Global IP Packet Switching - Time-driven Priority

Chung-Sheng Li, Yoram Ofek, Adrian Segall, Khosrow Sohraby, "Pseudo-Isochronous Cell Switching in ATM Networks," IEEE INFOCOM'94, Pages: 428-437, 1994.

ABSTRACT

This paper shows how to design an ATM network, for real-time traffic, such that under full network load (i) the maximum delay of a low-rate voice connection is minimized, (ii) the delay uncertainty or jitter is a fixed network parameter - independent of the network size, and (iii) the required buffer sizes (inside the network) to ensure loss-free routing is minimized. In addition, this design can be generalized to accommodate either variable bit rate (VBR) traffic with statistical multiplexing or the integration of available bit rate (ABR) traffic.
The isochronous timing information is used only for regulating and pacing the traffic flow inside the network, rather than routing as in traditional circuit switching networks. This means that an ATM cell is sent from one switch to another not at a specific time but within a time frame of a relatively long duration as compared to the cell transmission time. This time frame is an independent network parameter, which determines the delay and jitter inside the network. The routing of cells of each connection is based on VCI and VPI, and as a result, timing errors do not affect the ATM routing. A study of the blocking probability of this approach is presented. The study includes both analytical and simulation results, which demonstrate a trade-off between the blocking probability and the jitter, delay and scheduling complexity.

Yoram Ofek, "Generating a Fault Tolerant Global Clock using High-speed Control Signals for the MetaNet Architecture," IEEE T. on Communications, Volume: 42, Number: 5, Pages: 2179-2188, May 1994. Also published in: The 9th International Conference on Distributed Systems, June 1989, pp. 218-226. Also published in: IBM Research Report: RC 16873, May 1991.

ABSTRACT

This work describes a new technique, based on exchanging control signals, for constructing a fault tolerant global clock in a point-to-point distributed system with an arbitrary topology. The approach taken in this work is to generate a global clock from the ensemble of the high-speed local transmission clocks, and not to directly synchronize these clocks. The steady-state algorithm which generates the global clock is executed in hardware by the physical network interface of each node. At the network interface it is possible to measure accurately the propagation delay between neighboring nodes with a very small error or uncertainty, and thereby to achieve global synchronization that is proportional to this measurement error. It is shown that the local clock drift (or rate uncertainty) has only a secondary effect on the maximum global clock rate. The synchronization algorithm can tolerate any physical failure. It will continue to operate correctly on any connected segment of the network, i.e., it can tolerate any number of link and node failures, as long that the network remains connected. Furthermore, the algorithm can tolerate failures of the following types: (i) fast and slow clocks can be detected and isolated from the algorithm, (ii) changes in the value of link delays can be masked, and (iii) malicious changes of the global clock values can be detected and masked.

Yoram Ofek, Moti Yung, "The Integrated MetaNet Architecture: A Switch-based Multimedia LAN for Parallel Computing and Real-time Traffic," IEEE INFOCOM'94, Pages: 802-811, 1994.

ABSTRACT

This work presents a coherent system solution for combining with no loss due to congestion, on one hand, bursty traffic with unpredictable pattern in time and space, and on the other hand, periodic real-time or connection-oriented traffic with bandwidth guaranteed requirements over fixed routes. This solution facilitates the implementation of a scalable high performance multimedia system for distributed/parallel computing with real-time interactive voice and video.
The MetaNet real-time traffic has the following attributes: (1) guaranteed bandwidth with bounded delay, (2) fixed jitter - independent of the network size, (3) no loss due to congestion inside the network, (4) fixed path routing with FIFO order, and (5) support of complex periodicity scheduling. At the same time, the underlying MetaNet asynchronous (bursty) traffic properties (provided by previous works) remain unchanged. Namely, any node can try to transmit asynchronously (bursty traffic), without reservation as much as it can, and the network access and flow control ensure the following traditional LAN properties: (6) no loss with a single input buffer, (7) fair and deadlock-free access, and (8) self-routing with broadcast. The self-routing on the MetaNet is a variant of deflection routing. It makes on-line routing decisions based on the local flow of traffic (load conditions). Unlike other deflection techniques, the MetaNet routing is along a global sense of direction, which guarantees that cells will reach their destinations. Thus, this method is called convergence routing.

Yoram Ofek, "The Conservative Code for Bit Synchronization," IEEE Trans. on Communications, Volume: 38, Number: 7, Pages: 1107-1113, July 1990.

