A Novel Software-Based H.323 Gateway with
|
|
- Lucas Morris
- 5 years ago
- Views:
Transcription
1 A Novel Software-Based H.323 Gateway with Proxy-TC for VoIP Systems Presenter : Wei-Sheng Yin Advisor : Dr. Po-Ning Chen Institute of Communications Engineering National Chiao Tung University Agenda Introduction Motivation Proposed System Architecture System Architecture Proxy-Transcoder (Proxy-TC) The GW-TC GW-TC Procedure The Handoff Mechanism The Error Recovery Procedure Data Structures Simulation Model and Results Conclusions 1
2 Introduction Due to the constant growth of transmission bandwidth, real-time applications such as VoIP gradually become feasible on the Internet. VoIP can bring us advantages like toll-by-pass, coexistence of voice and data services, and easy deployment of telephony network between two branch offices. In techniques, VoIP carries voice traffic as data packets over a packet-switched network, instead of as a synchronous stream of binary sampled data over a circuit-switched, time-division multiplexed (TDM) network. Introduction - VoIP Protocols Features of the VoIP protocols, call control, and media transport Existing protocols for VoIP Recommendation H.323 by ITU-T Session Initiation Protocol and Session Description Protocol (SIP/SDP) by IETF Media Gateway Control Protocol (MGCP) by IETF Media Gateway Control (MEGACO) by IETF Been submitted to ITU-T SG-16 for decision as Recommendation H.248 Of the protocols above, H.323 is perhaps the most widely deployed; hence, our design is based on H.323 protocol stack. 2
3 Introduction - H.323 components Gatekeeper Manage a zone (i.e., a collection of H.323 devices). Required Functionality: Address translation, admissions control and bandwidth control. H.323 Terminal An endpoint on a LAN. Support real-time, 2-way communications with another H.323 entity. Must support voice (audio codec) and signaling (Q.931, H.245, RAS). Gateway Provide interoperability between different networks, and convert signaling and media. E.g., IP/PSTN gateway. Multi-point controller (MC), multi-point processor (MP) and Multi-point control Unit (MCU) are used to support conference calls. Introduction Gateway protocol stack Interworking Call Control (H.246) Gateway Call Manager Audio Codec G.723 G.729 RTP Video Codec H.261 H.263 RTCP H.225 RAS Control H Layer H.225 Call Control H.245 Control SCN Call Control Layer e.g., Q.931 SCN Link Control Layer e.g., LAPD Transport Protocols & Local Area Network Interface SCN Physical Interface 3
4 Motivations The telephony links maintained by a software-based gateway at times suffer short interrupts due to the possible uneven switching of PC platforms among several computation-bound tasks. We observe that the computation powers of the desktop PCs in use for general employees are sometimes not in full utilization. This leads us to the idea of gathering these excessive computation powers to share the load of the softwarebased gateway, and hence, reduce the number of computation-bound tasks on it. In such case, one may further increase the gateway capacity without introducing additional hardware cost. Proposed System Architecture Server PC1 Office Extension PC Telephone PBX PC2 R* Internet / Intranet PSTN / ISDN PC3 PC4 LAN Note: *The edge router must support QoS functionality. PC Telephone 4
5 Proposed System Architecture Office Extension G.711 Q.931 PBX PSTN / ISDN GW-TC GW TC2 TC3 G.711 TC1 G.723 or G.729 H.225 Call Control TC4 R* PC Telephone Internet / Intranet LAN Note: *The edge router must support QoS functionality. PC Telephone Proposed System Architecture - Modified Gateway Protocol Stack Interworking Call Control Gateway Call Manager GW-TC Payload (RTP/RTCP) GW-TC Control Audio Codec G.723 G.729 RTP Video Codec H.261 H.263 RTCP H.225 RAS Control H Layer H.225 Call Control H.245 Control SCN Call Control Layer e.g., Q.931 SCN Link Control Layer e.g., LAPD Transport Protocols & Local Area Network Interface SCN Physical Interface 5
6 Proposed System Architecture - Proxy-Transcoder (Proxy-TC) The Observer monitors local resource condition on PCs, such as available CPU power, and periodically updates the information for use by Controller. GW-TC proxy-tc Observer Controller Transcoder LAN/Gateway Internet/ RTP Logical Channel H.323 Endpoint G.711 G.723/G.729 RTCP Logical Channel CPU Proposed System Architecture - Proxy-Transcoder (Proxy-TC) The Controller maintains the communication between a proxy-tc and its associated gateway by using GW-TC signaling procedures. These control messages include: GW-TC proxy-tc Observer Controller (1) informing the gateway that LAN/Gateway Transcoder Internet/ the proxy-tc is still active RTP Logical Channel H.323 Endpoint upon the request of its G.711 G.723/G.729 associated gateway, RTCP Logical Channel receiving the request of the transcoding service from the gateway; and (3) updating the information of the CPU utilization. CPU 6
