Richard Paul Ejzak

A retransmission scheme for circuit-mode data on wireless links

The cellular radio link is characterized by deep fades leading to long error bursts (lasting hundreds of milliseconds). Data transmission over such links results in large packet error rates (in the range 10/sup -3/ to 10/sup -1/). We have designed a radio link protocol (RLP) to achieve high throughput on such links. The RLP is based on frequent, complete or partial feedback of the receiver state. Performance results for the US digital cellular TDMA standard show that the scheme can provide the equivalent of 9.6 kb/s service per full-rate TDMA channel above a carrier to interference ratio of 18 dB. >

GPRS-136: high-rate packet data service for North American TDMA digital cellular systems

This article provides an overview of the flexible, high-performance packet data channel that has been designed for high-rate packet data services over IS-136 TDMA channels. To achieve the highest data rates in the limited 30 kHz channel bandwidth, the packet data channel is designed for adaptive modulation and, in addition to a fixed coding mode, permits operation using an incremental redundancy mode.

Network overload and congestion: A comparison of ISUP and SIP

For years, telephony service providers have supported voice calls using circuit networks controlled by ISDN User Part (ISUP) signaling. ISUP signaling is based on a mature standard, is widely deployed, supports a variety of services, and has grown to include extensive congestion control mechanisms. With the move to packet networks, Session Initiation Protocol (SIP) signaling is being introduced to support voice over packet networks. As service providers begin this migration, comparisons between SIP and ISUP signaling are inevitable. Specifically, service providers want assurance that SIP provides congestion control mechanisms for packet networks similar to those ISUP provides for circuit networks. This paper is a result of a study to anticipate the congestion control needs of packet networks based on Signaling System 7 (SS7) network experience. It provides a comparison of SIP and ISUP congestion control functions as well as an analysis of mechanisms that could be deployed in SIP networks to enable them to achieve parity with ISUP networks.

Link layer retransmission schemes for circuit-mode data over the CDMA physical channel

In the last few years, wide-area data services over North American digital (TDMA and CDMA) cellular networks have been standardized. The standards were developed under three primary constraints: (i) compatibility with existing land-line standards and systems, (ii) compatibility with existing cellular physical layer standards that are optimized for voice, and (iii) market demands for quick solutions. In particular, the IS-95 CDMA air interface standard permits multiplexing of primary traffic (e.g., voice or circuit data) and secondary traffic (e.g., packet data) or in-band signaling within the same physical layer burst. In this paper, we describe two radio link protocols for circuit-mode data over IS-95. The first protocol, Protocol S, relies on a single level of recovery and uses a flexible segmentation and recovery (FSAR) sublayer to efficiently pack data frames into multiplexed physical layer bursts. We next describe Protocol T, that consists of two levels of recovery. Protocol T has been standardized for CDMA circuit-mode data as IS-99 (Telecommunications Industry Association, 1994). We provide performance comparisons of the two protocols in terms of throughput, delay and recovery from fades. We find that the complexity of the two level recovery mechanism can buy higher throughput through the reduced retransmission data unit size. However, the choice of TCP (and its associated congestion control mechanism) as the upper layer of recovery on the link layer, leads to long fade recovery times for Protocol T. The two approaches also have significant differences with respect to procedures and performance at handoff and connection establishment.

Seamless mobility across IMS and legacy circuit networks

The growing desire of network providers to introduce support for voice over IP (VoIP) has created interesting challenges in the area of interoperability with existing wireless circuit networks. The 3rd Generation Partnership Project (3GPP) and the 3rd Generation Partnership Project 2 (3GPP2) standards have defined the IP Multimedia Subsystem (IMS) as the platform for convergence. By definition, IMS is access agnostic; it provides services and features through a common core network, regardless of the means of transport. However, the IMS standards are just beginning to address the challenges associated with interworking with existing cellular circuit networks. Achieving seamless mobility involves supporting both roaming and handoff between networks. This paper discusses the issues involved in providing seamless mobility for subscribers across the packet and circuit domains and proposes network-based solutions.

