An Analysis of the Skype Peer-to-Peer Internet Telephony Protocol

Skype is a peer-to-peer VoIP client developed by KaZaa in 2003. Skype claims that it can work almost seamlessly across NATs and firewalls and has better voice quality than the MSN and Yahoo IM applications. It encrypts calls end-to-end, and stores user information in a decentralized fashion. Skype also supports instant messaging and conferencing. This report analyzes key Skype functions such as login, NAT and firewall traversal, call establishment, media transfer, codecs, and conferencing under three different network setups. Analysis is performed by careful study of Skype network traffic

A Common Protocol for Implementing Various DHT Algorithms

This document defines DHT-independent and DHT-dependent features of DHT algorithms and presents a comparison of Chord, Pastry and Kademlia. It then describes key DHT operations and their information requirements

TCP-Friendly Rate Control with Token Bucket for VoIP Congestion Control

TCP Friendly Rate Control (TFRC) is a congestion control algorithm that provides a smooth transmission rate for real-time network applications. TFRC refrains from halving the sending rate on every packet drop, instead it is adjusted as a function of the loss rate during a single round trip time. TFRC has been proven to be fair when competing with TCP flows over congested links, but it lacks quality-of-service parameters to improve the performance of real-time traffic. A problem with TFRC is that it uses additive increase to adjust the sending rate during periods with no congestion. This leads to short term congestion that can degrade the quality of voice applications. We propose two changes to TFRC that improve the performance of VoIP applications. Our implementation, TFRC with Token Bucket (TFRC-TB), uses discrete calculated bit rates based on audio codec bandwidth usage to increase the sending rate. Also, it uses a token bucket to control the sending rate during congestion periods. We have used ns2, the network simulator, to compare our implementation to TFRC in a wide range of network conditions. Our results suggest that TFRC-TB can provide a quality of service (QoS) mechanism to voice applications while competing fairly with other traffic over congested links

NetServ Framework Design and Implementation 1.0

Eyeball ISPs today are under-utilizing an important asset: edge routers. We present NetServ, a programmable node architecture aimed at turning edge routers into distributed service hosting platforms. This allows ISPs to allocate router resources to content publishers and application service pro\-vi\-ders motivated to deploy content and services at the network edge. This model provides important benefits over currently available solutions like CDN. Content and services can be brought closer to end users by dynamically installing and removing custom modules as needed throughout the network. Unlike previous programmable router proposals which focused on customizing features of a router, NetServ focuses on deploying content and services. All our design decisions reflect this change in focus. We set three main design goals: a wide-area deployment, a multi-user execution environment, and a clear economic benefit. We built a prototype using Linux, NSIS signaling, and the Java OSGi framework. We also implemented four prototype applications: ActiveCDN provides publisher-specific content distribution and processing; KeepAlive Responder and Media Relay reduce the infrastructure needs of telephony providers; and Overload Control makes it possible to deploy more flexible algorithms to handle excessive traffic

Protocols and System Design, Reliability, and Energy Efficiency in Peer-to-Peer Communication Systems

Modern Voice-over-IP (VoIP) communication systems provide a bundle of services to their users. These services range from the most basic voice-based services such as voice calls and voicemail to more advanced ones such as conferencing, voicemail-to-text, and online address books. Besides voice, modern VoIP systems provide video calls and video conferencing, presence, instant messaging (IM), and even desktop sharing services. These systems also let their users establish a voice, video, or a text session with devices in cellular, public switched telephone network (PSTN), or other VoIP networks. The peer-to-peer (p2p) paradigm for building VoIP systems involves minimal or no use of managed servers and is therefore attractive from an administrative and economic perspective. However, the benefits of using p2p paradigm in VoIP systems are not without their challenges. First, p2p communication (VoIP) systems can be deployed in environments with varying requirements of scalability, connectivity, security, interoperability, and performance. These requirements bring forth the question of designing open and standardized protocols for diverse deployments. Second, the presence of restrictive network address translators (NATs) and firewalls prevents machines from directly exchanging packets an

TCP-Friendly Rate Control with Token Bucket for VoIP Congestion Control

TCP Friendly Rate Control (TFRC) is a congestion control algorithm that provides a smooth transmission rate for real-time network applications. TFRC refrains from halving the sending rate on every packet drop, instead it is adjusted as a function of the loss rate during a single round trip time. TFRC has been proven to be fair when competing with TCP flows over congested links, but it lacks quality-of-service parameters to improve the performance of real-time traffic. A problem with TFRC is that it uses additive increase to adjust the sending rate during periods with no congestion. This leads to short term congestion that can degrade the quality of voice applications. We propose two changes to TFRC that improve the performance of VoIP applications. Our implementation, TFRC with Token Bucket (TFRC-TB), uses discrete calculated bit rates based on audio codec bandwidth usage to increase the sending rate. Also, it uses a token bucket to control the sending rate during congestion periods. We have used ns2, the network simulator, to compare our implementation to TFRC in a wide range of network conditions. Our results suggest that TFRC-TB can provide a quality of service (QoS) mechanism to voice applications while competing fairly with other traffic over congested links. I