McCanne, S., Scalable Compression and Transmission of Internet Multicast Video, Ph.D. thesis, University of California Berkeley, UCB/CSD-96-928, December 1996.
Abdella Battou, "Connection establishment latency: Measured results," Document 96-1472, ATM Forum, Oct. 1996.
Abstract: This contribution reports on performance measurements for ATM switches and networks. The measurements include call establishment latency of point-to-point and point-to-multipoint in LANs and WANs. These measurements also include the effect of peer groups on call establishment latency.
Keywords: ATM signaling; performance measurements
Scalable Reliable Multicast Using Multiple Multicast Groups TITLE2:., University of Massachussetts, Amherst, USA, 1996
Scalable Compression and Transmission of Internet Multicast Video., Technical Report CSD-96-928, University of California, Berkeley, USA, 1996
G. Armitage, "Support for multicast over UNI 3.0/3.1 based ATM networks.," Request for Comments (Proposed Standard) RFC 2022, Internet Engineering Task Force, Nov. 1996.
Abstract: Mapping the connectionless IP multicast service over the connection oriented ATM services provided by UNI 3.0/3.1 is a non-trivial task. This memo describes a mechanism to support the multicast needs of Layer 3 protocols in general, and describes its application to IP multicasting in particular. This RFC is a product of the IP/ATM working group of the IETF.
Arup Acharya Ajay Bakre B. R. Badrinath, "IP Multicast Extensions for Mobile Internetworking," in Proceedings of the Conference on Computer Communications (IEEE Infocom), (San Fransisco, California), Mar. 1996.
Abstract: This paper deals with multicasting in an internetwork with mobile hosts, particularly with regard to Mobile-IP [1] and Distance Vector Multicast Routing (DVMRP) protocols. When the source of a multicast datagram is a mobile host (MH), the datagram may not reach all group members to which the datagram is addressed, including other mobile hosts. When the source is a static host and the multicast group includes mobile hosts, a mobile group member may receive datagrams in one cell but not in another. Further, when a MH enters a cell which contains no other member of the same group, the MH will experience a delay before it starts receiving datagrams addressed to that group. Mobility between campuses, which result in a MH acquiring an additional unicast address, also has an effect on multicast routing. We propose enhancements to DVMRP executed at the Mobility Support Routers (MSR) that ensure correct forwarding of multicast datagrams to and from mobile hosts. Our solutions do not require any change at hosts and routers unaware of mobility, i.e. the modifications are limited to MSRs and MHs. We also describe an implementation incorporating a subset of our proposals. Lastly, we show that alternate styles of multicasting or mobile networking, viz. link-state (MOSPF ) and the IETF proposal [2], will face similar problems and our proposed solutions are still valid in their context. [1] J. Ioannidis, D. Duchamp and G. Q. Maguire. IP-based protocols for mobile internetworking. SIGCOMM '91 [2] IP Mobility Support. C. Perkins, editor. Internet Draft from the IETF Mobile-IP working group
A. Ballardie, "Scalable multicast key distribution," Request for Comments (Experimental) RFC 1949, Internet Engineering Task Force, May 1996.
Abstract: The benefits of multicasting are becoming ever-more apparent, and its use much more widespread. This is evident from the growth of the MBONE. Providing security services for multicast, such as traffic integrity, authentication, and confidentiality, is particularly problematic since it requires securely distributing a group (session) key to each of a group's receivers. Traditionally, the key distribution function has been assigned to a central network entity, or Key Distribution Centre (KDC), but this method does not scale for wide-area multicasting, where group members may be widely-distributed across the internetwork, and a wide-area group may be densely populated. Even more problematic is the scalable distribution of sender-specific keys. Sender-specific keys are required if data traffic is to be authenticated on a per-sender basis. This memo provides a scalable solution to the multicast key distribution problem.
Tom Billhartz, J. Bibb Cain, Ellen Farrey-Goudreau, Doug Fieg, and Steve Batsell, "Distributed Center-Location Algorithms: Proposals and Comparisons," in Proceedings of the Conference on Computer Communications (IEEE Infocom), (San Fransisco, California), Mar. 1996.
Abstract: Researchers have proposed the Core Based Trees (CBT) and Protocol Independent Multicasting (PIM) protocols to route multicast data on internetworks. The performance of these protocols is examined in the Distributed Interactive Simulation (DIS) environment using the OPNET simulation tool. The simulation results provide measures of important performance metrics including end-to-end delay, network resource usage, and join time. The size of the tables containing multicast routing information and the impact of the timers introduced by the protocols are also examined. Suggestions are offered to improve PIM Sparse while retaining the ability to offer both shared tree and source-based tree routing.
Anton Ballardie, Scott Reeve, and Nitin Jain, "Core based trees (CBT) multicast - protocol specification," Internet Draft, Internet Engineering Task Force, Sept. 1996. Work in progress.
Abstract: This document describes the Core Based Tree (CBT) network layer multicast protocol. CBT is a next-generation multicast protocol that makes use of a shared delivery tree rather than separate per-sender trees utilized by most other multicast schemes [1, 2, 3]. The CBT architecture is described in [4a]. This specification includes an optimization whereby unencapsulated (native) IP-style multicasts are forwarded by CBT routers, resulting in very good forwarding performance. This mode of operation is called CBT 'native mode'. Native mode can only be used in CBT-only domains.
