Understanding Voice over IP Protocols
|
|
|
- Horatio Leonard
- 10 years ago
- Views:
Transcription
1 Understanding Voice over IP Protocols Cisco Systems Service Provider Solutions Engineering February,
2 Topics to Discuss History of VoIP VoIP Early Adopters VoIP Standards and Standards Bodies VoIP Making Sense of the Protocols The Great Voice Myth VoIP Protocol Challenges Summary 2
3 Why Move to VoIP? Cost savings toll bypass Open standards H.323, SIP, MGCP Multi-vendor interoperability Integrated IP voice and data networks 3
4 Cisco Packet Voice Architecture TDM/ Circuit Switch Line Concentration Digital Trunk Subsystem Switching Network Call Control Connection Control Features Common Channel Signaling Complex Administration Maintenance Billing Open Service Application Layer (JAIN, AIN, TAPI, JTAPI, XML etc.) Open/Standard Interface Open Call Control Layer (SIP, H.323, MGCP, etc.) Open/Standard Interface Standards-Based Packet Infrastructure Layer (IP, ATM) 4
5 Topics to Discuss History of VoIP VoIP Early Adopters VoIP Standards and Standards Bodies VoIP Making Sense of the Protocols The Great Voice Myth VoIP Protocol Challenges Summary 5
6 Early Adopters Advanced Services and Toll-Bypass Regulatory opportunities allowed for toll-bypass PC-to-phone, calling-card and international fax services Cisco-based carriers used standard protocols, but not all carriers implemented standards Inter-carrier connections had protocol interoperability challenges 6
7 Topics to Discuss History of VoIP VoIP Early Adopters VoIP Standards and Standards Bodies VoIP Making Sense of the Protocols The Great Voice Myth VoIP Protocol Challenges Summary 7
8 Making the Rules for VoIP IETF (Internet Engineering Task Force) The community of engineers that standardizes the protocols that define how the Internet and Internet Protocols work. ITU (International Telecommunications Union) An international organization within the United Nations System where governments and the private sector coordinate global telecom networks and services. 8
9 Defining the VoIP Protocols H.323 SIP MGCP H.248 Megaco An ITU Recommendation that defines Packet-based multimedia communications systems. H.323 defines a distributed architecture for creating multimedia applications, including VoIP Defined as IETF RFC SIP defines a distributed architecture for creating multimedia applications, including VoIP Defined as IETF RFC MGCP defines a centralized architecture for creating multimedia applications, including VoIP An ITU Recommendation that defines Gateway Control Protocol. H.248 is the result of a joint-collaborate with the IETF. H.248 defines a centralized architecture, and is also known as Megaco Defined as IETF RFC Megaco defines a centralized architecture 9
10 Topics to Discuss History of VoIP VoIP Early Adopters VoIP Standards and Standards Bodies VoIP Making Sense of the Protocols The Great Voice Myth VoIP Protocol Challenges Summary 10
11 H.323 Components GK H.323 MCU Scope of H.323 e H.323 Gatekeeper Packet Network H.323 Terminal H.323 Gateway PSTN ISDN V.70 Terminal H.324 Terminal Speech Terminal H.320 Terminal Speech Terminal 11
12 Scope of H.323 Recommendation Video I/O Equipment Audio I/O Equipment Video Codec H.261, H.263 Audio Codec G.711, G.722, G.723, G.728, G.729 Receive Pain Delay (Sync) UDP RTP RTCP User Data Applications T.120, etc. System Control H.225 Layer IP H.245 Control TCP System Control User Interface Call Control H RAS Control H UDP 12
13 H.323 Signaling V H.323 Endpoint A Setup Alerting / Connect V H.323 Endpoint B H.225 (TCP Port 1720) Capabilities Exchange / MSD Open Logical Channel Open Logical Channel Acknowledge H.245 (TCP Dynamic Port) RTP Stream RTP Stream RTCP Stream Media (UDP) 13
14 Basic H.323 Call Gatekeeper A ACF LRQ LCF Gatekeeper B ACF RRQ/RCF IP Network RRQ/RCF ARQ V Gateway A H.225 (Q.931) Setup H.225 (Q.931) Alert and Connect H.245 RTP V ARQ Gateway B Phone A Phone B 14