7 Proposed System Architecture - Proxy-Transcoder (Proxy-TC) The Transcoder is responsible for voice stream transcoding between different speech coding algorithms, such as G.711, G and G.729. GW-TC proxy-tc Observer Controller Transcoder LAN/Gateway RTP Logical Channel G.711 G.723/G.729 RTCP Logical Channel CPU Internet/ H.323 Endpoint Proposed System Architecture - The GW-TC (1/2) The major goal of the GW-TC signaling is to enable the cooperation between the gateway and the proxy-tcs. There are six commands in the GW-TC signaling, which are described below: Register: The proxy-tcs send this command manually to the gateway (through web-browsing activity) to register to the gateway so that the gateway can create corresponding proxy-tc record. Setup: This command is used by the gateway to request one of the proxy-tcs to perform the transcoding service upon the occurrence of a new call. Release: The gateway uses this command to release the transcoding service either (1) upon the end of a call or (2) upon the receipt of the Emergency command from a proxy-tc. 7
8 Proposed System Architecture - The GW-TC (2/2) : Proxy-TCs use this command to update their latest statuses kept on the gateway. Query: If the update timer expires the gateway will send Query command to request the proxy-tc to update its status. Emergency: This command is sent by proxy-tcs to notify the gateway of their (sudden) shortage of transcoding resources. Acknowledge (): This command may be sent by either the gateway or proxy-tcs to inform the other side that the previous command has been successfully received. Proposed System Architecture - The GW-TC Procedure Normal call setup Normal call clearing procedure 8
9 Proposed System Architecture - The Handoff Mechanism Hard Handoff PSTN/ISDN Terminal Gateway proxy-tc1 proxy-tc2 H.323 Endpoint Voice stream transmitted through PC1 Emergency Release Voice stream communication interrupted (about 60~200ms) Close Logical Channel Setup Close Logical Channel Voice stream transfer to PC2 Open Logical Channel Open Logical Channel H.245 GW-TC Proposed System Architecture - The Handoff Mechanism Soft Handoff PSTN/ISDN Terminal Gateway TC1 TC2 H.323 Endpoint Emergency Setup Voice stream continued through PC1 Close Logical Channel Open Logical Channel Close Logical Channel Open Logical Channel Voice stream transfer to PC2 Release H.245 GW-TC 9
10 Proposed System Architecture - Error Recovery Procedure Situations concerned Setup Emergency Release Proposed System Architecture - Error Recovery Procedure -- Setup Gateway proxy-tc1 proxy-tc2 Successful Setup procedure Setup G.711 voice packet Setup Timer expiry Setup Setup Setup to another TC G.711 voice packet 10
11 Proposed System Architecture - Error Recovery Procedure -- Gateway TC timer reset timer expiry Query timer reset timer expires three times Query Query Discard TC Record Proposed System Architecture - Error Recovery Procedure -- Emergency Gateway TC1 TC2 Emergency Setup G.711 voice packet Gateway TC1 TC2 Emergency Emergency Emergency Setup G.711 voice packet 11
12 Proposed System Architecture - Error Recovery Procedure -- Release Gateway TC Release Release Release Release Proposed System Architecture - Data Structure The gateway needs to maintain two tables: The first one records the periodic update information of CPU utilization from each proxy-tc CPU utilization time The number of serving users The second one keeps the call information in the system The active proxy-tc Occupied port numbers Call party information 12
13 Simulation Model and Results - The Simulation Model Register (Web based) TC1 Arrival GW Setup Release TC2 Emergency TC record: TC1: TC resource : 169MHz num_custs_insys : 2 time of update : 11:05:22 TC2 : TC resource : 210MHz num_custs_insys : 1 time of update : 11:05:23 Call information record: Customer1: Active TC : TC2 Customer2 : Active TC : TC1 Customer3 : Active TC : TC2 TC3 We assume that both the call inter-arrival time and the call duration are exponentially distributed. Simulation Model and Results - CPU utilization records GW Average = TC1 Average = TC2 Average = TC3 Average =
14 Simulation Model and Results - Proxy-TC selection schemes Definition: Ri : available resource of i th proxy-tc Token(i) : selecting i th proxy-tc NR : necessary TC resource per call N : the number of proxy-tcs First-Fit-Round-Robin (FFRR) initial: i = 0 while( R { Token( i) } Maximum Available Resource R j = max { R } Token ( j ) i < NR) i = ( i + 1)mod( N i + i 1) Simulation Model and Results - Performance Indices (1/2) Definitions of call events Blocked call When upon receipt of a new call, the individual CPU resources of the gateway and all registered proxy-tcs are less than 25MIPS. Dropped call When the serving proxy-tc is occasionally running out of resources for at least one second and no other proxy-tc can take over this call. 14
15 Simulation Model and Results - Performance Indices (2/2) Definitions of probabilities The new call blocking probability Pb Number of blocked calls = Pr [ blocking ] = Total arrival calls The active call dropping probability Number of dropped calls Pd = Pr[ droping ] = Total served calls The handoff frequency Number of handoff calls Ph = Pr[ Handoff ] = Total served calls Simulation Model and Results - The Simulation Results (1/5) Case 1 : The relation between the number of registered proxy-tcs and the probabilities of call blocking and call dropping, when no handoff mechanism is implemented. Dropp The mean duration of each call is 3 minutes. 15
