Understanding Latency in IP Telephony

Save this PDF as:
 WORD  PNG  TXT  JPG

Size: px
Start display at page:

Download "Understanding Latency in IP Telephony"

Transcription

1 Understanding Latency in IP Telephony By Alan Percy, Senior Sales Engineer Brooktrout Technology, Inc. 410 First Avenue Needham, MA Phone: (781) Fax: (781) Internet:

2 Abstract With an increasing interest in implementing and deploying IP Telephony applications, there is a rising need to understand the cause and effect of latency in the deployed system. This paper addresses these needs by reviewing the effect of latency on human conversations, analyzing the system components that incur the latency, and methods of managing the latency to maintain sufficient quality of service. Introduction When building and deploying an IP Telephony solution, many different technical attributes will affect the quality of the final system. These attributes include the selection of voice coding (vocoder) algorithm, the system latency, link dependability, and others. Assuming the IP Telephony solution will use an industry standard vocoder, latency becomes the most important attribute that designers have control over and can have the greatest effect on the quality of service. Latency is the time delay incurred in speech by the IP Telephony system. Latency is typically measured in milliseconds from the moment that the speaker utters a word until the listener actually hears the word. This is termed as mouth-to-ear latency or the oneway latency that the users would realize when using the system. The round-trip latency is the sum of the two one-way latency figures that make up a telephone call. In the traditional Public Switched Telephone Network, the round-trip latency for domestic calls is virtually always under 150 milliseconds. At these levels, the latency is not noticeable to most people. Many international calls (especially calls carried via satellite) will have round-trip latency figures that can exceed 1 second, which can be very annoying for users. What is the effect of latency? A telephone conversation between two people depends on the timing of the speech more than most people realize. Most conversations include little utterances by the listener that serve as acknowledgements back to the speaker, confirming that the listener is actively engaged in the conversation. Listen to yourself carefully the next time you are on the phone with someone. Notice the small utterances that you will naturally say even though the other party is doing most of the talking. If you remove these utterances, the speaker will stop to wait for your feedback. If you delay the utterances, they will come to the speaker at the wrong time, resulting in confusion and an interruption to the flow of conversation. Try delaying or stopping these utterances and notice what happens to your telephone conversation. Brooktrout Technology, Inc. 2

3 What is considered an acceptable amount of latency? As with most human factors considerations, everyone has his or her own opinion on this issue, but based on feedback Brooktrout has received from early adopters of IP Telephony systems, there is a definite maximum latency that will be tolerated by users. The exact amount of latency that will be tolerated by users is hard to define because users will balance the degradation of added latency against the perceived value added by the system. Wireless telephone services are prime examples of where reduced connection quality will be accepted when balanced against the added value of high mobility. Assuming that an IP Telephony system is primarily used in a cost-reduction or toll bypass application, Brooktrout has developed the following chart, showing a relationship between user perceived link quality vs. the amount of one-way latency. Other applications with higher perceived value will surely accommodate greater latency figures. Perceived Link Quality Excellent Good Poor Unacceptable One-way latency in milliseconds Figure 1: Quality Perception vs. Latency As you can see, the user perception of the link quality deteriorates as the one-way latency exceeds 150 milliseconds. If the one-way latency exceeds 450 milliseconds, holding a conversation is very difficult and the latency becomes very annoying. If given a choice, most callers would choose to use a telephone line with less than 200 milliseconds of latency. This gives you a target figure for your IP telephony system. Keep the one-way latency under 200 milliseconds. Even if a caller can get great sound quality and much lower cost with your solution, that caller will typically go elsewhere if the latency is excessive. What are the causes of latency? Generally, an IP Telephony system is constructed using gateways to interface existing telephone equipment together over a wide-area-network (WAN). This typical deployment is shown in Figure 2, which shows the two end-point telephones connected to a WAN via gateways and routers. Brooktrout Technology, Inc. 3

4 Even if a system integrates the Gateway functionality into either the telephones or the router equipment, the same basic processing must take place. For all practical purposes, you can envision that every call using IP telephony requires two gateways, only the location of the gateways change. Telephone Gateway Incurred Latency Telephone Gateway Network Incurred Latency Gateway Router WAN Router Figure 2: Typical IP Telephony System Latency in an IP Telephony system is introduced by two primary sources. Some of the latency is incurred in the IP Telephony Gateways at either end, and the remainder is incurred by the IP network that connects the two gateways. Since latency is cumulative, any latency introduced by a component in an IP Telephony system will directly affect the total latency experienced by the user. Gateway-Incurred Latency Let us take a look inside a gateway and examine the origin of the latency introduced by the gateway. A high-level block diagram of the processing within a gateway is shown in Figure 3. The block diagram shows the high-level functions that occur in both gateway systems. The interface to the end-point telephone system is on the left side and the interface to the network is on the right side. Following the path of a voice conversation from one telephone to another, each of the functional blocks has an effect on the gateway-incurred latency. Each of these functions and the associated latency contribution is described in more detail in sections that follow below. Brooktrout Technology, Inc. 4

5 Network Interface Digital Signal Processing Packet Handling PCM DSP Coding Frames Buffering and Packetization Jitter Buffer Network Interface T1, E1, PRI, Loop-Start TCP/IP Protocol Stack T1 Ethernet Network Interface IP Figure 3: Gateway Processing Network Interface Latency The network interface in a gateway includes any hardware or software that connects the Gateway to the telephone system or network. The typical network interface frames and converts the network-side digitized audio PCM data streams into the internal PCM bus for transport to the DSP. There is typically very little latency induced in this process, with typical maximums well below 1 millisecond. Digital Signal Processing Latency The digital signal processing that occurs in an IP Telephony gateway is one of the more complex functions of the gateway. This functionality is typically achieved through the use of dedicated digital signal processor (DSP) hardware and associated software algorithms that compress or decompress the speech, detect tones, detect silence, generate tones, generate comfort noise, and cancel echo. This entire collection of processing is called voice coding or vocoding. Brooktrout Technology, Inc. 5