Keywords: CBT; multicast; routing; core-based trees
Fred Bauer and Anujan Varma, "ARIES: a rearrangeable inexpensive edge-based on-line steiner algorithm," in Proceedings of the Conference on Computer Communications (IEEE Infocom), (San Fransisco, California), Mar. 1996.
Abstract: In this paper, we propose and evaluate ARIES, a heuristic for updating multicast trees dynamically in large point-to-point networks. The algorithm is based on monitoring the accumulated damage to the multicast tree within local regions of the tree as nodes are added and deleted, and triggering a rearrangement when the number of changes within a connected subtree crosses a set threshold. We derive an analytical upper-bound on the competitiveness of the algorithm. We also present simulation results to compare the average-case performance of the algorithm with two other known algorithms for the dynamic multicast problem, GREEDY and EBA (Edge-Bounded Algorithm). Our results show that ARIES provides the best balance among competitiveness, computational effort, and changes in the multicast tree after each update.
Shun Yan Cheung, Mostafa H. Ammar, and Xui Li, "On the Use of Destination Set Grouping to Improve Fairness in Multicast Video Distribution," in Proceedings of the Conference on Computer Communications (IEEE Infocom), (San Fransisco, California), Mar. 1996.
Abstract: We address the problem of fairness in a feedback-controlled multicast video distribution scheme. In a fair scheme each receiver should receive a video stream with a quality that is commensurate with its capabilities or the capabilities of the path leading to it, regardless of other receivers or network paths. This fairness problem results from the fact that multicast communication trades economy of bandwidth with \em granularity of control. Distributing video using individual feedback-controlled point-to-point streams results in high bandwidth utilization but the granularity of control is high as communication parameters can be negotiated individually with each receiver. In contrast, using a single multicast stream has good bandwidth economy, but very low granularity of control. In this paper we propose, implement and experiment with a system that spans the spectrum represented by the two extremes above. In the scheme, called destination set grouping (DSG), a source maintains a small number of video streams, carrying the same video but each targeted at receivers with different capabilities. Each stream is feedback-controlled within prescribed limits by its group of receivers. Receivers may move among streams as their capabilities or the capabilities of the network paths leading to them change. The scheme is shown to improve fairness significantly at a small bandwidth cost.
Woohyong Choi, "A conference control model for light-weight sessions," Master's thesis, Korea Advanced Institute of Science and Technology, Korea, July 1996.
Abstract: The current model for light-weight multicast session based teleconferencing applications provide a very primitive set of control mechanisms such as net mutes mic and mic mutes net. Commercial products based on T.124 Recommendation of International Telecommunication Union(ITU) are being introduced, and it seems likely that a similar one based on T.124 will be derived for Internet usage. However, the current emphasis on wide area scalable multicast based conferencing in the Internet Engineering Task Force (IETF) is desirable, and we shouldn't sacrifice the benefits of multicast based sessions to conform to the tightly coupled model of T.124 Recommendation. This paper proposes a conference control model for light-weight sessions where media applications can collaborate with a coordination tool to provide a level of control over the light-weight sessions. This coordination tool provides a generic base to manage conferencing states, find agreements among the participants upon which a varying range of policies could be implemented without any changes to each applications. A prototype of the coordination tool has been built and is being used to provide conference control to existing applications.
Keywords: conference control; agreement protocol; conference bus
J. Cooperstock and S. Kotsopoulos, "Why use a fishing line when you have a net? an adaptive multicast data distribution protocol," in Proc. of Usenix Winter Conference, 1996.
Abstract: The design and implementation of a system to provide reliable and efficient distribution of large quantities of data to many hosts on a local area network or internetwork is described. By exploiting the one-to-many transmission capabilities of multicast and broadcast, it is possible to transmit data to multiple hosts simultaneously, using less bandwidth and thus obtaining greater efficiency than repeat unicasting. Although performance measurements indicate the superiority of multicast, we dynamically select from available transmission modes so as to maximize efficiency and throughput while providing reliable delivery of data to all hosts. Our results demonstrate that file-distribution programs based on our protocol can benefit from a substantial speed-up over TCP-based programs such as rdist. For example, our system has been used to distribute a 133 kByte password file to 68 hosts in 20 seconds, whereas the equivalent rdist took 251 seconds.
Keywords: multicast; network news; file distribution; rate-based flow control
Annotation: The flow control slowly the decreases packet interval to 10 ms, but ups that by 10 ms each time a resend request is received.
Stephen Deering, Deborah L. Estrin, Dino Farinacci, Van Jacobson, Liu Ching-Gung, and Liming Wei, "The PIM Architecture for Wide-Area Multicast Routing," IEEE/ACM Transactions on Networking, vol. 4, pp. 153-162, Apr. 1996.
Abstract: The purpose of multicast routing is to reduce the communication costs for applications that sent the same data to multiple recipients. Existing multicast routing mechanisms were intended for use within regions where a group is widely represented or bandwidth is universally plentiful. When group members, and senders to those group members, are distributed sparsely across a wide area, these schemes are not efficient; data packets or membership report information are occasionally sent over many links that do not lead to receivers or senders respectively. We have developed a multicast routing architecture that efficiently establishes distribution trees across wide area internets, where many groups will be sparsely represented. Efficiency is measured in terms of the router state, control message processing, and data packet processing, require across the entire network in order to deliver data packets to the members of the group. Our protocol independent multicast (PIM) architecture: a) maintains the traditional IP multicast service model of receiver-initiated membership, b) supports both shared and source-specific (shortest-path) distribution trees, c) is nor dependent on a specific unicast routing protocol, and d) uses soft-state mechanisms to adapt to underlying network conditions and group dynamics. The robustness, flexibility, and scaling properties of this architecture make it well-suited to large heterogeneous internetworks.