16 Simulation Model and Results - The Simulation Results (2/5) Case 2 :The probabilities of call blocking and call dropping, when the gateway cooperates with only one proxy-tc and handoff mechanism is implemented. Dropp The mean duration of each call is 3 minutes. Simulation Model and Results - The Simulation Results (3/5) Case 2 : The handoff frequency, when the gateway cooperates with only one proxy-tc and handoff mechanism is implemented. Handoff Frequency 16
17 Simulation Model and Results - The Simulation Results (4/5) Case 3 : The performance comparison between the two call placement schemes (i.e., FFRR and MAR) under the condition that the gateway cooperate with 10 proxy-tcs and handoff mechanism is implemented. Dropp The mean duration of each call is 15 minutes. Simulation Model and Results - The Simulation Results (5/5) Case 3 : The performance comparison between the two call placement schemes (i.e., FFRR and MAR) under the condition that the gateway cooperate with 10 proxy-tcs and handoff mechanism is implemented. Handoff Frequency 17
18 Conclusions We conclude that the proposed architecture achieves higher capacity and lower blocking probability with respect to the arrival calls, and the employment of the handoff mechanism leads to a drastic decrease in the active call dropping probability at the expense of a little increase in the new call blocking probability. Along this research direction, an interesting future work will be to find a method to predict the CPU utilization, as well as a good call placement approach to reduce the handoff frequency. 18
Introduction. H.323 Basics CHAPTER
CHAPTER 1 Last revised on: October 30, 2009 This chapter provides an overview of the standard and the video infrastructure components used to build an videoconferencing network. It describes the basics
More informationTroubleshooting Voice Over IP with WireShark
Hands-On Troubleshooting Voice Over IP with WireShark Course Description Voice over IP is being widely implemented both within companies and across the Internet. The key problems with IP voice services
More informationVoIP Basics. 2005, NETSETRA Corporation Ltd. All rights reserved.
VoIP Basics Phone Network Typical SS7 Network Architecture What is VoIP? (or IP Telephony) Voice over IP (VoIP) is the transmission of digitized telephone calls over a packet switched data network (like
More informationH.323. Definition. Overview. Topics
H.323 Definition H.323 is a standard that specifies the components, protocols and procedures that provide multimedia communication services real-time audio, video, and data communications over packet networks,
More informationTSIN02 - Internetworking
Lecture 8: SIP and H323 Litterature: 2004 Image Coding Group, Linköpings Universitet Lecture 8: SIP and H323 Goals: After this lecture you should Understand the basics of SIP and it's architecture Understand
More informationVoIP. ALLPPT.com _ Free PowerPoint Templates, Diagrams and Charts
VoIP ALLPPT.com _ Free PowerPoint Templates, Diagrams and Charts VoIP System Gatekeeper: A gatekeeper is useful for handling VoIP call connections includes managing terminals, gateways and MCU's (multipoint
More informationIP-Telephony Introduction
IP-Telephony Introduction Bernard Hammer Siemens AG, Munich Siemens AG 2001 1 Presentation Outline Why Internet Telephony Expectations Future Scenario Protocols & System Architectures Protocols Standardistion
More informationMedia Communications Internet Telephony and Teleconference
Lesson 13 Media Communications Internet Telephony and Teleconference Scenario and Issue of IP Telephony Scenario and Issue of IP Teleconference ITU and IETF Standards for IP Telephony/conf. H.323 Standard
More informationIntroduction. We have learned
H.323 Introduction We have learned IP, UDP, RTP (RTCP) How voice is carried in RTP packets between session participants How does one party indicate to another a desire to set up a call? How does the second
More informationBasic Architecture of H.323 C. Schlatter,
Basic Architecture of H.323 C. Schlatter, schlatter@switch.ch 2003 SWITCH Agenda Background to H.323 Components of H.323 H.323 Protocols Overview H.323 Call Establishment 2003 SWITCH 2 Background to H.323
More informationOverview of the Session Initiation Protocol
CHAPTER 1 This chapter provides an overview of SIP. It includes the following sections: Introduction to SIP, page 1-1 Components of SIP, page 1-2 How SIP Works, page 1-3 SIP Versus H.323, page 1-8 Introduction
More informationChapter 11: Understanding the H.323 Standard
Página 1 de 7 Chapter 11: Understanding the H.323 Standard This chapter contains information about the H.323 standard and its architecture, and discusses how Microsoft Windows NetMeeting supports H.323
More informationTODAY AGENDA. VOIP Mobile IP
