start of a talkspurt as given by (2). (1) (2)
|
|
- Gordon Quinn
- 8 years ago
- Views:
Transcription
1 An Evaluation of the Potential of Synchronized Time to Improve Voice Over IP Quality Hugh Melvin and Liam Murphy Abstract Delivering PSTN-like quality over current besteffort Internet infrastructure presents many technical challenges. Much research in recent years has focused on receiver-based approaches which adapt to varying network conditions in order to optimize playout quality. In this paper, we propose and evaluate a receiver-based approach that implements a hybrid adaptivefixed playout regime by integrating synchronized time into the playout algorithm. Such an approach can deliver significantly better quality than existing adaptive techniques particularly when the underlying network is not heavily congested and end-toend delays are not excessive. We present some initial results from our testbed system using the ITU-T E-model to quantify improvements. I. INTRODUCTION THE degree to which Internet Telephony has replaced the traditional PSTN has to date been very limited [1]. The Internet s best-effort service compares poorly with the deterministic, circuit switched PSTN network. The ITU-T recommendation G.114 specifies that one-way delays should not exceed 1 msec [2]. With the exception of satellite links, the PSTN easily meets these limits, presenting end-to-end delays of which propagation time forms a significant and consistent proportion. The Internet generally presents higher and more variable end-to-end delays where propagation time is often dwarfed by congestion delays. In addition, severe Internet congestion often leads to packet loss. The ITU-T E-model, though limited enables such delay and loss to be quantified on an additive Transmission Rating factor (R) - 1 scale, which can be mapped to the better known Mean Opinion Score (MOS) 1- scale [3]. It is significant that R (& MOS) values are more sensitive to an increase in packet loss than to an increase in delay [4]. Because of the commercial worth of voice services, significant research has been undertaken in attempts to improve quality of service (QoS). Such research can be categorized by whether it involves measures taken at the sender, in the network, or at the receiver []. In this paper we focus on receiver-based buffering approaches. These compensate for network jitter by delaying playout to facilitate the arrival of delayed packets, at the expense of adding to the overall end-toend delay. We summarize such work outlining both the merits and shortfalls of the differing approaches. We also propose, describe and evaluate a new hybrid adaptive-fixed playout strategy. In this approach, synchronized time, provided via the H. Melvin is a Ph.D. student at the Department of Computer Science, University College Dublin, Ireland ( hugh.melvin@nuigalway.ie). Liam Murphy is a lecturer at the Department of Computer Science, UCD and the director of UCD s Performance Engineering Laboratory. Network Time Protocol () is used to determine end-toend delays on a per-packet basis and this precise information is used to select an optimum playout strategy. Results were gathered from a testbed that linked Dublin City University with the National University of Ireland, Galway. Extensive tests were carried out over a number of days under varying network loads. Our results indicate an R-scale improvement of up to 2 when compared to baseline existing adaptive approaches. More generally we propose that a hybrid adaptivefixed strategy can significantly improve quality on Internet links where the network is not heavily congested, and actual end-to-end delays rarely exceed ITU-T G.114 requirements. The remainder of the paper is organized as follows. Section II outlines and evaluates some alternative receiver-based buffer strategies, introduces the issues surrounding accurate delay measurement, and describes the hybrid algorithm developed by the authors. Section III outlines the use and limitations of the ITU-T E-model. Section IV describes the test system in detail. Section V presents detailed results including an analysis of performance. Section VI concludes the paper, outlining further work currently being implemented. II. RECEIVER BUFFER STRATEGIES Jiang and Schulzrinne outline that the total end-to-end delay in a session includes [6]: Operating System delay. Host hardware input/output delay including packetization, encoding and decoding delays. Network delay comprising transmission delay, propagation delay and variable queuing delay within intermediate routers and switches. Application delay, introduced in the receiver to absorb the variation in end-to-end delay principally due to queuing delays. From a packet loss perspective, it is important to distinguish between link loss and late loss. The design of the Application delay involves a trade-off between total end-to-end delay and late packet loss. Different approaches have been taken in the implementation of application delay or buffering within the receiver. Reference [7] compared four playout delay adjustment algorithms, two of which (denoted Alg. 1 and 4 in [7] and known as Alg. A and B respectively in this paper) are based on stochastic gradient algorithms. They differ in that Alg. B additionally employs a spike detection mode. In non-spike mode, both algorithms use a linear filter mechanism that tracks network conditions (see 1) and adjusts playout time accordingly at the
2 start of a talkspurt as given by (2).!#"$ % & In the above, ' refers to packet ', is the estimated endto-end delay, is the filter gain, is the measured delay, is the playout time, is the send time, & is the estimated variation in delay and " is a multiplication factor. Adjusting on a per-talkspurt basis maintains the integrity of speech within talkspurts whilst altering the inter-talkspurt silence periods. Moon, Kurose and Towsley propose a different approach in [8] (denoted Alg. C in this paper) whereby delay percentile information for 1 packets is maintained and updated with each received packet and used to dynamically adjust playout delay, again on a per-talkspurt basis. They also incorporate a spike detection mode and report significant benefits from their approach when compared with Alg. A and B. They conclude that: Alg. A reacts too slowly to sudden network variations and thus is only suitable for slowly changing network conditions. Alg. B tends to underestimate playout delay particularly after exiting