Broadband communication systems

This thesis addresses a method to deduce the statistical bounds associated with the cell-transfer delay variations (CDVs) encountered by the cells of MPEG traffic, transmitted in the Asynchronous Transfer Mode (ATM) networks. This study focuses on: (1) Estimating CDV arising from multiplexing/switching for both constant bit rate (CBR) and variable bit rate (VBR) traffics via priority allocation based simulations. (2) Developing an information-theoretics based technique to get an insight of the combined BER-induced and multiplexing/switching-induced CDVs in ATM networks. Algorithms pertinent to CDV statistics are derived and the lower and upper bounds of the statistics are obtained via simulations in respect of CBR and VBR traffics. Ascertaining these bounds is useful in the cell admission control (CAC) strategies adopted in ATM transmissions. Inferential remarks indicating the effects of traffic parameters (such as bandwidth, burstiness etc.) on the values of the statistical bounds are presented, and scope for further work are presented.

This thesis addresses a method of deducing the statistical upper and lower bounds associated with the cell-transfer delay variations (CDVs) encountered by the cells transmitted in the asynchronous transfer mode (ATM) networks due to cell losses. This study focuses on: (1) Estimating CDV arising from multiplexing/switching for both constant bit rate and variable bit rate services via simulations. (2) Deducing an information-theoretics based new technique to get an insight of the combined BER-induced and multiplexing/switching-induced CDVs in ATM networks. Algorithms on the CDV statistics are derived and the lower and upper bounds of the statistics are obtained via simulations in respect of CBR and VBR traffics. These bounds bounds are useful in the cell-admission control (CAC) strategies adapted in ATM transmissions. Inferential remarks indicating the effects of traffic parameters (such as bandwidth, burstiness etc.) on the values of the statistical bounds are presented, and scope for further work is indicated.

Broadband-ISDN Network Architecture and Signaling concepts are described with particular emphasis on Asynchronous Transfer Mode (ATM) technology. A consolidated view of various aspects of B-ISDN/ATM Access and Network Signaling functions, architecture and protocols is presented. Additionally, a suggested evolutionary growth path for the B-ISDN signaling protocols is summarized. This is followed by a high-level comparison of two protocols under consideration for network signaling. The result of this evaluation indicates that the requirements of network signaling are best met by a protocol stack based on SS7 concepts. Finally, a set of future B-ISDN and Multimedia services is presented in context of the requirements they would impose on the signaling protocols. Enhancements to the access signaling protocol are proposed for the support of multiconnection and/or multiparty calls. These extensions/enhancements consist of a set of simplified messages, information elements and procedures based on message flows. Evolution and backward compatibility to existing protocols are taken into account while developing the extensions.

Rapid advances in technology have resulted in the evolution of Integrated Services Digital
Networks (ISDNs) to Broadband ISDNs (BISDNs). This thesis discusses and evaluates the
performance of a high speed and high capacity packet switching system architecture for
BISDNs. This system supports various high speed communication services like data, voice,
and video services. The main performance criterion is the mean switching delay, which
is defined as the packet transfer delay through the bus and through the loop connecting
various switch modules in the system. Other performance criteria are power and system
throughput. The performance parameters are evaluated separately for both data and control
(signaling) packets. The effects of the number of switch modules in the system, the data
packet length, and the ratio of arrival rate of data to control packets are examined.
It is observed that the switching delay and throughput increase with an increase in the
traffic intensity for any number of switching modules and the ratio of arrival rate of data
to control packets. The delay for data packets is found to be higher for lower values of
this ratio and vice versa. The power is found to be maximum for about 70% load. The
switching delay in this simulation model is found to be less than one millisecond for data
packets.

A new topology is proposed that is regular in structure and allows broadband Local Area Networks to be connected together to support a large number of users. The topology is based on connecting local area networks in cube fashion and allows for systematic growth as the need arises. These cubes are attached to each other as building blocks by adding cubes evenly in all three dimensions. In an operational network, irregularities occur due to failures at gateways. An addressing scheme and an automatic routing algorithm are developed that makes it possible to add new users without affecting the addresses of existing nodes. The capacity of the proposed networking architecture is determined and compares favorably to that of hypercube interconnected LANs when the number of rings in the network is small.