Steven Deering, Deborah Estrin, Dino Farinacci, Mark Handley Ahmed Helmy, Van Jacobson, Chinggung Liu, Puneet Sharma, David Thaler, and Liming Wei, "Protocol independent multicast-sparse mode (PIM-SM): Motivation and architecture," Internet Draft, Internet Engineering Task Force, Oct. 1996. Work in progress.
Abstract: Traditional multicast routing mechanisms (e.g. DVMRP and MOSPF) were intended for use within regions where groups are widely represented or bandwidth is universally plentiful. When group members, and senders to those group members, are distributed sparsely across a wide area, these schemes are not efficient; data packets or membership report information are periodically sent over many links that do not lead to receivers or senders, respectively. This characteristic lead the Internet community to investigate multicast routing architectures that efficiently establish distribution trees across wide-area internets, where many groups are sparsely represented and where bandwidth is not uniformly plentiful due to the distances and multiple administrations traversed. Efficiency is evaluated in terms of the state, control message processing, and data packet processing required across the entire network in order to deliver data packets to the members of the group. The Protocol Independent Multicast-Sparse Mode (PIM-SM) architecture: (a) maintains the traditional IP multicast service model of receiver-initiated membership; (b) uses explicit joins that propagate hop-by-hop from members' directly connected routers toward the distribution tree; (c) builds a shared multicast distribution tree centered at a Rendezvous Point, and then builds source-specific trees for those sources whose data traffic warrants it; (d) is not dependent on a specific unicast routing protocol; and (e) uses soft-state mechanisms to adapt to underlying network conditions and group dynamics. The robustness, flexibility, and scaling properties of this architecture make it well suited to large heterogeneous inter- networks. This document motivates and describes the PIM-SM architecture. Companion documents describe the detailed protocol mechanisms for PIM-SM and PIM-DM, respectively.
Keywords: multicast; routing; PIM; sparse mode; dense mode
Matthew B. Doar, "A better model for generating test networks," in Proceedings of Global Internet (Jon Crowcroft and Henning Schulzrinne, eds.), (London, England), pp. 86-93, IEEE, Nov. 1996.
Abstract: Much of the work on routing algorithms, particularly for multicast, which has been done in the past has used fairly simple models to generate the topological graph which represents the hosts in the network. Some such random graphs bear little resemblance to data communication networks which are actually deployed. This paper proposes a more realistic model, incorporating hierarchy and redundancy, and is developed into a network generation algorithm. The approach described here can be developed to provide more refined models in the future, and the source code of an implementation is freely available from ftp.nexen.com/pub/papers.
Keywords: Routing Topology Model Hierarchy
Andrew Findlay, "The multi-media telephone: directory service and session control for multi-media communications," in Proc. of the Third International Workshop on Services in Distributed and Networked Environments (SDNE), (Macau), June 1996.
Abstract: Multimedia communication tools now exist which have support for group communication using multicast protocols. The same tools can be used in unicast mode for one-to-one communication. The focus of development so far has been on group working and conference support. As a result, there is a session directory tool that allows users to advertise conferences and to join the ones that interest them. On the other hand, the support for setting up one-to-one sessions and close group conferences is rather limited. A new set of session control facilities is needed to create a multi-media telephone from the components now available. This paper lists some requirements and proposes mechanisms to address them.
Keywords: session control; MBONE; conference control; Internet telephony; X.500; LDAP; directory services
Victor Firoiu and Don Towsley, "Call Admission and Resource Reservation for Multicast Sessions," in Proceedings of the Conference on Computer Communications (IEEE Infocom), (San Fransisco, California), Mar. 1996.
Abstract: Many multicast applications, including audio and video, require that a quality of service (QoS) guarantee be made to the application by the network. Hence, multicast admission control and resource reservation procedures will be needed. In this paper we present a general framework for admission control and resource reservation for multicast sessions. Within this framework, efficient and practical algorithms that aim to efficiently utilize network resources are developed. The problem of admission control is decomposed into several subproblems that include: the division of end-to-end QoS requirements into local QoS requirements, the mapping of local QoS requirements into resource allocation, and the optimization of the resulting resource allocation. These are solved independently of each other yielding a set of mechanisms and policies that can be used to provide admission control and resource reservation for multicast connection establishment. The resource allocation algorithms we consider specifically accomodate and exploit receiver heterogeneity (in both end-to-end and per-hop QoS requirements) to optimize resource use by a multicast session. A comprehensive application of an instance of the algorithms in the context of packetized voice multicast connections over the Mbone is provided to illustrate their applicability.
E. Gauthier, J.Y. Le Boudec, and Ph. Oechslin, "SMART: A many-to-many multicast protocol for ATM," Contribution 96-1295, ATM Forum, 1996.
