Modern networks of Low Earth Orbit (LEO) satellites promise to offer low-latency connectivity between any pair of devices on Earth. However, as the length of inter-satellite links constantly changes, the lowest-delay paths between satellites change over time even if the network topology does not. In this paper, we characterize the lowest-delay paths in LEO satellite networks whose topologies are lattice graphs. We show how the shape of these paths can be determined by knowing the satellites’ orbits, their movement direction and the topological positions of source and destination satellites. Our characterization has the potential to inspire several practical applications, including fast, resource-efficient routing algorithms and protocols.

ISPs may notice that traffic from certain sources is entering their network at an unexpected location, but it is hard to know if this represents a problem or is just normal spoofed background noise. If such traffic is not spoofed, it would be useful to generate alerts, but alerting on background noise is not useful.

Data plane programmability is greatly improving network monitoring. Most new proposals rely on controllers pulling information (e.g., sketches or packets) from the data plane. This architecture is not a good fit for tasks requiring high reactivity, such as failure recovery, attack mitigation, and so on. Focusing on these tasks, we argue for a different architecture, where the data plane autonomously detects anomalies and pushes alerts to the controller. As a first step, we demonstrate that statistical checks can be implemented in P4 by revisiting definition and online computation of statistical measures. We collect our techniques in a P4 library, and showcase how they enable in-switch anomaly detection.

Large low earth orbit satellite constellations have the potential to provide global high-capacity low-latency Internet access, but early deployments lack dedicated inter-satellite links (ISLs). We investigate the use of ground-based relays as a substitute for ISLs to provide low-latency wide area networking. We examine how to route such huge networks, potentially using user-terminals as relays, and devise an enhanced routing algorithm to cope. We show that even without ISLs, such networks can still beat optical fiber networks for latency.

SpaceX has filed plans with the US Federal Communications Committee (FCC) to build a constellation of 4,425 low Earth orbit communication satellites. It will use phased array antennas for up and downlinks and laser communication between satellites to provide global low-latency high bandwidth coverage. To understand the latency propertes of such a network, we built a simulator based on public details from the FCC filings. We evaluate how to use the laser links to provide a network, and look at the problem of routing on this network. We provide a preliminary evaluation of how well such a network can provide low-latency communications, and examine its multipath properties. We conclude that a network built in this manner can provide lower latency communications than any possible terrestrial optical fiber network for communications over distances greater than about 3000 km.

An ISP's customers increasingly demand delivery of their traffic without congestion and with low latency. The ISP's topology, routing, and traffic engineering, often over multiple paths, together determine congestion and latency within its backbone. We first consider how to measure a topology's capacity to route traffic without congestion and with low latency. We introduce low-latency path diversity (LLPD), a metric that captures a topology's flexibility to accommodate traffic on alternative low-latency paths. We explore to what extent 116 real backbone topologies can, regardless of routing system, keep latency low when demand exceeds the shortest path's capacity. We find, perhaps surprisingly, that topologies with good LLPD are precisely those where routing schemes struggle to achieve low latency without congestion. We examine why these schemes perform poorly, and offer an existence proof that a practical routing scheme can achieve a topology's potential for congestion-free, low-delay routing. Finally we examine implications for the design of backbone topologies amenable to achieving high capacity and low delay.

Early in in the Internet's history, routing within a single provider's WAN centered on placing traffic on the shortest path. More recent traffic engineering efforts aim to reduce congestion and/or increase utilization within the status quo of greedy shortest-path first routing on a sparse topology. In this paper, we argue that this status quo of routing and topology is fundamentally at odds with placing traffic so as to minimize latency for users while avoiding congestion. We advocate instead provider backbone topologies that are more mesh-like, and hence better at providing multiple low-latency paths, and a routing system that directly considers latency minimization and congestion avoidance while dynamically placing traffic on multiple unequal-cost paths. We offer a research agenda for achieving this new low-latency approach to WAN topology design and routing.

Modern datacenter networks provide very high capacity via redundant Clos topologies and low switch latency, but transport protocols rarely deliver matching performance. We present NDP, a novel data-center transport architecture that achieves near-optimal completion times for short transfers and high flow throughput in a wide range of scenarios, including incast. NDP switch buffers are very shallow and when they fill the switches trim packets to headers and priority forward the headers. This gives receivers a full view of instantaneous demand from all senders, and is the basis for our novel, high-performance, multipath-aware transport protocol that can deal gracefully with massive incast events and prioritize traffic from different senders on RTT timescales. We implemented NDP in Linux hosts with DPDK, in a software switch, in a NetFPGA-based hardware switch, and in P4. We evaluate NDP's performance in our implementations and in large-scale simulations, simultaneously demonstrating support for very low-latency and high throughput.