spike mode due to its gain values. More recent work by Liang, Farber and Girod [9], denoted Alg. D in this paper, propose yet another mechanism that differs significantly from the above. Similar to Alg. C, it maintains a continuously updated histogram of previous packet delays to predict future playout delay. In contrast to Algorithms A, B and C, the playout adjustment of Alg. D is made on a per-packet basis. This is achieved by compressing or elongating packets (known as scaling) using a technique based on the Waveform Similarity Overlap-Add (WSOLA) algorithm. They report from tests that such scaling of packets results in little degradation of audio quality (.3-. in the DCR (Degradation Category Rating [1])) though qualify this by noting that scaling occurs infrequently during the reported tests. They note also that per-talkspurt algorithms will fail to react to short spikes where such spikes are contained within a single talkspurt. Considering the composition of human speech [11], this limitation can be very significant. Finally in [4], further comparisons are made between fixed and per-talkspurt adaptive playout algorithms (Alg. B above). The main points of this analysis are as follows: Adaptive algorithms that utilize the TCP-like formula such as (1) and (2) tend to overestimate delays. The selection of parameters such as and " is a nontrivial matter and tuning is required to suit network characteristics. Such characteristics may however change with time. Note that Alg. A-D are designed to operate with some late loss although the degree to which they do so varies. Additionally Alg. A-C suffer from distortion of inter-talkspurt silence periods and Alg. D suffers from packet scaling (distortion of both silence periods and talkspurts). Adaptive algorithms are thus most useful when receivers have no knowledge of actual one-way delays which is usually the case. If actual delays are (1) (2) known, and are within G.114 requirements, a fixed playout delay can avoid many of the problems associated with adaptive algorithms. However a fixed playout delay, imposed at the start of a session may quickly become inappropriate resulting in high late losses or unnecessarily high delays. A. DELAY MEASUREMENT Accurate delay measurement within the Internet is a nontrivial issue. Montgomery evaluated a number of solutions in [18], including the use of Round Trip Times (RTT), distributed synchronized time and a novel approach involving a variable delay estimation mechanism within routers, requiring a specific protocol format. More recently, RFC 2679 [19] examines the core issues surrounding accurate delay measurement. RTT remains an unreliable delay estimation mechanism but the availability of distributed synchronized time has greatly increased in recent years. This is due in part to the widespread deployment of the Network Time Protocol () but more importantly, to the availability of cheap yet highly accurate time sources such as GPS receivers. B. HYBRID ADAPTIVE-FIXED ALGORITHM The authors have implemented a hybrid adaptive-fixed algorithm that combines the useful characteristics of both adaptive and fixed buffer strategies. It operates as follows: Session commences implementing an adaptive buffer algorithm. The availability of synchronized time enables one-way packet delays to be precisely determined. Each receiver builds up a histogram of delays and from this extracts a delay estimate value (*) to meet target loss requirements. In the following pseudocode, +-, refers to the packetization delay and./ refers to a weight factor (-1) applied in determining the fixed playout delay. Recall that the G.114 limit is 1 msec. If (est < (1-Pkt)) { playout = est + (1-Pkt-est) * Wf Fixed Playout Mode = TRUE // switch to fixed mode } Maintain adaptive playout mode Maintain rolling histogram and at periodic intervals, recalculate delay estimate (*) : If (est < (1-Pkt)) playout = est + (1-Pkt-est) * Wf If (Fixed Playout Mode) If (conditions improved) Decrease fixed playout delay if (conditions deteriorated) Increase fixed playout delay Fixed Playout Mode = TRUE // switch to fixed mode // high delays If (Fixed Playout Mode) Fixed Playout Mode = FALSE // switch to adaptive mode Maintain adaptive playout mode The hybrid algorithm maintains a profile of network delays before determining a fixed playout delay, and thus avoids the
3 inflexibility usually associated with fixed buffer algorithms. The weight factor./ provides an extra delay margin by positioning the fixed playout delay between the extracted delay estimate (*) and the G.114 limit. Increasing./ will result in greater reductions in late losses at the expense of higher fixed playout delay. By implementing a fixed playout whenever possible, the integrity of speech both within and between talkspurts is maintained. Adaptive mode is thus used only when absolutely necessary. III. ITU E-MODEL The E-model is an ITU-T standardized tool for predicting how the average user rates the voice quality of a phone call with known transmission parameters. Technical details are specified in [3]. The model returns a transmission rating factor R, defined as: The factors of interest are (delay impairment) and (loss impairment). includes the distortion caused by low bit rate codec operation as well as the effect of packet loss (both link and late loss). Both and are intrinsic to the voice signal whereas factor is an advantage factor. All three are not considered in this analysis. In the context of evaluating real voice sessions, the E-model is limited in that it delivers an instantaneous rating based on singular loss and delay figures. Other research has thus examined issues such as bursty versus random loss, recency [12] [4], perceived versus instantaneous quality [12] and rating of entire voice calls rather than segments [4]. For the purposes of this paper, we use a simplified yet conservative E-model analysis to compare the performance of the hybrid algorithm with pertalkspurt adaptive algorithms. As such, our analysis is aimed at extracting approximate relative values rather than detailed absolute values. In considering, we simplify the effect of both talker and listener echo by applying a 1dB loss value which corresponds to reasonable echo cancellation. As such, the impairment can be approximated by 1 units per 1 msec [4] [13]. Note that acoustic echo is a serious problem with PC-based speaker/mic kits [13] leading to much worse impairment values than those assumed