6 Figure 4: DSP Voice Compression Subsystem Framing Latency To most efficiently perform vocoding, DSP implementations depend on processing entire frames (or batches) of data at one time. This allows the DSP to use special instructions that result in the high efficiencies needed for high-density IP Telephony applications. Next Sample Frame Figure 5: Framing Process The side effect of processing data in frames is that none of the data can be processed until the frame is completely full. Since the rate that the digitized audio comes in from the telephone network is typically at a fixed rate of 8,000 samples per second, the size of the frame used to process the data will directly affect the amount of latency. A 100 sample frame would take 12.5 milliseconds to fill, while a 1000 sample frame would take 125 milliseconds to fill. Deciding on the frame sizes is a compromise: the larger the frame, the greater DSP efficiency, but with that comes greater latency. Fortunately (or unfortunately depending on your point of view), you don t need to make this decision. Each of the standard voice coding methods uses a standard frame size. The maximum latency incurred by the framing process is directly dependent on the selection of vocoder. Voice Coder Bandwidth in bits/sec Frame Duration in milliseconds Frame Size In bytes G.711* G G.729a SX SX Table 1: Voice Coder Frame Sizes * While G.711 is technically not a vocoder, we list it here for comparative purposes. G.711 data streams have greater flexibility when specifying frame sizes, the figures listed here are just one example. Brooktrout Technology, Inc. 6

7 Processing Time After the collection of an entire frame is completed, the DSP algorithms must be run on the newly created frame. The time required to complete the processing varies considerably, but never exceeds the frame collection time. (If it did, the DSP would never complete processing one frame before the next frame arrived). Since most high-density IP Telephony gateway systems will process multiple channels of voice on each DSP, calculating the latency induced by processing the coding or decoding of the speech is rather complex. In this situation, each DSP will process some number of frames from different channels, one after the other in a sequential process. This means that the first channel will be completed much earlier than the later channels. In a fully utilized DSP subsystem, the processing for the last channel would occur just before the data for the first channel begins to arrive. As a result, you can t use the number of milliseconds used by the DSP to vocode any single channel in calculating the latency added by processing. Instead the latency incurred due to processing is typically specified as the frame size in milliseconds. This means the total latency from framing and processing can be no more than twice the frame size. Packet Handling Latency Between the DSP processing and passing the data to the WAN, there are a number of packet handling processes that will occur that will affect the system latency. Buffering After the voice coding processes, many IP Telephony systems then further buffer the resulting compressed voice data frames before passing them to the network software. This additional buffering many times is done to reduce the number of times the DSP needs to communicate to the main CPU in the gateway. In other situations it is done to make the result of coding algorithms fit into one common frame duration (not length). For example: If a system is running with G on one channel and G.729a on another, the frame sizes are different (see Table 1). A system may, by design, collect three G.729a frames into one buffer for every G frame. This would allow the system to transfer one buffer every 30 milliseconds, irrespective of the coding algorithm. Packetization As the coded voice is being prepared for transport over the WAN, it needs to be assembled into packets. This process is typically done by the TCP/IP protocol stack, using UDP (User Datagram Protocol) and RTP (Real Time Protocol). The selection of these protocols improves timely delivery of the voice data and eliminates the overhead of transmission acknowledgements and retries. Looking inside a typical IP telephony data packet, each packet starts with an IP, UDP, and RTP header that totals 40 bytes. The header contains the source and destination IP Brooktrout Technology, Inc. 7

8 addresses, the IP port number, packet sequence number and other protocol information needed to properly transport the data. After the IP header, one or more frames of coded voice data would follow. The decision of whether to pack more than one frame of data into a single packet is an important consideration for every IP Telephony system. If a system was using the G coder (which produces 24 byte frames every 30 milliseconds), each packet would have 40 bytes of header and 24 bytes of data. That would make the header 167% of the voice data payload! IP Header 20 Bytes UDP Header 8 Bytes RTP Header 12 Bytes Voice Data 40 Bytes Figure 6: Anatomy of an IP Telephony Packet The most common way to reduce the inefficiency of the IP packet overhead is to put more than one coded voice frame per IP packet. If two frames are passed per packet, the overhead figure drops to 83%, but with the side effect of adding yet another frame period of latency. This trade-off is another compromise that needs to be considered when deploying an IP Telephony system. An interesting trick that can reduce the overhead, but not increase the latency in systems is to let voice frames from other channels piggyback in the same packet. When voice from another channel in the originating gateway is going to the same destination gateway, the data can be combined into a single packet. This trick is not supported by the standard H.323 protocol, but can be used by proprietary solutions to improve efficiency. Jitter Buffer Latency Because IP networks cannot guarantee the delivery time of data packets (or their order, for that matter), the data will arrive at a very inconsistent rate. The variability in the arrival rate of data is called jitter. During the voice decoding process (data traveling from the network to a telephone), all systems need to compensate for jitter in the data arriving from the network. To compensate for jitter, most systems buffer at least one packet of data from the network before passing it to the DSP. Having these jitter buffers can significantly reduce the occurrence of data starvation and ensure the timing is correct when sending data to the DSP. Without jitter buffers, there is a very good chance that gaps in the data would be heard in the resulting speech. Brooktrout Technology, Inc. 8