Conventional operating systems used for video streaming employ an in-memory disk buffer cache to mask the high latency and low throughput of disks. However, data from Netflix servers show that this cache has a low hit rate, so does little to improve throughput. Latency is not the problem it once was either, due to PCIe-attached flash storage. With memory bandwidth increasingly becoming a bottleneck for video servers, especially when end-to-end encryption is considered, we revisit the interaction between storage and networking for video streaming servers in pursuit of higher performance.

As 802.11 wireless networks proliferate, interference becomes increasingly severe, particularly in dense, urban environments. These networks are usually operated by different users (e.g., tenants in apartments). In this paper, we develop techniques for mitigating interference between such loosely cooperating 802.11 MIMO APs and clients, which do not share a high-speed wired backplane or central controller. We propose CoOperative Power Allocation (COPA), an approach to concurrent wireless medium access that combines fine-grained, per-subcarrier power allocation, nulling, and multi-stream transmission to claim capacity that status-quo approaches cannot. Jointly turning these knobs allows COPA to allocate subcarriers to senders partially, rather than all-or-nothing, and to embrace a measure of interference when doing so increases capacity.

Network Function Virtualization is pushing network operators to deploy commodity hardware that will be used to run middlebox functionality and processing on behalf of third parties: in effect, network operators are slowly but surely becoming in-network cloud providers. The market for innetwork clouds is large, ranging from content providers, mobile applications and even end-users.

We present FUBAR, a system that reduces congestion and maximizes the utility of the entire network by installing new routes and changing the traffic load on existing ones. FUBAR works offline to periodically adjust the distribution of traffic on paths. It requires neither changes to end hosts nor precise prior knowledge of the traffic matrix. We demonstrate that even in the presence of traffic from all network devices to all other devices, FUBAR can optimize a real-world core-level network in a matter of minutes.

Contemporary network stacks are masterpieces of generality, supporting many edge-node and middle-node functions. Generality comes at a high performance cost: current APIs, memory models, and implementations drastically limit the effectiveness of increasingly powerful hardware. Generality has historically been required so that individual systems could perform many functions. However, as providers have scaled services to support millions of users, they have transitioned toward thousands (or millions) of dedicated servers, each performing a few functions. We argue that the overhead of generality is now a key obstacle to effective scaling, making specialization not only viable, but necessary.

Contemporary network stacks are masterpieces of generality, supporting a range of edge-node and middle-node functions. This generality comes at significant performance cost: current APIs, memory models, and implementations drastically limit the effectiveness of increasingly powerful hardware. Generality has historically been required to allow individual systems to perform many functions. However, as providers have scaled up services to support hundreds of millions of users, they have transitioned toward many thousands (or even millions) of dedicated servers performing narrow ranges of functions. We argue that the overhead of generality is now a key obstacle to effective scaling, making specialization not only viable, but necessary.

We consider the intra-AS route dissemination problem from first principles, and illustrate that when known route dissemination techniques propagate even a single external routing change, they can cause transient anomalies. These anomalies are not fundamental; they are artifacts of the order in which existing proposals disseminate routes. We show that carefully ordering route updates avoids transient looping and black holes. Perhaps surprisingly, this ordering may be enforced in a completely distributed fashion, while retaining familiar correctness, scalability, and convergence properties.

We've known for a while that the Internet has ossified as a result of the race to optimize existing applications or enhance security. NATs, performance-enhancing-proxies,firewalls and traffic normalizers are only a few of the middleboxes that are deployed in the network and look beyond the IP header to do their job. IP itself can't be extended because "IP options are not an option". Is the same true for TCP?

The latest large-scale data centers offer higher aggregate bandwidth and robustness by creating multiple paths in the core of the net- work. To utilize this bandwidth requires different flows take different paths, which poses a challenge. In short, a single-path transport seems ill-suited to such networks.

Recently new data center topologies have been proposed that offer higher aggregate bandwidth and location independence by creating multiple paths in the core of the network. To effectively use this bandwidth requires ensuring different flows take different paths, which poses a challenge.