here. Regarding the factor, all tests utilized the G.711 codec which introduces almost no distortion, compared to lower bitrate codecs such as GSM or G.723. For the purposes of quantifying per percentage packet loss, we presume firstly that PLC (Packet Loss Concealment) is applied thus reducing the impact of packet loss [see G.113 [14] for details]. Secondly we determine the impairment value on the basis of bursty rather than random loss. From [14], for packet loss in the range -% can therefore be approximated by a dualslope curve set at 1 units per % above 3% and at 3 units per % below 3% (packet loss rarely exceeded % during the tests). Note also that values from [14] are based on 1 msec G.711 packets rather than the 3 msec packets used in our tests. As such we underestimate the impairment as [16] illustrates for G.723. A critical factor for the hybrid algorithm GPS Satellite Local Client NUI,G Testbed Implementation GPS Receiver NUI,G LAN GPS Satellite Server, NUI,G Fig. 1. Internet HEANET Remote Servers Internet Testbed system Remote Client is the lower sensitivity of to increased delay than that of to packet loss. Practically speaking, this means that users are more tolerant of higher delay than higher loss. This justifies the use of the weight factor./ in calculating the fixed playout delay from the delay estimate (*). A simple mapping exists from values to the 1- MOS scale [3]. DCU IV. IMPLEMENTATION DETAILS Fig. 1 describes the test system developed for evaluating the hybrid algorithm. The end-hosts, one at the National University of Ireland, Galway (NUI,G) and the other at Dublin City University (DCU) are 22km apart and connected by the Higher Education Authority network ( HEAnet is Ireland s academic and research network and is linked to similar networks in Europe and the US. HEAnet is a well-provisioned network and considering its limited geographical size, internal network delays are usually well within the G.114 limit. Synchronized time was implemented via. A stratum 1 server was set up at NUI,G to strengthen the local infrastructure. A GPS clock provides its reference source. Six other stratum 1 servers, located in the UK, France, Germany and Switzerland were included in the client configuration files. For more detail on the implementation of within the test-system see []; for greater detail on operation see [17]. An open source implementation of H.323 ( was used to deliver pre-recorded voice streams. This code was modified to extract trace data for simulation and analysis, and indeed more fundamentally, to implement both adaptive and hybrid algorithms in realtime. A Matlab based simulator was also used to verify the correct operation of the modified code. The adaptive algorithms chosen for comparison were Alg. A (with =.98 and " =4) and Alg. B ( varied from.87 to.98, " set to 4). The weight factor./ used within the hybrid algorithm was set to.33, based on network characteristics. Its value represents a trade-off between total end-to-end delay and the acceptable degree of late loss and is influenced by network jitter. Each end-host advertises its implementation of. This is verified by sending an query to the remote end and
4 Offset of NUIG client to candidate servers 23: hrs 4: hrs 9: hrs 14: hrs 19: hrs 24: hrs 1 Offset of DCU client to candidate servers 23: hrs 4: hrs 9: hrs 14: hrs 19: hrs 24: hrs NUIG Server ntp sop.inria.fr ntp1 rz.rzze.uni erlangen.de ntp2.ja.net hora.cs.tu berlin.de swisstime.ee.ethz.ch ntp2.ptb.de Offset (msec) NUIG Server ntp sop.inria.fr ntp1 rz.rzze.uni erlangen.de ntp2.ja.net hora.cs.tu berlin.de swisstime.ee.ethz.ch ntp2.ptb.de Offset (msec) 1 2 Sample Number (taken every 1 min) Sample Number (taken every 1 min) Fig. 2. Performance of the NUI,G client Fig. 3. Performance of the DCU client analyzing the response. As detailed in [17], a robust subnet requires careful consideration and design. The integration of synchronized time within the hybrid algorithm is done through the Real-Time Transport Protocol Control Protocol (RTCP) Sender Report (SR) packets. RTCP is a companion protocol to RTP and packets are generated periodically during a session [2]. On receipt of the 1st RTCP SR packet, each receiver extracts the and RTP timestamps contained within the packet. As system clocks are synchronized through, the receiver can therefore determine all subsequent incoming RTP packet transit times. V. RESULTS Before evaluating the relative performance of the algorithms, it is important to assess the performance of as synchronized time is a critical requirement for the hybrid algorithm. Fig. 2 & 3 outline the performance of the NUI,G and DCU client hosts respectively over a 36 hr period, during which the tests were carried out. The legend outlines which of the seven servers were candidates at each sample instant plus the clock offset of each of the candidates relative to the client clock. The NUI,G client clock offset remained within a +/- 1 msec band for 97% of the time whereas the DCU client clock offset remained within a +/- 7. msec band for 9% of the time. As expected, the NUI,G client remained tightly coupled to the NUI,G server, selecting it as a candidate 93% of the time. The DCU client selected the NUI,G client as a candidate 6% of the time, disregarding it whenever the DCU-NUIG link became congested, relative to the other servers. This is reflected in the range of offset values and confirms the need for robust local infrastructure. Overall within a context, the performance of both the NUI,G and DCU clients was very satisfactory. In all 4 tests were carried out over a 3 day period. Figures 4, and 6 summarize results for one of the 3 days whereas Fig. 7 outlines a single test result during a period of significant congestion. The latter compares the performance of the hybrid algorithm with that of Alg. A. In all but 3 of the tests, the hybrid switched from adaptive to fixed at the 1st opportunity (after 1st RTCP Sender Report is received) and remained in fixed playout mode. As illustrated in Figures 4 and, the hybrid algorithm resulted in a late loss reduction of up to 4 percent whilst incurring an increase in end-to-end delay of 3-4 msec. Note that Alg. B, though reacting well to spikes, performed poorly, resulting Late loss rate % Late loss rate for different algorithms 1: 11: 12: 13: 14: 1: 16: 17: 18: 19: 2: 21: Fig. 4. Late Loss Reduction due to hybrid algorithm Algorithm A Algorithm B Hybrid in high late losses when exiting spike