9 The side effect of jitter buffers is (you guessed it) more latency. The larger the jitter buffers, the more tolerant the system is of jitter in the data from the network, but the additional buffering causes more latency. Network-Incurred Latency Now that the Gateway has the voice data compressed and packetized, the data is passed to the Wide Area Network for transport to the far-end gateway. Passing data over the WAN introduces yet another set of potential latency additions that will affect the total latency. Media Access Latency For each point where data is passed to or from physical media, there is a Media Access Delay added to the total latency. Since there are many different physical media used to inter-connect gateways, routers, and other networking equipment, these delays need to be considered. If a connection to the WAN were to use low-speed serial connections like RS-232 or dialup modems, the transfer time of the data can add latency to transmission at far greater amounts than higher speed media. Example: If a WAN link were using a 28,800 bits per second connection, each byte transferred requires.35 milliseconds. This would yield a total transfer time for a 100 byte packet of 35 milliseconds. If instead of an RS-232 connection, the WAN interface were using a 1.54 megabits per second dedicated T-1 connection, the transmission latency for the same 100 byte packet drops to.5 milliseconds. If instead of a RS-232 or T1 connection, the WAN interface were using a 100 megabits per second Ethernet connection, the transmission latency for the same 100 byte packet drops even further to.008 milliseconds. Although the transfer time for Ethernet is very fast, remember that Ethernet is a Carrier Sense Multiple Access (CSMA) media, which means that every computer on the same network has to share the same physical layer. Any collisions or congestion results in increased latency. Routing Latency Since IP is a routed protocol, all packets simply have a source and destination IP address. This simple design requires that routers examine each packet and, depending on the destination address, direct the packet via the proper route. The queuing logic used by most routers was designed before the concept of IP Telephony existed and therefore has certain weaknesses with respect to the real-time nature of IP Telephony. Many existing routers use Best-Effort routing, which is far from ideal for latency-sensitive voice traffic. Brooktrout Technology, Inc. 9

10 The key missing piece is a priority attribute, the absence of which results in the router delaying all data during congestion situations, irrespective of the application. The Resource Reservation Protocol (RSVP) has been defined by the IETF as a means of creating and managing resouces within routers and gateways. RSVP allows a gateway-togateway connection to establish a guaranteed bandwidth commitment on the intermediate network equipment, which would dramatically reduce the variability in packet delivery and improve the quality of the service. Since RSVP is a relatively recent development, the vast majority of existing equipment that is deployed in the public network cannot yet support RSVP. At this point in time, RSVP can only be used in closed systems where the network administrator has control over the equipment from end-to-end. Further developments by the WAN carriers in the future may change this situation. Firewalls and Proxy Servers Many networks use firewalls or proxy servers to provide security between the corporate LAN and the Internet. Since both of these security devices must examine every incoming and outgoing IP packet, they can incur a sizeable amount of latency, so their use is almost always avoided in IP Telephony applications. Packet Filter features that are built into routers are typically simple in design compared to stand-alone firewalls or proxy servers and therefore can typically offer some network security without significant added latency. Stand-alone firewalls or proxy servers must receive, decode, examine, validate, encode, and send every packet. All of which comes with increased latency. Ask your router vendor about packet filter capabilities and specifically inquire about the latency incurred by the router. Proxy servers provide even greater network security, but at an even greater cost in network latency. It is not uncommon for a busy proxy server to incur over 500 milliseconds of latency. This is not a problem to the web-browsing applications for which proxy servers were designed, but it is clearly unacceptable for real-time media. How can I manage latency? Managing the latency in a deployed IP Telephony system is key to the success of the resulting service. Some key steps that can be taken to reduce and manage the latency are: Know the sources of latency in your system (do a latency budget). Having a latency budget helps you set a target and identify areas that can be improved. Without a complete understanding of the various components that contribute to the total latency, you won t have a clear picture of where latency can be trimmed. Use routing equipment that supports prioritization of selected ports or provides RSVP to guarantee a certain level of packet throughput. Carefully selecting and managing your routing equipment is key to the success of your deployed system. Ensure that your network has sufficient bandwidth to avoid congestion. An important tool in managing bandwidth and congestion is the selection of proper vocoders. Use IP telephony platforms that allow dynamic switching of vocoders on a call-by-call Brooktrout Technology, Inc. 10

11 basis or even within a call. This will allow the network to respond to available bandwidth conditions in real-time. Stay away from equipment and media that you do not have control over (the public Internet). Having the ability to set priorities, meter link throughput, and adjust priorities are all required in maintaining a minimum of latency in an IP Telephony system. If you use a network carrier, ask for a guaranteed route. This will eliminate many time-of-day variables in the system. Reduce packet overhead. If feasible, use piggybacking in your design to send multiple channels of voice data to the same destination. Efficient use of piggybacking can reduce total network traffic by over 50%, leaving more room for growth. A Sample Latency Budget Just like in your personal finances, create a latency budget with a target figure in mind. In the example below, the one-way latency is calculated. Source Latency (in milliseconds) Network Interface 1 (1.54 Mbps T1) Framing 30 (G.723.1) Processing Time 10 (worst case) Buffering 0 (no additional buffering) Packetization 30 (two frames per packet) Media Access Delay 10 (5 2msec hops) Routing 50 (router dependent) Jitter Buffering 30 msec (one buffer) Total One-Way Latency 161 msec Table 2: Example Latency Budget Use the latency budget to target areas that can be reduced or critical bottlenecks. Brooktrout Technology, Inc. 11