Keywords: multicast; ATM
M. Grossglauser and K.K. Ramakrishnan, "SEAM: A scheme for scalable and efficient ATM multipoint-to-multipoint communication," Contribution 96-1142, ATM Forum, 1996. To appear in Infocom'97.
Keywords: ATM; multicast
Matthias Grossglauser, "Optimal Deterministic Timeouts for Reliable Scalable Multicast," in Proceedings of the Conference on Computer Communications (IEEE Infocom), (San Fransisco, California), Mar. 1996.
Abstract: Reliable multicast suffers from the problem of feedback implosion. To achieve scalability, the number of receivers sending feedback in case of loss must remain small. However, losses experienced by receivers are strongly correlated due to the resource sharing in the multicast tree. We present DTRM (Deterministic Timeouts for Reliable Multicast), a distributed algorithm to compute deterministic timeouts for each receiver in a multicast tree as a function of the tree topology and sender-to-receiver delays. DTRM has several desirable properties. First, the computation of the timeouts is entirely distributed; receivers and routers only rely on local topology information. Second, NACK implosion is provably avoided for a single loss anywhere in the tree if delay jitter is bounded. Third, feedback information does not need to be processed by routers, and receivers do not have to collaborate. We forsee two possible uses for DTRM. In networks providing hard delay bounds, timeouts can be computed once at session set-up time. In networks with unbounded delays, such as the Internet, timeouts can be adaptively recomputed in response to changes in estimated round-trip times.
R. Hinden and S. Deering, "IP version 6 addressing architecture," Request for Comments (Proposed Standard) RFC 1884, Internet Engineering Task Force, Jan. 1996.
Abstract: This specification defines the addressing architecture of the IP Version 6 protocol. The document includes the IPv6 addressing model, text representations of IPv6 addresses, definition of IPv6 unicast addresses, anycast addresses, and multicast addresses, and an IPv6 nodes required addresses. This document is the product of the IPNG Working Group of the IETF.
Sang Baeg Kim, "An Optimal VP-based Multicast Routing in ATM Networks," in Proceedings of the Conference on Computer Communications (IEEE Infocom), (San Fransisco, California), Mar. 1996.
Abstract: Their approach to modeling the problem has some drawbacks in the applications for ATM Networks since only a single multipoint connection request is considered in the presence of link capacity constraints. In this paper, we present a detailed mathematical model for the generalized multicast arborescence(i.e., a rooted out-tree) problem in which there are a number of concurrent multipoint connections of various traffic types. Point-to-point connections are also considered as a special case of multipoint connections. A solution to the model determines, in the most economical way, an establishment of VPCs with reserved capacity and multicast arborescences using the established VPCs for multipoint connections. This model is a natural extension of the Virtual Channel Assignment Problem introduced by the author. Algorithms and numerical results are presented.
Alex Koifman and Stephen Zabele, "RAMP: A Reliable Adaptive Multicast Protocol," in Proceedings of the Conference on Computer Communications (IEEE Infocom), (San Fransisco, California), Mar. 1996.
Abstract: The specifications and performance of RAMP, a Reliable Adaptive Multicast Protocol, are presented. Initially described in IETF RFC 1458, RAMP has been enhanced for use over an all-optical, circuit-switched, gigabit network under our ARPA-sponsored Testbed for Optical NEtworking (TBONE) project. RAMP uses immediate, receiver-initiated, NAK-based, unicast error notification combined with originator based unicast retransmission. The approach is motivated by the loss characteristics of the TBONE network, where extremely low bit-error rates (10-12 or better) and the absence of any store-and-forward capabilities in the switches make packet losses almost entirely a result of receiver buffer overflows. As receiver losses are largely independent, use of unicast over multicast for NAKs and retransmission eliminates unnecessary receiver processing overhead associated with reading and discarding redundant packets. Use of immediate rather than delayed NAKs further improves performance by reducing both latency and the likelihood of buffer overflow. The effectiveness of this combined error control approach has been verified by other researchers, as well as through our own investigations. Interestingly, TBONE loss characteristics resemble those of switched virtual circuit ATM networks and packet-switched networks employing reservation services. As these networks provide Quality of Service guarantees in the switches, the likely source of packet loss is again due to receiver errors and buffer overflows. Hence, RAMP's design is also relevant for the next generation of packet switched networks.
John C. Lin and Sanjoy Paul, "RMTP: a reliable multicast transport protocol," in Proceedings of the Conference on Computer Communications (IEEE Infocom), (San Fransisco, California), Mar. 1996.
Abstract: This paper describes the design and implementation of a reliable multicast transport protocol called RMTP. RMTP provides sequenced, lossless delivery of bulk data from one sender to a group of receivers. It allows a receiver to join or leave a group without notifying the sender or other receivers. Protocol state maintained at each multicast participant is independent of the number of participants. The sender knows neither the identity of each receiver nor the number of receivers in the multicast group. RMTP is based on a multi-level hierarchical approach, in which the receivers are grouped into a hierarchy of local regions, with a Designated Receiver (DR) in each local region. Receivers in each local region periodically send acknowledgments (ACKs) to their corresponding DR, DRs send ACKs to the higher-level DRs, until the DRs in the highest level send ACKs to the sender, thereby avoiding the ACK-implosion problem. DRs cache received data and respond to retransmission requests of the receivers in their corresponding local regions, thereby decreasing end-to-end latency. RMTP uses a packet-based selective repeat retransmission scheme for higher throughput. This paper also provides the measurements of RMTP's performance with receivers located at various sites in the Internet.