VOIP & MOBILE IP PREVIOUS LECTURE Why Networks? And types of Networks Network Topologies Protocols, Elements and Applications of Protocols TCP/IP and OSI Model Packet and Circuit Switching 2 TODAY AGENDA
More informationProblem verification during execution of H.323 signaling
Problem verification during execution of H.323 signaling 1 ESAD KADUSIC & 2 NATASA ZIVIC & 3 NARCIS BEHLILOVIC & 4 ALIJA VEGARA 1,3,4 Faculty of Electrical Engineering in Sarajevo Zmaja od Bosne, 71 000
More informationPilsung Taegyun A Fathur Afif A Hari A Gary A Dhika April Mulya Yusuf Anin A Rizka B Dion Siska Mirel Hani Airita Voice over Internet Protocol Course Number : TTH2A3 CLO : 2 Week : 7 ext Circuit Switch
More informationIntroduction. We have learned
H.323 Chapter 4 Introduction We have learned IP, UDP, RTP (RTCP) How does one party indicate to another a desire to set up a call? How does the second party indicate a willingness to accept the call? The
More informationSeminar report IP Telephony
A Seminar report On IP Telephony Submitted in partial fulfillment of the requirement for the award of degree of Bachelor of Technology in Computer Science SUBMITTED TO: www.studymafia.org SUBMITTED BY:
More informationOutline Overview Multimedia Applications Signaling Protocols (SIP/SDP, SAP, H.323, MGCP) Streaming Protocols (RTP, RTSP, HTTP, etc.) QoS (RSVP, Diff-S
Internet Multimedia Architecture Outline Overview Multimedia Applications Signaling Protocols (SIP/SDP, SAP, H.323, MGCP) Streaming Protocols (RTP, RTSP, HTTP, etc.) QoS (RSVP, Diff-Serv, IntServ) Conclusions
More informationProvide a generic transport capabilities for real-time multimedia applications Supports both conversational and streaming applications
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 Performance Tests Build-out Delay
More informationCourse 20337B: Enterprise Voice and Online Services with Microsoft Lync Server 2013 Exam Code: Duration:40 Hrs
Course 20337B: Enterprise Voice and Online Services with Microsoft Lync Server 2013 Exam Code: 70-337 Duration:40 Hrs Course Outline Module 1: Voice Architecture This module introduce Enterprise Voice
More informationCommunications Transformations 2: Steps to Integrate SIP Trunk into the Enterprise
Communications Transformations 2: Steps to Integrate SIP Trunk into the Enterprise The Changing Landscape IP-based unified communications is widely deployed in enterprise networks, both for internal calling
More informationVoIP Core Technologies. Aarti Iyengar Apricot 2004
VoIP Core Technologies Aarti Iyengar Apricot 2004 Copyright 2004 Table Of Contents What is Internet Telephony or Voice over IP? VoIP Network Paradigms Key VoIP Protocols Call Control and Signaling protocols
More informationPacketizer. Overview of H.323. Paul E. Jones. Rapporteur, ITU-T Q2/SG16 April 2007
Overview of H.323 Paul E. Jones Rapporteur, ITU-T Q2/SG16 paulej@packetizer.com April 2007 Copyright 2007 Executive Summary H.323 was first approved in February 1996, the same month that the first SIP
More information20337-Enterprise Voice and Online Services with Microsoft Lync Server 2013
Course Outline 20337-Enterprise Voice and Online Services with Microsoft Lync Server 2013 Duration: 5 day (30 hours) Target Audience: This course is intended for IT Consultants and Telecommunications Consulting
More informationActively Managing Multimedia Telchemy Actively Managing Multimedia
Actively Managing Multimedia VoIP Fault and Performance Management Alan Clark, CEO Incorporated alan.clark@telchemy.com About Leading provider of core technology for VoIP fault and performance management
More informationSecure Telephony Enabled Middle-box (STEM)
Report on Secure Telephony Enabled Middle-box (STEM) Maggie Nguyen 04/14/2003 Dr. Mark Stamp - SJSU - CS 265 - Spring 2003 Table of Content 1. Introduction 1 2. IP Telephony Overview.. 1 2.1 Major Components
More informationVoice over IP (VoIP)
Voice over IP (VoIP) David Wang, Ph.D. UT Arlington 1 Purposes of this Lecture To present an overview of Voice over IP To use VoIP as an example To review what we have learned so far To use what we have
More informationNetwork+ Guide to Networks 6th Edition. Chapter 12 Voice and Video Over IP
Network+ Guide to Networks 6th Edition Chapter 12 Voice and Video Over IP Objectives Use terminology specific to converged networks Explain VoIP (Voice over IP) services, PBXs, and their user interfaces
More informationA common issue that affects the QoS of packetized audio is jitter. Voice data requires a constant packet interarrival rate at receivers to convert
A common issue that affects the QoS of packetized audio is jitter. Voice data requires a constant packet interarrival rate at receivers to convert data into a proper analog signal for playback. The variations
More informationCisco Unified MeetingPlace Integration
CHAPTER 14 This chapter covers system-level design and implementation of Cisco Unified MeetingPlace 5.4 in a Cisco Unified Communications Manager 5.x environment. The following aspects of design and configuration
More informationMulti-Service Access and Next Generation Voice Service