12 About Brooktrout Technology As you can see, building an effective and high-quality IP Telephony solution requires that each component in the system be carefully selected to avoid unnecessary and irritating latency in voice calls. Brooktrout Technology s TR2001 helps ensure that your final solution minimizes latency by closely coupling some of the vocoding and network access portions of an IP Telephony solution. Some of the advantages of the Brooktrout TR2001 are: Industry standard vocoding algorithms that are optimized to minimize processing latency Efficient schemes that reduce the accumulative latency effect of buffering. Efficient driver for reduced host-processor load Real-time embedded TCP/IP protocol Single PCI slot solution that includes all network and telephony interfaces A power-efficient board that runs cooler than other solutions. Support for dynamic vocoder selection APIs for precise control of telephony, vocoders and H.323 protocol stack In addition to providing high-performance telephony platforms, Brooktrout also works closely with development partners to help them optimize and refine their systems. Brooktrout Technology, Inc. 12

Requirements of Voice in an IP Internetwork

Requirements of Voice in an IP Internetwork Requirements of Voice in an IP Internetwork Real-Time Voice in a Best-Effort IP Internetwork This topic lists problems associated with implementation of real-time voice traffic in a best-effort IP internetwork.

More information

Network administrators must be aware that delay exists, and then design their network to bring end-to-end delay within acceptable limits.

Network administrators must be aware that delay exists, and then design their network to bring end-to-end delay within acceptable limits. Delay Need for a Delay Budget The end-to-end delay in a VoIP network is known as the delay budget. Network administrators must design a network to operate within an acceptable delay budget. This topic

More information

Clearing the Way for VoIP

Clearing the Way for VoIP Gen2 Ventures White Paper Clearing the Way for VoIP An Alternative to Expensive WAN Upgrades Executive Overview Enterprises have traditionally maintained separate networks for their voice and data traffic.

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

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

Application Notes. Introduction. Sources of delay. Contents. Impact of Delay in Voice over IP Services VoIP Performance Management.

Application Notes. Introduction. Sources of delay. Contents. Impact of Delay in Voice over IP Services VoIP Performance Management. Application Notes Title Series Impact of Delay in Voice over IP Services VoIP Performance Management Date January 2006 Overview This application note describes the sources of delay in Voice over IP services,

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

QoS issues in Voice over IP

QoS issues in Voice over IP COMP9333 Advance Computer Networks Mini Conference QoS issues in Voice over IP Student ID: 3058224 Student ID: 3043237 Student ID: 3036281 Student ID: 3025715 QoS issues in Voice over IP Abstract: This

More information

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 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

More information

Broadband Networks. Prof. Dr. Abhay Karandikar. Electrical Engineering Department. Indian Institute of Technology, Bombay. Lecture - 29.

Broadband Networks. Prof. Dr. Abhay Karandikar. Electrical Engineering Department. Indian Institute of Technology, Bombay. Lecture - 29. Broadband Networks Prof. Dr. Abhay Karandikar Electrical Engineering Department Indian Institute of Technology, Bombay Lecture - 29 Voice over IP So, today we will discuss about voice over IP and internet

More information

VoIP Bandwidth Considerations - design decisions

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

More information

Troubleshooting Common Issues in VoIP

Troubleshooting Common Issues in VoIP Troubleshooting Common Issues in VoIP 2014, SolarWinds Worldwide, LLC. All rights reserved. Voice over Internet Protocol (VoIP) Introduction Voice over IP, or VoIP, refers to the delivery of voice and

More information

Voice over IP Technology

Voice over IP Technology Voice over Technology 2/4/2004 At World Telecom Labs, we have identified several limitations of the existing VO technology available on the market. We have addressed these one by one and finally produced

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

Glossary of Terms and Acronyms for Videoconferencing

Glossary of Terms and Acronyms for Videoconferencing Glossary of Terms and Acronyms for Videoconferencing Compiled by Irene L. Ferro, CSA III Education Technology Services Conferencing Services Algorithm an algorithm is a specified, usually mathematical

More information

Fundamentals of VoIP Call Quality Monitoring & Troubleshooting. 2014, SolarWinds Worldwide, LLC. All rights reserved. Follow SolarWinds:

Fundamentals of VoIP Call Quality Monitoring & Troubleshooting. 2014, SolarWinds Worldwide, LLC. All rights reserved. Follow SolarWinds: Fundamentals of VoIP Call Quality Monitoring & Troubleshooting 2014, SolarWinds Worldwide, LLC. All rights reserved. Introduction Voice over IP, or VoIP, refers to the delivery of voice and multimedia

More information

5. DEPLOYMENT ISSUES Having described the fundamentals of VoIP and underlying IP infrastructure, let s address deployment issues.

5. DEPLOYMENT ISSUES Having described the fundamentals of VoIP and underlying IP infrastructure, let s address deployment issues. 5. DEPLOYMENT ISSUES Having described the fundamentals of VoIP and underlying IP infrastructure, let s address deployment issues. 5.1 LEGACY INTEGRATION In most cases, enterprises own legacy PBX systems,

More information

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

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

More information

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

Introduction to Packet Voice Technologies and VoIP

Introduction to Packet Voice Technologies and VoIP Introduction to Packet Voice Technologies and VoIP Cisco Networking Academy Program Halmstad University Olga Torstensson 035-167575 olga.torstensson@ide.hh.se IP Telephony 1 Traditional Telephony 2 Basic

More information

Voice over IP (VoIP) for Telephony. Advantages of VoIP Migration for SMBs BLACK BOX. 724-746-5500 blackbox.com

