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Selected Areas in Communications, IEEE Journal on

Issue 5 • Date Jun 1989

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  • Fixed distortion subband coding of images for packet-switched networks

    Publication Year: 1989 , Page(s): 789 - 800
    Cited by:  Papers (14)  |  Patents (13)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1276 KB)  

    A subband image codec is presented that approximately attains a user-prescribed fidelity by allowing the encoder's compression rate to vary. The fixed distortion subband coding (FDSBC) system is suitable for use with future of packet-switched networks. The codec's design is based on an algorithm that allocates distortion among the subbands to minimize channel entropy. By coupling this allocation procedure with judiciously selected subband quantizers, an elementary four-band codec was obtained. Additional four-band structures may be nested in a hierarchical configuration for improved performance. Each of the configurations tested attains mean square distortions within 2.0 dB of the user-specific value over a wide range of distortion for several standard test images. Rate versus mean-square distortion performance rivals that of fixed-rate systems having similar complexity. The encoder's output is formatted to take advantage of prioritized packet networks. Simulations show that FDSBC is robust with respect to packet loss and may be used effectively for progressive transmission applications View full abstract»

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  • Missing packet recovery techniques for low-bit-rate coded speech

    Publication Year: 1989 , Page(s): 707 - 717
    Cited by:  Papers (30)  |  Patents (8)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (916 KB)  

    Since missing-packet recovery techniques for conventional PCM speech are not applicable to packetized speech communication systems with low-bit-rate coding schemes, quality degradation mechanisms are presented for missing-packet recovery techniques. These mechanisms are least significant bit (LSB) dropping, waveform substitution, and odd-even sample-interpolation schemes. A comparison of these techniques in terms of signal-to-noise ratio and perceptual distortion under packet loss conditions shows that the LSB-dropping scheme with embedded coding is the most promising technique for recovering missing packets View full abstract»

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  • An integrated services token-controlled ring network

    Publication Year: 1989 , Page(s): 670 - 679
    Cited by:  Papers (11)  |  Patents (5)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (748 KB)  

    A ring protocol is proposed that allows voice and data traffic to coexist within the ring. This ring has the following distinct features: (1) it allows synchronous traffic such as voice and video to have a definite access to the channel within each packetization period (or frame); (2) it allows data messages to have a higher channel access priority provided that the synchronous traffic is not delayed by more than one frame; (3) it supports variable rate data circuits. Simulation results show that the data-message delay is much smaller than for other integrated services schemes. Urgent messages can be transmitted with a higher priority over voice. Since the voice packet delay is bounded within one packetization period, no time-stamping is needed and the voice loss can be completely avoided by reserving a sufficient number of slots. Continual speech reception is possible by synchronizing the speech regeneration process to the end of each frame. Since the ring is synchronized, gateway switching to external circuit-switched and packet-switched networks is very simple View full abstract»

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  • Packetized radiographic image transfers over local area networks for diagnosis and conferencing

    Publication Year: 1989 , Page(s): 842 - 856
    Cited by:  Papers (6)  |  Patents (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1260 KB)  

    The authors present a multimedia medical communications system based on packet local area networks. This system is intended to facilitate the communications between radiologists and physicians within a hospital. Radiographs are taken, digitized, and then stored in the database. Radiologists retrieve radiographic images from the database for diagnosis, and make diagnostic reports to be stored in the database. Physicians can then consult these radiographic images and diagnostic reports. A real-time conference can be set up between a radiologist and a physician who are in different locations, for discussing a particular case or radiographic images can be retrieved from the database for diagnosis and conferencing. Simulations indicate the delay for displaying an image of 1 Mb from the local storage (hard disk) at a workstation is approximately 6.5 s View full abstract»

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  • Encoding facsimile images for packet-switched networks

    Publication Year: 1989 , Page(s): 857 - 864
    Cited by:  Papers (5)  |  Patents (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (656 KB)  

    An image compression algorithm is presented that is suitable for transmitting a high-resolution (400 dot/in) black-and-white (two-tone) facsimile image over a packet network. The algorithm decomposes the image into nondroppable (essential) and droppable (supplementary) bits. The nondroppable bits describe the image at low resolution (200 dot/in) and can be coded with any of the standard CCITT Group 3 or Group 4 techniques. Thus, the algorithm is compatible with the large installed base of Group 3 facsimile machines. The droppable (supplementary) bits are approximately 45% of the total number of bits and can be dropped by the network to relieve congestion. The supplementary bits are encoded with a predictive Huffman method. Resynchronization after missing some of the supplementary information presents no serious difficulties. The algorithm is information preserving, assuming that all droppable bits are available, and its compression efficiency is equal to or better than that of direct two-dimensional Group 3 or 4 CCITT encoding of the high-resolution (400 dot/in) image. Frame memory is not needed View full abstract»

