Steven McCanne

Advances in network simulation

Network researchers must test Internet protocols under varied conditions to determine whether they are robust and reliable. The paper discusses the Virtual Inter Network Testbed (VINT) project which has enhanced its network simulator and related software to provide several practical innovations that broaden the conditions under which researchers can evaluate network protocols.

vic : a flexible framework for packet video

The deployment of IP Multicast has fostered the development of a suite of applications, collectively known as the MBone tools, for real-time multimedia conferencing over the Internet. Two of these tools — nv from Xerox PARC and ivs from INRIA — provide video transmission using software-based codecs. We describe a new video tool, vic , that extends the groundbreaking work of nv and ivs with a more flexible system architecture. This flexibility is characterized by network layer independence, support for hardware-based codecs, a conference coordination model, an extensible user interface, and support for diverse compression algorithms. We also propose a novel compression scheme called “Intra-H.261”. Created as a hybrid of the nv and ivs codecs, Intra-H.261 provides a factor of 2–3 improvement in compression gain over the nv encoder (6 dB of PSNR) as well as a substantial improvement in run-time performance over the ivs H.261 coder.

An application level video gateway

The current model for multicast transmission of video over the Internet assumes that a fixed average bandwidth is uniformly present throughout the network. Consequently, sources limit their transmission rates to accommodate the lowest bandwidth links, even though high-bandwidth connectivity might be available to many of the participants. We propose an architecture where a video transmission can be decomposed into multiple sessions with different bandwidth requirements using an application-level gateway. Our video gateway transparently connects pairs of sessions into a single logical conference by manipulating the data and control information of the video streams. In particular, the gateway performs bandwidth adaptation through transcoding and rate-control. We describe an efficient algorithm for transcoding Motion-JPEG to H.261 that runs in real-time on standard workstations. By making the Real-time Transport Protocol (RTP) an integral component of our architecture, the video gateway interoperates with the current Internet video tools in a transparent fashion. We have built a prototype of the video gateway and used it to redistribute multi-megabit JPEG video seminars from the Bay Area Gigabit Network as 128 kb/s H.261 MBone sessions.

RMX: reliable multicast for heterogeneous networks

Although IP multicast is an effective network primitive for best-effort, large-scale, multi-point communication, many multicast applications such as shared whiteboards, multi-player games and software distribution require reliable data delivery. Building services like reliable sequenced delivery on top of IP multicast has proven to be a hard problem. The enormous extent of network and end-system heterogeneity in multipoint communication exacerbates the design of scalable end-to-end reliable multicast protocols. In this paper, we propose a radical departure from the traditional end-to-end model for reliable multicast and instead propose a hybrid approach that leverages the successes of unicast reliability protocols such as TCP while retaining the efficiency of IP multicast for multi-point data delivery. Our approach splits a large heterogeneous reliable multicast session into a number of multicast data groups of co-located homogeneous participants. A collection of application-aware agents-reliable multicast proxies (RMX)-organizes these data groups into a spanning tree using an overlay network of TCP connections. Sources transmit data to their local group, and the RMX in that group forwards the data towards the rest of the data groups. RMX use detailed knowledge of application semantics to adapt to the effects of heterogeneity in the environment. To demonstrate the efficacy of our architecture, we have built a prototype implementation that can be customized for different kinds of applications.

Inference of multicast routing trees and bottleneck bandwidths using end-to-end measurements

The efficacy of end-to-end multicast transport protocols depends critically upon their ability to scale efficiently to a large number of receivers. Several research multicast protocols attempt to achieve this high scalability by identifying sets of co-located receivers in order to enhance loss recovery, congestion control and so forth. A number of these schemes could be enhanced and simplified by some level of explicit knowledge of the topology of the multicast distribution tree, the value of the bottleneck bandwidth along the path between the source and each individual receiver and the approximate location of the bottlenecks in the tree. In this paper, we explore the problem of inferring the internal structure of a multicast distribution tree using only observations made at the end hosts. By noting correlations of loss patterns across the receiver set and by measuring how the network perturbs the fine-grained timing structure of the packets sent from the source, we can determine both the underlying multicast tree structure as well as the bottleneck bandwidths. The simulations show that the algorithm is robust and appears to converge to the correct tree with high probability.