Voice over IP (VoIP) for Telephony. Advantages of VoIP Migration for SMBs BLACK BOX. 724-746-5500 blackbox.com Voice over IP (VoIP) for Telephony Advantages of VoIP Migration for SMBs BLACK BOX Hybrid PBX VoIP Gateways SIP Phones Headsets 724-746-5500 blackbox.com Table of Contents Introduction...3 About Voice

More information

Encapsulating Voice in IP Packets

Encapsulating Voice in IP Packets Encapsulating Voice in IP Packets Major VoIP Protocols This topic defines the major VoIP protocols and matches them with the seven layers of the OSI model. Major VoIP Protocols 15 The major VoIP protocols

More information

Voice Over IP Per Call Bandwidth Consumption

Voice Over IP Per Call Bandwidth Consumption Over IP Per Call Bandwidth Consumption Interactive: This document offers customized voice bandwidth calculations with the TAC Bandwidth Calculator ( registered customers only) tool. Introduction Before

More information

12 Quality of Service (QoS)

12 Quality of Service (QoS) Burapha University ก Department of Computer Science 12 Quality of Service (QoS) Quality of Service Best Effort, Integrated Service, Differentiated Service Factors that affect the QoS Ver. 0.1 :, prajaks@buu.ac.th

More information

Network Simulation Traffic, Paths and Impairment

Network Simulation Traffic, Paths and Impairment Network Simulation Traffic, Paths and Impairment Summary Network simulation software and hardware appliances can emulate networks and network hardware. Wide Area Network (WAN) emulation, by simulating

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

Application Note How To Determine Bandwidth Requirements

Application Note How To Determine Bandwidth Requirements Application Note How To Determine Bandwidth Requirements 08 July 2008 Bandwidth Table of Contents 1 BANDWIDTH REQUIREMENTS... 1 1.1 VOICE REQUIREMENTS... 1 1.1.1 Calculating VoIP Bandwidth... 2 2 VOIP

More information

Question: 3 When using Application Intelligence, Server Time may be defined as.

Question: 3 When using Application Intelligence, Server Time may be defined as. 1 Network General - 1T6-521 Application Performance Analysis and Troubleshooting Question: 1 One component in an application turn is. A. Server response time B. Network process time C. Application response

More information

Unified Communications Group. Designing for Adoption: Real-time Audio in the Real World

Unified Communications Group. Designing for Adoption: Real-time Audio in the Real World Unified Communications Group Designing for Adoption: Real-time Audio in the Real World Information in this document, including URL and other Internet Web site references, is subject to change without notice.

More information

Chapter 3 ATM and Multimedia Traffic

Chapter 3 ATM and Multimedia Traffic In the middle of the 1980, the telecommunications world started the design of a network technology that could act as a great unifier to support all digital services, including low-speed telephony and very

More information

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 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

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

Performance Evaluation of AODV, OLSR Routing Protocol in VOIP Over Ad Hoc

Performance Evaluation of AODV, OLSR Routing Protocol in VOIP Over Ad Hoc (International Journal of Computer Science & Management Studies) Vol. 17, Issue 01 Performance Evaluation of AODV, OLSR Routing Protocol in VOIP Over Ad Hoc Dr. Khalid Hamid Bilal Khartoum, Sudan dr.khalidbilal@hotmail.com

More information

Voice over IP. Overview. What is VoIP and how it works. Reduction of voice quality. Quality of Service for VoIP

Voice over IP. Overview. What is VoIP and how it works. Reduction of voice quality. Quality of Service for VoIP Voice over IP Andreas Mettis University of Cyprus November 23, 2004 Overview What is VoIP and how it works. Reduction of voice quality. Quality of Service for VoIP 1 VoIP VoIP (voice over IP - that is,

More information

ESSENTIALS. Understanding Ethernet Switches and Routers. April 2011 VOLUME 3 ISSUE 1 A TECHNICAL SUPPLEMENT TO CONTROL NETWORK

ESSENTIALS. Understanding Ethernet Switches and Routers. April 2011 VOLUME 3 ISSUE 1 A TECHNICAL SUPPLEMENT TO CONTROL NETWORK VOLUME 3 ISSUE 1 A TECHNICAL SUPPLEMENT TO CONTROL NETWORK Contemporary Control Systems, Inc. Understanding Ethernet Switches and Routers This extended article was based on a two-part article that was

More information

APTA TransiTech Conference Communications: Vendor Perspective (TT) Phoenix, Arizona, Tuesday, 3.19.13. VoIP Solution (101)

APTA TransiTech Conference Communications: Vendor Perspective (TT) Phoenix, Arizona, Tuesday, 3.19.13. VoIP Solution (101) APTA TransiTech Conference Communications: Vendor Perspective (TT) Phoenix, Arizona, Tuesday, 3.19.13 VoIP Solution (101) Agenda Items Introduction What is VoIP? Codecs Mean opinion score (MOS) Bandwidth

More information

Performance Evaluation of VoIP Services using Different CODECs over a UMTS Network

Performance Evaluation of VoIP Services using Different CODECs over a UMTS Network Performance Evaluation of VoIP Services using Different CODECs over a UMTS Network Jianguo Cao School of Electrical and Computer Engineering RMIT University Melbourne, VIC 3000 Australia Email: j.cao@student.rmit.edu.au

More information

VegaStream Information Note Considerations for a VoIP installation

VegaStream Information Note Considerations for a VoIP installation VegaStream Information Note Considerations for a VoIP installation To get the best out of a VoIP system, there are a number of items that need to be considered before and during installation. This document

More information

WHITEPAPER. Quality of Service Testing in the VoIP Environment

WHITEPAPER. Quality of Service Testing in the VoIP Environment Quality of Service Testing in the VoIP Environment March 2005 In recent years, the business world has reaped tremendous benefits from the many exciting products and applications made possible by the marriage