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  • Comparative performance of voice/data local area networks

    Publication Year: 1989 , Page(s): 657 - 669
    Cited by:  Papers (14)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1160 KB)  

    Using simulation, a network-independent framework compares the performance of contention-based Ethernet and two contention-free round-robin schemes, namely Expressnet and the IEEE 802.4 token bus. Two priority mechanisms for voice/data traffic on round-robin networks are studied: the alternating-rounds mechanism of the Expressnet, and the token rotation timer mechanism of the token bus, which restricts access rights based on the time taken for a token to make one round. It is shown that the deterministic schemes almost always perform better than the contention-based scheme. Design issues such as the choice of minimum voice packet length, priority parameters, and voice encoding rate are investigated. An important aspect that is noted is the accurate characterization of performance over a wide region of the design space of voice/data networks View full abstract»

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  • Priority discarding of speech in integrated packet networks

    Publication Year: 1989 , Page(s): 644 - 656
    Cited by:  Papers (35)  |  Patents (14)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1160 KB)  

    The authors discuss the control of short-term congestion, which is referred to as overload, in integrated packet networks (IPNs) containing a mix of data, speech, and possibly other types of signals. A system model is proposed that assigns a delivery priority to each packet (speech or otherwise) at the transmitter and discards speech packets according to delivery priority at any point in the network in response to overload. This model attempts to minimize per-packet processing at networks nodes. The research described is guided by two principles for IPN design: minimal per-packet processing and flexibility due to signal structure. The quality of the received speech is maintained by classifying speech segments according to their structure and coding them in a way that ensures ease of lost-packet regeneration at the receiver. The results of an experiment are reported that confirmed the general validity of this model from the standpoint of transmitter and receiver processing and subjective quality View full abstract»

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  • A compatible fixed-frame ISDN gateway for broadband metropolitan area networks

    Publication Year: 1989 , Page(s): 680 - 689
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (756 KB)  

    A method (referred to as I-SWIFT) is presented for integrating the flow of synchronous voice streams into existing broadband cable systems. The interface between the integrated voice subnetwork and the cable system is realized by a metropolitan area gateway (MAG) implemented at the broadband cable headend. In I-SWIFT, an intelligent scheduling algorithm in the MAG dynamically allocates channel bandwidth using modified data switch-filtering techniques. The MAG accomplishes this by establishing a fixed voice frame (with realignment) which is implemented in an upwardly compatible fashion with respect to the existing carrier-sense multiaccess communication with collision detection (CSMA/CD) data stations. Thus, previously installed stations require no modification. It is shown that for typical system design parameters, the I-SWIFT gateway approach can achieve much improved performance over previous compatible voice/data integration methods View full abstract»

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  • Implementation mechanisms for packet switched voice conferencing

    Publication Year: 1989 , Page(s): 698 - 706
    Cited by:  Papers (10)  |  Patents (14)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (672 KB)  

    A distributed control mechanism for managing a packet-switched voice conference connection is presented. The principal concept introduced is the idea of viewing a conference connection as a logical ring of participants. Alternative methods for implementing voice conferencing on both broadcast and point-to-point networks are introduced, analyzed, and compared. Tradeoffs between the two methods with respect to station workload and maximum number of conference participants are discussed. Experimental implementations on both a carrier-sense multi-access with collision detection (CSMA/CD) Ethernet and a token-ring ProNet are described. The mechanisms presented can be used as part of a packet-switched voice communications protocol that includes conferencing capabilities View full abstract»

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  • End-to-end performance models for variable bit rate voice over tandem links in packet networks

    Publication Year: 1989 , Page(s): 718 - 728
    Cited by:  Papers (5)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (888 KB)  