ABSTRACT

A novel coding scheme for bit synchronization the conservative code, is presented and analyzed in this paper. Each codeword is characterized by having a predefined number of transitions, with a known delimiting transition at the end, i.e., the number of transitions is conserved. As a result, it is possible to decode the incoming serial bit-stream without explicitly recovering the receiving clock with a phase-locked loop. Thus, it is possible to receive messages from asynchronous sources without any training period. This code can be extremely useful in communication over a passive optical star.
In the analysis, the efficiency or capacity of the code is computed with two additional constraints (i) limited run-length and (ii) balancing. It will be shown that this coding scheme is efficient when both constraints are applied. The importance of this analysis is to show, that the additional constraints imposed on the encoding, do not cause a significant reduction in its efficiency. Hence, the realization of the encoder and decoder is possible with current technology.
The design of a serial electronic interface is described. Its goal is to maximize the interface bandwidth (greater than 1 gigabit/sec with GaAs technology), which is usually the bottleneck of an optical communication system. It is achieved by reducing the critical timing path to one flip-flop and one NOR gate, and by having a hierarchical interface design of four levels.

Chung-Sheng Li, Yoram Ofek, Moti Yung, "Time-driven Priority Flow Control for Real-time Heterogeneous Internetworking," IEEE INFOCOM'96, March 1996.

ABSTRACT

We consider real-time traffic in a heterogeneous internetworking environment with IP routers, MAC bridges, Hubs, Switched LANs etc. We assume that the current routing protocols remain unchanged. However, in this environment, in order to provide quality of service (QoS): bandwidth, delay, constant-bounded jitter and no-loss due to congestion, we suggest a new flow control function called time-driven priority, which is an internal traffic shaping mechanism.
We show how it supports two classes of connections: constant bit rate (CBR) with deterministic guarantees, and variable bit rate (VBR) with statistical multiplexing. The mechanism does not require to identify and separate the packet flows of different real-time sessions/connections inside the network. As a result, it achieves lower switching complexity when compared with other internal traffic shaping methods. As consequences of the time-driven priority mechanism we further achieve: (1) QoS parameters which are independent of the connection bandwidth, (2) QoS parameters which are independent of the existing heterogeneous internetworking asynchronous data traffic, and (3) the capability for policing and securing the network QoS.

Yoram Ofek, "Integration of Voice Communication on a Synchronous Optical Hypergraph," IEEE INFOCOM'88, 1988. Also published in: IBM Research Report: RC 13341, December 1987.

ABSTRACT

The optical hypergraph is a novel network architecture in which each edge of the hypergraph is a multiple-access broadcast medium constructed as a passive optical star coupler. Access to each net (edge) is time-slotted, and the system maintains global slot synchronization. The integration of voice into the system is done by reserving time slots in a periodic manner. A packet which contains several voice parcels from different phone conversations is transferred in these slots. These parcels may have different destinations on the optical net. As a result of the global end to end synchronization, the delay from the source to the destination is a known constant, with accuracy of plus or minus half a time slot.
In the analysis, it is shown that the system improves its operation as the communication bandwidth increases. In other words, the algorithms and protocols improve in performance as the communication bandwidth increases. Two criteria are used to exhibit this phenomenon (i) the utilization efficiency and (ii) the end to end delay.

Chung-Sheng Li, Yoram Ofek, "Distributed Source-Destination Synchronization in ATM Using Inband Clock Distribution," IEEE J. on Selected Areas in Comm., Volume: 14, Number: 1, Pages: 153-161, January 1996. Also published in: ICC ???????

ABSTRACT

This paper presents a new distributed methodology for source-destination synchronization for interactive teleconferencing. The method is based on a reference clock, which is synthesized from a distributed global clock. The global clock is generated by periodically exchanging inband synchronization signals with neighboring nodes. The timing jitter achieved with this method can be arbitrarily close to the jitter obtained by the centralized synchronous methods which usually use an out-of-band, hard-wired reference clock.
The global clock synchronization algorithm, used in this work, guarantees frequency locking of all the network nodes to the slowest clock in the system. As a result, the slowest clock can be used as an implicit reference clock for source-destination synchronization protocols, such as, Synchronous Frequency Encoding Technique (SFET) and Synchronous Residual Time Stamp (SRTS). This inband synchronization method does not require the explicit knowledge of which clock is actually the slowest in the system. Therefore, if the slowest clock fails, then another clock on a different node will be the slowest, and the nodes will use it as a reference clock for the source-destination synchronization protocol. The existing out-of-band reference clock techniques do not have this strong fault tolerant property.