15 Deploying H.323 Networks DGK Minimizes GK configuration Addition of new zones Addition of new NPAs Addition of new rate centers GK GK GK LA West Zone LA GW #1 GW #2 Rate Intra-LATA Rate Center #1 Toll Center #1 Chicago GW Local Midwest PSTN Zone NY GW Local East PSTN Zone 15
16 MGCP/H.248/Megaco Architectures Call Agent Call Agent SS7 P S T N IMT PRI PSTN P S T N Access Gateway MGCP / H.248 / Megaco RTP 16
17 Deploying MGCP/H.248/Megaco Networks STP PSTN SS7 SS7 Backhaul SLT IMTs MG CA Traditional TDM Traffic Modem Dial-up Traffic MGCP and ISDN Backhaul MG VoIP ISDN/PRI OSS TDM Voice NAS/VoIP Billing and Measurement Server Service Provider's TDM Network Service Provider's Packet Network 17
18 SIP Architecture I N T E L L I G E N T LDAP Oracle XML SIP Proxy, Registrar & Redirect Servers SIP CPL 3pcc Application Services S E R V I C E S SIP SIP SIP User Agents (UA) RTP (Media) PSTN Legacy PBX CAS or PRI 18
19 SIP Signaling PSTN Calling Party SIP Signaling and SDP Signaling (UDP or TCP) SIP VoIP Network INVITE 100 Trying INVITE 100 Trying 180 Ringing 180 Ringing 200 OK 200 OK PSTN Signaling Called Party ACK ACK Media (UDP) RTCP Stream Bearer Or Media 19
20 SIP Servers/Services Registrar Redirect Location Database SIP Servers/ Services Where is this name/phone#? REGISTER Here I am 3xx Redirection They moved, try this address INVITE I want to talk to another UA SIP Proxy Proxied INVITE I ll handle it for you SIP User Agents SIP User Agents SIP-GW 20
21 Deploying SIP Networks PSTN 312 Chicago POP Central Zone PSTN 212 NY POP East Zone IP Network West Zone SF POP PSTN
22 Topics to Discuss History of VoIP VoIP Early Adopters VoIP Standards and Standards Bodies VoIP Making Sense of the Protocols The Great Voice Myth VoIP Protocol Challenges Summary 22
23 Voice Myths Myths Networks can only be built one way Networks will only use one protocol All networks will converge Facts VoIP allows centralized or distributed architectures H.323, SIP, MGCP and H.248/Megaco will all be present in VoIP networks Networks will converge to IP 23
24 Topics to Discuss History of VoIP VoIP Early Adopters VoIP Standards and Standards Bodies VoIP Making Sense of the Protocols The Great Voice Myth VoIP Protocol Challenges Summary 24
25 Interconnecting VoIP Networks H.323 SIP? MGCP H.248 Megaco 25
26 Connecting VoIP to SS7/C7 Networks IAM H.323 H.225 Setup (ANI,DN) Proceeding H.245 MGCP CRCX ACK SDP SIP INVITE ACK SDP 26
27 VoIP Interworking Issues Service interworking E.g.: H.450 <-> SIP <-> MGCP Media interworking End-to-end codec negotiation Bearer interworking End-to-end fax, modem, DTMF 27
28 VoIP Interworking Bearer level Modem (relay/passthru) Fax (relay/passthru) T.38 T.37 DTMF (relay/passthru) Media level Codec (negotiation, selection) Service translation issues Call deflection Park/hold Signal issues SDP H
29 Fax and Modem Passthru Mechanisms Modem and fax are control mechanisms based on PLL (Phase Locked Loops) They are both time sensitive Highly sensitive to packet network impairments: Jitter Packet loss Delay Susceptible to clock slew (clock sync differences between gateways) 29
30 Passthru Simplified Voice Gateway PCM G.711m DSP G.711m G.729 IP Cloud Voice Gateway G.711m G.729 DSP PCM G.711m 30
31 What Is Modem Passthru? It is the transport of modem signals (modulation, error correction and compression) through a packet network using PCM encoded packets 31
32 Modem Passthru (Cont.) Modem tone detection (<= V.90) Switchover signaling No VAD EC off RTP payload redundancy (10ms packetization) RFC2198 (optional) 32
33 Modem Passthru Issues Consecutive packet drops (loss) cause retrain Consecutive drops during retrain causes disconnect Variation of delay (jitter) has quite an effect Jitter (at 10%) is a conservative estimate Since jitter mostly impacts performance with packet loss 33
34 What Is Modem Relay? Modem relay involves demodulating the modem signal at ingress gateway Passing this data as packet data to terminating gateway Re-modulating the data and passes it to the receiving modem 34