mode. As outlined in section III, a simplified E-model analysis was applied which focuses on the relative performance of the algorithms rather than absolute R or MOS scores. Applying the approximate and impairment formulas from section III to Figures 4 and, Fig. 6 shows that the net effect of the hybrid algorithm was an R-factor improvement of between 2-2. Although Fig. 7 shows a high variance in network delays over a specific test period, the long term variation in delay over the testbed link was much less dramatic. Fig 8 shows the delay seen by the DCU client host whilst querying the NUI,G server over a 36 hr period, during which the tests were carried out. Congestion was not a serious problem on this link and other than very infrequent spikes, the delay remained around 16 msec. In fact, analysis of the delays to the other six servers, none of which is located in Ireland, indicates that the delay pattern was quite similar although the baseline delay ranged from 3-6 msec depending on location. In [21], the Delay Penalty (msec) Additional delay due to hybrid algorithm Hybrid delay penalty relative to Alg. A Hybrid delay penalty relative to Alg. B 1: 11: 12: 13: 14: 1: 16: 17: 18: 19: 2: 21: Fig.. Increased delay due to hybrid algorithm
5 1: 11: 12: 13: 14: 1: 16: 17: 18: 19: 2: 21: R factor improvement Delay (msec) Fig. 6. R factor performance comparison R factor improvement : Hybrid v Alg. A R factor improvement : Hybrid v Alg. B R-factor improvement due to the hybrid algorithm Performance of Hybrid vs Adaptive algorithms Hybrid commences in adaptive mode Hybrid changeover to fixed playout Subsequent c/o to higher fixed playout Adaptive playout 1st RTCP packet received => analysis of actual delays commences Fig. 7. Packet Number Typical test result authors extend the evaluation of the hybrid to more diverse networks and network conditions. Results are consistently encouraging, indicating that the hybrid algorithm is a viable alternative over much of the Internet s infrastructure. VI. CONCLUSIONS In this paper we propose and evaluate a new hybrid adaptive-fixed playout algorithm for. We describe and evaluate existing adaptive algorithms. Developments in and GPS technologies have greatly facilitated the implementation of distributed synchronized time and thus accurate delay measurement. Where network delays are known, and are within G.114 requirements, adaptive playout is largely unnecessary, and results in unnecessary late packet loss and voice distortion. In such cases, a fixed playout delay, though increasing overall end-to-end delay, will significantly reduce Delay (msec) Fig. 8. Delay seen by DCU to NUI,G 23: hrs 4: hrs 9: hrs 14: hrs 19: hrs 24: hrs Sample Number (taken every 1 min) Delay to NUI,G server seen from DCU late packet loss and fully preserve speech integrity. This tradeoff can often result in improved quality as users are more sensitive to increased loss than delay. Our hybrid algorithm combines the useful characteristics of both adaptive and fixed buffer schemes, estimating and implementing an optimum fixed playout delay whenever possible whilst utilizing an adaptive playout scheme only when necessary. Our results indicate an R-factor improvement of up to 2 over a series of tests carried out within the Irish research network. Finally, we show that, though requiring careful subnet design, more than adequately supports the implementation of the hybrid algorithm. Current work is examining the benefit of dynamically varying, both the weight factor and the frequency of playoutmode re-evaluation, according to network conditions. Due to the E-model s limitations, more analysis including subjective testing is also required to accurately quantify relative performance gains. REFERENCES [1] S. Bradner, Internet Telephony-progress along the road, IEEE Internet Computing, vol. 6,no. 3,May-June 22, pp [2] Recommendation G.114, One way transmission time, ITU, May 2. [3] Recommendation G.17, The E-model, a computational model for use in transmission planning, ITU, May 2. [4] A. Markopoulou, F. Tobagi, and M. Karam, Assessment of quality over Internet backbones, IEEE Proc of Infocom 22. [] H. Melvin and L. Murphy, Time synchronization for Quality of Service, IEEE Internet Computing, vol. 6,no. 3,May-June 22, pp [6] W. Jiang and H. Schulzrinne, QoS measurement of Internet real-time multimedia services, tech. report CUCS--99m, Columbia Univ., New York, Dec [7] R. Ramjee, J. Kurose, D. Towsley, and H. Schulzrinne, Adaptive playout mechanisms for packetized audio applications in wide-area networks, Proc. Conf. Comp. Comm. (IEEE Infocom), IEEE CS Press, Los Alamitos, Calif., June 1994, pp [8] S. Moon, J. Kurose, and D. Towsley, Packet audio playout delay adjustment:performance bounds and algorithms, ACM/Springer Multimedia Systems, vol. 6, pp , January [9] Y. Liang, N. Farber, and B. Girod, Adaptive playout scheduling using time-scale modification in packet voice communications, Proc. of ICASSP 21. [1] Recommendation P.8, Methods for Subjective Determination of Transmission Quality, Aug [11] J. Daigle and J. Langford, Models for analysis of packet voice communications systems, IEEE Journal on Selected Areas in Comm., vol. SAC-4, no. 6, pp. 847-, Sept [12] A. Clarke, Modeling the effects of burst packet loss and recency on subjective voice quality, Proc. of IP Telephony Workshop, Mar. 21. [13] J. Janssen, D. De Vleeschauwer, M. Buchli, and G. Petit, Assessing voice quality in packet-based telephony, IEEE Internet Computing, vol. 6,no. 3,May-June 22, pp [14] Recommendation G.113, Transmission impairments due to speech processing, ITU, Feb. 21. [1] W. Jiang and H. Schulzrinne, Modeling of packet loss and delay and their effect on real-time multimedia service quality, NOSSDAV 2, Chapel Hill,NC, Jun. 2. [16] S. Voran, Speech quality of G coding with added temporal discontinuity impairments, Proc. of ICASSP, May 21. [17] D. Mills, Network time protocol: specification, implementation, and analysis, IETF RFC 13, Mar [18] W. Montgomery, Techniques for Packet Voice Synchronization, IEEE Journal on Selected Areas in Comm., vol. SAC-1, no. 6, Dec [19] G. Almes, S. Kalidindi, M. Zekauskas, A One-way Delay Metric for IPPM, IETF RFC 2679, Sept [2] H.Schulzrinne, S.Casner, R.Frederick, and V.Jacobson RTP: A Transport Protocol for Realtime Applications, IETF RFC 1889, Jan [21] H. Melvin and L. Murphy, Implementation of a Hybrid Playout Algorithm for with Clock Skew Compensation, unpublished.