It is becoming the norm for small mobile devices to have access to multiple technologies for connecting to the Internet. This gives researchers an increasing interest for solutions allowing to use efficiently several communication mediums. We propose a demonstration of our Multipath TCP implementation for Linux, that allows spreading a single TCP flow across multiple Internet paths, without requiring any change to applications. The demonstration will involve a real Internet communication with MPTCP, with simultaneous use of several paths, as well as a demonstration of MPTCP failover capability.

Recent technological advances in commodity server architectures, with multiple multi-core CPUs, integrated memory controllers, high-speed interconnects and enhanced network interface cards, provide substantial computational capacity and thus an attractive platform for packet forwarding. However, to exploit this available capacity, we need a suitable software platform that allows effective parallel packet processing and resource management. In this paper, we at first introduce an improved forwarding architecture for software routers that enhances parallelism by exploiting hardware classification and multi-queue support, already available in recent commodity network interface cards. After evaluating the original scheduling algorithm of the widely-used Click modular router, we propose solutions for extending this scheduler for improved fairness, throughput and more precise resource management. To illustrate the potential benefits of our proposal, we implement and evaluate a few key elements of our overall design.

To search the web quickly, search engines partition the web index over many machines, and consult every partition when answering a query. To increase throughput, replicas are added for each of these machines. The key parameter of these algorithms is the trade-off between replication and partitioning: increasing the partitioning level improves query completion time since more servers handle the query, but may incur non-negligible startup costs for each sub-query. Finding the right operating point and adapting to it can significantly improve performance and reduce costs.

Modern commodity hardware architectures, with their multiple multi-core CPUs and high-speed system interconnects, exhibit tremendous power. In this paper, we study performance limitations when building both software routers and software virtual routers on such systems. We show that the fundamental performance bottleneck is currently the memory system, and that through careful mapping of tasks to CPU cores, we can achieve forwarding rates of 7 million minimum-sized packets per second on mid-range server-class systems, thus demonstrating the viability of software routers. We also find that current virtualisation systems, when used to provide forwarding engine virtualisation, yield aggregate performance equivalent to that of a single software router, a tenfold improvement on current virtual router platform performance. Finally, we identify principles for the construction of high-performance software router systems on commodity hardware, including full router virtualisation support.

We present TFWC, a TCP-friendly window-based congestion control mechanism for real-time multimedia streaming applications. Although TFRC is regarded as a de facto standard for those types of applications, under low stat-mux conditions fairness can be an issue. We re-introduce a TCP-like ACK mechanism while retaining the TCP throughput equation to compute the sending rate. We show that our proposed protocol is fairer than TFRC when competing with TCP flows under certain circumstances.

In this paper, we evaluate the performance of a software IP router forwarding plane inside the Xen virtual machine monitor environment with a view to identifying (some) design issues in Virtual Routers. To this end, we evaluate and compare the forwarding performance of two identical Linux software router configurations, run either above the Xen hypervisor or within vanilla Linux. Even with minimal sized packets, we show that the Xen DomO privileged domain offers near native forwarding performance at the condition that the sollicitation to unpriviledged domains stay minimal, whereas Xen unprivileged domains offer very poor performance in every cases. This shows that an important design principle for virtual router platforms must be to handle all forwarding, for all virtual routers, onto the same forwarding engine, in order to avoid much detrimental per-packet context switching.

Fast-growing Internet applications like streaming media and telephony prefer timeliness to reliability, making TCP a poor fit. Unfortunately, UDP, the natural alternative, lacks congestion control. High-bandwidth UDP applications must implement congestion control themselves-a difficult task-or risk rendering congested networks unusable. We set out to ease the safe deployment of these applications by designing a congestion-controlled unreliable transport protocol. The outcome, the Datagram Congestion Control Protocol or DCCP, adds to a UDP-like foundation the minimum mechanisms necessary to support congestion control. We thought those mechanisms would resemble TCP's, but without reliability and, especially, cumulative acknowledgements, we had to reconsider almost every aspect of TCP's design. The resulting protocol sheds light on how congestion control interacts with unreliable transport, how modern network constraints impact protocol design, and how TCP's reliable bytestream semantics intertwine with its other mechanisms, including congestion control.