Hands-On Multi-Service Access and Next Generation Voice Service Course Description The next generation of telecommunications networks is being deployed using VoIP technology and soft switching replacing
More informationH.323 Tutorial Realsoft Corporation January 12, 2000
H.323 Tutorial 2000 Realsoft Corporation http://www.realsoft-corp.com/ January 12, 2000 Abstract: This document summarizes the H.323 (H.225, H.245) Recommendation into an understandable tutorial. Much
More informationAT&T IP Flexible Reach And IP Toll Free Cisco Call Manager Configuration Guide. Issue /5/2007
And IP Toll Free Cisco Call Manager Configuration Guide Issue 2.13 6/5/2007 Page 1 of 38 TABLE OF CONTENTS 1 Introduction... 3 2 Special Notes... 3 3 Overview... 4 3.1 Call Manager Site... 4 3.2 TFTP and
More informationZyXEL V120 Support Notes. ZyXEL V120. (V120 IP Attendant 1 Runtime License) Support Notes
ZyXEL V120 (V120 IP Attendant 1 Runtime License) Support Notes Version 1.00 April 2009 1 Contents Overview 1. Overview of V120 IP Attendant...3 2. Setting up the V120...4 3. Auto Provision...7 4. V120
More informationEnterprise Voice and Online Services with Microsoft Lync Server 2013
Enterprise Voice and Online Services with Microsoft Lync Server 2013 Course # Exam: Prerequisites Technology: Delivery Method: Length: 20337 70-337 20336 Microsoft Lync Server Instructor-led (classroom)
More informationTC32 presentation to ECMA General Assembly, Edinburgh, 22nd June 2000
TC32 presentation to ECMA General Assembly, Edinburgh, 22nd June 2000 John Elwell, Chairman ECMA TC32 Siemens Communications (International) Limited john.elwell@siemenscomms.co.uk ECMA/TC32/2000/103 ECMA/GA/2000/69
More informationB.Eng. (Hons.) Telecommunications. Examinations for / Semester 1
B.Eng. (Hons.) Telecommunications Cohort: BTEL/12/FT Examinations for 2014-2015 / Semester 1 MODULE: IP TELEPHONY MODULE CODE: TELC 3107 Duration: 3 Hours Instructions to Candidates: 1. Answer all questions.
More informationTransporting Voice by Using IP
Transporting Voice by Using IP National Chi Nan University Quincy Wu Email: solomon@ipv6.club.tw 1 Outline Introduction Voice over IP RTP & SIP Conclusion 2 Digital Circuit Technology Developed by telephone
More informationThe Interworking of IP Telephony with Legacy Networks
The Interworking of IP Telephony with Legacy Networks Yang Qiu Valmio 0/ 0080 Helsinki Yang.Qiu@nokia.com Abstract This document describes the Interworking of IP Telephony networks with legacy networks.
More informationABSTRACT. that it avoids the tolls charged by ordinary telephone service
ABSTRACT VoIP (voice over IP - that is, voice delivered using the Internet Protocol) is a term used in IP telephony for a set of facilities for managing the delivery of voice information using the Internet
More informationCCIE Voice v3.0 Quick Reference
Table of Contents Chapter 1 Unified Communications Manager...3 CCIE oice v3.0 Quick Reference Chapter 2 Understanding Quality of Service... 42 Chapter 3 Telephony Protocols... 69 Chapter 4 Unity/Unity
More informationPerformance Management: Key to IP Telephony Success
Performance Management: Key to Telephony Success Alan Clark, President & CEO Telchemy, Incorporated http://www.telchemy.com Vo Performance Management Voice, Video and other Real Time Applications Vo Performance
More informationAVANTUS TRAINING PTE PTE LTD LTD
[MS20337]: Enterprise Voice and Online Services with Microsoft Lync Server 2013 Length : 5 Days Audience(s) : IT Professionals Level : 300 Technology : Microsoft Lync Server Delivery Method : Instructor-led
More informationDigital Advisory Services Professional Service Description SIP IP Trunk with Field Trial for Legacy PBX Model
Digital Advisory Services Professional Service Description SIP IP Trunk with Field Trial for Legacy PBX Model 1. Description of Services. 1.1 SIP IP Trunk with Field Trial for Legacy PBX Verizon will assist
More informationApproaches to Deploying VoIP Technology Instead of PSTN Case Study: Libyan Telephone Company to Facilitate the Internal Work between the Branches
Approaches to Deploying VoIP Technology Instead of PSTN Case Study: Libyan Telephone Company to Facilitate the Internal Work between the Branches Dr. Elmabruk M Laias * Department of Computer, Omar Al-mukhtar
More informationDigital Advisory Services Professional Service Description SIP Centralized IP Trunk with Field Trial Model
Digital Advisory Services Professional Service Description SIP Centralized IP Trunk with Field Trial Model 1. Description of Services. 1.1 SIP Centralized IP Trunk with Field Trial Verizon will assist
More informationSummary of last time " " "
Summary of last time " " " Part 1: Lecture 3 Beyond TCP TCP congestion control Slow start Congestion avoidance. TCP options Window scale SACKS Colloquia: Multipath TCP Further improvements on congestion
More informationAT&T IP Flexible Reach And IP Toll Free Cisco Unified Communication Manager H.323 Configuration Guide. Issue /3/2008
AT&T IP Flexible Reach And IP Toll Free Cisco Unified Communication Manager H.323 Configuration Guide Issue 2.17 3/3/2008 Page 1 of 49 TABLE OF CONTENTS 1 Introduction... 4 2 Special Notes... 4 3 Overview...