35 Fax Relay T.38 T.30 UDP T.30 PSTN PSTN Real-time Also called demod/remod Can be used in H.323/MGCP/SIP signaling Delivers fax data over UDP streams (uses same RTP port) reuses voice UDP ports Fallback to proprietary mode Method of encoding the T.30 and T.4 into packets IP 35
36 DTMF What is DTMF Why is it required? and where is it used? How do you transport it in IP? DTMF implementation 36
37 DTMF (Cont.) In TDM world, all voice traffic is sent as uncompressed 64Kbs PCM streams; anything sent on that circuit is an untouched stream of bits; (e.g., voice speech, modem tones, fax tones, and DTMF digits) DSP codecs designed to interpret human speech, can distort DTMF tones (machine-tones) High b/w codecs less likely to distort Distortion causes problems with voic and IVR systems 37
38 DTMF Schemes with VoIP Protocols H.323 MGCP, H.248, Megaco SIP In-Band In-Band In-Band In-Band Out-of- Band Cisco RTP, H.245 Alphanum, H.245 Signal, AVT Tones RFC2833 Cisco RTP, NSE, NTE,RFC2833 RFC
39 Topics to Discuss History of VoIP VoIP Early Adopters VoIP Standards and Standards Bodies VoIP Making Sense of the Protocols The Great Voice Myth VoIP Protocol Challenges Summary 39
40 Summary Understand the possibilities and the issues Avoid protocol/product based bias Decide on application Consider market and business drivers Deploy what s possible today Choose signaling protocol depending on services intended to be offered Many possibilities stay tuned 40
41 Crystal Ball on VoIP All three protocols (or its variations) are here for the long run Changes/enhancements will be made IP will be the core 41
42 Reference URLs ITU: IETF: SIP: H.323: MGCP: _protocol.asp?page=techlibrary 42
43 43
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
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
Curso de Telefonía IP para el MTC. Sesión 2 Requerimientos principales. Mg. Antonio Ocampo Zúñiga
Curso de Telefonía IP para el MTC Sesión 2 Requerimientos principales Mg. Antonio Ocampo Zúñiga Factors Affecting Audio Clarity Fidelity: Audio accuracy or quality Echo: Usually due to impedance mismatch
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
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
Hands on VoIP. Content. Tel +44 (0) 845 057 0176 [email protected]. 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
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.
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
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
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
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
Unit 23. RTP, VoIP. Shyam Parekh
Unit 23 RTP, VoIP Shyam Parekh Contents: Real-time Transport Protocol (RTP) Purpose Protocol Stack RTP Header Real-time Transport Control Protocol (RTCP) Voice over IP (VoIP) Motivation H.323 SIP VoIP
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
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
SIP-H.323 Interworking
SIP-H.323 Interworking Phone (408) 451-1430 1762 Technology Drive Suite 124 Fax (408) 451-1440 San Jose CA 95110-1307 USA URL www.ipdialog.com Joon Maeng [email protected] SIP and H.323! IETF SIP! Session
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,
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
An Introduction to Voice over the IP. Test1 Pool Questions
Dr. Mona Cherri Business and Technology North Lake College/DCCCD An Introduction to Voice over the IP I. True and False Questions Test1 Pool Questions 1. The first Internet-telephony software, Internet
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
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
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
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
VoIP. Overview. Jakob Aleksander Libak [email protected]. Introduction Pros and cons Protocols Services Conclusion
VoIP Jakob Aleksander Libak [email protected] 1 Overview Introduction Pros and cons Protocols Services Conclusion 2 1 Introduction Voice over IP is routing of voice conversations over the internet or
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.