Dave Meyer, "Dvmrp/pim pilot project," in North American Network Operators' Group Meeting Notes (Stan Barber, ed.), (San Diego, California), North American Network Operators' Group, NANOG, Feb. 1996.
Keywords: DVMRP; PIM; multicast routing
Nick McKeown and Balaji Prabhakar, "Scheduling Multicast Cells in an Input-Queued Switch," in Proceedings of the Conference on Computer Communications (IEEE Infocom), (San Fransisco, California), Mar. 1996.
Abstract: In this paper we consider policies for scheduling cells in an input-queued cell switch. It is assumed that each input maintains a single queue for arriving multicast cells and that only the cell at the head of line (HOL) can be observed and scheduled at one time. The policies are assumed to be work-conserving, which means that cells may be copied to the outputs that they request over several cell times. When a scheduling policy decides which cells to schedule, contention may require that it leave a \em residue of cells to be scheduled in the next cell time. The selection of where to place the residue uniquely defines the scheduling policy. We prove that for a 2xN switch, a policy that always concentrates the residue, subject to a natural fairness constraint, always outperforms all other policies. Simulation results indicate that this policy also performs well for more general MxN switches. We present a heuristic round-robin policy called mRRM that is simple to implement in hardware, fair, and performs almost as well as the concentrating policy.
Srini Tridandapani Biswanath Mukherjee, "Multicast traffic in multi-hop lightwave networks: performance analysis and an argument for channel sharing," in Proceedings of the Conference on Computer Communications (IEEE Infocom), (San Fransisco, California), Mar. 1996.
Abstract: A local lightwave network can be constructed by employing two-way fibers to connect nodes in a passive-star physical topology, and the available optical bandwidth may be effectively accessed by the nodal transmitters and receivers at electronic rates using wavelength division multiplexing (WDM). The number of channels, w, in a WDM network is limited by technology and is usually less than the number of nodes, N, in the network. We provide a general method using channel-sharing to construct practical multi-hop networks under this limitation. Channel-sharing may be achieved through time-division-multiplexing. The method is applied to a generalized shuffle-exchange-based architecture. Multicasting-the ability to transmit information from a single source node to multiple destination nodes-is becoming an important requirement in high-performance networks. Multicasting, if improperly implemented, can be bandwidth-abusive. Channel sharing is one approach towards efficient management of multicast traffic. We develop a general modeling procedure for the analysis of both unicast (point-to-point) and multicast (point-to-multipoint) traffic in shared-channel, multi-hop WDM networks. The analysis is comprehensive in that it considers all components of delay that packets in the network experience-namely, synchronization, queueing, transmission, and propagation. The results show that, in the presence of multicast traffic, WDM networks with w less than N channels may actually perform better than if w=N channels are used. This work was performed while Srini Tridandapani was with UC Davis. He is currently with Iowa State University.
Yoram Ofek and Bulent Yener, "Reliable Concurrent Group Multicast from Bursty Sources," in Proceedings of the Conference on Computer Communications (IEEE Infocom), (San Fransisco, California), Mar. 1996.
Abstract: This paper presents a protocol and design for concurrent and reliable group multicast (many-to-many) from bursty data sources in general networks. In a group multicast, any node in the group can be a multicast source and multiple nodes in the group may start to multicast simultaneously, i.e., an asynchronous access to the network. The reliable multicast protocol presented in this work is window based with a combined sender and receiver initiation of the reliable transmission protocol. In reliable multicasting usually there is no hard real-time requirement, and the necessary requirement is to ensure that data is received correctly by all the active members of the group. The approach taken in this work is to combine the multicast operation with the internal flow control, as a result, it is possible to provide: (i) loss-free multicast routing with immediate feedback to the sender, with (ii) single ACKnowledgement message. Furthermore, in every multicast, (iii) a node can access all the capacity allocated to its group with no delay, however, if several nodes are active in the same group then the capacity will be shared fairly. In addition, (iv) each sender in the multicast group uses a single timer, and (v) a node can join and leave a multicast group in a transparent fashion, i.e., there is no need to explicitly notify the members of the group. A multiple criteria optimization study of the bandwidth allocation to each multicast group is presented. The optimization problem has two min-max objective functions: (i) for delay, which caused by the number of links needed to connect the group, and (ii) for congestion, which is caused by sharing a link among multiple multicast groups.
Colin Perkins and Jon Crowcroft, "Real time audio and video transmission of IEEE GLOBECOM'96 over the Internet," technical report, University College London, London, England, Nov. 1996.