    Analytical models are presented for computing the end-to-end voice call performance in a packet network that drops the less significant bits in voice packets during periods of congestion. These models provide information about the end-to-end quality likely to be experienced in future packet-switched integrated services networks. An existing single-node bit-dropping model is modified to include the situation resulting when the overall arrival process at an internal node consists of a mix of packets of different sizes due to bit dropping at previous nodes. A detailed model to capture bit-dropping effects in a tandem connection of nodes is presented. The model includes the effect of load fluctuations at each node, and also takes into account the dependencies in bit dropping experienced by a voice packet at successive nodes in a tandem connection. The model also incorporates the internodal dependence when reductions in packet service times occur at intermediate nodes due to bit dropping at previous nodes. Two approximation procedures are discussed that serve as upper and lower bounds. In particular, the upper bound is shown to be very tight for a practical range of loads, and hence serves as a good approximation with significant computational simplicity View full abstract»

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  • Shared access packet transmission systems for compressed digital video

    Publication Year: 1989 , Page(s): 815 - 825
    Cited by:  Papers (11)  |  Patents (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1012 KB)  

    The problem of designing shared access packet-transport-based transmission systems for compressed video signals is studied. The feasibility of using conventional link-level and transport-level protocol services to transmit compressed video is examined by focusing on two practically important scenarios for compressed video transmission: (1) multipoint-to-multipoint video transmission using a 200 Mb/s implicit token passing (ITP) fiber-optic local area network (LAN); and (2) point-to-multipoint broadcast video distribution using a 90 Mb/s packet time-division multiplexing (packet-TDM) direct-broadcast satellite channel. To evaluate the performance of such shared-access broadband packet video systems accurate simulation models were developed that were driven by realistic `broadcast quality' compressed video sources for the ITP-LAN and packet-TDM systems. The models were used to determine design tradeoffs between channel throughput, video quality (measured by clipping probability), and the transport-level and media-access-level protocol features and parameters implemented in the packet video network interface unit View full abstract»

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  • Queueing analysis of delay constrained voice traffic in a packet switching system

    Publication Year: 1989 , Page(s): 729 - 738
    Cited by:  Papers (22)  |  Patents (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (728 KB)  

    Unlike data traffic, the voice packet stream from a node has very high correlation between consecutive packets. In addition, in order for the speech to be properly reconstructed, a delay constraint must be satisfied. A queueing model that accurately predicts packet loss probabilities for such a system is presented. Analytical results are obtained from an embedded bivariate Markov chain and are validated by a simulation program. Based on this model, the impact of the delay constraint, talkspurt detection thresholds, and packet size on packet loss are studied. Two schemes, named `instant' and `random', for discarding late packets are considered. Simulation results show that better performance can be obtained by using the latter scheme View full abstract»

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  • Models for packet switching of variable-bit-rate video sources

    Publication Year: 1989 , Page(s): 865 - 869
    Cited by:  Papers (100)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (420 KB)  

    The authors extend earlier work (ibid., vol.36, p.834-44, Jul. 1988) in modeling video sources using interframe coding schemes and in carrying out buffer queueing analysis for the multiplexing of several such sources. The previous models and analysis were suitable for relatively uniform activity scenes. Here, models are considered for scenes with multiple activity levels which lead to sudden changes in the coder output bit rates. Such models apply to talker-listener alternating scenes, as well as to situations where there is a mix of dissimilar services, e.g., television and videotelephony. Correlated Markov models for the corresponding sources are given. A flow-equivalent queueing analysis is used to obtain common buffer queue distributions and probabilities of packet loss. The results demonstrate the efficiency of packet video on a single link, due to the smoothing effect of multiplexing several variable-bit-rate video sources View full abstract»

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  • Two-layer coding of video signals for VBR networks

    Publication Year: 1989 , Page(s): 771 - 781
    Cited by:  Papers (142)  |  Patents (33)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (2364 KB)  

    Two-layer conditional-replenishment coding of video signals over a variable-bit-rate (VBR) network is described. A slotted-ring network based on an Orwell protocol is assumed, where transmission of certain packets is guaranteed. The two-layer coder produces two output bit streams: the first bit stream contains all the important structural information in the image and is accommodated in the guaranteed capacity of the network, while the second adds the necessary quality finish. The performance of the coder is tested with CIF standard sequences and broadcast-quality pictures. The portion of the VBR channel allocated to the lower layer as guaranteed bandwidth is examined. Using broadcast-quality pictures, statistics were obtained on the performance of this system for different choices of bit rate in the lower layer. The effect of lost packets is shown on CIF standard picture sequences. It is shown that the coder performs well for a guaranteed channel rate as low as 10-20% of the total bit rate View full abstract»

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  • Selective recovery of video packet loss using error concealment

