IP TELEPHONY DCT2000 TEST SOLUTIONS. Focus on Digital Telecom Test

Size: px
Start display at page:

Download "IP TELEPHONY DCT2000 TEST SOLUTIONS. Focus on Digital Telecom Test"

Transcription

1 TELEPHONY DCT2000 TEST SOLUTIONS Focus on Digital Telecom Test

2 Telephony Challenges The telecommunications industry has experienced a marked proliferation of -based networks. This phenomenal growth in the sector has reshaped the future of voice traffic, encouraged the integration of multimedia and led to the evolution of Telephony. The rise of Telephony signals the migration of voice traffic from traditional circuit-switched networks to packet-switched networks. The Benefits of Telephony Telephony eliminates the cost of maintaining multiple networks each dedicated to the traffic of a single medium. By utilizing as a common transport layer, networks dedicated to voice, video, and data effectively merge into a single unified network. As legacy circuit-switched networks are integrated with emerging packet-switched networks, an open and scalable multimedia infrastructure emerges. Today s businesses employ Telephony networks to reduce costs as well as to broaden the means of communication. Communication across networks has been elevated from simple Web and access to multimedia conferencing sessions and real-time document collaboration. As a wealth of features available with circuit switched voice systems are added to future Internet Telephony products and services, international and domestic callers alike will turn to Telephony providers for an integrated solution. This integration across network environments increases reliability and interoperability while offering greater economies of scale. The Challenges of Telephony In a packet-switched network, real-time streaming data such as voice and video are compressed into packetized data and sent out over physical links to be reassembled and re-sequenced on the other end. This process results in an effective use of network bandwidth. In order to deliver high quality voice solutions, Telephony incorporates families of multimedia specifications. A major challenge is the explosion in specifications. Today s telecommunications standards bodies have numerous standing committees dedicated to delivering voice-overpacket solutions. The dominant Telephony standards are and, which have been published by the International Telecommunication Union (ITU) and the Internet Engineering Task Force (IETF) respectively. standards include the protocol suite,, and /RTCP while standards include, /RTCP, H.428/Megaco and other networking protocols. Meanwhile, manufacturers and providers attempt to rapidly produce and deploy Telephony products and services to the market even as the standards continue to evolve. To be successful, manufacturers must ensure that new products and services are able to interwork with other networks such as PSTN and Cellular networks. Therefore, Telephony products must undergo comprehensive testing to verify conformance to interworking protocols (such as -T, SIGTRAN and BICC).

3 Catapult Telephony Test Solutions Catapult has been meeting the testing needs of the telecommunications industry since With long-term expertise in building test solutions for cellular, SS7 and other networks, Catapult offers a broad range of solutions to help solve the most demanding Telephony test challenges. The Catapult DCT2000 Digital Communications Test System provides a powerful and versatile platform allowing developers to test almost any and network nodes. A Powerful Platform Leveraging the strengths of a UNIX -based workstation and the power of Catapult PowerPCI cards, the DCT2000 offers a multi-user testing environment. This accessibility enables all members of a test group to simultaneously test a given product. DCT2000 users have the option of running their Internet Telephony tests over the workstation sockets or the Catapult Power- PCI Ethernet cards. The multi-port capability of the DCT2000 allows multiple-protocol testing on a single system. As a result, the DCT2000 can simulate and monitor multiple nodes in the Telephony network simultaneously. Catapult customers can test almost any or network node, such as a, Gatekeeper, Multi-point Control Unit (MCU), Signaling, Media, Media Controller (MGC), Call Agent and Terminal Equipment. The modular architecture of the DCT2000 delivers a powerful platform that can be adapted to evolving test needs. A Versatile Platform The DCT2000 supports standards including the protocol suite, and /RTCP protocols. The DCT2000 also supports standards including, /RTCP and H.248/Megaco. Additionally, the DCT2000 supports the inter-working protocols between telephony and PSTN/Cellular networks, including -T, SIGTRAN (M3UA, M2PA, SUA,, etc.) and BICC (CS1 and CS2) protocols. Supporting over 200 protocol and variant modules, the DCT2000 provides the encoders, decoders and underlying state machines for testing Telephony, 3G (UMTS and cdma2000), GPRS, SS7, ISDN, QSIG, V5 and many more. The DCT2000 delivers an expandable and comprehensive telecom test solution that can reduce product development time and costs. DCT2000 Test Applications The DCT2000 supports a number of applications that span the product life cycle from initial development to final deployment in the field. Customers can utilize the DCT2000 to perform the following applications: Design and Feature Verification Conformance Testing Interoperability Testing Load and Stress Testing Installation and Acceptance Testing Throughout product development and various testing cycles, expert Catapult Customer Application Engineers (CAEs) provide world-class support for the DCT2000 test system. The DCT2000 is multi-user, multi-protocol, programmable test system that can reduce product development time and costs. PowerPCI cards can run multiple protocols, providing a scalable solution that is not limited by workstation performance. 1