Abstract: This article is about the experiences that we had in transmitting the proceedings of some events at the IEEE Globecom '96 in London, England, in the week of 17-22 November, 1996. Live Video and Audio of all of the events in the Churchill Auditorium of the Queen Elizabeth II Conference Center we captured and transmitted, in real time, as well as stored and transmitted later, for remote participants in 3 continents, over the Internet. Two independent systems were used simultaneuously, one supplied by researchers from NTT laboratories in Japan, and the other by researchers from UCL. The former system is based on a server model of distribution1, whilst the latter is based on the use of network level packet multicast. Both systems employ compression algorithms so that the network capacity requirement in each case was of the order of 100 kbps to 200 kbps total, thus enabling remote participants without very high end network connectivity to take part. Receivers neeed only software for a PC runing most popular versions of Windows or a Unix workstation to be able to receive either type of transmission, or to retrieve the recorded sessions from NTT laboratories' servers. The multimedia transmission was carried over carefully engineeredlinks that traversed many different subnet technologies, including point-to-point circuits, SMDS networks, ATM networks, and fast ethernet switches. This was both to give a high level of assurance that the traffic would not experience too much interference from other traffic at the site and elsewhere, and to ensure very low packet store and forward delays. The system ran for 4 days continuously, and was generally very succesful. In the future, it should be possible to have remote paying attendees.
Keywords: Mbone; Globecom; multicast; video conferencing; packet audio; packet video
Ram Ramanathan, "An Algorithm for Multicast Tree Generation in Networks with Asymmetric Links," in Proceedings of the Conference on Computer Communications (IEEE Infocom), (San Fransisco, California), Mar. 1996.
Abstract: We formulate the problem of multicast tree generation as one of computing, in a directed graph, a Directed Steiner tree of minimal cost. In this context, we present a polynomial-time algorithm that provides for tradeoff selection, using a single parameter K, between the tree-cost (Steiner cost) and the runtime efficiency. Further, the same algorithm may be used for delay optimization or tree-cost minimization simply by configuring the value of K appropriately. We present theoretical and experimental analysis characterizing the problem and the performance of our algorithm. Theoretically, we (1) show that it is highly unlikely that there exists a polynomial-time algorithm with a performance guarantee of constant times optimum cost (2) introduce metrics for measuring the asymmetry of graphs and (3) show that the worst-case cost of the tree produced by our algorithm is at most twice the optimum cost times tow of these asymmetry measures. For graphs with bounded asymmetry, this gives constant times optimum performance guarantee and is significant in light of (1). We also show that three well-known algorithms for (undirected) Steiner trees are but particular cases of our algorithm. Our experimental study shows that operating at a low K gives nearly best possible average tree cost while maintaining acceptable runtime efficiency.
George N. Rouskas and Ilia Baldine, "Multicast Routing with End-to-End Delay and Delay Variation Constraints," in Proceedings of the Conference on Computer Communications (IEEE Infocom), (San Fransisco, California), Mar. 1996.
Abstract: We study the problem of constructing multicast trees to meet the quality of service requirements of real-time, interactive applications operating in high-speed packet-switched environments. In particular, we assume that multicast communication depends on (a) bounded delay along the paths from the source to each destination, and (b) bounded variation among the delays along these paths. We first establish that the problem of determining such a constrained tree is \cal NP-complete. We then derive heuristics that demonstrate good average case behavior in terms of the maximum inter-destination delay variation of the final tree. We also show how to dynamically reorganize the initial tree in response to changes in the destination set, in a way that is minimally disruptive to the multicast session.
Anup Rao and Rob Lanphier, "Real time streaming protocol (RTSP)," Internet Draft, Internet Engineering Task Force, Nov. 1996. Work in progress.
Abstract: The Real Time Streaming Protocol, or RTSP, is an application-level protocol for control over the delivery of data with real-time properties. RTSP provides an extensible framework to enable controlled, on-demand delivery of real- time data, such as audio and video. Sources of data can include both live data feeds and stored clips. This protocol is intended to control multiple data delivery sessions, provide a means for choosing delivery channels such as UDP, multicast UDP and TCP, and delivery mechanisms based upon RTP (RFC 1889). RTSP uses the Session Control Protocol (SCP) (see appendix) to allow the use of a single TCP connection between the client and server for controlling delivery of one or more streams of data.
Keywords: stream control; RTSP; packet audio; packet video; media-on-demand; VOD; VCR
H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP: a transport protocol for real-time applications," Request for Comments (Proposed Standard) RFC 1889, Internet Engineering Task Force, Jan. 1996.
Abstract: This memo describes RTP, the real-time transport protocol. RTP provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services. RTP does not address resource reservation and does not guarantee quality-of-service for real-time services. The data transport is augmented by a control protocol (RTCP) to allow monitoring of the data delivery in a manner scalable to large multicast networks, and to provide minimal control and identification functionality. RTP and RTCP are designed to be independent of the underlying transport and network layers. The protocol supports the use of RTP-level translators and mixers. This RFC is the product of the Audio/Video Transport Working Group of the IETF.
Henning Schulzrinne, "Dynamic configuration of conferencing applications using pattern-matching multicast," Multimedia Systems, vol. 2, Mar. 1996.
Abstract: Multimedia conferencing systems are usually large, complex software systems. We describe a local control architecture and communication protocols called pattern-matching multicast (PMM) that allow media agents, controllers and auxiliary applications such as media recorders and management proxies to be tied together into a single conference application. Unlike other systems, control of a single conference can be shared between several controllers. Each medium can be handled by one or more independent media agents. Parts of the system have been implemented using an IP-multicast-based audio conferencing tool (NeVoT). The communicating applications disseminate state and control information through a distributor. The distributor mainly limits distribution of messages based on expressed interest of other applications, thus implementing an application-level, receiver-driven local multicast. It also automatically starts applications as needed. The same inter-application protocol was also implemented using IP multicast restricted to the local host and can be based on other inter-process messaging services such as ToolTalk.