Yoram Ofek, Michael Faiman, "Distributed Global Event Synchronization in a Fiber Optic Hypergraph Network," The 7th International Conference on Distributed Computing Systems, IEEE, Pages: 307-314, 1987.

ABSTRACT

The principles and the design of global event synchronization in a large area network (about 10,000 nodes within 1000$km sup 2$), are presented and nalyzed in this paper. The network architecture is a hypergraph, its edges are nets or buses, and its vertices are nodes.
The communication over each net is time slotted, and the event duration is one time slot. Synchronization among all the system's nets is maintained and, as a result, this distributed system preserves a total ordering of all the events in the system. It is shown that this total ordering is achieved with small communication overhead (less than 5%). Each net is a passive, centralized optical star, with a bandwidth of about 1 gigabit/second. The high bandwidth enables a wide multiple-access nets, therefore, the dimension of the hypergraph is low. The low dimension allows the timing and state information to propagate quickly through the system, and simplifies the distributed switching in the netwok.

Yoram Ofek, Moshe Sidi, "Design and Analysis of a Hybrid Access Control to an Optical Star using WDM," Journal of Parallel and Distributed Computing, Volume 17, Pages: 259-265, April 1993. Also published in: IEEE INFOCOM'91. Also published in: IBM Research Report: RC 16059, August 1990.

ABSTRACT

A passive optical star is an ideal shared medium, from both fault tolerant and access synchronization point of views. The communication over an optical star merges to a single point in space and then broadcast back to all the nodes. This circular symmetry facilitates the solution for two basic distributed synchronization problems, which are presented in this work: (i) generating global event clock for synchronizing the nodes' operation, and (ii) distributed scheduling for accessing the shared passive medium, which is a hybrid (deterministic and random) technique.
We present, prove and analyze this hybrid scheduling algorithm which is equivalent to a distributed queue, and therefore, is also algorithmically fair. Furthermore, our solution has two additional properties: destination overflow prevention and destination fairness. The effective solution of these problems can be used for efficiently implementing a local area network based on a passive optical star.

M. Baldi, Y. Ofek, "End-to-end Delay Analysis of Videoconferencing over Packet Switched Networks," IEEE/ACM Transactions on Networking, Vol. 8, No. 4, Aug. 2000, pp. 479-492.

ABSTRACT

Videoconferencing is an important global application -- it enables people around the globe to interact when distance separates them. In order for the participants in a videoconference call to interact naturally, the end-to-end delay should be below human perception; even though an objective and unique figure cannot be set, 100 ms is widely recognized as the desired one-way delay requirement for interaction. Since the global propagation delay can be about 100 ms, the actual end-to-end delay budget available to the system designer (excluding propagation delay) can be no more than 10 ms. We identify the components of the end-to-end delay in various configurations with the objective of understanding how it can be kept below the desired 10-ms bound.
We analyze these components step-by-step through six system configurations obtained by combining three generic network architectures with two video encoding schemes. We study the transmission of raw video and variable bit rate (VBR) MPEG video encoding over 1) circuit switching; 2) synchronous packet switching; and 3) asynchronous packet switching. In addition, we show that constant bit rate (CBR) MPEG encoding delivers unacceptable delay -- on the order of the group of pictures (GOP) time interval -- when maximizing quality for static scenes.
This study aims at showing that having a global common time reference, together with time-driven priority (TDP) and VBR MPEG video encoding, provides adequate end-to-end delay, which is 1) below 10 ms; 2) independent of the network instant load; and 3) independent of the connection rate. The resulting end-to-end delay (excluding propagation delay) can be smaller than the video frame period, which is better than what can be obtained with circuit switching.

M. Baldi, Y. Ofek, B. Yener "Adaptive Group Multicast with Time-Driven Priority," IEEE/ACM Transactions on Networking, Vol. 8, No.1, Feb. 2000, pp. 31-43.