4 Alerting Connect Telephony Network Infrastructure Internet Telephony delivers a flexible network infrastructure for transmitting voice calls across the integrated and traditional voice networks. Depending on the call scenario, various network nodes and Telephony protocols may be employed. Ca Catapult Supported Telephony Protocols * Protocol Spec Function Signaling Encoder/Decoder H.245 ITU-T Call Control H CS ITU-T Call Signaling H RAS ITU-T Registration, Admission and Status H.235 ITU-T Security and Encryption H.450 ITU-T Supplementary Services Annex E Encoder/Decoder/S.M. ITU-T Multiplexed Call Signaling Transport Encoder/Decoder IETF Media Services Encoder/Decoder IETF Session Initiation -T Encoder/Decoder IETF Session Initiation for telephone Encoder/Decoder IETF Transport functions for real-time data RTCP Encoder/Decoder IETF Control of Communication H.248 (Megaco) Encoder/Decoder IETF Control v6 & v4 Encoder/Decoder/S.M. IETF Protocol Encoder/Decoder IETF Stream Control Transmission Protocol M2PA Encoder/Decoder IETF MTP2 Peer-to-Peer Adaptation M3UA Encoder/Decoder/S.M. IETF MTP3 User Adaptation SUA Encoder/Decoder IETF SCCP User Adaptation BICC Encoder/Decoder ITU-T Bearer Independent Call Control CALL FLOW 1: To PSTN/Cellular Networks H RAS Messages Q.931 (H.225.0) Call Signalling Messages To PSTN/Cellular Networks Signaling Network Ga 3 * Contact Catapult Communications for the most current listing of supported telephony specifications. Network Key Abbrev. Network Node MCU MGC Multipoint Control Unit Media Controller TE PC Terminal Equipment Personal Computer 2

5 To PSTN/Cellular Networks ENDPOINT 1 ENDPOINT 2 GATEKEEPER Admission Request Admission Confirmed ll Proceeding Admission Request Setup Admission Confirmed To PSTN/Cellular Networks Gatekeeper Gatekeeper MCU Network CALL FLOW 2: AND - Caller Off-Hook Dialing RGW AGENT RQNT NOTIFY ACK 200 RQNT CALL AGENT Soft Switch NOTIFY ACK 200 CRCX+RQNT Media teway Ringing RQNT SIGTRAN 31 H.248 (MEGACO) Agent 33 Free Dial Tone Connection Hang Up MDCX+RQNT NOTIFY ACK 200 DLCX+RQNT resp. 200 resp. 180 resp. 200 ACK INVITE Ring Callee Off Hook Caller Heard 4 Residential Abbrev. Request Example Acknowledgment OK ACK Acknowledgement ACK info@catapult.com /2.0 BYE BYE BYE sip:info@catapult.com /2.0 CRCX Create Connection CRCX 1204 person@rgw-1.catapult.jp.co 1.0 DLCX Delete Connection DLCX 1210 person@rgw-1.catapult.jp.co 1.0 Invite Invitation INVITE sip:info@catapult.com /2.0 MDCX Modify Connection MDCX 1209 person@rgw-1.catapult.jp.co 1.0 NOTIFY Notify Call Agent NTFY 2005 person@rgw-1.catapult.jp.co 1.0 RQNT Request Notification RQNT 1206 person@rgw-1.catapult.jp.co 1.0 resp.xxx Acceptation Message / OK BYE MGC MGC SIGTRAN -T -Based Database (e.g. HLR) 35 BICC 33 Free Dial Tone Residential Network H RASH Call Signaling H.245 Control Signaling UDP TCP RTCP /-T UDP TCP UDP H.248 UDP H.248/MEGACO /-T UDP RANAP RNSAP TCAP ISUP SCCP M3UA M3UA TUP ISDN QSIG TCAP RANAP RNSAP IUA SUA IUA SUA SIGTRAN RANAP RNSAP TCAP ISUP SCCP TUP MTP3 M2PA M2UA M3UA VUA VUA MTP3 MTP2 MTP1 STCMTP, MTP-3b & M3UA MTP-3b M2PA SSCF SSCOP AAL5 ATM BICC STCSSCOP M3UA SSCOP SSCOPMCE BICC AAL5 ATM STC

6 DCT2000 Telephony Test Applications In addition to supporting communications within and networks, Telephony also allows the inter-working between these networks and PSTN/Cellular networks. Manufacturers and service providers may demand diverse types of Telephony applications. PSTN/Cellular TE TE TE Network Origination Termination TE TE TE TE TE PSTN/Cellular Origination Termination Origination PSTN/Cellular Termination PSTN/Cellular Origination -PSTN/Cellular Termination Network PSTN/Cellular DCT2000 PBX Media SIGTRAN H.248/Megaco MGC DUT DCT2000 INTEROPERABILITY TESTING With the flexibility of the DCT2000 test platform, a manufacturer is able to perform both load and stress testing on a combined Gatekeeper/ product. DCT QSIG DCT H225RAS H245 PBX DCT H225CS QSIG Gatekeeper DUT DCT LOAD AND STRESS TESTING H225RAS Utilizing the DCT2000 to simulate both a and PBX, a manufacturer is able to successfully verify features, functionality and interoperabilty of Gatekeeper and products. Customer s Device Under Test Simulated on DCT2000 DCT2000 Application: Interoperability Testing within an Telephony Network A manufacturer wants to verify the interoperability of an and a Gatekeeper. The testing requires the devices under test to inter-work with other network devices: a PBX (Private Branch exchanges) and an. The manufacturer wants to avoid incurring the expense of procuring the external and PBX and turns to Catapult for a testing solution. To perform this interoperability testing, a single DCT2000 is used to simulate the required network devices. By simulating the PBX, the DCT2000 emulates QSIG (Q reference SIGnaling) protocol to act as a real PBX. By simulating the, the DCT2000 also emulates the protocol suite, including H CS for call setup, H RAS for call registration, admission and status report and H.245 for control signaling. Meanwhile, the DCT2000 supports (Real Time Transport Protocol) and RTCP (Real Time Transport Control Protocol) to handle the traffic. Additionally, the multi-user access capability of the DCT2000 eliminates the need to purchase multiple test systems. DCT2000 Application: Load and Stress Testing within a Telephony Network A manufacturer is developing a Media Controller (MGC). The manufacturer wants to verify the MGC s performance under heavy traffic. The simulated environment should include a Agent, a Signaling and a Media. The manufacturer decides to purchase a Catapult DCT2000 to perform the Load and Stress testing. The DCT2000 provides the simulation functions of the required network devices in a single platform. The DCT2000 generates the desired message traffic, including signaling for session establishment, H.248/Megaco media gateway control signaling and SIGTRAN (SIGnaling TRANsport) signaling for SS7 over. Under different load levels, the manufacturer can verify the performance of the MGC under test. Specifically, the DCT2000 can generate the extreme signaling load to the MGC under test to see how this MGC will respond. With the DCT2000, the manufacturer can be confident that the MGC under heavy traffic performs exactly as expected. 4