Assessment of VoIP Quality over Internet Backbones
Assessment of VoIP Quality over Internet Backbones Athina Markopoulou & Fouad Tobagi Electrical Engineering Dept. Stanford University Mansour Karam RouteScience Technologies, Inc. INFOCOM 2002, 06/25/02
More informationAssessing the quality of VoIP transmission affected by playout buffer scheme and encoding scheme
Assessing the quality of VoIP transmission affected by playout buffer scheme and encoding scheme Miroslaw Narbutt, Mark Davis Communications Network Research Institute Dublin Institute of Technology Wireless
More informationApplication 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 informationPerformance 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 informationMeasurement of V2oIP over Wide Area Network between Countries Using Soft Phone and USB Phone
The International Arab Journal of Information Technology, Vol. 7, No. 4, October 2010 343 Measurement of V2oIP over Wide Area Network between Countries Using Soft Phone and USB Phone Mohd Ismail Department
More informationImplementation of Video Voice over IP in Local Area Network Campus Environment
Implementation of Video Voice over IP in Local Area Network Campus Environment Mohd Nazri Ismail Abstract--In this research, we propose an architectural solution to integrate the video voice over IP (V2oIP)
More informationMULTI-STREAM VOICE OVER IP USING PACKET PATH DIVERSITY
MULTI-STREAM VOICE OVER IP USING PACKET PATH DIVERSITY Yi J. Liang, Eckehard G. Steinbach, and Bernd Girod Information Systems Laboratory, Department of Electrical Engineering Stanford University, Stanford,
More informationPlayout Controller QoS at the IP Edge Points For networks with packet loss and jitter Henrik Åström GLOBAL IP SOUND
Playout Controller QoS at the IP Edge Points For networks with packet loss and jitter Henrik Åström GLOBAL IP SOUND Speech Processing, Transmission and Quality Aspects (STQ) Workshop Feb 2003 Outline VoIP
More informationCOMPARISONS OF FEC AND CODEC ROBUSTNESS ON VOIP QUALITY AND BANDWIDTH EFFICIENCY
COMPARISONS OF FEC AND CODEC ROBUSTNESS ON VOIP QUALITY AND BANDWIDTH EFFICIENCY WENYU JIANG AND HENNING SCHULZRINNE Columbia University, Department of Computer Science 121 Amsterdam Ave, Mail Code 001,
More informationSTUDIES TOWARD IMPROVED VoIP SERVICES FOR FUTURE COMBAT SYSTEMS. Robert G. Cole and Subramaniam Kandaswamy JHU Applied Physics Laboratory Laurel, MD
STUDIES TOWARD IMPROVED VoIP SERVICES FOR FUTURE COMBAT SYSTEMS Robert G. Cole and Subramaniam Kandaswamy JHU Applied Physics Laboratory Laurel, MD Alan Clark Telchemy, Inc. Suwanee, GA ABSTRACT In this
More informationAn Analysis of Error Handling Techniques in Voice over IP
An Analysis of Error Handling Techniques in Voice over IP Martin John Lipka ABSTRACT The use of Voice over IP (VoIP) has been growing in popularity, but unlike its wired circuit-switched telephone network
More informationQoS in VoIP. Rahul Singhai Parijat Garg
QoS in VoIP Rahul Singhai Parijat Garg Outline Introduction The VoIP Setting QoS Issues Service Models Techniques for QoS Voice Quality Monitoring Sample solution from industry Conclusion Introduction
More informationMonitoring VoIP Call Quality Using Improved Simplified E-model
Monitoring VoIP Call Quality Using Improved Simplified E-model Haytham Assem, David Malone Hamilton Institute, National University of Ireland, Maynooth Hitham.Salama.2012, David.Malone@nuim.ie Jonathan
More informationVoice 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 informationBroadband 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 informationImpact of link failures on VoIP performance
1 Impact of link failures on VoIP performance Catherine Boutremans, Gianluca Iannaccone and Christophe Diot Abstract We use active and passive traffic measurements to identify the issues involved in the
More informationIntroduction. Impact of Link Failures on VoIP Performance. Outline. Introduction. Related Work. Outline
Impact of Link Failures on VoIP Performance International Workshop on Network and Operating System Support for Digital Audio and Video (NOSSDAV) C. Boutremans, G. Iannaccone and C. Diot Sprint ATL May
More informationUsing Optimization to Achieve Efficient Quality of Service in Voice over IP Networks
Using Optimization to Achieve Efficient Quality of Service in Voice over IP Networks Michael Todd Gardner*, Victor S. Frost**, and David W. Petr** **Information and Telecommunications Technology Center
More informationGauging VoIP call quality from 802.11 WLAN resource usage
Gauging VoIP call quality from 82.11 WLAN resource usage Miroslaw Narbutt and Mark Davis Communications Network Research Institute School of Electronic and Communications Engineering Dublin Institute of
More informationANALYSIS 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 informationBasic principles of Voice over IP
Basic principles of Voice over IP Dr. Peter Počta {pocta@fel.uniza.sk} Department of Telecommunications and Multimedia Faculty of Electrical Engineering University of Žilina, Slovakia Outline VoIP Transmission
More informationPerceived Speech Quality Prediction for Voice over IP-based Networks
Perceived Speech Quality Prediction for Voice over IP-based Networks Lingfen Sun and Emmanuel C. Ifeachor Department of Communication and Electronic Engineering, University of Plymouth, Plymouth PL 8AA,
More informationNETWORK REQUIREMENTS FOR HIGH-SPEED REAL-TIME MULTIMEDIA DATA STREAMS
NETWORK REQUIREMENTS FOR HIGH-SPEED REAL-TIME MULTIMEDIA DATA STREAMS Andrei Sukhov 1), Prasad Calyam 2), Warren Daly 3), Alexander Iliin 4) 1) Laboratory of Network Technologies, Samara Academy of Transport