ABSTRACT

This paper shows how to provide an adaptive real-time group multicast (many-to-many) communication service. Adaptive means that the number of nodes that transmit to the multicast group is continuously changing. In order to meet deterministic quality-of-service (QoS) requirements of a real-time group multicast, some communication resources are reserved. In this work we show 1) how bandwidth is reserved for each multicast group and 2) how an active source can dynamically share the bandwidth allocated to this multicast group with other active group members.
Quality-of-service support for a real-time multicast group is based on time-driven priority [9]. In this scheme the time is divided into time frames of fixed duration, and all the time frames are aligned by using a common global time reference, which can be obtained from the global positioning system. Bandwidth is allocated to a multicast group as a whole, rather than individually to each user. The allocation is done by reserving time intervals within time frames in a periodic fashion.
This type of allocation raises two problems that are studied in this paper: 1) scheduling: how time intervals are reserved to each multicast group and 2) adaptive sharing: how the active (transmitting) participants can dynamically share the time intervals that have been reserved for their multicast group. The proposed approach is based on the embedding of multiple virtual rings, one for each multicast group. By using the virtual rings, it is simple to route messages to all the participants while minimizing the bound on the buffer sizes and queueing delays. The finalpart of this paper introduces a scalable growth of the multicast group by adding multiple subtrees to the virtual ring.

M. Baldi, Y. Ofek, "Common Time Reference for Interactive Multimedia Applications," IEEE International Conference on Multimedia & Expo (ICME2000), New York, NY, USA, July-Aug. 2000, pp. 1679-1682.

ABSTRACT

A delay of about 100 ms gives human communicators the feeling of live interaction. Since in a global network the propagation delay alone is about 100 ms, every other delay component, such as processing and queuing, should be kept as short as possible.
Moreover, the deployment of new high bandwidth multimedia applications will boost network traffic and consequently the demand for very high capacity transmission technologies, such as Wavelength Division Multiplexing (WDM). Networks will suffer (i) electronic switching bottlenecks among high-speed links and (ii) communications link bottlenecks between high capacity core technologies and low speed access technologies.
This paper addresses the design of interactive systems for applications such as toll quality telephony, videotelephony and videoconferencing, highlighting the benefits brought by the availability of global common time reference derived from GPS (Global Positioning System). Common time reference is essential to keep the user perceived delay within the 100 ms bound while avoiding the two above mentioned bottlenecks. The proposed solution can be applied to both IP and ATM networks, does not require changes to any of the existing protocols, and enables traffic aggregation in the core of the network–thus not requiring nodes to keep state information on microflows-while providing a guaranteed quality service to individual applications.

M. Baldi, Y. Ofek, "Blocking Probability with Time-driven Priority Scheduling," SCS Symposium on Performance Evaluation of Computer and Telecommunication Systems (SPECTS 2000), Vancouver, BC, Canada, July 2000.

ABSTRACT

This paper evaluates call blocking probability over packet-switched networks with Time-driven priority (TDP) (Li et al. 1998; Li, Ofek and Yung 1996). The presented work is novel since call blocking is typically studied in the context of circuit-switched networks. TDP together with resource reservation enables real-time delivery of packets with no loss due to congestion and constant jitter of one time frame (TF).typically between 12.5 µs and 125 µs. Resource reservation for a call (or multiple calls) over a TDP network requires finding a schedule. A call may not be accepted for two reasons (i) there is no capacity - the call is rejected - or (ii) there is capacity but no schedule.the call is blocked. This work studies the call blocking probability as a function of the link utilization, since call blocking can possibly lead to low link utilization. In other words, it may not be possible to fully utilize the network because of unschedulability (i.e., the inability to find a schedule).
The results show that it is possible to achieve high link utilization and that call blocking, in most cases, is negligible. Moreover, the result shows how the blocking problem diminishes as the link bandwidth increases. This is achieved without increasing the complexity of the schedule computation and the scheduler run-time operation in TDP switches.

M. Baldi, Y. Ofek, "Ring versus Tree Embedding for Real-time Group Multicast," 18th Joint Conference of the IEEE Computer and Communication Societies (INFOCOM '99), New York, NY, USA, March 1999, pp. 1099-1106, vol. 3.