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
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
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
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
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
This specification this document to get an official version of this User Network Interface Specification
This specification describes the situation of the Proximus network and services. It will be subject to modifications for corrections or when the network or the services will be modified. Please take into
How To Understand The Differences Between A Fax And A Fax On A G3 Network
The Fax on IP Networks White Paper February 2011 2 The Fax on IP Networks Contents Overview... 3 Group 3 Fax Technology... 4 G.711 Fax Pass-Through... 5 T.38 IP Fax Relay... 6 Network Design Considerations...
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
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.
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
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
Voice over IP (VoIP) Part 2
Kommunikationssysteme (KSy) - Block 5 Voice over IP (VoIP) Part 2 Dr. Andreas Steffen 1999-2001 A. Steffen, 10.12.2001, KSy_VoIP_2.ppt 1 H.323 Network Components Terminals, gatekeepers, gateways, multipoint
B12 Troubleshooting & Analyzing VoIP
B12 Troubleshooting & Analyzing VoIP Phillip Sherlock Shade, Senior Forensics / Network Engineer Merlion s Keep Consulting [email protected] Phillip Sherlock Shade (Phill) [email protected] Phillip
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
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
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
SIP A Technology Deep Dive
SIP A Technology Deep Dive Anshu Prasad Product Line Manager, Mitel June 2010 Laith Zalzalah Director, Mitel NetSolutions What is SIP? Session Initiation Protocol (SIP) is a signaling protocol for establishing
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
VoIP Analysis Fundamentals with Wireshark. Phill Shade (Forensic Engineer Merlion s Keep Consulting)
VoIP Analysis Fundamentals with Wireshark Phill Shade (Forensic Engineer Merlion s Keep Consulting) 1 Phillip D. Shade (Phill) [email protected] Phillip D. Shade is the founder of Merlion s Keep Consulting,
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
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
ENTERPRISE SOLUTION FOR DIGITAL AND ANALOG VOICE TRANSPORT ACROSS IP/MPLS
SOLUTION BRIEF ENTERPRISE SOLUTION FOR DIGITAL AND ANALOG VOICE TRANSPORT ACROSS IP/MPLS IT Organizations Can Reduce Costly TDM Leased Line Fees Challenge IP networks were not designed to transport bit-synchronous
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
Project Code: SPBX. Project Advisor : Aftab Alam. Project Team: Umair Ashraf 03-1853 (Team Lead) Imran Bashir 02-1658 Khadija Akram 04-0080
Test Cases 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:23-11-2007 SPBX
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 [email protected] June 2004 Copyright 2004 Executive Summary
Optimizing Converged Cisco Networks (ONT)
Optimizing Converged Cisco Networks (ONT) Module 2: Cisco VoIP Implementations (Deploy) Calculating Bandwidth Requirements for VoIP Objectives Describe factors influencing encapsulation overhead and bandwidth
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
Cisco Networks (ONT) 2006 Cisco Systems, Inc. All rights reserved.
Optimizing Converged Cisco Networks (ONT) reserved. Lesson 2.4: Calculating Bandwidth Requirements for VoIP reserved. Objectives Describe factors influencing encapsulation overhead and bandwidth requirements
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
A Comparative Study of Signalling Protocols Used In VoIP
A Comparative Study of Signalling Protocols Used In VoIP Suman Lasrado *1, Noel Gonsalves *2 Asst. Prof, Dept. of MCA, AIMIT, St. Aloysius College (Autonomous), Mangalore, Karnataka, India Student, Dept.