More informationSources: Chapter 6 from. Computer Networking: A Top-Down Approach Featuring the Internet, by Kurose and Ross
Multimedia Communication Multimedia Systems(Module 5 Lesson 2) Summary: H Internet Phone Example Making the Best use of Internet s Best-Effort Service. Sources: H Chapter 6 from Computer Networking: A
More informationQOS Requirements and Service Level Agreements. LECTURE 4 Lecturer: Associate Professor A.S. Eremenko
QOS Requirements and Service Level Agreements LECTURE 4 Lecturer: Associate Professor A.S. Eremenko Application SLA Requirements Different applications have different SLA requirements; the impact that
More informationA New Adaptive Redundancy Control Algorithm For VoIP Applications
A New Adaptive Redundancy Control Algorithm For VoIP Applications Haytham Assem, David Malone Hamilton Institute, National University of Ireland Maynooth, Ireland Email: {Hitham.Salama.2012, David.Malone}@nuim.ie
More informationRequirements 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 informationA New Adaptive FEC Loss Control Algorithm for Voice Over IP Applications
A New Adaptive FEC Loss Control Algorithm for Voice Over IP Applications Chinmay Padhye and Kenneth J. Christensen Computer Science and Engineering University of South Florida Tampa, FL 336 {padhye, christen}@csee.usf.edu
More informationClearing 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 informationNew Models for Perceived Voice Quality Prediction and their Applications in Playout Buffer Optimization for VoIP Networks
New Models for Perceived Voice Quality Prediction and their Applications in Playout Buffer Optimization for VoIP Networks Dr. Lingfen Sun Prof Emmanuel Ifeachor University of Plymouth United Kingdom {L.Sun;
More informationMeasurement of IP Transport Parameters for IP Telephony
Measurement of IP Transport Parameters for IP Telephony B.V.Ghita, S.M.Furnell, B.M.Lines, E.C.Ifeachor Centre for Communications, Networks and Information Systems, Department of Communication and Electronic
More informationApplication 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 informationPerformance Analysis of Interleaving Scheme in Wideband VoIP System under Different Strategic Conditions
Performance Analysis of Scheme in Wideband VoIP System under Different Strategic Conditions Harjit Pal Singh 1, Sarabjeet Singh 1 and Jasvir Singh 2 1 Dept. of Physics, Dr. B.R. Ambedkar National Institute
More informationScheduling for VoIP Service in cdma2000 1x EV-DO
Scheduling for VoIP Service in cdma2000 1x EV-DO Young-June Choi and Saewoong Bahk School of Electrical Engineering & Computer Science Seoul National University, Seoul, Korea E-mail: {yjchoi, sbahk}@netlab.snu.ac.kr
More informationVoice over IP Quality of Service Using Active Queue Management
VI International Telecommunications Symposium (ITS2006), September 3-6, 2006, Fortaleza-CE, Brazil 1 Voice over IP Quality of Service Using Active Queue Management Vitalio Alfonso Reguera, Félix F. Álvarez
More informationQoS 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 informationHow To Understand The Differences Between A Fax And A Fax On A G3 Network
The Fax on IP Networks White Paper February 2011 2 The Fax on IP Networks Contents Overview... 3 Group 3 Fax Technology... 4 G.711 Fax Pass-Through... 5 T.38 IP Fax Relay... 6 Network Design Considerations...
More informationGoal 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 informationIndepth 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 informationAdaptive Rate Voice over IP Quality Management Algorithm
98 Adaptive Rate Voice over IP Quality Management Algorithm Eugene S. Myakotnykh Centre for Quantifiable Quality of Service in Communication Systems (Q2S) 1, Norwegian University of Science and Technology,
More informationNew Models for Perceived Voice Quality Prediction and their Applications in Playout Buffer Optimization for VoIP Networks
New Models for Perceived Voice Quality Prediction and their Applications in Playout Buffer Optimization for VoIP Networks Lingfen Sun and Emmanuel Ifeachor Centre for Signal Processing & Multimedia Communication
More informationETSI TS 101 329-2 V1.1.1 (2000-07)
TS 101 329-2 V1.1.1 (2000-07) Technical Specification Telecommunications and Internet Protocol Harmonization Over Networks (TIPHON); End to End Quality of Service in TIPHON Systems; Part 2: Definition
More informationVoice over IP. Presentation Outline. Objectives
Voice over IP Professor Richard Harris Presentation Outline Brief overview of VoIP and applications Challenges of VoIP IP Support for Voice Protocols used for VoIP (current views) RTP RTCP RSVP H.323 Semester
More informationSIP 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 informationHow To Determine The Capacity Of An 802.11B Network
Capacity of an IEEE 802.11b Wireless LAN supporting VoIP To appear in Proc. IEEE Int. Conference on Communications (ICC) 2004 David P. Hole and Fouad A. Tobagi Dept. of Electrical Engineering, Stanford
More informationEstimation of Voice over IP Quality in the Netherlands
Estimation of Voice over IP Quality in the Netherlands X. Zhou,F.Muller,R.E.Kooij,,andP.VanMieghem Delft University of Technology, P.O. Box 0, 00 GA Delft, The Netherlands {X.Zhou, F.Muller, R.E.Kooij,
More informationAssessing the Quality of Voice Communications over Internet Backbones
Assessing the Quality of Voice Communications over Internet Backbones Athina P. Markopoulou, Member, IEEE, Fouad A. Tobagi, Fellow, IEEE, Mansour J. Karam, Member, IEEE Abstract As the Internet evolves