Keywords: conference control; PMM; local control; multicast
Henning Schulzrinne, "Internet telephony - towards the integrated services internet," in Proc. of IEEE Workshop on Internet Telephony, (Utrecht, The Netherlands), Feb. 1996.
Abstract: Currently, the Internet is mostly used for non-real time, data services such as electronic mail, news groups or WWW browsing. Increased availability of high-speed modems and ISDN as well as audio-equipped workstations and PCs have made it feasible to use the Internet for telephony, as well as an alternative for circuit-switched multimedia conferencing applications. Besides possible economic advantages, the Internet allows much easier addition of advanced functionalities and user interfaces. However, a large number of technical and infrastructure problems remain to be solved before Internet telephony becomes viable on a large scale. We present measurement results on Internet behavior, and algorithms to compensate for the Internet-specific impairments, in particular, large delay variations. Bandwidth control adapts encodings to the available bandwidth. A multicast-based signaling protocol allows to set up connections to the callee's email address, without having to know the callee current network location. An example research application, NeVoT, incorporates these algorithms and protocols.
Keywords: Internet telephony; packet audio; packet video; conference control; NeVoT
Henning Schulzrinne, "A real-time stream control protocol (RTSP')," Internet Draft, Internet Engineering Task Force, Dec. 1996. Work in progress.
Abstract: The Real Time Streaming Protocol, or RTSP, is an application-level protocol for control over the delivery of data with real-time properties. RTSP provides an extensible framework to enable controlled, on-demand delivery of real- time data, such as audio and video. Sources of data can include both live data feeds and stored clips. This protocol is intended to control multiple data delivery sessions, provide a means for choosing delivery channels such as UDP, multicast UDP and TCP, and delivery mechanisms based upon RTP (RFC 1889).
Keywords: RTSP; streaming protocol
Krishna Sivalingam and Patrick Dowd, "A Lightweight Media Access Protocol for WDM-Based Distributed Shared Memory System," in Proceedings of the Conference on Computer Communications (IEEE Infocom), (San Fransisco, California), Mar. 1996.
Abstract: LIGHTNING, a WDM testbed currently under construction which has been designed for high-performance supercomputer interconnection. The architecture is based on a dynamically reconfigurable hierarchical WDM network that is being constructed to interconnect a large number of supercomputers and create a distributed shared memory (DSM) environment. This paper describes the network media access protocol based on a single tunable transmitter and single tunable receiver (TT-TR) per node which exploits the bimodal traffic characteristics of a DSM system. The primary objectives of the protocol design are reduced average latency per packet, support of broadcast/multicast, and support of collision-less communication. The proposed approach is compared to an earlier protocol based on one tunable transmitter and one fixed receiver (TT-FR) per node. The performance of the protocol in terms of average latency and channel utilization is analyzed for varying system characteristics such as number of nodes, channels, and levels. Both uniform and non-uniform traffic patterns are considered in the performance analysis.
Michael Smirnov, "Efficient multicast routing in high speed networks," in Computer Communications, (Leics, UK), Jan. 1996.
Abstract: The paper presents jointly two algorithms: for a minimal cost shared multicast tree construction (MCTC) and for a scalable multicast session call establishment, called multicast tree development (MCTD). Both algorithms are of original design based on a network simplex presentation, providing new effective approach to the MCTC, and object-oriented modelling framework providing highest scalability for sparse groups with high dynamics of membership, different quality of service requirements and multiple senders. A new approach is proposed for the MCT construction: bidirectional growing of the MCT without pruning; instead the MCTD makes use of extra branches to improve reliability and advertise new receivers.
Keywords: high-speed networks, routing; multimedia streams; multicast tree; object-oriented framework; call model; scalability
Henning Schulzrinne, Michael Smirnov, Rudolf Roth, and Adam Wolisz, "Ip multicasting over atm: The multicube approach," in Proceedings of Third International Symposium on Interworking - INTERWORKING'96: High-Speed Networking and Interoperability (J. C. Luetchford S.~Rao, H.~Uose, ed.), (Nara, Japan), pp. 181-190, IOS Press/Ohmsha, Oct. 1996.
Abstract: This paper gives practical information along with theoretical background on Multicube project design decisions for scalable and reliable multicast service facilitating multimedia interworking of real users over the European ATM pilot. IP multicast mapping is proposed to be performed with the use of a minimal address resolution server serving a special multicast LIS within the ATM LAN. The feasibility of this choice is proved partly by practical experiment partly via simulation. Data applications for CSCW scenarios depend on a reliable information transfer. The Multicube project selected Xy, an X application sharing tool, to study the reliable multicast services provided by the protocols MTP, RMP and SRM.
Keywords: IP over ATM, multicast, address resolution, reliable transport protocols
Dorgham Sisalem, Henning Schulzrinne, and Christian Sieckmeyer, "The network video terminal," in HPDC Focus Workshop on Multimedia and Collaborative Environments (Fifth IEEE International Symposium on High Performance Distributed Computing), (Syracuse, New York), IEEE Computer Society, Aug. 1996.