More information

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) VoIP Analysis Fundamentals with Wireshark Phill Shade (Forensic Engineer Merlion s Keep Consulting) 1 Phillip D. Shade (Phill) phill.shade@gmail.com Phillip D. Shade is the founder of Merlion s Keep Consulting,

More information

Voice Over Internet Protocol(VoIP)

Voice Over Internet Protocol(VoIP) Voice Over Internet Protocol(VoIP) By Asad Niazi Last Revised on: March 29 th, 2004 SFWR 4C03 Major Project Instructor: Dr. Kartik Krishnan 1. Introduction The telecommunications companies around the world

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

Discussion Paper Category 6 vs Category 5e Cabling Systems and Implications for Voice over IP Networks

Discussion Paper Category 6 vs Category 5e Cabling Systems and Implications for Voice over IP Networks Discussion Paper Category 6 vs Category 5e Cabling Systems and Implications for Voice over IP Networks By Galen Udell Belden CDT Networking 2006 Category 6 vs Category 5e Cabling Systems and Implications

More information

Delivering reliable VoIP Services

Delivering reliable VoIP Services QoS Tips and Tricks for VoIP Services: Delivering reliable VoIP Services Alan Clark CEO, Telchemy alan.d.clark@telchemy.com 1 Objectives Clear understanding of: typical problems affecting VoIP service

More information

VOICE OVER IP AND NETWORK CONVERGENCE

VOICE OVER IP AND NETWORK CONVERGENCE POZNAN UNIVE RSITY OF TE CHNOLOGY ACADE MIC JOURNALS No 80 Electrical Engineering 2014 Assaid O. SHAROUN* VOICE OVER IP AND NETWORK CONVERGENCE As the IP network was primarily designed to carry data, it

More information

Simple Voice over IP (VoIP) Implementation

Simple Voice over IP (VoIP) Implementation Simple Voice over IP (VoIP) Implementation ECE Department, University of Florida Abstract Voice over IP (VoIP) technology has many advantages over the traditional Public Switched Telephone Networks. In

More information

Knowledge Is Power: Do what s best for the client.

Knowledge Is Power: Do what s best for the client. Knowledge Is Power: Do what s best for the client. 1. Understanding Voice and Data Differences Even when they are carried on the same network, voice traffic and data traffic cannot be handled the same

More information

Troubleshooting VoIP and Streaming Video Problems

Troubleshooting VoIP and Streaming Video Problems Using the ClearSight Analyzer to troubleshoot the top five VoIP problems and troubleshoot Streaming Video With the prevalence of Voice over IP and Streaming Video applications within the enterprise, it

More information

Protocols. Packets. What's in an IP packet

Protocols. Packets. What's in an IP packet Protocols Precise rules that govern communication between two parties TCP/IP: the basic Internet protocols IP: Internet Protocol (bottom level) all packets shipped from network to network as IP packets

More information

Technote. SmartNode Quality of Service for VoIP on the Internet Access Link

Technote. SmartNode Quality of Service for VoIP on the Internet Access Link Technote SmartNode Quality of Service for VoIP on the Internet Access Link Applies to the following products SmartNode 1000 Series SmartNode 2000 Series SmartNode 4520 Series Overview Initially designed

More information

Project Code: SPBX. Project Advisor : Aftab Alam. Project Team: Umair Ashraf 03-1853 (Team Lead) Imran Bashir 02-1658 Khadija Akram 04-0080

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

More information

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

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

More information

Bandwidth Security and QoS Considerations

Bandwidth Security and QoS Considerations This chapter presents some design considerations for provisioning network bandwidth, providing security and access to corporate data stores, and ensuring Quality of Service (QoS) for Unified CCX applications.

More information

Plain-old-telephone-service (POTS) networks

Plain-old-telephone-service (POTS) networks INTERNATIONAL JOURNAL OF NETWORK MANAGEMENT Int. J. Network Mgmt., 8, 227 234 (1998) Transporting Voice Traffic Over Packet Networks POTS networks are being rapidly superceded by newer, packet-based ones,

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

Is Your Network Ready for VoIP? > White Paper

Is Your Network Ready for VoIP? > White Paper > White Paper Tough Questions, Honest Answers For many years, voice over IP (VoIP) has held the promise of enabling the next generation of voice communications within the enterprise. Unfortunately, its

More information

IP Telephony Basics. Part of The Technology Overview Series for Small and Medium Businesses

IP Telephony Basics. Part of The Technology Overview Series for Small and Medium Businesses IP Telephony Basics Part of The Technology Overview Series for Small and Medium Businesses What is IP Telephony? IP Telephony uses the Internet Protocol (IP) to transmit voice or FAX traffic over a public

More information

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

IP Telephony v1.0 Scope and Sequence. Cisco Networking Academy Program IP Telephony v1.0 Scope and Sequence Cisco Networking Academy Program Table of Content COURSE OVERVIEW...4 Course Description...4 Course Objectives...4 Target Audience...5 Prerequisites...5 Lab Requirements...5

More information

RTP Performance Enhancing Proxy

RTP Performance Enhancing Proxy PACE RTP Performance Enhancing Proxy V2 Whilst the above information has been prepared by Inmarsat in good faith, and all reasonable efforts have been made to ensure its accuracy, Inmarsat makes no warranty

More information

Overcoming Barriers to High-Quality Voice over IP Deployments. White Paper

Overcoming Barriers to High-Quality Voice over IP Deployments. White Paper Overcoming Barriers to High-Quality Voice over IP Deployments White Paper White Paper Overcoming Barriers to High-Quality Voice over IP Deployments Executive Summary Quality of Service (QoS) issues are