Receiver-driven bandwidth adaptation for light-weight sessions

Current Internet multicast conferencing tools treat all sources with equal importance in that they either statically allocate a fixed bandwidth to each source in a session, or they automatically adapt each source’s transmission rate independently of all other sources. But not all sources are of equal interest to all receivers. We believe that to effectively support human to human communication, this disparity in receiver interest should be reflected in the rate-adaptation process. To this end, we propose a protocol called “SCUBA” that enables media sources to intelligently account for receiver interest in their rate-adjustment algorithms. SCUBA is orthogonal to and complements existing rate-adaptation schemes and can interoperate with either senderor receiverdirected control systems. To scale the SCUBA protocol with multicast session size, we decouple the receiver-feedback process from the session size through sampling. This approach introduces a “tunable” tradeoff between convergence time and sampling accuracy that for large sessions is solely dependent on the control traffic bandwidth. In addition to its applicability in video conferencing, our control scheme can be combined with media transcoders to intelligently manage a bottleneck link at a well-known and fixed location in the network. We implemented SCUBA within our video conferencing tool vie and our media gateway rtpgw and feedback from their preliminary deployment indicates that the efficacy of the overall multimedia communication system has been greatly enhanced.

Optimal routing table design for IP address lookups under memory constraints

The design of lookup tables for fast IP address lookup algorithms using a general processor is formalized as an optimization problem. A cost model that models the access times and sizes of the hierarchical memory structure of the processor is formulated. Two algorithms, using dynamic programming and Lagrange multipliers, solve the optimization problem optimally and approximately respectively. Experimental results show our algorithms have visible improvements over existing ones in the literature.

A layered DCT coder for Internet video

Several compression schemes for Internet video utilize block-based conditional replenishment (CR) where block updates are coded independently of the past. In the current Internet video tools, blocks are compressed with a single-layer representation. We propose a new approach that compresses image blocks using a layered representation. Our layered-DCT (LDCT) compression algorithm, derived from progressive JPEG, has been combined with CR and optimized for efficient software implementation to provide an improved solution for Internet packet video. Although LDCT is constrained to a layered representation, its compression performance is as good or better than the single layer Intra-H.261 and baseline JPEG coding schemes.

Joint source/channel coding for multicast packet video

Current schemes for multicast packet video cope with congestion by adapting the transmission rate of a variable rate codec at the source. We propose a new approach, based on receiver- rather than source-adaptation, where a video source is encoded hierarchically with each layer of hierarchy distributed on a separate network channel. Each receiver can then dynamically adapt to local network capacity by adjusting the number of layers it receives. In order to deploy such a system, we must at the same time develop a layered codec tailored for this model. We present a prototype coder that has been designed specifically for our receiver-based congestion avoidance scheme. In order to evaluate the efficacy of our approach, we have implemented it in an existing Internet remote conferencing application and constrained the complexity of our design to run in real-time on standard workstations. Even with this constraint, our codec can generate a flexible range of layers while exhibiting reasonable compression performance.

Search party: using randomcast for reliable multicast with local recovery

IP multicast is an efficient means of sending to a group, but the packets are sent unreliably. Some applications, like distributed whiteboard and news articles, require detection and retransmission of lost packets. In order to scale to large groups, local recovery is necessary to avoid involving the entire group in the repair process for packet losses affecting small regions of the distribution tree. While many current research efforts have attempted to devise local recovery schemes that rely only on the existing service model, we believe that extending the multicast forwarding service could enable viable and highly scalable local recovery mechanisms. To investigate this open issue, we propose a new randomized forwarding service called randomcast, and build upon it a loss recovery protocol called search party. Starting with the local recovery structure of the very scalable lightweight multicast services (LMS) scheme, we use randomized forwarding to greatly improve robustness at a modest cost in overhead and/or retransmission delay (the trade off between the two costs is fine-tunable). Analysis predicts that as the group size N increases, overhead will increase by at most log N and retransmission delay will be unaffected. Simulation experiments show that both increase very little as N grows from 8 to 64, and confirm the tunability of the trade-off.