The 802.11 encryption standard Wired Equivalent Privacy (WEP) is still widely used today despite the numerous discussions on its insecurity. In this paper, we present a novel vulnerability which allows an attacker to send arbitrary data on a WEP network after having eavesdropped a single data packet. Furthermore, we present techniques for real-time decryption of data packets, which may be used under common circumstances. Vendor produced mitigation techniques which cause frequent WEP re-keying prevent traditional attacks, whereas our attack remains effective even in such scenarios. We implemented a fully automatic version of this attack which demonstrates its practicality and feasibility in real networks. As even rapidly re-keyed networks can be quickly compromised, we believe WEP must now be abandoned rather than patched yet again.

It is well-known that BGP, the current inter-domain routing protocol, has many deficiencies. This paper describes a hybrid link-state and path-vector protocol called HLP as an alternative to BGP that has vastly better scalability, isolation and convergence properties. Using current BGP routing information, we show that HLP, in comparison to BGP, can reduce the churn-rate of route updates by a factor 400 as well as isolate the effect of routing events to a region 100 times smaller than that of BGP. For a majority of Internet routes, HLP guarantees worst-case linear-time convergence. We also describe a prototype implementation of HLP on top of the XORP router platform. HLP is not intended to be a finished and final proposal for a replacement for BGP, but is instead offered as a starting point for debates about the nature of the next-generation inter-domain routing protocol.

A number of large-scale distributed Internet applications could potentially benefit from some level of knowledge about the relative proximity between its participating host nodes. For example, the performance of large overlay networks could be improved if the application-level connectivity between the nodes in these networks is congruent with the underlying IP-level topology. Similarly, in the case of replicated Web content, client nodes could use topological information in selecting one of multiple available servers. For such applications, one need not find the optimal solution in order to achieve significant practical benefits. Thus, these applications, and presumably others like them, do not require exact topological information and can instead use sufficiently informative hints about the relative positions of Internet hosts. In this paper, we present a binning scheme whereby nodes partition themselves into bins such that nodes that fall within a given bin are relatively close to one another in terms of network latency. Our binning strategy is simple (requiring minimal support from any measurement infrastructure), scalable (requiring no form of global knowledge, each node only needs knowledge of a small number of well-known landmark nodes) and completely distributed (requiring no communication or cooperation between the nodes being binned). We apply this binning strategy to the two applications mentioned above: overlay network construction and server selection. We test our binning strategy and its application using simulation and Internet measurement traces. Our results indicate that the performance of these applications can be significantly improved by even the rather coarse-grained knowledge of topology offered by our binning scheme.

Theory and experiments show that as the per-flow product of bandwidth and latency increases, TCP becomes inefficient and prone to instability, regardless of the queuing scheme. This failing becomes increasingly important as the Internet evolves to incorporate very high-bandwidth optical links and more large-delay satellite links.To address this problem, we develop a novel approach to Internet congestion control that outperforms TCP in conventional environments, and remains efficient, fair, scalable, and stable as the bandwidth-delay product increases. This new eXplicit Control Protocol, XCP, generalizes the Explicit Congestion Notification proposal (ECN). In addition, XCP introduces the new concept of decoupling utilization control from fairness control. This allows a more flexible and analytically tractable protocol design and opens new avenues for service differentiation.Using a control theory framework, we model XCP and demonstrate it is stable and efficient regardless of the link capacity, the round trip delay, and the number of sources. Extensive packet-level simulations show that XCP outperforms TCP in both conventional and high bandwidth-delay environments. Further, XCP achieves fair bandwidth allocation, high utilization, small standing queue size, and near-zero packet drops, with both steady and highly varying traffic. Additionally, the new protocol does not maintain any per-flow state in routers and requires few CPU cycles per packet, which makes it implementable in high-speed routers.

Hash tables - which map "keys" onto "values" - are an essential building block in modern software systems. We believe a similar functionality would be equally valuable to large distributed systems. In this paper, we introduce the concept of a Content-Addressable Network (CAN) as a distributed infrastructure that provides hash table-like functionality on Internet-like scales. The CAN is scalable, fault-tolerant and completely self-organizing, and we demonstrate its scalability, robustness and low-latency properties through simulation.

In this paper we introduce TFMCC, an equation-based multicast congestion control mechanism that extends the TCP-friendly TFRC protocol from the unicast to the multicast domain. The key challenges in the design of TFMCC lie in scalable round-trip time measurements, appropriate feedback suppression, and in ensuring that feedback delays in the control loop do not adversely affect fairness towards competing flows. A major contribution is the feedback mechanism, the key component of end-to-end multicast congestion control schemes. We improve upon the well-known approach of using exponentially weighted random timers by biasing feedback in favor of low-rate receivers while still preventing a response implosion. We evaluate the design using simulation, and demonstrate that TFMCC is both TCP-friendly and scales well to multicast groups with thousands of receivers. We also investigate TFMCC's weaknesses and scaling limits to provide guidance as to application domains for which it is well suited.