    Publication Year: 1989 , Page(s): 807 - 814
    Cited by:  Papers (55)  |  Patents (50)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1076 KB)  

    An efficient recovery method using error concealment is proposed for video packet loss in fast packet switching networks. In this method, the receiver detects the damaged picture area caused by packet loss from the structured picture data received, makes error concealments, notifies the transmitter, and continues decoding. The transmitter, having received the notice, calculates the affected picture area in the local decoded picture and continues encoding without using this affected area. In this selective recovery method, video signals are not stopped even if a long propagation delay exists, no additional information is transmitted to error recovery and conventional coding algorithms can be used. The proposed method is suitable for multipoint communication. Simulation results show the affected picture area is localized for a considerable time attesting to the method's effectiveness View full abstract»

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  • Error control techniques for integrated services packet networks

    Publication Year: 1989 , Page(s): 690 - 697
    Cited by:  Papers (6)  |  Patents (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (560 KB)  

    The design of a multiservice packet network must ensure that delays to speech packets are minimized while data and other types of packets are delivered without error. The author suggests some ways in which error detection, forward error correction (FEC), and automatic repeat request (ARQ) schemes may be utilized in integrated services packet networks (ISPNs) to ensure that satisfactory error performance and reliability standards are achieved. The results show that ARQ schemes combined with reliable error detection are the most practical way of achieving reliable error control in integrated services networks. Also, such error control schemes can have performance advantages if applied on a region-by-region basis rather than simple end-to-end View full abstract»

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  • A variable bit rate video codec for asynchronous transfer mode networks

    Publication Year: 1989 , Page(s): 761 - 770
    Cited by:  Papers (78)  |  Patents (20)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (600 KB)  

    The bandwidth flexibility offered by the asynchronous transfer mode (ATM) technique makes it possible to select picture quality and bandwidth over a wide range in a simple and straightforward manner. A prototype model of a video codec was developed that demonstrates the feasibility of both variable bit rate (VBR) coding and user-selectable picture quality. The VBR coding algorithm is discussed and it is shown how a stabilized quality is achieved and how this quality and associated bandwidth can be selected by the user. How error propagation is limited to reduce the visibility of cell losses is also discussed. Interfaces with the ATM network are analyzed, with emphasis on decoder synchronization and absorption of cell delay jitter. The VBR codec offers very good picture quality for videophony applications at an equivalent load of 5.9 Mb/s. Picture quality remains relatively constant, even for heavy motion View full abstract»

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  • Dynamic priority protocols for packet voice

    Publication Year: 1989 , Page(s): 632 - 643
    Cited by:  Papers (35)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (916 KB)  

    Since the reconstruction of continuous speech from voice packets is complicated by the variable delays of the packets through the network, a dynamic priority protocol is proposed to minimize the variability of packet delays. The protocol allows the priority of a packet to vary with time. After a discussion of the concept of dynamic priorities, two examples of dynamic priorities are studied through queueing analysis and simulations. Optimal properties of the oldest customer first (OCF) and earliest deadline first (EDF) disciplines are proven, suggesting that they may be theoretically effective in reducing the variability of packet delays. Simulation results of the OCF discipline indicate that the OCF discipline is most effective under conditions of long routes and heavy traffic, i.e., the conditions when delay variability is most likely to be significant. Under OCF, the delays of packets along long routes are improved at the expense of packets along short routes. It is noted that more complex and realistic simulations, including simulations of the EDF discipline, are needed View full abstract»

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  • Generation interval distribution characteristics of packetized variable rate video coding data streams in an ATM network

    Publication Year: 1989 , Page(s): 833 - 841
    Cited by:  Papers (15)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (816 KB)  

    A description is given of the packetized data stream characteristics of variable-rate video coding in an asynchronous transfer mode (ATM) network. Variable-rate video coding characteristics are presented using motion-compensated adaptive intra-interframe prediction and entropy coding. The peak/mean entropy ratio, i.e., the burstiness of the TV conference video signals, ranges from 3 to 4. A method is proposed to transmit the packets of the three outputs of the variable-rate video coder: prediction errors, motion vector information, and quantizer selection information. The packet generation interval distribution characteristics of the prediction errors and motion vector information are clarified. It is shown that the generation interval distribution of prediction errors and motion vector information are not characterized by a simple Erland distribution, but by a combination of different distributions View full abstract»

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  • Variable bit-rate coding of video signals for ATM networks

