@INPROCEEDINGS{Marsh03:Wide, AUTHOR = "Ian Marsh and Fengyi Li and Gunnar Karlsson", TITLE = "Wide Area Measurements of VoIP Quality", BOOKTITLE = "Quality of Future Internet Services", PUBLISHER = "Springer", MONTH = Oct, YEAR = 2003, ADDRESS = "Stockholm, Sweden", ABSTRACT = "Time, day, location and instantaneous network conditions largely dictate the quality of Voice over IP calls. In this paper we present the results of over 18000 VoIP measurements, taken from nine sites connected in a full-mesh configuration. We measure the quality of the routes on a hourly basis by transmitting a pre-recorded call between a pair of sites. We repeat the procedure for all nine sites during the one hour interval. Based on the obtained jitter, delay and loss values as defined in RFC 1889 (RTP) we conclude that the VoIP quality is acceptable for all but one of the nine sites we tested. We also conclude that VoIP quality has improved marginally since we last conducted a similar study in 1998.", KEYWORDS = "VoIP, WAN, Delay, Loss, Jitter", NOTE = "ISSN 0302--9743, ISBN 3-540-20192-0" ENTRYBY = Im } @InProceedings{hagsand03:low, author = {Olof Hagsand and Ian Marsh and Kjell Hanson}, title = {Sics\textsl{o}phone: A Low-delay Internet Telephony Tool}, OPTcrossref = {}, OPTkey = {}, booktitle = {29th Euromicro Conference}, pages = {189--195}, year = {2003}, OPTeditor = {Ralf Steinmetz and Andreas Mauthe}, OPTvolume = {}, OPTnumber = {}, OPTseries = {}, OPTaddress = {Belek, Turkey}, month = {Sep}, organization = {IEEE}, publisher = {IEEE}, OPTnote = {}, OPTannote = {} abstract = "The end to end delay is a critical factor in the perceived quality of service for Voice over IP applications. Sics\textsl{o}phone is a complete VoIP system that couples the low level features of audio hardware with a standard jitter buffer playout algorithm. Using the sound card directly eliminates intermediate buffering as well as providing fine control over timers needed by a soft real-time application such as VoIP. A statistical based approach for inserting packets into audio buffers is used in conjunction with a scheme for inhibiting unnecessary fluctuations in the system. We also present mouth-to-ear delay measurements for selected VoIP applications and show that several hundreds of milliseconds can be saved by using the techniques described in this paper. A prototype for both UNIX and Windows platforms has been implemented, demonstrating that our system adapts to network conditions whilst maintaining low delays.", Entryby = Im } @INPROCEEDINGS{Mars0104:Dimensioning, AUTHOR = "Bengt, Ahlgren and Anders, Andersson and Olof, Hagsand and Marsh, Ian", TITLE = "Dimensioning Links for {IP} Telephony", BOOKTITLE = "Internet Telephony Workshop 2001", ADDRESS = "New York", DAYS = "2--3", PAGES = "14--24", MONTH = apr, YEAR = 2001, ABSTRACT = "Packet loss is an important parameter for dimensioning network links or traffic classes carrying IP telephony traffic. We present a model based on the Markov modulated Poisson process (MMPP) which calculates packet loss probabilities for a set of superpositioned voice input sources and the specified link properties. We do not introduce another new model to the community, rather try and verify one of the existing models via extensive simulation and a real world implementation. A plethora of excellent research on queuing theory is still in the domain of ATM researchers and we attempt to highlight it's validity to the IP Telephony community. Packet level simulations show very good correspondence with the predictions of the model. Our main contribution is the verification of the MMPP model with measurements in a laboratory environment. The loss rates predicted by the model are in general close to the measured loss rates and the loss rates obtained with simulation. The general conclusion is that the MMPP-based model is a tool well suited for dimensioning links carrying packetized voice in a system with limited buffer space.", URL = "http://www.cs.columbia.edu/~hgs/papers/iptel2001/14.ps", } @INPROCEEDINGS{Mars03:VoIP, AUTHOR = "Ian Marsh and Fengyi Li", TITLE = "A VoIP Measurement Infrastructure", BOOKTITLE = "Nordic Teletraffic Seminar", MONTH = Aug, YEAR = 2002, PAGES = "337-348", ADDRESS = "Espoo, Finland", ABSTRACT = "Time, day, location and instantaneous network conditions largely dictate the quality of Voice over IP calls. In this paper we describe a VoIP measurement infrastructure to measure the delay, loss and jitter of simulated phone calls on the Internet. We measure the quality by transmitting a simulated voice call between chosen sites and carefully recording the subsequent packet arrivals at the receiver. We have gathered more than 25,000 sample VoIP sessions from ten global sites. This is our second and more detailed attempt at measuring VoIP quality. This second phase has also focused on the effects of packet size, network asymmetry and silence suppression on measuring jitter, delay and loss. We have made the sessions and tools available for future investigations. Generally the quality of VoIP is excellent within the US and Europe and has improved since our last measurements. Finally this paper concludes with what we have learnt from two efforts of measuring VoIP quality on Wide Area Networks. ", KEYWORDS = "VoIP, WAN, Delay, Loss, Jitter", URL = "http://www.sics.se/~ianm/Papers/nts16.pdf", ENTRYBY = Im } @INPROCEEDINGS{kaj:qosip03, AUTHOR = {Ingemar Kaj and Ian Marsh}, TITLE = {Modelling the Arrival Process for Packet Audio}, BOOKTITLE = {Quality of Service in Multiservice IP Networks}, PAGES = "35--49", ADDRESS = {Milan, Italy}, YEAR = 2003, MONTH = FEB, ABSTRACT = {Packets in an audio stream can be distorted relative to one another during the traversal of a packet switched network. This distortion can be mainly attributed to queues in routers between the source and the destination. The queues can consist of packets either from our own flow, or from other flows. The contribution of this work is a Markov model for the time delay variation of packet audio in this scenario. Our model is extensible, and show this by including sender silence suppression and packet loss into the model. By comparing the model to wide area traffic traces we show the possibility to generate an audio arrival process similar to those created by real conditions. This is done by comparing the probability density functions of our model to the real captured data.}, URL = {http://www.sics.se/~ianm/Publications/audio_delay_cr.pdf} } @InProceedings{, author = {Henrik Abrahmsson, Olof Hagsand and Ian Marsh}, title = {TCP over Variable Capacity Links: A Simulation Study}, OPTcrossref = {}, OPTkey = {}, OPTbooktitle = {7th IFIP/IEEE International Workshop on Protocols for High Speed Networks}, pages = {117--129}, year = {2002}, editor = {Georg Carle and Martina Zitterbart}, volume = {LNCS 2334}, OPTnumber = {}, OPTseries = {}, OPTaddress = {Berlin, Germany}, OPTmonth = {Apr}, OPTorganization = {}, OPTpublisher = {Springer}, OPTnote = {}, OPTannote = {} abstract = "New optical network technologies provide opportunities for fast, controllable bandwidth management. These technologies can now explicitly provide resources to data paths, creating demand driven bandwidth reservation across networks where an applications bandwidth needs can be meet almost \textsl{exactly}. Dynamic synchronous Transfer Mode (DTM) is a gigabit network technology that provides channels with dynamically adjustable capacity. TCP is a reliable end-to-end transport protocol that adapts its rate to the available capacity. Both TCP and the DTM bandwidth can react to changes in the network load, creating a complex system with inter-dependent feedback mechanisms. The contribution of this work is an assessment of a bandwidth allocation scheme for TCP flows on variable capacity technologies. We have created a simulation environment using ns-2 and our results indicate that the allocation of bandwidth maximises TCP throughput for most flows, thus saving valuable capacity when compared to a scheme such as link over-provisioning. We highlight one situation where the allocation scheme might have some deficiencies against the static reservation of resources, and describe its causes. This type of situation warrants further investigation to understand how the algorithm can be modified to achieve performance similar to that of the fixed bandwidth case." } @INPROCEEDINGS{MarshI:globecom99, AUTHOR = {Ian Marsh}, TITLE = {Measuring {I}nternet Telephony Quality: Where are we today?