The Internet has witnessed a rapid growth in deployment of Web-based streaming applications during recent years. In these applications, the server should be able to perform end-to-end congestion control and quality adaptation to match the delivered stream quality to the average available bandwidth. Thus the delivered quality is limited by the bottleneck bandwidth on the path to the client. This paper proposes a proxy caching mechanism for layered-encoded multimedia streams in the Internet to maximize the delivered quality of popular streams to interested clients. The main challenge is to replay a quality-variable cached stream while performing quality adaptation effectively in response to the variations in available bandwidth. We present a prefetching mechanism to support higher quality cached streams during subsequent playbacks and improve the quality of the cached stream with its popularity. We exploit inherent properties of multimedia streams to extend the semantics of popularity and capture both level of interest among clients and usefulness of a layer in the cache. We devise a fine-grain replacement algorithm suited for layered-encoded streams. Our simulation results show that the interaction between the replacement algorithm and prefetching mechanism causes the state of the cache to converge to an efficient state such that the quality of a cached stream is proportional to its popularity, and the variations in quality of a cached stream are inversely proportional to its popularity. This implies that after serving several requests for a stream, the proxy can effectively hide low bandwidth paths to the original server from interested clients.

It has been claimed that multicast state cannot be aggregated. In this paper, we will debunk this myth and present a simple technique that can be used to aggregate multicast forwarding state. In particular, we present an interface-centric data structure model which allows aggregation of ranges of multicast addresses in the forwarding table. Understanding the limits of possible aggregation is critical to our knowledge of how IP multicast will scale as it becomes widely deployed. We show through analysis and simulation that some aggregation is possible, even under purely random address allocation and purely random group membership distribution. We further show how other methods of allocation can significantly improve the ability to aggregate, and how non-random distributions of membership can affect aggregation both positively and negatively.

This paper proposes a mechanism for equation-based congestion control for unicast traffic. Most best-effort traffic in the current Internet is well-served by the dominant transport protocol, TCP. However, traffic such as best-effort unicast streaming multimedia could find use for a TCP-friendly congestion control mechanism that refrains from reducing the sending rate in half in response to a single packet drop. With our mechanism, the sender explicitly adjusts its sending rate as a function of the measured rate of loss events, where a loss event consists of one or more packets dropped within a single round-trip time. We use both simulations and experiments over the Internet to explore performance.

Streaming audio and video applications are becoming increasingly popular on the Internet, and the lack of effective congestion control in such applications is now a cause for significant concern. The problem is one of adapting the compression without requiring video servers to reencode the data, and fitting the resulting stream into the rapidly varying available bandwidth. At the same time, rapid fluctuations in quality will be disturbing to the users and should be avoided. We present a mechanism for using layered video in the context of unicast congestion control. This quality adaptation mechanism adds and drops layers of the video stream to perform long-term coarse-grain adaptation, while using a TCP-friendly congestion control mechanism to react to congestion on very short timescales. The mismatches between the two timescales are absorbed using buffering at the receiver. We present an efficient scheme for the distribution of available bandwidth among the active layers. Our scheme allows the server to trade short-term improvement for long-term smoothing of quality. We discuss the issues involved in implementing and tuning such a mechanism, and present our simulation results.

Current multicast routing protocols can be classified into three types according to how the multicast tree is established: broadcast and prune (e.g., DVMRP, PIM-DM), membership advertisement (e.g., MO-SPF), and rendezvous-based (e.g., CBT, PIM-SM). Rendezvous-based protocols associate with each logical multicast group address, a physical unicast address, referred to as the 'core' or 'rendezvous point' (RP). Members first join a multicast tree rooted at this rendezvous point in order to receive data packets sent to the group. Rendezvous mechanisms are well suited to large wide-area networks because they distribute group-specific data and membership information only to those routers that are on the multicast distribution tree. However, rendezvous protocols require a bootstrap mechanism to map each logical multicast address to its current physical rendezvous point address. The bootstrap mechanism must adapt to network and router failures but should minimize unnecessary changes in the group-to-RP mapping. In addition, the bootstrap mechanism should be transparent to the hosts. This paper describes and analyzes the bootstrap mechanism developed for PIM-SM. The mechanism employs an algorithmic mapping of multicast group to rendezvous point address, based on a set of available RPs distributed throughout a multicast domain. The primary evaluation measures are convergence time, message distribution overhead, balanced assignment of groups to RPs, and host impact. The mechanism as a whole, and the design lessons in particular, are applicable to other rendezvous-based multicast routing protocols as well.