More informationAgilent 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 informationPERFORMANCE ANALYSIS OF VOIP TRAFFIC OVER INTEGRATING WIRELESS LAN AND WAN USING DIFFERENT CODECS
PERFORMANCE ANALYSIS OF VOIP TRAFFIC OVER INTEGRATING WIRELESS LAN AND WAN USING DIFFERENT CODECS Ali M. Alsahlany 1 1 Department of Communication Engineering, Al-Najaf Technical College, Foundation of
More informationSynchronization Essentials of VoIP WHITE PAPER
Synchronization Essentials of VoIP WHITE PAPER Synchronization Essentials of VoIP Introduction As we accelerate into the New World of VoIP we assume we can leave some of the trappings of wireline telecom
More informationChapter 2. Literature Review
Chapter 2 Literature Review This chapter presents a literature review on Load balancing based Traffic Engineering, VoIP application, Hybrid Neuro-Fuzzy System, and Intra & Inter Domain Networks. 2.1 Load
More informationVoice Quality Planning for NGN including Mobile Networks
Voice Quality Planning for NGN including Mobile Networks Ivan Pravda 1, Jiri Vodrazka 1, 1 Czech Technical University of Prague, Faculty of Electrical Engineering, Technicka 2, 166 27 Prague 6, Czech Republic
More informationTuning the VoIP Gateways to Transport International Voice Calls over a Best-Effort IP Backbone
Tuning the VoIP Gateways to Transport International Voice Calls over a Best-Effort IP Backbone A. Van Moffaert, D. De Vleeschauwer, J. Janssen, M.J.C. Büchli, G.H. Petit Alcatel Bell, Network Strategy
More informationDelivering 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 informationAssessment of VoIP Quality over Internet Backbones
Assessment of VoIP Quality over Internet Backbones Athina P. Markopoulou, Fouad A. Tobagi, Mansour J. Karam Abstract As the Internet evolves into a ubiquitous communication infrastructure and provides
More informationAdaptive Playout for VoIP based on the Enhanced Low Delay AAC Audio Codec
Adaptive Playout for VoIP based on the Enhanced Low Delay AAC Audio Codec Jochen Issing 1, Nikolaus Färber 1, and Manfred Lutzky 1 1 Fraunhofer IIS, Erlangen, 91058, Germany Correspondence should be addressed
More informationPerformance 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 informationPerformance Measurement and Analysis of H.323 Traffic
Performance Measurement and Analysis of H.323 Traffic Prasad Calyam 1, Mukundan Sridharan 2, Weiping Mandrawa 1, and Paul Schopis 1 1 OARnet, 1224 Kinnear Road,Columbus, Ohio 43212. {pcalyam,wmandraw,pschopis}@oar.net
More informationIntroduction to VoIP. 陳 懷 恩 博 士 副 教 授 兼 所 長 國 立 宜 蘭 大 學 資 訊 工 程 研 究 所 Email: wechen@niu.edu.tw TEL: 03-9357400 # 255
Introduction to VoIP 陳 懷 恩 博 士 副 教 授 兼 所 長 國 立 宜 蘭 大 學 資 訊 工 程 研 究 所 Email: wechen@niu.edu.tw TEL: 3-93574 # 55 Outline Introduction VoIP Call Tpyes VoIP Equipments Speech and Codecs Transport Protocols
More informationTraffic Characterization and Perceptual Quality Assessment for VoIP at Pakistan Internet Exchange-PIE. M. Amir Mehmood
Traffic Characterization and Perceptual Quality Assessment for VoIP at Pakistan Internet Exchange-PIE M. Amir Mehmood Outline Background Pakistan Internet Exchange - PIE Motivation Preliminaries Our Work
More informationQuality 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 informationMOS Technology Brief Mean Opinion Score Algorithms for Speech Quality Evaluation
MOS Technology Brief Mean Opinion Score Algorithms for Speech Quality Evaluation Mean Opinion Score (MOS) is commonly used to rate phone service speech quality, expressed on a scale from 1 to 5, where
More informationVoIP 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 informationClasses of multimedia Applications
Classes of multimedia Applications Streaming Stored Audio and Video Streaming Live Audio and Video Real-Time Interactive Audio and Video Others Class: Streaming Stored Audio and Video The multimedia content
More informationPacket Loss Distributions and Packet Loss Models ABSTRACT
UIT - Secteur de la normalisation des télécommunications ITU - Telecommunication Standardization Sector UIT - Sector de Normalización de las Telecomunicaciones Study Period 2001-2004 Study Group 12 Lannion,
More informationAutomated analysis of network QoS parameters for Voice over IP applications
Automated analysis of network QoS parameters for Voice over IP applications I. Miloucheva 1, A. Nassri 2, A. Anzaloni 3 Salzburg Research, Austria 1, SIEMENS Austria 2, Instituto Technologico and Aeronautico
More informationImproving VoIP Quality Through Path Switching
University of Pennsylvania ScholarlyCommons Departmental Papers (ESE) Department of Electrical & Systems Engineering March 2005 Improving VoIP Quality Through Path Switching Shu Tao University of Pennsylvania
More informationNetwork 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 informationTroubleshooting 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 informationUnderstanding Latency in IP Telephony
Understanding Latency in IP Telephony By Alan Percy, Senior Sales Engineer Brooktrout Technology, Inc. 410 First Avenue Needham, MA 02494 Phone: (781) 449-4100 Fax: (781) 449-9009 Internet: www.brooktrout.com
More informationVoice and Fax/Modem transmission in VoIP networks
Voice and Fax/Modem transmission in VoIP networks Martin Brand A1Telekom Austria ETSI 2011. All rights reserved Name : Martin Brand Position: Senior IT Specialist at A1 Telekom Vice Chairman ETSI TC INT
More information12 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 informationFundamentals 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 information1-800-CALL-H.E.P. - Experiences on a Voice-over-IP Test Bed.