ABSTRACT

In general topology networks, routing from one node to another over a tree embedded in the network is intuitively a good strategy, since it typically results in a route length of O(\log n) links, being n the number of nodes in the network. Routing from one node to another over a ring embedded in the network would result in route length of O(n) links. However, in group (many-to-many) multicast, the overall number of links traversed by each packet, i.e., the networks elements on which resources must be possibly reserved, is typically O(N) for both tree and ring embedding, where N is the size of the group. This paper focuses on the tree versus ring embedding for real-time group multicast in which all packets should reach all other nodes in the group with a bounded end-to-end delay. In this work, real-time properties are guaranteed by the deployment of time-driven priority in network nodes.
In order to have a better understanding of the non-trivial problem of ring versus tree embedding, we consider the following group multicast scenarios: (i) static - fixed subset of active nodes, (ii) dynamic - fixed number of active nodes (i.e., the identity of active nodes is changing over time, but its size remains constant), and (iii) adaptive - the number and identity of active nodes change over time.
Tree and ring embedding are compared using the following metrics: (i) end-to-end delay bound, (ii) overall bandwidth allocated to the multicast group, and (iii) signaling overhead for sharing of the resources allocated to the group. The results are interesting and counter-intuitive, since, as shown, embedding a tree is not always tbe best strategy. In particular, dynamic and adaptive multicast on a tree requires a protocol for updating state information and coordinates the operation of the group. Such a protocol is not required on the ring where the circular tapology, and implicit token passing mechanisms are sufficient. Moreover, the bandwidth allocation on the ring for the three multicast scenarios is O(N); while on a general tree it is O(N) for the static multicast scenario and O(N^{2}) for the dynamic and adaptive multicast scenarios.

M. Baldi, Y. Ofek, "End-to-end Delay of Videoconferencing over Packet Switched Networks," 9th IEEE Workshop on Local and Metropolitan Area Networks, Banff, Alberta, Canada, May 1998.

ABSTRACT

Videoconferencing is an important global application -- it enables people around the globe to interact when distance separates them. In order for the participants in a videoconference call to interact naturally, the end-to-end delay should be below human perception; even though an objective and unique figure cannot be set, 100 ms is widely recognized as the desired one-way delay requirement for interaction. Since the global propagation delay can be about 100 ms, the actual end-to-end delay budget available to the system designer (excluding propagation delay) can be no more than 10 ms. We identify the components of the end-to-end delay in various configurations with the objective of understanding how it can be kept below the desired 10-ms bound.
We analyze these components step-by-step through six system configurations obtained by combining three generic network architectures with two video encoding schemes. We study the transmission of raw video and variable bit rate (VBR) MPEG video encoding over 1) circuit switching; 2) synchronous packet switching; and 3) asynchronous packet switching. In addition, we show that constant bit rate (CBR) MPEG encoding delivers unacceptable delay -- on the order of the group of pictures (GOP) time interval -- when maximizing quality for static scenes.
This study aims at showing that having a global common time reference, together with time-driven priority (TDP) and VBR MPEG video encoding, provides adequate end-to-end delay, which is 1) below 10 ms; 2) independent of the network instant load; and 3) independent of the connection rate. The resulting end-to-end delay (excluding propagation delay) can be smaller than the video frame period, which is better than what can be obtained with circuit switching.

M. Baldi, Y. Ofek, "End-to-end Delay of Videoconferencing over Packet Switched Networks," 17th Joint Conference of the IEEE Computer and Communication Societies (INFOCOM '98), San Francisco, CA, USA, Apr. 1998.

ABSTRACT

Videoconferencing is an important global application -- it enables people around the globe to interact when distance separates them. In order for the participants in a videoconference call to interact naturally, the end-to-end delay should be below human perception; even though an objective and unique figure cannot be set, 100 ms is widely recognized as the desired one-way delay requirement for interaction. Since the global propagation delay can be about 100 ms, the actual end-to-end delay budget available to the system designer (excluding propagation delay) can be no more than 10 ms. We identify the components of the end-to-end delay in various configurations with the objective of understanding how it can be kept below the desired 10-ms bound.
We analyze these components step-by-step through six system configurations obtained by combining three generic network architectures with two video encoding schemes. We study the transmission of raw video and variable bit rate (VBR) MPEG video encoding over 1) circuit switching; 2) synchronous packet switching; and 3) asynchronous packet switching. In addition, we show that constant bit rate (CBR) MPEG encoding delivers unacceptable delay -- on the order of the group of pictures (GOP) time interval -- when maximizing quality for static scenes.
This study aims at showing that having a global common time reference, together with time-driven priority (TDP) and VBR MPEG video encoding, provides adequate end-to-end delay, which is 1) below 10 ms; 2) independent of the network instant load; and 3) independent of the connection rate. The resulting end-to-end delay (excluding propagation delay) can be smaller than the video frame period, which is better than what can be obtained with circuit switching.