MAC layer design for statistical multiplexing of voice and data over EGPRS

The General Packet Radio Service (GPRS) was introduced by ETSI to meet the ever-increasing demand for wireless data over the GSM radio interface. Enhanced GPRS (EGPRS Phase 1) was later introduced in order to increase the spectrum efficiency and data throughput by using 8-PSK modulation and link adaptation techniques. While (E)GPRS uses packet-switched technology to efficiently transmit best-effort data, real-time services (e.g. voice) are currently supported only via circuit-switched systems. This service separation reduces the potential spectrum efficiency gain resulting from multiplexing different services on the same radio channel. This paper describes the key new concepts needed to support statistical multiplexing of different radio access bearer classes over the GSM/EDGE radio access network (GERAN) air interface. The focus is on the definition of new traffic and control channels to support statistical multiplexing of speech, real-time data, and non-real-time data, and the corresponding new MAC procedures that are needed to guarantee quality of service (QoS).

Support for CDMA circuit mobiles in a SIP-based packet core network: A step on the path to IMS

This paper introduces an architecture based on the Internet Protocol Multimedia Subsystem (IMS) standards that provides packet transport for wireless circuit mode voice services. The motivating force behind this architecture is transcoder-free operation (TrFO), which improves voice quality for mobile-to-mobile calls while reducing the cost of the equipment required to support them. Included in this paper are a description of the architecture, examples of call scenarios supported by the architecture, examples of call flows that demonstrate the use of the Session Initiation Protocol (SIP) in the architecture, and a description of the IMS components that are reused in the architecture.

A common SIP profile for next-generation networks

The telephony network is evolving to an all Internet Protocol (IP)-based network. The 3rd Generation Partnership Project (3GPP) and 3rd Generation Partnership Project 2 (3GPP2) standards organizations are currently defining the network architecture for the next generation IP Multimedia Subsystem (IMS) network, which is being applied to multiple access technologies and networks, including the next-generation network (NGN). Concurrently, an effort lead by International Telecommunications Union Telecommunications standardization sector study group 13 (ITU-T SG 13), and presently carried in the focus group on NGN, is adopting IMS for wireline networks. The European Telecommunications Standards Institutes (ETSI) Telecommunication and Internet converged Services and Protocols for Advanced Networks (TISPAN) group is also developing NGN specifications. An essential part of the next-generation network is the seamless interaction with the existing telephony network, enabled by interworking between integrated services digital network user part (ISUP) in the public switched telephone network (PSTN) and Session Initiation Protocol (SIP) in the IMS. Several SIP profiles have been defined in the Internet Engineering Task Force (IETF) and ITU-T for specific interworking applications. SIP is a very flexible protocol with numerous options and extensions. A SIP profile is a specific combination of options and extensions as used by an application. No profile currently supports the interworking of all existing services between SIP networks. This paper analyzes existing SIP profiles against requirements for interworking between different networks and identifies the main SIP profile characteristics necessary to effectively interoperate between multiple networks while maintaining existing services. Two SIP profiles used in ITU-T discussions — IMS SIP and SIP with encapsulated ISUP (SIP-I) — are highlighted to explain ISUP signaling situations. Requirements for a common SIP profile are then described as they relate to these two profiles.

Issues in supporting multimedia services in SIP networks with mixed endpoint types

The evolution from traditional circuit-based networks to packet-based networks is a salient feature of todays telecommunications industry. As new types of Session Initiation Protocol (SIP) networks and endpoints (e.g., 3rd Generation Partnership Project Internet Protocol Multimedia Subsystem [3GPP IMS], cable, public switched telephone network [PSTN] gateways, and 802.11-capable personal digital assistant [PDAs]) are introduced, SIP extensions are being introduced to solve the problems unique to each of them. As SIP networks mature to support mixed endpoint types, interesting issues relating to interoperability among the various endpoints arise. This paper includes a discussion of some of these issues, including out-of-band versus in-band call progress information, handling early media, support of Internet Protocol (IP) and asynchronous transfer mode (ATM) transport networks, interworking with ISDN User Part (ISUP), and support of different SIP extensions. It also explores issues of performance, reliability, and service quality that arise as networks serving mixed endpoints are deployed.