Multimedia applications incorporate the use of more than one type of media, i.e., voice, video, data, text and image. With the advances in high-speed communication, the ability to transmit multimedia is becoming widely available. One of the means of transport for multimedia in distributed networks is Broadband Integrated Services Digital Network (B-ISDN). B-ISDN supports the transport of large volumes of data with a low error rate. It also handles the burstiness of multimedia traffic by providing dynamic bandwidth allocation. When multimedia is requested for transport in a distributed network, different Quality of Service (QOS) may be required for each type of media. For example, video can withstand more errors than voice. In order to provide, the most efficient form of transfer, different QOS media are sent using different channels. By using different channels for transport, jitter can impose skews on the temporal relations between the media. Jitter is caused by errors and buffering delays. Since B-ISDN uses Asynchronous Transfer Mode (ATM) as its transfer mode, the jitter that is incurred can be assumed to be bounded if traffic management principles such as admission control and resource reservation are employed. Another network that can assume bounded buffering is the 16 Mbps token-ring LAN when the LAN Server (LS) Ultimedia(TM) software is applied over the OS/2 LAN Server(TM) (using OS/2(TM)). LS Ultimedia(TM) reserves critical resources such as disk, server processor, and network resources for multimedia use. In addition, it also enforces admission control(1). Since jitter is bounded on the networks chosen, buffers can be used to realign the temporal relations in the media. This dissertation presents a solution to this problem by proposing a Feedback-based Multimedia Synchronization Technique (FMST) to correct and compensate for the jitter that is incurred when media are received over high speed communication channels and played back in real time. FMST has been implemented at the session layer for the playback of the streams. A personal computer was used to perform their synchronized playback from a 16 Mbps token-ring and from a simulated B-ISDN network.

In order to guarantee a committed Quality of Service (QoS) to the users of a Broadband Integrated Services Digital Network (B-ISDN), preventive congestion control becomes critical, and is implemented through Call Acceptance Control (CAC) and Usage Parameter Control (UPC) functions. Currently, Asynchronous Transfer Mode (ATM) cells are equipped with a 1-bit Cell Loss Priority (CLP) field, which can be used for service-oriented and/or UPC marking. This creates a conflict, since these two marking approaches may have contradicting objectives, and are designed to operate independently. Moreover, by admitting excessive cells as marked traffic, this group is allowed to grow uncontrollably, thereby jeopardizing the QoS committed to other marked cells. This dissertation presents a solution to these problems by proposing a new 4-class priority strategy that unifies the two marking approaches, and is based on a 2-bit CLP field. The impacts of the new priority scheme are triple-fold: (I) For the UPC, a new scheme, the Forgiving Leaky Bucket (FLB), not only carries priorities through subnetwork boundaries, but also has the power of unmarking, i.e. forgiving, previously marked cells, depending on the bandwidth availability in the entering subnetwork. Forgiving will correct access-point bias, a phenomenon observed in internetworked ATM subnetworks of different congestion conditions. (II) At ATM switching nodes, a new space priority scheme is based on a hybrid of the Nested Threshold, and Push-Out cell discarding methods. This scheme is designed for the 4-class priority strategy, and improves the quality of the low priority traffic. (III) In interfacing High Speed Local Area Networks and Metropolitan Area Networks, idle bandwidth due to STM multiplexing is utilized to carry marked excessive cells of connection-oriented variable bit rate traffic, in addition to the service-oriented marking performed at transmitting stations. The resulting stream is then carried through internetworking points, subject to FLB adjustments. As a result, the STM and ATM subnetworks will support a uniform end-to-end priority strategy, essential for a B-ISDN. The proposed impacts are analyzed and compared with conventional implementations, and future directions are indicated.

In last few years there has been significant growth in the area of wireless communication. Quality of Service (QoS) has become an important consideration for supporting variety of applications that utilize the network resources. These applications include voice over IP, multimedia services, like, video streaming, video conferencing etc. IEEE 802.16/WiMAX is a new network which is designed with quality of service in mind. This thesis focuses on analysis of quality of service as implemented by the WiMAX networks. First, it presents the details of the quality of service architecture in WiMAX network. In the analysis, a WiMAX module developed based on popular network simulator ns-2, is used. Various real life scenarios like voice call, video streaming are setup in the simulation environment. Parameters that indicate quality of service, such as, throughput, packet loss, average jitter and average delay, are analyzed for different types of service flows as defined in WiMAX. Results indicate that better quality of service is achieved by using service flows designed for specific applications.