7 DCT2000 Test Implementations The DCT2000 supports test applications that span the entire product life cycle from initial development to final deployment. The flexibility of the DCT2000 permits the implementation of numerous test scenarios. During the initial stages of development, developers can use the DCT2000 to perform design and feature verification as well as conformance testing. As the product evolves, the DCT2000 can be used for interoperability testing as well as stress and load testing. In the final stages of development, the DCT2000 allows developers to perform installation and acceptance testing. Catapult Telephony Solutions Catapult delivers this expertise. Working with customers to develop innovative solutions, Catapult supports the continual evolution of the telecom industry and the dynamic progress of the Telephony market. Catapult is uniquely positioned to provide telecom test solutions that enable the products and services of tomorrow s telephony networks. Catapult Support With one of the highest industry ratios of application engineers to sales engineers, Catapult provides a global service infrastructure that delivers world-class customer support. Catapult Application Engineers (AEs) are committed to helping customers meet their telecom test challenges. Comprehensive Training Training is available to ensure the rapid and smooth installation of each Catapult test system. Training materials are tailored to the needs and experience level of the users and, during the training process, demonstration scripts specific to the customer s test environment may be generated. Class sizes are usually small to allow maximum interaction between the students and instructor. Catapult AEs provide both theoretical and practical knowledge in the telecom test arena. Customer Satisfaction Catapult is committed to providing quality support services and has the resources in place to help maximize the utility and value of the DCT2000 in almost any telecom test environment. Catapult Communications understands telephony technology and provides the world s leading telephony companies with testing tools. To find out whether the protocol stacks you require for testing are available please contact your local Catapult Communication sales office for further assistance.

8 North America Corporate Headquarters Western Region 160 S. Whisman Road Mountain View, CA tel: fax: Central and Eastern Regions HillCrest Commons Suite North Roselle Road Schaumburg, IL 60l95 tel: fax: Southern Region 4011 W. Plano Parkway, Suite 105 Plano, TX tel: fax: Canada Bell Mews Plaza 39 Robertson Road, Suite 261 Nepean, Ontario, Canada K2H 8R2 tel: fax: canada@catapult.com Japan Int s Nakano Bldg. 8F Nakano Nakano-ku, Tokyo , Japan tel: +81 (0) fax: +81 (0) japan@catapult.com YRP Venture Bldg. Room Hikarinooka Yokosuka-shi, Kanagawa , Japan tel: +81 (0) fax: +81 (0) Europe European Headquarters 1 Lansdowne Court, Bumpers Way Chippenham, England SN14 6RZ tel: +44 (0) fax: +44 (0) uk@catapult.com Central Region Talhofstrasse a D Gilching, Germany tel: +49 (0) fax: +49 (0) germany@catapult.com Southern Region Centre d Entreprises CGIA 5 & 7 Rue Marcelin Berthelot Antony Cedex, France tel: +33 (0) fax: +33 (0) france@catapult.com Nordic Region Tekniikantie Espoo, Finland tel: +358 (0) fax: +358 (0) nordic@catapult.com Catapult Communications International Limited 105 North West Business Park Ballycoolin, Dublin 15 Republic of Ireland tel: +353 (0) fax: +353 (0) ireland@catapult.com DCT2000, PowerPCI and the Catapult logo are registered trademarks of Catapult Communications Corporation. The names of other companies and products herein are trademarks or registered trademarks of their respective trademark owners Catapult Communications K JULY 2002

Chapter 10 VoIP for the Non-All-IP Mobile Networks

Chapter 10 VoIP for the Non-All-IP Mobile Networks Chapter 10 VoIP for the Non-All-IP Mobile Networks Prof. Yuh-Shyan Chen Department of Computer Science and Information Engineering National Taipei University Outline 10.1 GSM-IP: VoIP Service for GSM 256

More information

Master Kurs Rechnernetze Computer Networks IN2097

Master Kurs Rechnernetze Computer Networks IN2097 Chair for Network Architectures and Services Institute for Informatics TU München Prof. Carle, Dr. Fuhrmann Master Kurs Rechnernetze Computer Networks IN2097 Prof. Dr.-Ing. Georg Carle Dr. Thomas Fuhrmann

More information

TSIN02 - Internetworking

TSIN02 - Internetworking TSIN02 - Internetworking Lecture 9: SIP and H323 Literature: Understand the basics of SIP and it's architecture Understand H.323 and how it compares to SIP Understand MGCP (MEGACO/H.248) SIP: 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

Demystifying Protocols:

