports...1 Introduction...1 Before You Begin...1 Conventions...1 Prerequisites...1 Components Used...1 Configure...2 Network Diagram...2

More information

Convergence Technologies Professional (CTP) Course 1: Data Networking

Convergence Technologies Professional (CTP) Course 1: Data Networking 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

More information

ADTRAN SBC and Avaya IP Office PBX SIP Trunk Interoperability

ADTRAN SBC and Avaya IP Office PBX SIP Trunk Interoperability 6AOSSG001-42B March 2014 Interoperability Guide ADTRAN SBC and Avaya IP Office PBX SIP Trunk Interoperability This guide describes an example configuration used in testing the interoperability of an ADTRAN

More information

Understanding IP Faxing (Fax over IP)

Understanding IP Faxing (Fax over IP) Understanding IP Faxing (Fax over IP) A detailed technical overview of how VoIP technology and IP Faxing (Fax over IP) are changing the way organizations utilize existing network infrastructures for voice

More information

VOICE over IP H.323 Advanced Computer Network SS2005 Presenter : Vu Thi Anh Nguyet

VOICE over IP H.323 Advanced Computer Network SS2005 Presenter : Vu Thi Anh Nguyet 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

More information

VoIP Configuration Examples

VoIP Configuration Examples APPENDIX C This section uses four different scenarios to demonstrate how to configure Voice over IP (VoIP). The actual VoIP configuration procedure depends on the topology of your voice network. The following

More information

AT&T IP Flex Reach/ IP Toll Free Configuration Guide IC 3.0 with Interaction SIP Proxy

AT&T IP Flex Reach/ IP Toll Free Configuration Guide IC 3.0 with Interaction SIP Proxy INTERACTIVE INTELLIGENCE AT&T IP Flex Reach/ IP Toll Free Configuration Guide IC 3.0 with Interaction SIP Proxy Version 1.7 9/2/2009 TABLE OF CONTENTS 1 AT&T... 5 1.1 Introduction... 5 1.2 Product Descriptions...

More information

Voice over IP Probe! for Network Operators and! Internet Service Providers

Voice over IP Probe! for Network Operators and! Internet Service Providers Voice over IP Probe! for Network Operators and! Internet Service Providers Product Presentation September 2011 2011 ADVENAGE GmbH Agenda Voice over IP Probe Key Facts VoIP Probe in a Nutshell Use Cases

More information

AudioCodes Gateway in the Lync Environment

AudioCodes Gateway in the Lync Environment AudioCodes Gateway in the Lync Environment Course: Audience: Prerequisites: Products: Four days hands-on, technical instruction covering installation, configuration, maintenance, troubleshooting and administration

More information

Nationwide WAN + VoIP connectivity

Nationwide WAN + VoIP connectivity Nationwide WAN + VoIP connectivity Client: Multi-state network of universities based in a Southern state. Customer's requirement: The customer wanted to establish WAN connectivity between the Head office

More information

SIP: Ringing Timer Support for INVITE Client Transaction

SIP: Ringing Timer Support for INVITE Client Transaction SIP: Ringing Timer Support for INVITE Client Transaction Poojan Tanna (poojan@motorola.com) Motorola India Private Limited Outer Ring Road, Bangalore, India 560 037 Abstract-The time for which the Phone

More information

Data Communications & Computer Networks. Circuit and Packet Switching

Data Communications & Computer Networks. Circuit and Packet Switching Data Communications & Computer Networks Chapter 9 Circuit and Packet Switching Fall 2008 Agenda Preface Circuit Switching Softswitching Packet Switching Home Exercises ACOE312 Circuit and packet switching

More information

How to use wired (Wireless) Phone to make off-net calls via Gateway

How to use wired (Wireless) Phone to make off-net calls via Gateway How to use wired (Wireless) Phone to make off-net calls via Gateway In the following samples, we ll introduce VIP-154T and VIP-192 makes off-net Calls (PSTN calls) via VIP-480FO applications. Installation

More information

AT&T IP Flexible Reach Service

AT&T IP Flexible Reach Service I. Service Overview II. Service Components, standard and options I. Service Overview AT&T s Business Voice over IP ( AT&T BVoIP ) portfolio of services enable the transmission of voice telephone calls

More information

Application Notes Rev. 1.0 Last Updated: February 3, 2015

Application Notes Rev. 1.0 Last Updated: February 3, 2015 SBC 1000/2000 Series Configuration Guide with Cisco Unified Call Manager v8.6 for Level 3 Voice Complete SM Deployments Application Notes Rev. 1.0 Last Updated: February 3, 2015 Contents 1 Document Overview...

More information

Application Note. Pre-Deployment and Network Readiness Assessment Is Essential. Types of VoIP Performance Problems. Contents

Application Note. Pre-Deployment and Network Readiness Assessment Is Essential. Types of VoIP Performance Problems. Contents Title Six Steps To Getting Your Network Ready For Voice Over IP Date January 2005 Overview This provides enterprise network managers with a six step methodology, including predeployment testing and network

More information

Dialogic 4000 Media Gateway Series

Dialogic 4000 Media Gateway Series The Dialogic 4000 Media Gateways (DMG4000 Gateways) are integrated systems that can provide both local PSTN and voice endpoint connectivity at a branch office premise as well as base-level voice service

More information

Optimum Business SIP Trunk Set-up Guide

Optimum Business SIP Trunk Set-up Guide Optimum Business SIP Trunk Set-up Guide For use with IP PBX only. SIPSetup 07.13 FOR USE WITH IP PBX ONLY Important: If your PBX is configured to use a PRI connection, do not use this guide. If you need

More information

Integrating VoIP Phones and IP PBX s with VidyoGateway

Integrating VoIP Phones and IP PBX s with VidyoGateway Integrating VoIP Phones and IP PBX s with VidyoGateway Updated February 2011 INDEX: I. ABSTRACT.1 II. III. IV. VIDYOGATEWAY OVERVIEW.. 1 NETWORK TOPOLOGIES AND DEFINITIONS...2 CONNECTING TO VIDYOCONFERENCES

More information

Voice over IP Fundamentals

Voice over IP Fundamentals 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

More information