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
White Paper: Voice Over IP Networks
FREE FREE One One Hour Hour VoIPonline VoIPonline Seminar TM Seminar TM For additional information contact: Terry Shugart - [email protected] http://www.analogic.com/cti TEL: 978-977-3000 FAX: 978-977-6813
VA Enterprise Standard: VIDEO CODEC/RECORDING
DEPARTMENT OF VETERANS AFFAIRS (VA) OFFICE OF INFORMATION AND TECHNOLOGY (OIT) VA SERVICE DELIVERY ENGINEERING (SDE) ENTERPRISE SYSTEMS ENGINEERING (ESE) VA Enterprise Standard: VIDEO CODEC/RECORDING Version
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,
Dialogic Diva SIPcontrol Software
Dialogic Diva SIPcontrol Software converts Dialogic Diva Media Boards (Universal and V-Series) into SIP-enabled PSTN-IP gateways. The boards support a variety of TDM protocols and interfaces, ranging from
How to Configure the NEC SV8100 for use with Integra Telecom SIP Solutions
How to Configure the NEC SV8100 for use with Integra Telecom SIP Solutions Overview: This document provides a reference for configuration of the NEC SV8100 IP PBX to connect to Integra Telecom SIP trunks.
Chapter 10 Session Initiation Protocol. Prof. Yuh-Shyan Chen Department of Computer Science and Information Engineering National Taipei University
Chapter 10 Session Initiation Protocol Prof. Yuh-Shyan Chen Department of Computer Science and Information Engineering National Taipei University Outline 12.1 An Overview of SIP 12.2 SIP-based GPRS Push
Methods for Lawful Interception in IP Telephony Networks Based on H.323
Methods for Lawful Interception in IP Telephony Networks Based on H.323 Andro Milanović, Siniša Srbljić, Ivo Ražnjević*, Darryl Sladden*, Ivan Matošević, and Daniel Skrobo School of Electrical Engineering
Development of SIP-H.323 Gateway Project
Development of SIP-H.323 Gateway Project Ruston Hutchens QUESTnet 2005 Thursday 7 th July v2 SIP-H.323 Gateway project Motivation Large deployment base of H.323 terminals (over 2.9 million calls placed
VoIP Bandwidth Considerations - design decisions
VoIP Bandwidth Considerations - design decisions When calculating the bandwidth requirements for a VoIP implementation the two main protocols are: a signalling protocol such as SIP, H.323, SCCP, IAX or
AV@ANZA Formación en Tecnologías Avanzadas
SISTEMAS DE SEÑALIZACION SIP I & II (@-SIP1&2) Contenido 1. Why SIP? Gain an understanding of why SIP is a valuable protocol despite competing technologies like ISDN, SS7, H.323, MEGACO, SGCP, MGCP, and
SIP Essentials Training
SIP Essentials Training 5 Day Course Lecture & Labs COURSE DESCRIPTION Learn Session Initiation Protocol and important protocols related to SIP implementations. Thoroughly study the SIP protocol through
Enterprise Video Conferencing
Enterprise Video Conferencing When Voice Meets Video How SIP & H.323 Can Coexist SIPNOC 2014 Presented by: Gernot Scheichl June 2014 Agenda The Market The Challenges History Comparing the Protocols (H.323
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
SIP Trunking Manual 05.15. Technical Support Web Site: http://ws1.necii.com (registration is required)
SIP Trunking Manual 05.15 Technical Support Web Site: http://ws1.necii.com (registration is required) This manual has been developed by NEC Unified Solutions, Inc. It is intended for the use of its customers
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
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
NTP VoIP Platform: A SIP VoIP Platform and Its Services
NTP VoIP Platform: A SIP VoIP Platform and Its Services Speaker: Dr. Chai-Hien Gan National Chiao Tung University, Taiwan Email: [email protected] Date: 2006/05/02 1 Outline Introduction NTP VoIP
Ram Dantu. VOIP: Are We Secured?
