Asynchronous Scrambled Coded Multiple Access (ASCMA) technology presents several features that make it well suited for IoT and mMTC type applications over state-of-the-art terrestrial wireless networks. These include low-rate error correction coding combined with low complexity receiver processing and an ability to transmit data asynchronously using single as well as multiple-streams from a user terminal. The resulting advantages in terms of spectral efficiency, low-latency, and reduced power consumption have been quantified previously using link-level simulations. In this paper, we evaluate the system level performance of a 5G New Radio (NR) mobile network employing ASCMA for uplink mMTC type applications. Using an accurate system level simulator, we evaluate two different scheduling schemes for ASCMA, the first assumes a grant-based scheduling of user terminals over available network resources and the second assumes random access or grant-free scheduling of the same. A novel abstraction method is developed to aid the system level evaluations. Using extensive simulations, we demonstrate the ability of ASCMA technology to provide significant improvements to key network performance indicators such as packet loss rate, average retransmission rate, average cell throughput and cell-edge throughput.

Asynchronous Scrambled Coded Multiple Access (A- SCMA) has been shown to offer high spectral efficiencies on Random Access (RA) channels. By allowing unscheduled data transmission, latency is reduced and power can be conserved, making such techniques well suited for IoT type applications. In this paper multi-stream (multi- layer) transmission from a single terminal is demonstrated which enables higher per-terminal spectral efficiency. The application of ASCMA with single and multi-stream on both Additive White Gaussian Noise (AWGN) as well as terrestrial fading channels is presented.

A family of new non-orthogonal multiple access techniques named Scrambled Coded Multiple Access (SCMA) is described. It allows different users of a communications system to efficiently share the same bandwidth by using specific scramblers to separate user bursts. SCMA technology is designed for satellite communications with extremely small antenna terminals for potential applications such as asset tracking, sensor networking, and smart grid; and also for latency reduction. The latest variation, Asynchronous Scrambled Coded Multiple Access (A-SCMA) on the other hand, not only removes the need for timeslot synchronization and precise power control, but also offers an unprecedented system throughput of more than 4 bits/symbol at a modest per user signal to noise ratio (SNR).

Asynchronous Scrambled Coded Multiple Access (A- SCMA) has been shown to offer high spectral efficiencies on Random Access (RA) channels. This paper describes a new multiple access scheme, SA- SCMA, that adds spreading. Even modest spreading provides significant gains in spectral efficiency, while the use of large RRC rolloff combined with adjacent channel interference cancellation (ACIC) further enhances performance. Spectral efficiencies achieved are high enough to offer data transfer over RA, eliminating the usual 2-way propagation delay needed to assign resources, lowering latency, an important consideration for geostationary satellite inroutes. Additionally, spreading reduces flux density, enabling smaller antenna apertures needed for lower cost applications. The use of SA- SCMA on a saturated radio to further reduce user terminal cost is also discussed.

Fixed, portable and mobile terminals use Random Access (RA) to transfer data or request more spectrally efficient dedicated resources. This paper describes a new multiple access scheme, A- SCMA, that operates with unsynchronized transmission and low power, making it particularly suitable to small machine-to-machine communications in satellite or terrestrial applications. A multicarrier version operates with large and variable packet sizes without reduced throughput or large delays.

After ten years of successful deployment, the second generation satellite standard for digital video broadcasting, DVB-S2, has been updated to include very low signal-to-noise ratio modes to better support mobile and very small terminal applications without changing its fundamental structure. This paper provides a high level discussion on several of the most important aspects of the updated standard.

The advantages of Interleave Division Multiple Access (IDMA) are taken one step further by Scrambled Coded Multiple Access (SCMA) where user specific, long interleave tables are replaced by a common scrambling sequence that can be used for all users with a simple shift.

Unlike other conventional multiple access techniques such as TDMA or FDMA, this paper demonstrates that the proper choice of forward error control (FEC) codes for Interleave Division Multiple Access (IDMA) is a function of the number of users. By means of Extrinsic Information Transfer (EXIT) charts and Monte-Carlo simulations it is shown that extremely weak single user codes can surprisingly be the best choice when the number of users is high.

The performance of Interleave Division Multiple Access (IDMA) with a new set of low rate turbo codes is evaluated on Additive White Gaussian Noise (AWGN) and Rician fading channels. Despite the simplicity of the scheme and excessive multiuser interference, very good performance close to theoretical limit is demonstrated. The effect of varying number of users, code rate and channel characteristics on the performance is investigated.

A novel parity-check matrix design for low-density parity-check (LDPC) codes is described. By eliminating the routing problem associated with LDPC codes, the design results in a small implementation area, and the codes have outstanding error-rate performance close to the Shannon limit for a wide range of code rates, from 1/4 to 9/10, and for various modulation schemes such as binary phase-shift keying (PSK), quaternary PSK, 8-PSK, 16-amplitude PSK (APSK), and 32-APSK. As a result, LDPC codes designed with this method have been standardized for next-generation digital video broadcasting.