Demystifying Protocols: Demystifying Protocols: A Comparison of Protocols Suitable for IP Telephony By Tracy Venters Sonus Networks tracy.venters@sonusnet.com INTRODUCTION Often described as a "protocol soup," the dizzying array

More information

SIP : Session Initiation Protocol

SIP : Session Initiation Protocol : Session Initiation Protocol EFORT http://www.efort.com (Session Initiation Protocol) as defined in IETF RFC 3261 is a multimedia signaling protocol used for multimedia session establishment, modification

More information

Dialogic Distributed Signaling Interface Protocol Stacks

Dialogic Distributed Signaling Interface Protocol Stacks Dialogic Dialogic (DSI) support a range of Signaling System 7 (SS7) and IETF SIGTRAN specifications to provide solid building blocks for the most advanced applications. These signaling protocols have been

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

MED: Voice over IP systems

MED: Voice over IP systems www.ptt.co.uk Online course specification MED: Voice over IP systems Target audience: This online course is designed for those who will be responsible for the design or maintenance of Voice over IP (VoIP)

More information

VoIP. Overview. Jakob Aleksander Libak jakobal@ifi.uio.no. Introduction Pros and cons Protocols Services Conclusion

VoIP. Overview. Jakob Aleksander Libak jakobal@ifi.uio.no. Introduction Pros and cons Protocols Services Conclusion VoIP Jakob Aleksander Libak jakobal@ifi.uio.no 1 Overview Introduction Pros and cons Protocols Services Conclusion 2 1 Introduction Voice over IP is routing of voice conversations over the internet or

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

How To Interwork On An Ip Network

How To Interwork On An Ip Network An Overview of - Interworking 2001 RADVISION. All intellectual property rights in this publication are owned by RADVision Ltd. and are protected by United States copyright laws, other applicable copyright

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

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

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

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

Course 4: IP Telephony and VoIP

Course 4: IP Telephony and 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

More information

Understanding Voice over IP Protocols

Understanding Voice over IP Protocols 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

More information

VIDEOCONFERENCING. Video class

VIDEOCONFERENCING. Video class VIDEOCONFERENCING Video class Introduction What is videoconferencing? Real time voice and video communications among multiple participants The past Channelized, Expensive H.320 suite and earlier schemes

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

SS7 ISUP and Converged Networks

SS7 ISUP and Converged Networks SS7 ISUP and Converged Networks Our Spectra2 SS7 ISUP support provides VoIP users with an integrated platform for testing and analysis of converged networks. Spectra2 can monitor, test, and generate SS7

More information

Session Initiation Protocol (SIP) The Emerging System in IP Telephony

Session Initiation Protocol (SIP) The Emerging System in IP Telephony Session Initiation Protocol (SIP) The Emerging System in IP Telephony Introduction Session Initiation Protocol (SIP) is an application layer control protocol that can establish, modify and terminate multimedia

More information

Combining Voice over IP with Policy-Based Quality of Service

Combining Voice over IP with Policy-Based Quality of Service TechBrief Extreme Networks Introduction Combining Voice over IP with Policy-Based Quality of Service Businesses have traditionally maintained separate voice and data networks. A key reason for this is

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

An Introduction to VoIP Protocols

An Introduction to VoIP Protocols 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

More information

INTELLIGENT NETWORK SERVICES MIGRATION MORE VALUE FOR THE

INTELLIGENT NETWORK SERVICES MIGRATION MORE VALUE FOR THE INTELLIGENT NETWORK SERVICES MIGRATION MORE VALUE FOR THE Voice over LTE SUBSCRIBER TECHNOLOGY White Paper Mobile operators have invested a lot of time and money in Intelligent Network (IN) services for

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

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

Mobile Wireless Overview

Mobile Wireless Overview Mobile Wireless Overview A fast-paced technological transition is occurring today in the world of internetworking. This transition is marked by the convergence of the telecommunications infrastructure

More information

Introduction. Channel Associated Signaling (CAS) Common Channel Signaling (CCS) Still widely deployed today Considered as old technology

Introduction. Channel Associated Signaling (CAS) Common Channel Signaling (CCS) Still widely deployed today Considered as old technology VoIP and SS7 Introduction Channel Associated Signaling (CAS) Still widely deployed today Considered as old technology Common Channel Signaling (CCS) Separation of signaling and call paths Signaling System

More information

PacketizerTM. Overview of H.323 http://www.packetizer.com/voip/h323/papers/ Paul E. Jones. Rapporteur, ITU-T Q2/SG16 paulej@packetizer.

PacketizerTM. Overview of H.323 http://www.packetizer.com/voip/h323/papers/ Paul E. Jones. Rapporteur, ITU-T Q2/SG16 paulej@packetizer. A resource for packet-switched conversational protocols Overview of H.323 http:///voip/h323/papers/ Paul E. Jones Rapporteur, ITU-T Q2/SG16 paulej@packetizer.com June 2004 Copyright 2004 Executive Summary

More information

VoIP Technology Overview. Ai-Chun Pang Grad. Ins. of Networking and Multimedia Dept. of Comp. Sci. and Info. Engr. National Taiwan University

VoIP Technology Overview. Ai-Chun Pang Grad. Ins. of Networking and Multimedia Dept. of Comp. Sci. and Info. Engr. National Taiwan University VoIP Technology Overview Ai-Chun Pang Grad. Ins. of Networking and Multimedia Dept. of Comp. Sci. and Info. Engr. National Taiwan University Outline RTP (Real-Time Transport Protocol)/RTCP (RTP Control