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

INTRODUCTION TO VOICE OVER IP

INTRODUCTION TO VOICE OVER IP 52-30-20 DATA COMMUNICATIONS MANAGEMENT INTRODUCTION TO VOICE OVER IP Gilbert Held INSIDE Equipment Utilization; VoIP Gateway; Router with Voice Modules; IP Gateway; Latency; Delay Components; Encoding;

More information

High Performance VPN Solutions Over Satellite Networks

High Performance VPN Solutions Over Satellite Networks High Performance VPN Solutions Over Satellite Networks Enhanced Packet Handling Both Accelerates And Encrypts High-Delay Satellite Circuits Characteristics of Satellite Networks? Satellite Networks have

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

VoIP in 802.11. Mika Nupponen. S-72.333 Postgraduate Course in Radio Communications 06/04/2004 1

VoIP in 802.11. Mika Nupponen. S-72.333 Postgraduate Course in Radio Communications 06/04/2004 1 VoIP in 802.11 Mika Nupponen S-72.333 Postgraduate Course in Radio Communications 06/04/2004 1 Contents Introduction VoIP & WLAN Admission Control for VoIP Traffic in WLAN Voice services in IEEE 802.11

More information

Challenges and Solutions in VoIP

Challenges and Solutions in VoIP Challenges and Solutions in VoIP Challenges in VoIP The traditional telephony network strives to provide 99.99 percent uptime to the user. This corresponds to 5.25 minutes per year of down time. Many data

More information

Using the ClearSight Analyzer To Troubleshoot the Top Five VoIP Problems And Troubleshooting Streaming Video

Using the ClearSight Analyzer To Troubleshoot the Top Five VoIP Problems And Troubleshooting Streaming Video Using the ClearSight Analyzer To Troubleshoot the Top Five VoIP Problems And Troubleshooting Streaming Video With the prevalence of Voice over IP applications within the enterprise, it is important to

More information

ehealth and VoIP Overview

ehealth and VoIP Overview ehealth and VoIP Overview Voice over IP (VoIP) configurations can be very complex. Your network could contain a variety of devices, applications, and configuration capabilities to support voice traffic.

More information

This topic lists the key mechanisms use to implement QoS in an IP network.

This topic lists the key mechanisms use to implement QoS in an IP network. IP QoS Mechanisms QoS Mechanisms This topic lists the key mechanisms use to implement QoS in an IP network. QoS Mechanisms Classification: Each class-oriented QoS mechanism has to support some type of

More information

Implementing VoIP support in a VSAT network based on SoftSwitch integration

Implementing VoIP support in a VSAT network based on SoftSwitch integration Implementing VoIP support in a VSAT network based on SoftSwitch integration Abstract Satellite communications based on geo-synchronous satellites are characterized by a large delay, and high cost of resources.

More information

Voice over IP. Better answers

Voice over IP. Better answers This white paper provides an overview of voice and data convergence. It discusses enabling business drivers and technical factors, such as compression techniques and QOS parameters, that have complemented

More information

A SENSIBLE GUIDE TO LATENCY MANAGEMENT

A SENSIBLE GUIDE TO LATENCY MANAGEMENT A SENSIBLE GUIDE TO LATENCY MANAGEMENT By Wayne Rash Wayne Rash has been writing technical articles about computers and networking since the mid-1970s. He is a former columnist for Byte Magazine, a former

More information

Is Your Network Ready For IP Telephony?

Is Your Network Ready For IP Telephony? WHITE PAPER Is Your Network Ready For IP Telephony? Straight facts about IP telephony planning and deployment 1. Introduction Enterprises are rapidly adopting IP telephony for cost savings, productivity

More information

The Fax on IP Networks. White Paper February 2011

The Fax on IP Networks. White Paper February 2011 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...

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

Testing Voice Service for Next Generation Packet Voice Networks

Testing Voice Service for Next Generation Packet Voice Networks Testing Voice Service for Next Generation Packet Voice Networks Next Generation voice networks combine voice and data on the same transmission path. The advantages are many, but because of the technology

More information

B12 Troubleshooting & Analyzing VoIP

B12 Troubleshooting & Analyzing VoIP B12 Troubleshooting & Analyzing VoIP Phillip Sherlock Shade, Senior Forensics / Network Engineer Merlion s Keep Consulting phill.shade@gmail.com Phillip Sherlock Shade (Phill) phill.shade@gmail.com Phillip

More information

Quality of Service Testing in the VoIP Environment

Quality of Service Testing in the VoIP Environment Whitepaper Quality of Service Testing in the VoIP Environment Carrying voice traffic over the Internet rather than the traditional public telephone network has revolutionized communications. Initially,

More information

VoIP from A to Z. NAEO 2009 Conference Cancun, Mexico

VoIP from A to Z. NAEO 2009 Conference Cancun, Mexico VoIP from A to Z NAEO 2009 Conference Cancun, Mexico VoIP glossary What is VoIP? Bandwidth Signaling Codecs Quality of Service (QoS) What is VoIP? Voice over Internet Protocol (VoIP) is the method of transmitting

More information

Distributed Systems 3. Network Quality of Service (QoS)

Distributed Systems 3. Network Quality of Service (QoS) Distributed Systems 3. Network Quality of Service (QoS) Paul Krzyzanowski pxk@cs.rutgers.edu 1 What factors matter for network performance? Bandwidth (bit rate) Average number of bits per second through

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

ACD: Average Call Duration is the average duration of the calls routed bya a VoIP provider. It is a quality parameter given by the VoIP providers.