More informationSignaling System 7 (SS7) By : Ali Mustafa
Signaling System 7 (SS7) By : Ali Mustafa Contents Types of Signaling SS7 Signaling SS7 Protocol Architecture SS7 Network Architecture Basic Call Setup SS7 Applications SS7/IP Inter-working VoIP Network
More informationMultimedia Applications. Classification of Applications. Transport and Network Layer
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
More informationPROTOCOLS FOR THE CONVERGED NETWORK
Volume 2 PROTOCOLS FOR THE CONVERGED NETWORK Mark A. Miller, P.E. President DigiNet Corporation A technical briefing from: March 2002 Table of Contents Executive Summary i 1. Converging Legacy Networks
More informationETSF10 Internet Protocols Transport Layer Protocols
ETSF10 Internet Protocols Transport Layer Protocols 2012, Part 2, Lecture 2.2 Kaan Bür, Jens Andersson Transport Layer Protocols Special Topic: Quality of Service (QoS) [ed.4 ch.24.1+5-6] [ed.5 ch.30.1-2]
More informationImplementing Cisco Voice Communications & QoS (CVOICE) 8.0 COURSE OVERVIEW: WHO SHOULD ATTEND: PREREQUISITES: Running on UC 9.
Implementing Cisco Voice Communications & QoS (CVOICE) 8.0 COURSE OVERVIEW: Running on UC 9.x Software Implementing Cisco Voice Communications and QoS (CVOICE) v8.0 is a 5-day training program that teaches
More informationReal-time Services BUPT/QMUL
Real-time Services BUPT/QMUL 2017-05-27 Agenda Real-time services over Internet Real-time transport protocols RTP (Real-time Transport Protocol) RTCP (RTP Control Protocol) Multimedia signaling protocols
More informationUnderstanding Cisco CallManager Trunk Types
42 CHAPTER In a distributed call-processing environment, Cisco CallManager communicates with other Cisco CallManager clusters, the public switched telephone network (PSTN), and other non-ip telecommunications
More informationMultimedia Networking. Protocols for Real-Time Interactive Applications
Multimedia Networking Protocols for Real-Time Interactive Applications Real Time Protocol Real Time Control Protocol Session Initiation Protocol H.323 Real-Time Protocol (RTP) RTP is companion protocol
More informationImplementing Cisco IP Telephony & Video, Part 1 (CIPTV1) 1.0
Implementing Cisco IP Telephony & Video, Part 1 (CIPTV1) 1.0 COURSE OVERVIEW: Implementing Cisco IP Telephony & Video, Part 1 (CIPTV1) v1.0 is a five-day course that prepares the learner for implementing
More informationWhat is NGN? Hamid R. Rabiee Mostafa Salehi, Fatemeh Dabiran, Hoda Ayatollahi Spring 2011
What is NGN? Hamid R. Rabiee Mostafa Salehi, Fatemeh Dabiran, Hoda Ayatollahi Spring 2011 Outlines Next Generation Network (NGN) Definition Applications Requirements Network Architecture QoS Issues 2 What
More informationINTERNATIONAL TELECOMMUNICATION UNION
INTERNATIONAL TELECOMMUNICATION UNION ITU-T H.323 TELECOMMUNICATION STANDARDIZATION SECTOR OF ITU Annex G (02/00) SERIES H: AUDIOVISUAL AND MULTIMEDIA SYSTEMS Infrastructure of audiovisual services Systems
More informationCourse Outline: Implementing Cisco IP Telephony & Video, Part 1 (CIPTV1)
Course Outline: Implementing Cisco IP Telephony & Video, Part 1 (CIPTV1) Learning Method: Instructor-led Classroom Learning Duration: 5.00 Day(s)/ 40 hrs : CIPTV1 v1.0 gives the learner all the tools they
More informationInternet Telephony. Definition. Overview. Topics. 1. Introduction
Internet Telephony Definition Internet telephony refers to communications services voice, facsimile, and/or voice-messaging applications that are transported via the Internet, rather than the public switched
More informationProtocols supporting VoIP
Protocols supporting VoIP Dr. Danny Tsang Department of Electronic & Computer Engineering Hong Kong University of Science and Technology 1 Outline Overview Session Control and Signaling Protocol H.323
More informationCisco H.323 Signaling Interface
CHAPTER 1 Introduction This chapter provides an overview of the (HSI) system and subsystems and contains the following sections: Cisco HSI Overview, page 1-1 Cisco HSI System Description, page 1-2 Operational
More informationReal Time Protocols. Overview. Introduction. Tarik Cicic University of Oslo December IETF-suite of real-time protocols data transport:
Real Time Protocols Tarik Cicic University of Oslo December 2001 Overview IETF-suite of real-time protocols data transport: Real-time Transport Protocol (RTP) connection establishment and control: Real
More informationApplication Notes. Introduction. Performance Management & Cable Telephony. Contents
Title Managing Cable Telephony Services Series VoIP Performance Management Date June 2004 Overview This application note describes the typical performance issues that cable operators encounter when deploying
More informationImproving QoS of VoIP over Wireless Networks (IQ-VW)
Improving QoS of VoIP over Wireless Networks (IQ-VW) Mona Habib & Nirmala Bulusu CS522 12/09/2002 1 Agenda Voice over IP (VoIP): Why? VoIP Protocols: H.323 and SIP Quality of Service (QoS) Wireless Networks
More informationnc. C o m m u n i c a t i o n S o l u t i o n s to provide a better solution to the problem. While the standard is not yet finalized, it is clear that
nc. V.MoIP: Modem over Packet Networks Technology C o m m u n i c a t i o n S o l u t i o n s Abstract The migration from circuit-switched to packet-based networks brings with it many new possibilities.