More information

Receiving the IP packets Decoding of the packets Digital-to-analog conversion which reproduces the original voice stream

Receiving the IP packets Decoding of the packets Digital-to-analog conversion which reproduces the original voice stream 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

More information

Operation Manual Voice Overview (Voice Volume) Table of Contents

Operation Manual Voice Overview (Voice Volume) Table of Contents Operation Manual Voice Over (Voice Volume) Table of Contents Table of Contents Chapter 1 Voice Over... 1-1 1.1 Introduction to VoIP... 1-1 1.1.1 VoIP System... 1-1 1.1.2 Basic VoIP Call Flow... 1-2 1.1.3

More information

Voice over IP (VoIP) Overview. Introduction. David Feiner ACN 2004. Introduction VoIP & QoS H.323 SIP Comparison of H.323 and SIP Examples

Voice over IP (VoIP) Overview. Introduction. David Feiner ACN 2004. Introduction VoIP & QoS H.323 SIP Comparison of H.323 and SIP Examples Voice over IP (VoIP) David Feiner ACN 2004 Overview Introduction VoIP & QoS H.323 SIP Comparison of H.323 and SIP Examples Introduction Voice Calls are transmitted over Packet Switched Network instead

More information

1. Public Switched Telephone Networks vs. Internet Protocol Networks

1. Public Switched Telephone Networks vs. Internet Protocol Networks Internet Protocol (IP)/Intelligent Network (IN) Integration Tutorial Definition Internet telephony switches enable voice calls between the public switched telephone network (PSTN) and Internet protocol

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

ARCHITECTURES TO SUPPORT PSTN SIP VOIP INTERCONNECTION

ARCHITECTURES TO SUPPORT PSTN SIP VOIP INTERCONNECTION ARCHITECTURES TO SUPPORT PSTN SIP VOIP INTERCONNECTION 10 April 2009 Gömbös Attila, Horváth Géza About SIP-to-PSTN connectivity 2 Providing a voice over IP solution that will scale to PSTN call volumes,

More information

Implementing a Voice Over Internet (Voip) Telephony using SIP. Final Project report Presented by: Md. Manzoor Murshed

Implementing a Voice Over Internet (Voip) Telephony using SIP. Final Project report Presented by: Md. Manzoor Murshed Implementing a Voice Over Internet (Voip) Telephony using SIP Final Project report Presented by: Md. Manzoor Murshed Objectives Voice Over IP SIP H.323 MGCP Simulation using Westplan Conclusion 5/4/2006

More information

A seminar on Internet Telephony

A seminar on Internet Telephony A seminar on Internet Telephony Presented by: Nitin Prakash Sharma M. Tech. I.T IIT Kharagpur Internet Telephony 1 Contents Introduction H.323 standard Classes of connections and billing Requirements for

More information

Voice over IP. Raj Jain. The Ohio State University Columbus, OH 43210 Jain@cse.ohio-State.Edu http://www.cse.ohio-state.edu/~jain/ Raj Jain

Voice over IP. Raj Jain. The Ohio State University Columbus, OH 43210 Jain@cse.ohio-State.Edu http://www.cse.ohio-state.edu/~jain/ Raj Jain Voice over IP IP The Ohio State University Columbus, OH 43210 Jain@cse.ohio-State.Edu http://www.cse.ohio-state.edu/~jain/ 1 Overview Sample Products and Services 13 Technical Issues 4 Other Issues H.323

More information

00:00:40 00:01:00 00:01:20 00:01:40 00:02:00

00:00:40 00:01:00 00:01:20 00:01:40 00:02:00 2 1 2 1 ::4 :1: :1:2 :1:4 :2: - 3-4 - 2 2-6 1 1-7 ::4 :1: :1:2 :1:4 :2: - 8-9 2 2 3 3 - Performs Comprehensive Voice, VoIP, and PSTN Testing Simulate VoIP, TDM and Analog End-Devices Perform Protocol Functionality

More information

Multimedia Communications Voice over IP

Multimedia Communications Voice over IP Multimedia Communications Voice over IP Anandi Giridharan Electrical Communication Engineering, Indian Institute of Science, Bangalore 560012, India Voice over IP (Real time protocols) Internet Telephony

More information

By Paolo Galtieri The public switched telephone network The Internet Convergence

By Paolo Galtieri The public switched telephone network The Internet Convergence By Paolo Galtieri This article provides an overview of Voice over Internet Protocol (VoIP), one of the many applications taking advantage of the enormous growth of the Internet over the last several years.

More information

VoIP Signaling and Call Control

VoIP Signaling and Call Control VoIP Signaling and Call Control Cisco Networking Academy Program 1 Need for Signaling and Call Control 2 Model for VoIP Signaling and Call Control VoIP signaling components Endpoints Common control Common

More information

Network Technologies

Network Technologies Network Technologies Telephone Networks IP Networks ATM Networks Three Network Technologies Telephone Network The largest worldwide computer network, specialized for voice ing technique: Circuit-switching

More information

Software Engineering 4C03 VoIP: The Next Telecommunication Frontier

Software Engineering 4C03 VoIP: The Next Telecommunication Frontier Software Engineering 4C03 VoIP: The Next Telecommunication Frontier Rudy Muslim 0057347 McMaster University Computing and Software Department Hamilton, Ontario Canada Introduction Voice over Internet Protocol

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

Nokia Siemens Networks his 700 Integrated signaling application solutions

Nokia Siemens Networks his 700 Integrated signaling application solutions Nokia Siemens Networks his 700 Integrated signaling application solutions Today s and the next generation networks use the Signaling System No. 7 (SS7) to exchange information within and between the networks.