Tremendous progress has been made on channel coding techniques during the last decade, starting from the invention of turbo codes and the rediscovery of the low-density parity check (LDPC) codes. These codes essentially closed the gap between the Shannon capacity limit and practical implementation. Commercial adoption of these new coding techniques, however, depends heavily on the introduction of the technology into communication standards. In this paper, we briefly review the considerations when turbo codes and LDPC codes were introduced to the 3/sup rd/ generation wireless standards (3GPP and 3GPP2), and the digital video broadcast for satellite standard (DVB-S2), respectively. We then examine a few possible approaches to take advantage of these powerful codes for future wireless channels.

Wireless data applications have just begun to gain popularity in the marketplace. The rollout of 3rd generation (3G) wireless networks is anticipated in the next few years. We examine wireless data networks that are complementary to the 3G wireless networks, their technologies and the challenges. In particular, we present some of the industry initiatives in the millimeter wave frequency bands for broadband and satellite delivered fixed wireless data network. Furthermore, we offer our perspectives on how these different networks will evolve and inter-work together to provide a seamless experience for the end users. Fixed data networks need to be competitive with wire-line services, this requires us to push advanced signal processing technologies further. We present the latest advances that we have made in channel coding and signal processing technologies to approach that goal.

Enhanced Data for GSM Evolution (EDGE) and IS-136HS have been considered as the high-speed data air-interface for GSM and North America TDMA evolution. Deploying such a high-speed data air-interface in a very limited bandwidth requires a very small reuse factor. This paper describes the challenges brought on by the small reuse factor and several approaches to enhance the throughput and performance of such systems. Simulation and analysis results are used to demonstrate the tradeoffs.

The telecommunication industry is evolving at an unprecedented pace. As the PCS operators are just rolling out their network, the ITU (International Telecommunication Union) has already initiated the development of the third generation mobile communication standards. This paper presents an overview of the design issues and technical innovations as being developed during the process of the standard developments.

A bandwidth-efficient television transmission technique called time-multiplexed video transmission (TMVT) is described. The TMVT system uses a two-dimensional filtering and interpolation technique to reduce the transmission bandwidth of TV signals, thereby increasing the utilization efficiency of a satellite transponder by approximately 50%. The system adopts the time-multiplexed component transmission scheme with a single carrier to eliminate interference between different TV signals transmitted through the same satellite transponder, and to avoid the intermodulation between the multiple FM carriers, as in the conventional frequency division multiplexing (FDM) approach. A prototype hardware system has been built and tested, with results closely matching the theoretical analysis.>

This paper describes a TV signal compression scheme which may become a common practice for analog video transmission and broadcast via satellite by the year 2000. The technique uses twodimensional interfield processing to reduce the video bandwidth with minimum degradation in subjective quality, and time multiplexed with analog multiplexing technique for transmission in component format. Computer simulations of the processing technique on the test slides, and analysis of subjective quality andS/Nexpected over a typical satellite link, are reported. A proof-of-concept prototype being developed at COMSAT Laboratories is also outlined.

A new contention-based demand assignment protocol (CPODA) is described. The protocol is designed to handle packetized data and voice traffic in a multiple access satellite broadcast channel, where the channel is typically shared by hundreds of earth stations. A sophisticated traffic environment is assumed, including multiple priority and delay class distinctions, variable message lengths, and an arbitrary load distribution among the stations. In addition, stations may have different receiving rate capabilities. The CPODA protocol is part of a broader research project concerned with developing protocols for a one-hop satellite network (SATNET). The CPODA and other protocols are currently undergoing experimental evaluation using an INTELSAT IV-A satellite and four earth stations in the Atlantic region.

Concatenated coding systems utilizing a convolutional code as the inner code and a Reed-Solomon code as the outer code are considered. In order to obtain very reliable communications over a very noisy channel with relatively modest coding complexity, it is proposed to concatenate a byte-oriented unit-memory convolutional code with an RS outer code whose symbol size is one byte. It is further proposed to utilize a real-time minimal-byte-error probability decoding algorithm, together with feedback from the outer decoder, in the decoder for the inner convolutional code. The performance of the proposed concatenated coding system is studied, and the improvement over conventional concatenated systems due to each additional feature is isolated.

A recursive procedure is derived for decoding of rateR = 1/nbinary convolutional codes which minimizes the probability of the individual decoding decisions for each information bit, subject to the constraint that the decoding delay be limited to Δ branches. This new decoding algorithm is similar to, but somewhat more complex than, the Viterbi decoding algorithm. A "real-time," i.e., fixed decoding delay, version of the Viterbi algorithm is also developed and used for comparison to the new algorithm on simulated channels. It is shown that the new algorithm offers advantages over Viterbi decoding in soft-decision applications, such as in the inner coding system for concatenated coding.