SLAC-PUB-8384 February 2000 1-800-CALL-H.E.P. - Experiences on a Voice-over-IP Test Bed. W. Matthews, L. Cottrell, R. Nitzan Presented at International Conference On Computing In High Energy Physics And
More informationImproving VoIP Quality Through Path Switching
Improving VoIP Quality Through Path Switching Shu Tao, Kuai Xu, Antonio Estepa, Teng Fei Lixin Gao, Roch Guérin, Jim Kurose, Don Towsley, Zhi-Li Zhang Abstract The current best-effort Internet cannot readily
More informationVoIP over P2P networks
VoIP over P2P networks Víctor Ramos UAM-Iztapalapa Redes y Telecomunicaciones Victor.Ramos@ieee.org http://laryc.izt.uam.mx/~vramos What is the Internet? The IP protocol suite and related mechanisms and
More informationVoIP QoS on low speed links
Ivana Pezelj Croatian Academic and Research Network - CARNet J. Marohni a bb 0 Zagreb, Croatia Ivana.Pezelj@CARNet.hr QoS on low speed links Julije Ožegovi Faculty of Electrical Engineering, Mechanical
More informationVoIP 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 informationKnowledge 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 informationImpact of link failures on VoIP performance
Impact of link failures on VoIP performance Catherine Boutremans LCA-I&C, EPFL Lausanne, Switzerland catherine.boutremans@epfl.ch Gianluca Iannaccone Sprint ATL Burlingame, CA gianluca@sprintlabs.com Christophe
More informationSTANDPOINT 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 informationMeasuring VoIP QoS. Going Beyond Can You Hear Me Now? Kaynam Hedayat. MPLScon May 2005
Measuring VoIP QoS Going Beyond Can You Hear Me Now? Kaynam Hedayat MPLScon May 2005 Overview Why Real-Time VoIP Quality Assurance Matters? Challenges - Going Beyond Can You Hear Me Now? Monitoring Technologies
More informationMonitoring and Managing Voice over Internet Protocol (VoIP)
Network Instruments White Paper Monitoring and Managing Voice over Internet Protocol (VoIP) As with most new technologies, Voice over Internet Protocol (VoIP) brings new challenges along with the benefits.
More informationHow to Measure Network Performance by Using NGNs
Speech Quality Measurement Tools for Dynamic Network Management Simon Broom, Mike Hollier Psytechnics, 23 Museum Street, Ipswich, Suffolk, UK IP1 1HN Phone +44 (0)1473 261800, Fax +44 (0)1473 261880 simon.broom@psytechnics.com
More informationAchieving 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 informationIndex Terms Audio streams, inactive frames, steganography, Voice over Internet Protocol (VoIP), packet loss. I. Introduction
Volume 2, Issue 2, February 2012 ISSN: 2277 128X International Journal of Advanced Research in Computer Science and Software Engineering Research Paper Available online at: Advanced Integrated Steganographic
More informationEvaluating 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 informationApplication 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 informationThe Analysis and Simulation of VoIP
ENSC 427 Communication Networks Spring 2013 Final Project The Analysis and Simulation of VoIP http://www.sfu.ca/~cjw11/427project.html Group #3 Demet Dilekci ddilekci@sfu.ca Conrad Wang cw11@sfu.ca Jiang
More informationAudio and Video for the Internet
RTP Audio and Video for the Internet Colin Perkins TT rvaddison-wesley Boston San Francisco New York Toronto Montreal London Munich Paris Madrid Capetown Sydney 'lokyo Singapore Mexico City CONTENTS PREFACE
More informationWhite Paper. PESQ: An Introduction. Prepared by: Psytechnics Limited. 23 Museum Street Ipswich, Suffolk United Kingdom IP1 1HN
PESQ: An Introduction White Paper Prepared by: Psytechnics Limited 23 Museum Street Ipswich, Suffolk United Kingdom IP1 1HN t: +44 (0) 1473 261 800 f: +44 (0) 1473 261 880 e: info@psytechnics.com September
More informationIntroduction to VoIP. 陳 懷 恩 博 士 助 理 教 授 兼 計 算 機 中 心 資 訊 網 路 組 組 長 國 立 宜 蘭 大 學 資 工 系 Email: wechen@niu.edu.tw TEL: 03-9357400 # 340
Introduction to VoIP 陳 懷 恩 博 士 助 理 教 授 兼 計 算 機 中 心 資 訊 網 路 組 組 長 國 立 宜 蘭 大 學 資 工 系 Email: wechen@niu.edu.tw TEL: 3-93574 # 34 Outline Introduction VoIP Call Tpyes VoIP Equipments Speech and Codecs Transport
More informationTesting VoIP on MPLS Networks
Application Note Testing VoIP on MPLS Networks Why does MPLS matter for VoIP? Multi-protocol label switching (MPLS) enables a common IP-based network to be used for all network services and for multiple
More informationComparison of Voice over IP with circuit switching techniques
Comparison of Voice over IP with circuit switching techniques Author Richard Sinden Richard Sinden 1 of 9 Abstract Voice-over-IP is a growing technology. Companies are beginning to consider commercial
More informationPerformance Analysis of VoIP Codecs over BE WiMAX Network
Performance Analysis of VoIP Codecs over BE WiMAX Network Muhammad Imran Tariq, Muhammad Ajmal Azad, Razvan Beuran, Yoichi Shinoda Japan Advanced Institute of Science and Technology, Ishikawa, Japan National
More informationApplication 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 information1 There are also private networks including leased-lines, analog PBXs
Application Notes Title Series Overview Data and Fax modem performance on VoIP services VoIP Performance Management Date November 2005 This application note describes the typical performance issues that
More informationVoice 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 informationCommunications Test Equipment White Paper
XPI Solutions for Multi-Service/Access Communications Test Equipment White Paper s Addressing VoIP Speech Quality with Non-intrusive Measurements by John Anderson Product Marketing Manager Agilent Technologies
More informationAdaptive Coding and Packet Rates for TCP-Friendly VoIP Flows
Adaptive Coding and Packet Rates for TCP-Friendly VoIP Flows C. Mahlo, C. Hoene, A. Rostami, A. Wolisz Technical University of Berlin, TKN, Sekr. FT 5-2 Einsteinufer 25, 10587 Berlin, Germany. Emails:
More information