More information

Efficient evolution to all-ip

Efficient evolution to all-ip Press information June 2006 Efficient evolution to all-ip The competitive landscape for operators and service providers is constantly changing. New technologies and network capabilities enable new players

More information

Internet, Part 2. 1) Session Initiating Protocol (SIP) 2) Quality of Service (QoS) support. 3) Mobility aspects (terminal vs. personal mobility)

Internet, Part 2. 1) Session Initiating Protocol (SIP) 2) Quality of Service (QoS) support. 3) Mobility aspects (terminal vs. personal mobility) Internet, Part 2 1) Session Initiating Protocol (SIP) 2) Quality of Service (QoS) support 3) Mobility aspects (terminal vs. personal mobility) 4) Mobile IP Session Initiation Protocol (SIP) SIP is a protocol

More information

Mobile VoIP: Managing, scheduling and refining voice packets to and from mobile phones

Mobile VoIP: Managing, scheduling and refining voice packets to and from mobile phones Mobile VoIP: Managing, scheduling and refining voice packets to and from mobile phones MOHAMMAD ABDUS SALAM Student ID: 01201023 TAPAN BISWAS Student ID: 01201003 \ Department of Computer Science and Engineering

More information

PARAMETERS TO BE MONITORED IN THE PROCESS OF OPERATION WHEN IMPLEMENTING NGN TECHNICAL MEANS IN PUBLIC TELECOMMUNICATION NETWORKS

PARAMETERS TO BE MONITORED IN THE PROCESS OF OPERATION WHEN IMPLEMENTING NGN TECHNICAL MEANS IN PUBLIC TELECOMMUNICATION NETWORKS Draft Recommendation Q.3902 PARAMETERS TO BE MONITORED IN THE PROCESS OF OPERATION WHEN IMPLEMENTING NGN TECHNICAL MEANS IN PUBLIC TELECOMMUNICATION NETWORKS Summary This Recommendation describes the main

More information

SIP-Based Solutions in the Contact Center: Using Dialogic Media Gateways with the Genesys Voice Platform

SIP-Based Solutions in the Contact Center: Using Dialogic Media Gateways with the Genesys Voice Platform -Based Solutions in the Contact Center: To stay competitive and keep their customers happy and loyal, companies are working hard to enhance customer service as costeffectively as possible. Contact centers

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

How To Connect Gsm To Ip On A Gsm Network On A Pnet On A Microsoft Cell Phone On A Pc Or Ip On An Ip Onc (Gsm) On A Network On An Iph (Gms) On An

How To Connect Gsm To Ip On A Gsm Network On A Pnet On A Microsoft Cell Phone On A Pc Or Ip On An Ip Onc (Gsm) On A Network On An Iph (Gms) On An THE STUDY OF THE INTERCONNECTION OF GSM MOBILE COMMUNCATION SYSTEM OVER IP BASED NETWORK+ Le-Pond Chin. Jyh-Hong Wen2, and Ting-Way Liu Department of Information Management Shih-Chien University, Taipei,

More information

Online course syllabus. MAB: Voice over IP

Online course syllabus. MAB: Voice over IP Illuminating Technology Course aim: Online course syllabus MAB: Voice over IP This course introduces the principles and operation of telephony services that operate over Internet Protocol (IP) networks

More information

An Integrated Call Agent of the Converged VoIP Network

An Integrated Call Agent of the Converged VoIP Network JOURNAL OF INFORMATION SCIENCE AND ENGINEERING 23, 787-799 (2007) An Integrated Call Agent of the Converged VoIP Network Department of Computer Science and Information Engineering National Chiao Tung University

More information

VoIP Glossary. Client (Softphone client): The software installed in the userâ s computer to make calls over the Internet.

VoIP Glossary. Client (Softphone client): The software installed in the userâ s computer to make calls over the Internet. VoIP Glossary Analog audio signals: Analog audio signals are used to transmit voice data over telephone lines. This is done by varying or modulating the frequency of sound waves to accurately reflect the

More information

Secure VoIP for optimal business communication

Secure VoIP for optimal business communication White Paper Secure VoIP for optimal business communication Learn how to create a secure environment for real-time audio, video and data communication over IP based networks. Andreas Åsander Manager, Product

More information

TECHNICAL CHALLENGES OF VoIP BYPASS

TECHNICAL CHALLENGES OF VoIP BYPASS TECHNICAL CHALLENGES OF VoIP BYPASS Presented by Monica Cultrera VP Software Development Bitek International Inc 23 rd TELELCOMMUNICATION CONFERENCE Agenda 1. Defining VoIP What is VoIP? How to establish

More information

NTS Testing Labs Voice over IP (VoIP) Product Test Lab Compatibility & Functionality Test Outline

NTS Testing Labs Voice over IP (VoIP) Product Test Lab Compatibility & Functionality Test Outline NTS Testing Labs Voice over IP (VoIP) Product Test Lab Compatibility & Functionality Test Outline Revision 1.0 NATIONAL TECHNICAL SYSTEMS The NTS Mission: Assisting our Clients in Navigating a Short Course

More information

Voice Over IP. Priscilla Oppenheimer www.priscilla.com

Voice Over IP. Priscilla Oppenheimer www.priscilla.com Voice Over IP Priscilla Oppenheimer www.priscilla.com Objectives A technical overview of the devices and protocols that enable Voice over IP (VoIP) Demo Packet8 and Skype Discuss network administrator