Ram Dantu Professor, Computer Science and Engineering Director, Center for Information and Computer Security University of North Texas [email protected] www.cse.unt.edu/~rdantu VOIP: Are We Secured? 04/09/2012
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...
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
SIP (Session Initiation Protocol) Technical Overview. Presentation by: Kevin M. Johnson VP Engineering & Ops
SIP (Session Initiation Protocol) Technical Overview Presentation by: Kevin M. Johnson VP Engineering & Ops Page 1 Who are we? Page 2 Who are we? Workforce Automation Software Developer Page 3 Who are
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 [email protected]
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
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
Understanding Session Initiation Protocol (SIP)
CHAPTER 42 This chapter describes SIP and the interaction between SIP and Cisco Unified Communications Manager. This section covers the following topics: SIP Trunk Configuration Checklist, page 42-1 SIP
Voice over IP Solutions
White Paper Voice over IP Solutions Sean Christensen Professional Services Juniper Networks, Inc. 1194 North Mathilda Avenue Sunnyvale, CA 94089 USA 408 745 2000 or 888 JUNIPER www.juniper.net Part Number
Goal We want to know. Introduction. What is VoIP? Carrier Grade VoIP. What is Meant by Carrier-Grade? What is Meant by VoIP? Why VoIP?
Goal We want to know Introduction What is Meant by Carrier-Grade? What is Meant by VoIP? Why VoIP? VoIP Challenges 2 Carrier Grade VoIP Carrier grade Extremely high availability 99.999% reliability (high
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
Applied Networks & Security
Applied Networks & Security VoIP with Critical Analysis http://condor.depaul.edu/~jkristof/it263/ John Kristoff [email protected] IT 263 Spring 2006/2007 John Kristoff - DePaul University 1 Critical analysis
SIP: Ringing Timer Support for INVITE Client Transaction
SIP: Ringing Timer Support for INVITE Client Transaction Poojan Tanna ([email protected]) Motorola India Private Limited Outer Ring Road, Bangalore, India 560 037 Abstract-The time for which the Phone
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
NAT TCP SIP ALG Support
The feature allows embedded messages of the Session Initiation Protocol (SIP) passing through a device that is configured with Network Address Translation (NAT) to be translated and encoded back to the
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
Lehrstuhl für Informatik 4 Kommunikation und verteilte Systeme
Chapter 2: Representation of Multimedia Data Chapter 3: Multimedia Systems Communication Aspects and Services Multimedia Applications and Communication Protocols Quality of Service and Resource Management
Application Notes. Introduction. Contents. Managing IP Centrex & Hosted PBX Services. Series. VoIP Performance Management. Overview.
Title Series Managing IP Centrex & Hosted PBX Services Date July 2004 VoIP Performance Management Contents Introduction... 1 Quality Management & IP Centrex Service... 2 The New VoIP Performance Management
Simulation of SIP-Based VoIP for Mosul University Communication Network
Int. J. Com. Dig. Sys. 2, No. 2, 89-94(2013) 89 International Journal of Computing and Digital Systems http://dx.doi.org/10.12785/ijcds/020205 Simulation of SIP-Based VoIP for Mosul University Communication
Real-time communication on IP networks
Real-time communication on IP networks TROND ULSETH AND FINN STAFSNES Standardisation work on VoIP protocols has now been going on for almost 10 years. The basic protocols can be considered as mature,
Voice over IP Protocols And Compression Algorithms
University of Tehran Electrical and Computer Engineering School SI Lab. Weekly Presentations Voice over IP Protocols And Compression Algorithms Presented by: Neda Kazemian Amiri Agenda Introduction to
(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
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
VoIP QoS. Version 1.0. September 4, 2006. AdvancedVoIP.com. [email protected] [email protected]. Phone: +1 213 341 1431
VoIP QoS Version 1.0 September 4, 2006 AdvancedVoIP.com [email protected] [email protected] Phone: +1 213 341 1431 Copyright AdvancedVoIP.com, 1999-2006. All Rights Reserved. No part of this