}, BOOKTITLE = {Proceedings of IEEE Globecom: Global Internet}, ADDRESS = {Rio De Janeiro, Brazil}, YEAR = {1999}, MONTH = DEC, URL = {http://www.sics.se/~ianm/Publications/ipquality.ps} } @INPROCEEDINGS{Marshi:VoIP, AUTHOR = {Ian Marsh and Fengyi Li}, TITLE = {A VoIP Measurement Infrastructure}, BOOKTITLE = {Sixteenth Nordic Teletraffic Seminar}, ADDRESS = {Espoo, Finland}, YEAR = {2002}, MONTH = AUG, PAGES = 337--348, URL = {http://www.sics.se/~ianm/Publications/ipquality.ps} } @INPROCEEDINGS{bg:96, AUTHOR = {Björn Grönvall and Ian Marsh and Steve Pink}, TITLE = {A Multicast-based Distributed File System for the Internet}, BOOKTITLE = {Proceedings of the Seventh ACM SIGOPS European Workshop}, ADDRESS = {Connemara, Ireland}, YEAR = {1996}, MONTH = SEP, URL = {http://www.sics.se/cna/publications/sigops96.ps} } @techreport{hagsand:sicstech02, author = {Olof Hagsand and Ian Marsh and Olof Hagsand}, title = {Sicsophone: A Low-Delay Internet Telephony Tool}, institution = {SICS -- Swedish Institute of Computer Science}, year = {2002}, number = {T2002:26}, month = {December}, abstract = { The end to end delay is a critical factor in the perceived quality of service for Voice over IP applications. The described solution is a complete system-level platform and complements QoS work in the network and application areas. We describe a VoIP system that couples the low level features of audio hardware with a jitter buffer playout algorithm. Using the sound card directly eliminates intermediate buffering as well as providing fine control over timers needed by a soft real-time application such as VoIP. A statistical based approach for inserting packets into audio buffers is used in conjunction with a scheme for inhibiting unnecessary fluctuations in the system. We give comparisons for the performance of the playout algorithm against idealised playout conditions. We also present mouth to ear delay measurements for selected VoIP applications and show that several hundreds of milliseconds can be saved by using the techniques described in this paper. A prototype for both UNIX and Windows platforms (NT and 9X) has been implemented, demonstrating that our system adapts to network conditions whilst maintaining low delays.}, url = {ftp://ftp.sics.se/pub/SICS-reports/Reports/SICS-T--2002-26--SE.pdf} } @InProceedings{biyani:early2003, author = {Pravesh Biyani and Olof Hagsand and Gunnar Karlsson and Ian Marsh and Ignacio Mas}, title = {Early Estimation of Voice over IP Quality}, OPTcrossref = {}, OPTkey = {}, booktitle = {21st NORDUnet Networking Conference}, OPTpages = {}, year = {2003}, OPTeditor = {}, OPTvolume = {}, OPTnumber = {}, OPTseries = {}, address = {Reykavik, Iceland}, month = {Aug}, organization = {Nordunet}, OPTpublisher = {}, OPTnote = {}, OPTannote = {} } @InProceedings{Marsh:wide2002, author = {Ian Marsh and Fengyi Li}, title = {Wide Area Measurements of Voice over IP Revisted}, OPTcrossref = {}, OPTkey = {}, booktitle = {RadioVetenskap ock Kommunikation 02}, pages = {42}, year = {2002}, OPTeditor = {}, OPTvolume = {}, OPTnumber = {}, OPTseries = {}, address = {Stockholm, Sweden}, month = {Aug}, organization = {Nordunet}, OPTpublisher = {}, OPTnote = {}, OPTannote = {} } @Misc{marsh:quality2003, OPTkey = {}, author = {Ian Marsh}, title = {Quality aspects of audio communication}, school = {KTH}, month = {May}, year = {2003}, note = {TRITA-IMIT-LCN AVH 03:01, ISSN 1651-4106, ISRN KTH/IMIT/LCN/AVH-03/01 SE}, OPTannote = {} }