Amit Mondal

Removing exponential backoff from TCP

The well-accepted wisdom is that TCPs exponential backoff mechanism, introduced by Jacobson 20 years ago, is essential for preserving the stability of the Internet. In this paper, we show that removing exponential backoff from TCP altogether can be done without inducing any stability side-effects. We introduce the implicit packet conservation principle and show that as long as the endpoints uphold this principle, they can only improve their end-to-end performance relative to the exponential backoff case. By conducting large-scale simulations, modeling, and network experiments in Emulab and the Internet using a kernel-level FreeBSD TCP implementation, realistic traffic distributions, and complex network topologies, we demonstrate that TCPs binary exponential backoff mechanism can be safely removed. Moreover, we show that insuitability of TCPs exponential backoff is fundamental, i.e., independent from the currently-dominant Internet traffic properties or bottleneck capacities. Surprisingly, our results indicate that a path to incrementally deploying the change does exist.

A Case for WiFi Relay: Improving VoIP Quality for WiFi Users

Voice over Internet (VoIP) has been experiencing enormous growth in recent years. While posed to replace traditional PSTN for both enterprise and residential customers, VoIP has yet to achieve the same level of quality and reliability as PSTN. One key challenge is that a growing segment of customers is increasingly relying on WiFi connections. VoIP over WiFi (VoWiFi) experiences significant degradation in quality because of packet losses, mostly due to WiFis low capacity, varying signal strength, interference, etc. To understand this problem, we have developed and deployed a comprehensive measurement platform in a global enterprise network. From large-scale real-world traces, we quantitatively analyze the impact of WiFi connections and study measures to mitigate such impact. Our results confirm that WiFi connections incur significantly more packet losses than wirelines, but these losses can be effectively concealed by sending each packet up to five times (heavy replication). Due to WiFis inherent overhead, heavy replication only marginally increases WiFi airtime. To avoid the overhead on wirelines, we further propose a relay-based solution, where heavy replication only occurs between endpoints and nearby relays, and is removed before packets are transmitted on inter-branch long haul links or the public Internet. The solution has been implemented and deployed in the global enterprise network, and measurement results confirm that it can indeed greatly improve the performance of VoIP for WiFi users. In particular, it reduces the percentage of poor calls from 35% to 10%; and increases the percentage of acceptable ones from 45% to 70%.

Understanding end-user perception of network problems

It is widely assumed that certain network characteristics cause end-user irritation with network performance. These assumptions then drive the selection of quality of service parameters or the goals of adaptive systems. We have developed a methodology and toolchain, SoylentLogger, that employs user studies to empirically investigate such assumptions. SoylentLogger collects client-centric network measurement data that is labeled by the end-user as being associated with irritation at perceived network performance (or not). The data collection and labeling occurs in real-time as the user normally uses the network. We conducted a study that tracked 32 ordinary users over a period of 3 weeks and then used that data to test common assumptions about network sources of user irritation.

Upgrading mice to elephants: effects and end-point solutions

Short TCP flows may suffer significant response-time performance degradations during network congestion. Unfortunately, this creates an incentive for misbehavior by clients of interactive applications (e.g., gaming, telnet, web): to send ¿dummy¿ packets into the network at a TCP-fair rate even when they have no data to send, thus improving their performance in moments when they do have data to send. Even though no ¿law¿ is violated in this way, a large-scale deployment of such an approach has the potential to seriously jeopardize one of the core Internets principles-statistical multiplexing. We quantify, by means of analytical modeling and simulation, gains achievable by the above misbehavior. Our research indicates that easy-to-implement application-level techniques are capable of dramatically reducing incentives for conducting the above transgressions, still without compromising the idea of statistical multiplexing.

SureCall: Towards glitch-free real-time audio/video conferencing

Global enterprises are increasingly adopting unified communication solutions over traditional telephone systems. Such solutions provide integrated audio/video conferencing and messaging services, and enable flexible working environments by allowing mobile and dispersed users to communicate and collaborate easily and efficiently. The ultimate goal of unified communications is to ensure a smooth and best possible user experience across all scenarios. To address this challenge and understand the impact of various network scenarios on unified audio/video conferencing, we have developed a distributed experimental platform – SureCall – and deployed it on over 80 machines across a global enterprise and many residential networks. SureCall has collected worth of more than 6 months of packet-level audio/video conferencing traces. Through in-depth analysis of these traces, we have quantitatively compared how key performance metrics, such as packet loss and jitter, as well as the correlation between them, are affected by the enterprise and residential networks, by WiFi connections and VPN links, etc. In addition, we show how SureCall can serve as an ideal platform to design, experiment and validate new schemes and algorithms. We have developed a new audio quality classifier using the SureCall platform, which is being experimented with the recent release of Office Communicator solution for large-scale validation.

Supporting application network flows with multiple QoS constraints

There is a growing need to support real-time applications over the Internet. Real-time interactive applications often have multiple quality-of-service (QoS) requirements which are application specific. Traditional provisioning of QoS in the Internet through IP routing — Intserv or Diffserv — faces many technical challenges, and is also deterred by the huge deployment issues. As an alternative, application providers often build their own application-specific overlay networks to meet their QoS requirements. In this paper, we present a unified framework which can serve diverse applications with multiple QoS constraints. Our scalable flow route management architecture, called MCQoS, employs a hybrid approach using a path vector protocol to disseminate aggregated path information combined with on-demand path discovery to find paths that match the diverse QoS requirements. It uses a distributed algorithm to dynamically adapt to an alternate path when the current path fails to satisfy the required QoS constraints. We do large-scale simulation and analysis to show that our approach is both efficient and scalable, and that it substantially outperforms the state of the art protocols in accuracy. Our simulation results show that MCQoS can reduce the false negative percentage to less than 1% compared with 5–10% in other approaches, and eliminates false positives, whereas other schemes have false positive rates of 10–20% with minimal increase in protocol overhead. Finally, we implemented and deployed our system on the Planetlab testbed for evaluation in a real network environment.