End-to-end congestion control mechanisms have been critical to the robustness and stability of the Internet. Most of today's Internet traffic is TCP, and we expect this to remain so in the future. Thus, having "TCP-friendly" behavior is crucial for new applications. However, the emergence of non-congestion-controlled realtime applications threatens unfairness to competing TCP traffic and possible congestion collapse. We present an end-to-end TCP-friendly rate adaptation protocol (RAP), which employs an additive-increase, multiplicative-decrease (AIMD) algorithm. It is well suited for unicast playback of realtime streams and other semi-reliable rate-based applications. Its primary goal is to be fair and TCP-friendly while separating network congestion control from application-level reliability. We evaluate RAP through extensive simulation, and conclude that bandwidth is usually evenly shared between TCP and RAP traffic. Unfairness to TCP traffic is directly determined by how TCP diverges from the AIMD algorithm. Basic RAP behaves in a TCP-friendly fashion in a wide range of likely conditions, but we also devised a fine-grain rate adaptation mechanism to extend this range further. Finally, we show that deploying RED queue management can result in an ideal fairness between TCP and RAP traffic.

Streaming audio and video applications are becoming increasingly popular on the Internet, and the lack of effective congestion control in such applications is now a cause for significant concern. The problem is one of adapting the compression without requiring video-servers to re-encode the data, and fitting the resulting stream into the rapidly varying available bandwidth. At the same time, rapid fluctuations in quality will be disturbing to the users and should be avoided.In this paper we present a mechanism for using layered video in the context of unicast congestion control. This quality adaptation mechanism adds and drops layers of the video stream to perform long-term coarse-grain adaptation, while using a TCP-friendly congestion control mechanism to react to congestion on very short timescales. The mismatches between the two timescales are absorbed using buffering at the receiver. We present an efficient-scheme for the distribution of buffering among the active layers. Our scheme allows the server to trade short-term improvement for long-term smoothing of quality. We discuss the issues involved in implementing and tuning such a mechanism, and present our simulation results.

A multicast session directory is a mechanism by which users can discover the existence of multicast sessions. In the Mbone, session announcements have also served as multicast address reservations - a dual purpose that is efficient, but which may cause some side-affects as session directories scale.In this paper we examine the scaling of multicast address allocation when it is performed by such a multicast session directory. Despite our best efforts to make such an approach scale, this analysis ultimately reveals significant scaling problems, and suggests a new approach to multicast address allocation in the Internet environment.

Multicast routing enables efficient data distribution to multiple recipients. However, existing work has concentrated on extending single-domain techniques to wide-area networks, rather than providing mechanisms to realize inter-domain multicast on a global scale in the Internet. We describe an architecture for inter-domain multicast routing that consists of two complementary protocols. The Multicast Address-Set Claim (MASC) protocol forms the basis for a hierarchical address allocation architecture. It dynamically allocates to domains multicast address ranges from which groups initiated in the domain get their multicast addresses. The Border-Gateway Multicast Protocol (BGMP), run by the border routers of a domain, constructs inter-domain bidirectional shared trees, while allowing any existing multicast routing protocol to be used within individual domains. The resulting shared tree for a group is rooted at the domain whose address range covers the group's address; this domain is typically the group initiator's domain. We demonstrate the feasibility and performance of these complementary protocols through simulation. This architecture, together with existing protocols operating within each domain, is intended as a framework in which to solve the problems facing the current multicast addressing and routing infrastructure.

IP Multicast, Lightweight Sessions and Application Level Framing provide guidelines by which multimedia conferencing tools can be designed, but they do not provide specific solutions. In this paper, we use these design principles to guide the design of a multicast based shared editor, and examine the consequences of taking a loose consistency approach to achieve good performance in the face of network failures and losses.

This paper presents the Conference Control Channel Protocol (CCCP), a new scheme intended for controlling conferences ranging from small, tightly coupled meetings, to extremely large loosely coupled seminars. We describe the requirements of such a scheme, and present a framework for building systems that connect together new and existing applications.