ACD: Average Call Duration is the average duration of the calls routed bya a VoIP provider. It is a quality parameter given by the VoIP providers. ACD: Average Call Duration is the average duration of the calls routed bya a VoIP provider. It is a quality parameter given by the VoIP providers. API: An application programming interface (API) is a source

More information

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

Application Notes. Introduction. Contents. Managing IP Centrex & Hosted PBX Services. Series. VoIP Performance Management. Overview.

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

More information

STANDPOINT FOR QUALITY-OF-SERVICE MEASUREMENT

STANDPOINT FOR QUALITY-OF-SERVICE MEASUREMENT STANDPOINT FOR QUALITY-OF-SERVICE MEASUREMENT 1. TIMING ACCURACY The accurate multi-point measurements require accurate synchronization of clocks of the measurement devices. If for example time stamps

More information

Achieving PSTN Voice Quality in VoIP

Achieving PSTN Voice Quality in VoIP Achieving PSTN Voice Quality in VoIP Voice over Internet Protocol (VoIP) is being widely deployed to offer users low-cost alternatives for long-distance and international telephone calls. However, for

More information

IMPLEMENTING VOICE OVER IP

IMPLEMENTING VOICE OVER IP 51-20-78 DATA COMMUNICATIONS MANAGEMENT IMPLEMENTING VOICE OVER IP Gilbert Held INSIDE Latency is the Key; Compression; Interprocessing Delay; Network Access at Origin; Network Transmission Delay; Network

More information

CS 520: Network Architecture I Winter Lecture 12: The Internet Control Message Protocol and Layering.

CS 520: Network Architecture I Winter Lecture 12: The Internet Control Message Protocol and Layering. CS 520: Network Architecture I Winter 2007 Lecture 12: The Internet Control Message Protocol and Layering. The previous lecture completed a discussion of the IP address space and the latest attempts to

More information

Frequently Asked Questions

Frequently Asked Questions Frequently Asked Questions 1. Q: What is the Network Data Tunnel? A: Network Data Tunnel (NDT) is a software-based solution that accelerates data transfer in point-to-point or point-to-multipoint network

More information

Data Networking and Architecture. Delegates should have some basic knowledge of Internet Protocol and Data Networking principles.

Data Networking and Architecture. Delegates should have some basic knowledge of Internet Protocol and Data Networking principles. Data Networking and Architecture The course focuses on theoretical principles and practical implementation of selected Data Networking protocols and standards. Physical network architecture is described

More information

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

Voice over IP. Abdus Salam ICTP, February 2004 School on Digital Radio Communications for Research and Training in Developing Countries Voice over IP Abdus Salam ICTP, February 2004 School on Digital Radio Communications for Research and Training in Developing Countries Ermanno Pietrosemoli Latin American Networking School (Fundación EsLaRed)

More information

Computer Network. Interconnected collection of autonomous computers that are able to exchange information

Computer Network. Interconnected collection of autonomous computers that are able to exchange information Introduction Computer Network. Interconnected collection of autonomous computers that are able to exchange information No master/slave relationship between the computers in the network Data Communications.

More information

Application Notes. Contents. Overview. Introduction. Echo in Voice over IP Systems VoIP Performance Management

Application Notes. Contents. Overview. Introduction. Echo in Voice over IP Systems VoIP Performance Management Application Notes Title Series Echo in Voice over IP Systems VoIP Performance Management Date January 2006 Overview This application note describes why echo occurs, what effects it has on voice quality,

More information

Evaluating Data Networks for Voice Readiness

Evaluating Data Networks for Voice Readiness Evaluating Data Networks for Voice Readiness by John Q. Walker and Jeff Hicks NetIQ Corporation Contents Introduction... 2 Determining Readiness... 2 Follow-on Steps... 7 Summary... 7 Our focus is on organizations

More information

R2. The word protocol is often used to describe diplomatic relations. How does Wikipedia describe diplomatic protocol?

R2. The word protocol is often used to describe diplomatic relations. How does Wikipedia describe diplomatic protocol? Chapter 1 Review Questions R1. What is the difference between a host and an end system? List several different types of end systems. Is a Web server an end system? 1. There is no difference. Throughout

More information

ANALYSIS OF LONG DISTANCE 3-WAY CONFERENCE CALLING WITH VOIP

ANALYSIS OF LONG DISTANCE 3-WAY CONFERENCE CALLING WITH VOIP ENSC 427: Communication Networks ANALYSIS OF LONG DISTANCE 3-WAY CONFERENCE CALLING WITH VOIP Spring 2010 Final Project Group #6: Gurpal Singh Sandhu Sasan Naderi Claret Ramos (gss7@sfu.ca) (sna14@sfu.ca)

More information

Streaming Audio and Video

Streaming Audio and Video Streaming Audio and Video CS 360 Internet Programming Daniel Zappala Brigham Young University Computer Science Department Streaming Audio and Video Daniel Zappala 1/27 Types of Streaming stored audio and

More information

Optimizing Converged Cisco Networks (ONT)

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

More information

The Conversion Technology Experts. Quality of Service (QoS) in High-Priority Applications

The Conversion Technology Experts. Quality of Service (QoS) in High-Priority Applications The Conversion Technology Experts Quality of Service (QoS) in High-Priority Applications Abstract It is apparent that with the introduction of new technologies such as Voice over IP and digital video,

More information

Real-Time Broadcast Video Services over the Internet using MPEG-DASH

Real-Time Broadcast Video Services over the Internet using MPEG-DASH over the Internet using MPEG-DASH Real-Time Broadcast Video Services over the Internet using MPEG-DASH Backhaul and Primary Distribution over the Internet does not require service contracts, special IT

More information