M. Baldi, Y. Ofek, B. Yener, "Adaptive Real Time Group Multicast," 16th Joint Conference of the IEEE Computer and Communication Societies (INFOCOM '97), Kobe, Japan, Apr. 1997.

ABSTRACT

This paper shows how to provide an adaptive real-time group multicast (many-to-many) communication service. Such a service can be used by applications, like audio/video tele-conferencing, that require low loss, and bounded delay and jitter. In order to meet deterministic quality of service (QoS) requirements of a real-time group multicast, some communication resources are reserved. In this work we show (i) how bandwidth is reserved for each group, and (ii) how an active user in a multicast group can dynamically share, in an efficient and fair manner, the bandwidth allocated to its group.Quality of service support for a real-time multicast group is based on time driven priority. In this scheme the time is divided into time frames of fixed duration and all the time frames are aligned by using a global time reference which can be obtained from GPS (global positioning system). Bandwidth is allocated to a multicast group as a whole, rather than individually to each user. The allocation is done by reserving time intervals within time frames in some periodic fashion.This sort of allocation raises two problems that are studied in this paper: (1) Scheduling: how time intervals are reserved to each multicast group, and (2) Adaptive sharing: how the participants dynamically share the time intervals that have been reserved for their multicast group. The proposed approach is based on embedding multiple virtual rings, one for each multicast group. By using the virtual rings it is simple to route messages to all the participants, while minimizing the bound on the buffer sizes and queuing delays.

M. Baldi, Y. Ofek, B. Yener, "Adaptive Real-Time Group Multicast," RC 20686 (91481), IBM - T. J. Watson Research Center, Yorktown Heights, NY, USA, Dec. 1996.

ABSTRACT

This paper shows how to provide an adaptive real-time group multicast (many-to-many) communication service. Such a service can be used by applications, like audio/video tele-conferencing, that require low loss, and bounded delay and jitter. In order to meet deterministic quality of service (QoS) requirements of a real-time group multicast, some communication resources are reserved. In this work we show (i) how bandwidth is reserved for each group, and (ii) how an active user in a multicast group can dynamically share, in an efficient and fair manner, the bandwidth allocated to its group.Quality of service support for a real-time multicast group is based on time driven priority. In this scheme the time is divided into time frames of fixed duration and all the time frames are aligned by using a global time reference which can be obtained from GPS (global positioning system). Bandwidth is allocated to a multicast group as a whole, rather than individually to each user. The allocation is done by reserving time intervals within time frames in some periodic fashion.This sort of allocation raises two problems that are studied in this paper: (1) Scheduling: how time intervals are reserved to each multicast group, and (2) Adaptive sharing: how the participants dynamically share the time intervals that have been reserved for their multicast group. The proposed approach is based on embedding multiple virtual rings, one for each multicast group. By using the virtual rings it is simple to route messages to all the participants, while minimizing the bound on the buffer sizes and queuing delays.

M. Baldi, Y. Ofek, "End-to-end Delay of Videoconferencing over Packet Switched Networks," RC 20669 (91480), IBM - T. J. Watson Research Center, Yorktown Heights, NY, USA, Dec. 1996.

ABSTRACT

Videoconferencing is an important global application -- it enables people around the globe to interact when distance separates them. In order for the participants in a videoconference call to interact naturally, the end-to-end delay should be below human perception; even though an objective and unique figure cannot be set, 100 ms is widely recognized as the desired one-way delay requirement for interaction. Since the global propagation delay can be about 100 ms, the actual end-to-end delay budget available to the system designer (excluding propagation delay) can be no more than 10 ms. We identify the components of the end-to-end delay in various configurations with the objective of understanding how it can be kept below the desired 10-ms bound.
We analyze these components step-by-step through six system configurations obtained by combining three generic network architectures with two video encoding schemes. We study the transmission of raw video and variable bit rate (VBR) MPEG video encoding over 1) circuit switching; 2) synchronous packet switching; and 3) asynchronous packet switching. In addition, we show that constant bit rate (CBR) MPEG encoding delivers unacceptable delay -- on the order of the group of pictures (GOP) time interval -- when maximizing quality for static scenes.
This study aims at showing that having a global common time reference, together with time-driven priority (TDP) and VBR MPEG video encoding, provides adequate end-to-end delay, which is 1) below 10 ms; 2) independent of the network instant load; and 3) independent of the connection rate. The resulting end-to-end delay (excluding propagation delay) can be smaller than the video frame period, which is better than what can be obtained with circuit switching.