| 1. |
- Abrahamsson, H, et al.
(författare)
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TCP over high speed variable capacity links: A simulation study for bandwidth allocation
- 2002
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Konferensbidrag (refereegranskat)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 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.
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| 2. |
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| 3. |
- Ahlgren, Bengt, et al.
(författare)
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Dimensioning links for IP telephony
- 2001. - 1
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Ingår i: Proceedings of the 2nd IP-Telephony Workshop (IPtel 2001), 2-3 April 2001, New York City, New York, USA.
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Konferensbidrag (refereegranskat)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 super positioned 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 its 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.
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| 4. |
- Ahlgren, Bengt, et al.
(författare)
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Dimensioning Links for IP Telephony
- 2000. - 1
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Rapport (övrigt vetenskapligt/konstnärligt)abstract
- Transmitting telephone calls over the Internet causes problems not present in current telephone technology such as packet loss and delay due to queueing in routers. In this undergraduate thesis we study how a Markov modulated Poisson process is applied as an arrival process to a multiplexer and we study the performance in terms of loss probability. The input consists of the superposition of independent voice sources. The predictions of the model is compared with results obtained with simulations of the multiplexer made with a network simulator. The buffer occupancy distribution is also studied and we see how this distribution changes as the load increases.
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| 5. |
- Biyani, Pravesh, et al.
(författare)
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Early Estimation of Voice over IPQuality
- 2003
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Konferensbidrag (refereegranskat)abstract
- Users of Voice over IP (VoIP) applications are sensitive to the quality of an ongoing call. We hypothesize that the quality of a VoIP session can be estimated from the first few seconds of the session and this can be generalized to other VoIP calls. Our approach is an in-band probing mechanism and does not require any external monitoring schemes or network support. We show by post processing VoIP data from globally distributed sites that it is possible to determine the quality after an initial number of seconds. One application is admission control, where it would be possible to reject poor quality calls before they are fully established.
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| 6. |
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| 7. |
- Fu, Jing, et al.
(författare)
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A Programming Model for a Forwarding Element
- 2004
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Ingår i: 2nd Swedish National Computer Networking Workshop, SNCNW 2004, Karlstad, Sweden. ; , s. 59-65
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Konferensbidrag (refereegranskat)abstract
- The architectural complexity and diversity of current network devices make them complex to manage and difficult to program. In this work, we specify a programming model for network devices that function as forwarding elements. We first identify the key packet processing functions and analyze current network devices. Next, we derive a model that consists of processing blocks to represent simple packet processing functions running on a forwarding element. We further use a processing block topology to represent how the individual packet processing functions are interconnected on the datapath. We also demonstrate how to program a forwarding element and show an example IPv4 forwarding service implementation. Finally, we evaluate programmability and the performance of the FE model.
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| 8. |
- Hagsand, Olof, et al.
(författare)
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Self-Admission Control for IP Telephony using Early Quality Estimation
- 2004
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Ingår i: Lecture Notes in Computer Science. - Berlin, Heidelberg : Springer Berlin Heidelberg. - 0302-9743 .- 1611-3349. ; 3042, s. 381-391
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Tidskriftsartikel (refereegranskat)abstract
- If quality of service could be provided at the transport or the application layer, then it might be deployed simply by software upgrades, instead of requiring a complete upgrade of the network infrastructure. In this paper, we propose a self-admission control scheme that does not require any network support or external monitoring schemes. We apply the admission control scheme to IP telephony as it is an important application benefiting from admission control. We predict the quality of the call by observing the packet loss over a short initial period using an in-band probing mechanism. The quality prediction is then used by the application to continue or to abort the call. Using over 9500 global IP telephony measurements, we show that it is possible to accurately predict the quality of a call. Early rejection of sessions has the advantage of saving valuable network resources plus not disturbing the on-going calls.
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| 9. |
- Hagsand, Olof, et al.
(författare)
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Self-admission control for ip telephony using early quality estimation
- 2004. - 1
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Ingår i: NETWORKING 2004, Networking Technologies, Services, and Protocols; Performance of Computer and Communication Networks; Mobile and Wireless Communications (Third International IFIP-TC6 Networking Conference, Athens, Greece, May 9-14, 2004. Proceedings). - Berlin, Heidelberg : Springer. - 9783540219590 ; , s. 381-391
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Bokkapitel (refereegranskat)abstract
- If quality of service could be provided at the transport or the application layer, then it might be deployed simply by software upgrades, instead of requiring a complete upgrade of the network infrastructure. In this paper, we propose a self-admission control scheme that does not require any network support or external monitoring schemes. We apply the admission control scheme to IP telephony as it is an important application benefiting from admission control. We predict the quality of the call by observing the packet loss over a short initial period using an in-band probing mechanism. The quality prediction is then used by the application to continue or to abort the call. Using over 9500 global IP telephony measurements, we show that it is possible to accurately predict the quality of a call. Early rejection of sessions has the advantage of saving valuable network resources plus not disturbing the on-going calls.
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| 10. |
- Hagsand, Olof, et al.
(författare)
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Sicsophone : A low-delay Internet telephony tool
- 2003
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Ingår i: PROCEEDINGS OF THE 29TH EUROMICRO CONFERENCE - NEW WAVES IN SYSTEM ARCHITECTURE. - Belek, Turkey : IEEE. - 0769519962 ; , s. 189-195
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Konferensbidrag (refereegranskat)abstract
- The end to end delay is a critical factor in the perceived quality of service for Voice over IP applications. Sicsophone 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.
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