More informationMicrosoft Enterprise Voice and Online Services with Microsoft Lync Server 2013
1800 ULEARN (853 276) www.ddls.com.au Microsoft 20337 - Enterprise Voice and Online Services with Microsoft Lync Server 2013 Length 5 days Price $4290.00 (inc GST) Version B Overview This five-day instructor-led
More informationH.323 / B-ISDN Signalling Interoperability
ECMA Technical Report TR/73 December 1998 Standardizing Information and Communication Systems H.323 / B-ISDN Signalling Interoperability Phone: +41 22 849.60.00 - Fax: +41 22 849.60.01 - URL: http://www.ecma.ch
More informationSecurity and Lawful Intercept In VoIP Networks. Manohar Mahavadi Centillium Communications Inc. Fremont, California
Security and Lawful Intercept In VoIP Networks Manohar Mahavadi Centillium Communications Inc. Fremont, California Agenda VoIP: Packet switched network VoIP devices VoIP protocols Security and issues in
More informationKeep Calm and Call On! IBM Sametime Communicate Softphone Made Simple. Frank Altenburg, IBM
Keep Calm and Call On! IBM Sametime Communicate Softphone Made Simple Frank Altenburg, IBM Agenda Voice and Video an effective way to do business! Sametime Softphone Computer is your phone! Sametime Voice
More informationITU-APT Workshop on NGN Planning March 2007, Bangkok, Thailand
ITU-APT Workshop on NGN Planning 16 17 March 2007, Bangkok, Thailand 1/2 Riccardo Passerini, ITU-BDT 1 Question 19-1/2: Strategy for migration from existing to next-generation networks (NGN) for developing
More informationAvaya Communication Manager Network Region Configuration Guide
Communication Manager 2.1 Avaya Labs ABSTRACT This application note is a tutorial on network regions. Two basic configuration examples are covered in detail, along with explanations of the concepts and
More informationConfiguring VoIP Gatekeeper Registration Delay Operations
Configuring VoIP Gatekeeper Registration Delay Operations This document describes how to configure an Cisco IOS IP Service Level Agreements (SLAs) Voice over IP (VoIP) gatekeeper registration delay operation
More informationNon. Interworking between SIP and H.323, MGCP, Megaco/H.248 LS'LDORJ,QF 7HFKQRORJ\ 'ULYH 6XLWH 3KRQH )D[
Non Interworking between SIP and H.323, MGCP, Megaco/H.248 7HFKQRORJ\ 'ULYH 6XLWH 3KRQH )D[ 6DQ -RVH &$ 86$ 85/ ZZZLSGLDORJFRP Joon Maeng Jörg Ott jmaeng@ipdialog.com jo@ipdialog.com The Starting Point
More informationZ24: Signalling Protocols
Z24: Signalling Protocols Mark Handley H.323 ITU protocol suite for audio/video conferencing over networks that do not provide guaranteed quality of service. H.225.0 layer Source: microsoft.com 1 H.323
More informationCCVP CIPT2 Quick Reference
Introduction...3...4 Centralized Call Processing Redundancy...11 CCVP CIPT2 Quick Reference Bandwidth Management and Call Admission Control...17 Applications for Multisite Deployments...21 Security...31
More informationSynopsis of Basic VoIP Concepts
APPENDIX B The Catalyst 4224 Access Gateway Switch (Catalyst 4224) provides Voice over IP (VoIP) gateway applications for a micro branch office. This chapter introduces some basic VoIP concepts. This chapter
More informationAdvanced Networking Voice over IP: Introduction and H.323 standard
Advanced Networking Voice over IP: Introduction and H.323 standard Renato Lo Cigno VoIP: Integrating Services Voice on IP Networks is just another application Nothing special or specialized as traditional
More informationVoice Over IP. Marko Leppänen Helsinki University of Technology Department of Computer Science Abstract
Voice Over IP Marko Leppänen Helsinki University of Technology Department of Computer Science Marko.Leppanen@hut.fi Abstract Voice Over IP (VoIP) has been in heavy investigation during recent years. VoIP
More informationConfiguring T.38 Fax Relay
Configuring T38 Fax Relay Configuring T38 Fax Relay, page 1 Configuring T38 Fax Relay This chapter describes configuration for T38 fax relay on an IP network T38 is an ITU standard that defines how fax
More informationProtocol & Port Information for the deployment of. and. within IP Networks
Protocol & Port Information for the deployment of and within IP Networks SpliceCom Ltd The Hall Business Centre Berry Lane Chorleywood Herts WD3 5EX Phone: 01923-287700 Fax: 01923-287722 E-mail: info@splicecom.com
More informationCisco Webex Cloud Connected Audio
White Paper Cisco Webex Cloud Connected Audio Take full advantage of your existing IP telephony infrastructure to help enable a Webex integrated conferencing experience Introduction Cisco Webex Cloud Connected
More informationMobile MOUSe CONVERGENCE+ ONLINE COURSE OUTLINE