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

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

Voice Over Internet Protocol (VoIP) Issues and Challenges William McCrum mccrum.william@ic.gc.ca

Voice Over Internet Protocol (VoIP) Issues and Challenges William McCrum mccrum.william@ic.gc.ca Voice Over Internet Protocol (VoIP) Issues and Challenges William McCrum Phone: +1 613-990-4493 Fax: Email: +1 613-957-8845 mccrum.william@ic.gc.ca Content Network Evolution and drivers VoIP Realizations

More information

How To Make A Cell Phone Converged Into A Cell Network

How To Make A Cell Phone Converged Into A Cell Network MPLS: Enabling Fixed-Mobile Convergence Barry M. Tishgart Vice President, Managed Services 2006 11 10 SPRINT, the "Going Forward" logo, the NEXTEL name and logo and other trademarks are trademarks of Sprint

More information

Implementing SIP and H.323 Signalling as Web Services

Implementing SIP and H.323 Signalling as Web Services Implementing SIP and H.323 Signalling as Web Services Ge Zhang, Markus Hillenbrand University of Kaiserslautern, Department of Computer Science, Postfach 3049, 67653 Kaiserslautern, Germany {gezhang, hillenbr}@informatik.uni-kl.de

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

Functional Specifications Document

Functional Specifications Document Functional Specifications Document VOIP SOFT PBX Project Code: SPBX Project Advisor : Aftab Alam Project Team: Umair Ashraf 03-1853 (Team Lead) Imran Bashir 02-1658 Khadija Akram 04-0080 Submission Date:19-10-2007

More information

Mobile Packet Backbone Network Training Programs. Catalog of Course Descriptions

Mobile Packet Backbone Network Training Programs. Catalog of Course Descriptions Mobile Packet Backbone Network Training Programs Catalog of Course Descriptions Page 2 Catalog of Course Descriptions INTRODUCTION... 6 MOBILE PACKET BACKBONE NETWORK (M-PBN) R5.1 DELTA... 7 MOBILE PACKET

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

OPNET Implementation of the Megaco/H.248 Protocol: Multi-Call and Multi-Connection. Scenarios

OPNET Implementation of the Megaco/H.248 Protocol: Multi-Call and Multi-Connection. Scenarios OPNET Implementation of the Megaco/H.248 Protocol: Multi-Call and Multi-Connection Scenarios Edlic Yiu, Edwood Yiu, and Ljiljana Trajković Simon Fraser University Vancouver, British Columbia, Canada E-mail:

More information

Voice Over Internet Protocol (VoIP)

Voice Over Internet Protocol (VoIP) Voice Over Internet Protocol (VoIP) Submitted By: Amit Prakash Computer Communication Networks- II ECE 436 University of Illinois at Chicago Abstract: This paper discuses the Voice Over Internet Protocol,

More information

Integrating Voice over IP services in IPv4 and IPv6 networks

Integrating Voice over IP services in IPv4 and IPv6 networks ARTICLE Integrating Voice over IP services in IPv4 and IPv6 networks Lambros Lambrinos Dept.of Communication and Internet studies Cyprus University of Technology Limassol 3603, Cyprus lambros.lambrinos@cut.ac.cy

More information

Whitepaper: Microsoft Office Communications Server 2007 R2 and Cisco Unified Communications Manager Integration Options

Whitepaper: Microsoft Office Communications Server 2007 R2 and Cisco Unified Communications Manager Integration Options Whitepaper: Microsoft Office Communications Server 2007 R2 and Cisco Unified Communications Manager Integration Options Document Summary This document provides information on several integration scenarios

More information

ALCATEL CRC Antwerpen Fr. Wellesplein 1 B-2018 Antwerpen +32/3/240.8550; Suresh.Leroy@alcatel.be +32/3/240.7830; Guy.Reyniers@alcatel.

ALCATEL CRC Antwerpen Fr. Wellesplein 1 B-2018 Antwerpen +32/3/240.8550; Suresh.Leroy@alcatel.be +32/3/240.7830; Guy.Reyniers@alcatel. Contact: ALCATEL CRC Antwerpen Fr. Wellesplein 1 B-2018 Antwerpen +32/3/240.8550; Suresh.Leroy@alcatel.be +32/3/240.7830; Guy.Reyniers@alcatel.be Voice over (Vo) was developed at some universities to diminish

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

Topologies, Equipment, and Communications

Topologies, Equipment, and Communications 4801_CH02 Page 31 Tuesday, August 7, 2001 11:41 AM 2 Topologies, Equipment, and Communications In order for the reader to appreciate the level of complexity involved with the definition of a robust and

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

ABSTRACT. Keywords: VoIP, PSTN/IP interoperability, SIP, H.323, RTP, PBX, SDP, MGCP, Westplan. 1. INTRODUCTION

ABSTRACT. Keywords: VoIP, PSTN/IP interoperability, SIP, H.323, RTP, PBX, SDP, MGCP, Westplan. 1. INTRODUCTION Implementing a Voice Over Internet (Voip) Telephony System Md. Manzoor Murshed Final Project Report for the course CprE550: Distributed Systems and Middleware ABSTRACT This Project is to describe the architecture

More information

Towards interoperability between existing VoIP systems

Towards interoperability between existing VoIP systems Towards interoperability between existing VoIP systems Thesis for a Master of Science degree in Telematics from the University of Twente, Enschede, the Netherlands Enschede, February 26, 2008 Lianne Meppelink