    Publication Year: 1989 , Page(s): 801 - 806
    Cited by:  Papers (73)  |  Patents (14)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (528 KB)  

    Statistical characteristics of video signals for video packet coding, are clarified and a variable-bit-rate coding method for asynchronous transfer mode (ATM) networks is described that is capable of compensating for packet loss. ATM capabilities are shown to be greatly affected by delay, delay jitter, and packet loss probability. Packet loss has the greatest influence on picture quality. Packets may be lost either due to random bit error in a cell header or to network control when traffic is congested. A layered coding technique using discrete-cosine transform (DCT) coding is presented which is suitable for packet loss compensation. The influence of packet loss on picture quality is discussed, and decoded pictures with packet loss are shown. The proposed algorithm was verified by computer simulations View full abstract»

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  • Basic characteristics of variable rate video coding in ATM environment

    Publication Year: 1989 , Page(s): 752 - 760
    Cited by:  Papers (102)  |  Patents (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (648 KB)  

    Basic characteristics of variable-rate video coders applied to asynchronous transfer mode (ATM) transmission are described. Burstiness of video information is evaluated for conference-type scenes using various coding algorithms. Three measures (distribution, autocorrelation, and coefficient of variation) are introduced to evaluate burstiness. Video sources are modeled and characterized by the autoregressive process and coefficient of variation. Video quality improvement achieved with variable rate transmission is evaluated using signal-to-noise ratio (SNR) and subjective ratings. An improvement of 5-10 dB in temporal SNR and 1 rank in mean opinion score are reported View full abstract»

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  • Packet video and its integration into the network architecture

    Publication Year: 1989 , Page(s): 739 - 751
    Cited by:  Papers (80)  |  Patents (14)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1568 KB)  

    Packet video is investigated from a systems point of view. The most important issues relating to its transmission are identified and studied in the context of a layered network architecture model, leading to a better understanding of the interactions between network and signal handling. The functions at a particular layer can thereby be made less dependent on network implementation and signal format. In the layered network model, the higher layers provide format conversion, hierarchical source coding, error recovery, resynchronization, cost/quality arbitration, session setup and tear-down, packetization, and multiplexing. Provisions from the network layers pertain mainly to real-time transmission. Special consideration is given to hierarchical source coding, error recovery, statistical behavior, and timing aspects. Simulation results indicating practical solutions to some of the issues raised are presented for a hierarchical packet-video subband coding system View full abstract»

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  • Statistics of video signals for viewphone-type pictures

    Publication Year: 1989 , Page(s): 826 - 832
    Cited by:  Papers (20)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (596 KB)  

    The authors describe a series of measurements of the statistics of viewphone-type video signals, with particular regard to the possible transmission of such signals over an asynchronous transfer mode (ATM) network. Measurements include the frame-to-frame differences and the cluster length distribution. It is known that with efficient picture coding, the information rate required for a constant picture quality varies significantly and creates problems in a constant-bit-rate system. The multiplexing of a number of sources in a variable-bit-rate (VBR) system is considered. It is shown that considerable reduction in the variability of the data rate is obtained. While the results are derived from one particular type of picture coder, it is expected that the conclusion will apply to other coding schemes View full abstract»

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  • Packet communication protocol for image services on a high-speed multimedia LAN

    Publication Year: 1989 , Page(s): 782 - 788
    Cited by:  Papers (6)  |  Patents (6)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (680 KB)  

    The authors describe the requirements for high-speed multimedia communication and propose a high-speed communication protocol to provide congestion-free access and efficient retransmission/flow control. Its usefulness is proved in a 400 Mb/s multiaccess loop local area network (LAN) with a 100 Mb/s user interface. The main characteristics of the protocol are separation of image communication handling, guarantee of no buffer overflow in a network, and end-to-end block-based transfer. Buffer reservation control in the user-network interface and a retransmission scheme based on a long-size block are used to realize high-speed congestion control and error recovery. A multimedia terminal architecture suitable for real-time image communication is also discussed. In the prototype system, a few frames of high-resolution image information can be transferred in a second. Around 26 Mb/s effective throughput between application entities has been obtained View full abstract»

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Aims & Scope

IEEE Journal on Selected Areas in Communications focuses on all telecommunications, including telephone, telegraphy, facsimile, and point-to-point television, by electromagnetic propagation.

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Meet Our Editors

Editor-in-Chief
Muriel Médard
MIT