Abstract: Currently, a variety of the MBONE video tools provide video conferencing capabilities on different platforms and with a variety of compression algorithms. However, most of these tools lack the ability to interact with other media agents that might be used during a conferencing session. Such interaction is required, for example, for achieving lip synchronisation between audio and video streams or for quality of service control. In this paper, we present a new video tool, NeViT. This tool provides the basic capabilities needed for video conferencing services such as video capturing, compression and decompression engines and multicasting and ATM network interfaces. To ease the interaction with other media agents, nevit incorporates a message handling facility to interact over a local conference bus with other media agents, a floor controller or the conference controller. Currently, we are working on adding lip synchronisation and quality-of-service control using this conference bus.
Keywords: video conferencing; NeViT; intermedia synchronisation; ATM; MBone
Francois Toutain and Oliver Huber, "A General Preemption-based Admission Policy Using A Smart Market Approach," in Proceedings of the Conference on Computer Communications (IEEE Infocom), (San Fransisco, California), Mar. 1996.
Abstract: QoS networks are more and more envisioned to provide preemptable communications. That is, a new connection may cause on-going ones to be interrupted, depending on the current network utilization. For instance, preemption combined with layered video encoding allows graceful degradation of the received stream to be performed, as more and more users compete for resources. We propose a general admission policy that solves the preemption-based resource allocation problem. This policy allows users to compete for connection setup and keeping, via a smart market approach. Algorithmic and accounting issues are investigated, inside a single node, in an end-to-end approach and for multicast communica- tions. Our approach must be appreciated as a building block for a higher-level administration policy, in which it provides controlled user competition.
David G. Thaler and Chinya V. Ravishankar, "Distributed Center-Location Algorithms: Proposals and Comparisons," in Proceedings of the Conference on Computer Communications (IEEE Infocom), (San Fransisco, California), Mar. 1996.
Abstract: Recent multicast routing protocol proposals such as PIM and CBT have been based on the notion of group-shared trees. Since construction of a minimal-cost tree spanning all members of a group is difficult, they rely on center-based trees, and distribute packets from all sources over a single shortest-path tree rooted at some center. While, PIM and CBT provisionally use administrative selection of centers or trivial heuristics for locating the center of a group, they do not preclude the use of other methods as long as they provide an ordered list of centers. Other previously proposed heuristics typically require knowledge of the complete network topology, a requirement which is not always practical for a distributed problem such as Internet routing. In this paper we investigate the problem of finding a good center in distributed fashion, and study various heuristics for automating center selection and examine their applicability to real-world networks. We also propose several new algorithms which we feel to be more practical than existing methods. We present simulation results showing that of the methods potentially feasible in the Internet Multicast Backbone, ours offer the best results in terms of cost and delay.
Brett J. Vickers, Meejeong Lee, and Tatsuya Suda, "Feedback control mechanisms for real-time multipoint video services," to appear in JSAC Network Support for Multipoint Communications, vol. ?, pp. -, ? 1996.
Abstract: While existing research shows that reactive congestion control mechanisms are capable of providing high video quality and channel utilization for point-to-point real-time video, there has been relatively little study of the reactive congestion control of point-to-multipoint video, especially in ATM networks. Problems complicating the provision of multipoint, feedback-based real-time video service include (1) implosion of feedback returning to the source as the number of multicast destinations increases, and (2) variance in the amount of available bandwidth on different branches in the multipoint connection. In this paper, a new service architecture is proposed for real-time multicast video, and two multipoint feedback mechanisms to support this service are introduced and studied. The mechanisms support a minimum bandwidth guarantee and the best effort support of video traffic exceeding the minimum rate. They both rely on adaptive, multi-layered coding at the video source and closed-loop feedback from the network in order to control both the high and low priority video generation rates of the video encoder. Simulation results show that the studied feedback mechanisms provide, at the minimum, a quality of video comparable to a CBR connection reserving the same amount of bandwidth. When unutilized network bandwidth becomes available, the mechanisms are capable of exploiting it to dynamically improve video quality beyond the minimum guaranteed level.
Keywords: ATM; ABR; packet video; congestion control
Robert Voigt, "Framework for a global hierarchical multicast," in North American Network Operators' Group Meeting Notes (Stan Barber, ed.), (San Diego, California), North American Network Operators' Group, NANOG, Feb. 1996.
Oliver T. W. Yu and Victor C. M. Leung, "Signaling Network Architecture and Transaction Protocols to Support Realtime Connection Rerouting in ATM/B-ISDNs," in Proceedings of the Conference on Computer Communications (IEEE Infocom), (San Fransisco, California), Mar. 1996.
Abstract: In future B-ISDN and PCN environments, an increasing number of network control services, such as dynamic multiparty conferences, mobile call handoff, etc., will require realtime connection rerouting services not currently supported by the B-ISDN signaling network architecture based on the SS7 CCS network and its transaction protocols (TCAP). In this paper, we propose a novel signaling network architecture employing the following new signaling transport protocols in the signaling ATM adaptation layer: (1) the associated-CCS transport protocols employing source-routing unicast and multicast to provide fast distributed signaling transport for reatime distributed transactions, and (2) the inband signaling transport protocol to provide inband synchronization data transport required by traffic synchronization transactions, to support realtime connection rerouting services in ATM/B-ISDNs realized by the following new transaction protocols in the TCAP layer: (1) the robust fast reservation transaction protocol for reservation of communication resources, (2) the packet-ordering synchronization transaction protocol for cell synchronization to minimize traffic disruptions.