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

Comparison of Voice over IP with circuit switching techniques

Comparison of Voice over IP with circuit switching techniques Comparison of Voice over IP with circuit switching techniques Author Richard Sinden Richard Sinden 1 of 9 Abstract Voice-over-IP is a growing technology. Companies are beginning to consider commercial

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

OVERVIEW OF ALL VOIP SOLUTIONS

OVERVIEW OF ALL VOIP SOLUTIONS OVERVIEW OF ALL VOIP SOLUTIONS Kovács Gábor Parnaki Zsolt Gergı 13/03/2009 TABLE OF CONTENTS Introduction Overview of VoIP protocols Standard based implementations: H.323 SIP Proprietary solutions: Skype

More information

Applied Networks & Security

Applied Networks & Security Applied Networks & Security VoIP with Critical Analysis http://condor.depaul.edu/~jkristof/it263/ John Kristoff jtk@depaul.edu IT 263 Spring 2006/2007 John Kristoff - DePaul University 1 Critical analysis

More information

IP Telephony and Network Convergence

IP Telephony and Network Convergence IP Telephony and Network Convergence Raimo.Kantola@hut.fi Rkantola/28.11.00/s38.118 1 Today corporations have separate data and voice networks Internet Corporate Network PSTN, ISDN Rkantola/28.11.00/s38.118

More information

White Paper: Voice Over IP Networks

White Paper: Voice Over IP Networks FREE FREE One One Hour Hour VoIPonline VoIPonline Seminar TM Seminar TM For additional information contact: Terry Shugart - tshugart@analogic.com http://www.analogic.com/cti TEL: 978-977-3000 FAX: 978-977-6813

More information

ISSUES IN PARALLEL DISCRETE EVENT SIMULATION FOR AN INTERNET TELEPHONY CALL SIGNALING PROTOCOL

ISSUES IN PARALLEL DISCRETE EVENT SIMULATION FOR AN INTERNET TELEPHONY CALL SIGNALING PROTOCOL ISSUES IN PARALLEL DISCRETE EVENT SIMULATION FOR AN INTERNET TELEPHONY CALL SIGNALING PROTOCOL Phillip M. Dickens Vijay K. Gurbani Paper code: S262 Department of Computer Science and Applied Mathematics

More information

ARIB STD-T64-C.S0042 v1.0 Circuit-Switched Video Conferencing Services

ARIB STD-T64-C.S0042 v1.0 Circuit-Switched Video Conferencing Services ARIB STD-T-C.S00 v.0 Circuit-Switched Video Conferencing Services Refer to "Industrial Property Rights (IPR)" in the preface of ARIB STD-T for Related Industrial Property Rights. Refer to "Notice" in the

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

Southern Methodist University. Department of Electrical Engineering. Telecommunications (EETS) Course Descriptions

Southern Methodist University. Department of Electrical Engineering. Telecommunications (EETS) Course Descriptions Southern Methodist University Department of Electrical Engineering Telecommunications (EETS) Course Descriptions 7301 Introduction to Telecommunications Overview of public and private telecommunications

More information

Testing IVR Systems White Paper

Testing IVR Systems White Paper Testing IVR Systems Document: Nexus8610 IVR 05-2005 Issue date: Author: Issued by: 26MAY2005 Franz Neeser Senior Product Manager Nexus Telecom AG, Switzerland We work to improve your network Abstract Interactive

More information

Advanced SIP Series: SIP and 3GPP Operations

Advanced SIP Series: SIP and 3GPP Operations Advanced S Series: S and 3GPP Operations, Award Solutions, Inc Abstract The Session Initiation Protocol has been chosen by the 3GPP for establishing multimedia sessions in UMTS Release 5 (R5) networks.

More information

Special Module on Media Processing and Communication

Special Module on Media Processing and Communication Special Module on Media Processing and Communication Multimedia Communication Fundamentals Dayalbagh Educational Institute (DEI) Dayalbagh Agra PHM 961 Indian Institute of Technology Delhi (IITD) New Delhi

More information

UK Interconnect White Paper

UK Interconnect White Paper UK Interconnect White Paper 460 Management Management Management Management 460 Management Management Management Management AI073 AI067 UK Interconnect White Paper Introduction The UK will probably have

More information

Overview ENUM ENUM. VoIP Introduction (2/2) VoIP Introduction (1/2)

Overview ENUM ENUM. VoIP Introduction (2/2) VoIP Introduction (1/2) Overview Voice-over over-ip (VoIP) ENUM VoIP Introduction Basic PSTN Concepts and SS7 Old Private Telephony Solutions Internet Telephony and Services VoIP-PSTN Interoperability IP PBX Network Convergence

More information

SP-000139. 3GPP SA#7 Meeting 15-17 17 March, 2000 Madrid. Megaco-H.248 / SIP. Giuseppe Ricagni

SP-000139. 3GPP SA#7 Meeting 15-17 17 March, 2000 Madrid. Megaco-H.248 / SIP. Giuseppe Ricagni 3GPP SA#7 Meeting 15-17 17 March, 2000 Madrid SP-000139 Megaco-H.248 / SIP Giuseppe Ricagni Session SIP: Initiation Protocol Developed by MMUSIC WG Work now carried on by SIP WG Lightweight signaling protocol

More information

Mobile Application Part protocol implementation in OPNET

Mobile Application Part protocol implementation in OPNET Mobile Application Part protocol implementation in OPNET Vladimir Vukadinovic and Ljiljana Trajkovic School of Engineering Science Simon Fraser University Vancouver, BC, Canada E-mail: {vladimir, ljilja}@cs.sfu.ca

More information