Mobile MOUSe CONVERGENCE+ ONLINE COURSE OUTLINE COURSE TITLE CONVERGENCE+ COURSE DURATION 13 Hour(s) of Self-Paced Interactive Training COURSE OVERVIEW This course will prepare you to design, implement
More informationConfiguration Guide IP-to-IP Application
Multi-Service Business Gateways Enterprise Session Border Controllers VoIP Media Gateways Configuration Guide IP-to-IP Application Version 6.8 November 2013 Document # LTRT-40004 Configuration Guide Contents
More informationCHAPTER. Introduction. Last revised on: February 13, 2008
CHAPTER 1 Last revised on: February 13, 2008 The Cisco Unified Communications System delivers fully integrated communications by enabling data, voice, and video to be transmitted over a single network
More informationINTERNATIONAL INTERCONNECTION FORUM FOR SERVICES OVER IP. (i3 FORUM) Interoperability Test Plan for International Voice services
INTERNATIONAL INTERCONNECTION FORUM FOR SERVICES OVER IP (i3 FORUM) Workstream Technical Aspects Workstream Operations Interoperability Test Plan for International Voice services (Release 3.0) May 2010
More informationInterworking Signaling Enhancements for H.323 and SIP VoIP
Interworking Signaling Enhancements for H.323 and SIP VoIP This feature module describes enhancements to H.323 and Session Initiation Protocol (SIP) signaling when interworking with ISDN, T1 channel associated
More informationMobile MOUSe IMPLEMENTING VOIP ONLINE COURSE OUTLINE
Mobile MOUSe IMPLEMENTING VOIP ONLINE COURSE OUTLINE COURSE TITLE IMPLEMENTING VOIP COURSE DURATION 13 Hour(s) of Self-Paced Interactive Training COURSE OVERVIEW The Cisco Implementing VoIP course validates
More informationEEG453 Multimedia systems Dr. Mohab Mangoud University of Bahrain. Lecture # (3) Communication networks
EEG453 Multimedia systems Dr. Mohab Mangoud University of Bahrain Lecture # (3) Introduction to Multimedia Communication networks Elements of Multimedia Systems Two key communication modes Person-to-person
More informationBT SIP Trunk Configuration Guide
CUCM 9.1 BT SIP Trunk Configuration Guide This document covers service specific configuration required for interoperability with the BT SIP Trunk service. Anything which could be considered as normal CUCM
More informationAP-GS1004 TM GSM Gateway
AP-GS1004 TM GSM Gateway High Performance VoIP Gateway Solution ITPP (IP multimedia Telephony Provision Platform) AddPac Technology 2011, Sales and Marketing www.addpac.com IP Multimedia Telephony Provision
More informationKommunikationssysteme [KS]
Kommunikationssysteme [KS] Dr.-Ing. Falko Dressler Computer Networks and Communication Systems Department of Computer Sciences University of Erlangen-Nürnberg http://www7.informatik.uni-erlangen.de/~dressler/
More informationVolume SUPPORTING THE CONVERGED NETWORK. Mark A. Miller, P.E. President DigiNet Corporation. A technical briefing from: July 2002
Volume 6 SUPPORTING THE CONVERGED NETWORK Mark A. Miller, P.E. President DigiNet Corporation A technical briefing from: July 2002 Table of Contents Executive Summary i 1. The Challenge of Supporting Converged
More informationOverview. Slide. Special Module on Media Processing and Communication
Overview Review of last class Protocol stack for multimedia services Real-time transport protocol (RTP) RTP control protocol (RTCP) Real-time streaming protocol (RTSP) SIP Special Module on Media Processing
More informationH.323 in Telecommunications
Teknillinen Korkeakoulu Teletekniikan laboratorio S-38.128 Teletekniikan erikoistyö H.323 in Telecommunications Tekijä: Kai Väänänen 44380T kai.vaananen@sonera.fi Ohjaaja: Vesa Kosonen Jätetty: 17.8.1999
More informationCisco HSI System Overview
CHAPTER 1 Introduction This chapter provides an overview of the Cisco H.323 Signaling Interface (HSI) system and subsystems and contains the following sections: Cisco HSI Overview, page 1-1 Cisco HSI System
More informationImpact of Voice Coding in Performance of VoIP
Impact of Voice Coding in Performance of VoIP Batoul Alia Baker Koko 1, Dr. Mohammed Abaker 2 1, 2 Department of Communication Engineering, Al-Neelain University Abstract: Voice over Internet Protocol
More informationVoIP Protocols and QoS
Announcements I. Times have been posted for demo slots VoIP Protocols and QoS II. HW5 and HW6 solutions have been posted HW6 being graded Internet Protocols CSC / ECE 573 Fall, 2005 N. C. State University
More informationMultimedia Networking
CMPT765/408 08-1 Multimedia Networking 1 Overview Multimedia Networking The note is mainly based on Chapter 7, Computer Networking, A Top-Down Approach Featuring the Internet (4th edition), by J.F. Kurose
More information