Telecommunications – Radiotelephone system – Zoned or cellular telephone system
Reexamination Certificate
2000-12-22
2004-10-26
Trost, William (Department: 2683)
Telecommunications
Radiotelephone system
Zoned or cellular telephone system
C455S436000, C455S438000, C379S220010
Reexamination Certificate
active
06810256
ABSTRACT:
BACKGROUND
The present invention generally relates to coding in the field of communication systems and, more particularly, to handling the signaling for controlling transcoding when a connection is handed off between mobile switching centers (MSCs) in radiocommunication systems.
The growth of commercial communication systems and, in particular, the explosive growth of cellular radiotelephone systems, have compelled system designers to search for ways to increase system capacity without reducing communication quality beyond consumer tolerance thresholds. One technique to achieve these objectives involved changing from systems wherein analog modulation was used to impress data onto a carrier wave, to systems wherein digital modulation was used to impress the data on carrier waves.
In wireless digital communication systems, standardized air interfaces specify most of the system parameters, including speech coding type(s), burst format, communication protocol, etc. For example, the European Telecommunication Standard Institute (ETSI) has specified a Global System for Mobile Communications (GSM) standard that uses time division multiple access (TDMA) to communicate control, voice and data information over radio frequency (RF) physical channels or links using a Gaussian Minimum Shift Keying (GMSK) modulation scheme at a symbol rate of 271 ksps. In the U.S., the Telecommunication Industry Association (TIA) has published a number of Interim Standards, such as IS-54 and IS-136, that define various versions of digital advanced mobile phone service (D-AMPS), a TDMA system that uses a differential quadrature phase shift keying (DQPSK) modulation scheme for communicating data over RF links.
TDMA systems subdivide the available frequency into one or more RF channels. The RF channels are further divided into a number of physical channels corresponding to time slots in TDMA frames. Logical channels are formed of one or several physical channels where modulation and coding is specified. In these systems, the mobile stations communicate with a plurality of scattered base stations by transmitting and receiving bursts of digital information over uplink and downlink RF channels.
The growing number of mobile stations in use today has generated the need for more voice and data channels within cellular telecommunication systems. As a result, base stations have become more closely spaced, with an increase in interference between mobile stations operating on the same frequency in neighboring or closely spaced cells. In fact, some systems now employ code division multiple access (CDMA), using a form of spread spectrum modulation wherein signals intentionally share the same time and frequency. Although digital techniques provide a greater number of useful channels from a given frequency spectrum, there still remains a need to maintain interference at acceptable levels, or more specifically to monitor and control the ratio of the carrier signal strength to interference, (i.e., carrier-to-interference (C/I) ratio).
Another factor which is increasingly important in providing various communication services is the desired/required user bit rate for data to be transmitted over a particular connection. For example, for voice and/or data services, user bit rate corresponds to voice quality and/or data throughput, with a higher user bit rate producing better voice quality and/or higher data throughput. The total user bit rate is determined by a selected combination of techniques for speech coding, channel coding, modulation, and resource allocation, e.g., for a TDMA system, this latter technique may refer to the number of assignable time slots per connection, for a CDMA system, this latter parameter may refer to the number of assignable codes per connection.
Speech coding (or more generally “source coding”) techniques are used to compress the input information into a format which uses an acceptable amount of bandwidth but from which an intelligible output signal can be reproduced. Many different types of speech coding algorithms exist, e.g., residual excited linear predictive (RELP), regular-pulse excitation (RPE), etc., the details of which are not particularly relevant to this invention. More significant in this context is the fact that various speech coders have various output bit rates and that, as one would expect, speech coders having a higher output bit rate tend to provide greater consumer acceptance of their reproduced voice quality than those having a lower output bit rate. As an example, consider that more traditional, wire-based telephone systems use PCM speech coding at 64 kbps, while GSM systems employ an RPE speech coding scheme operating at 13 kbps.
In addition to speech coding, digital communication systems also employ various techniques to handle erroneously received information. Generally speaking, these techniques include those which aid a receiver to correct the erroneously received information, e.g., forward error correction (FEC) techniques, and those which enable the erroneously received information to be retransmitted to the receiver, e.g., automatic retransmission request (ARQ) techniques. FEC techniques include, for example, convolutional or block coding (collectively referred to herein as “channel coding”) of the data prior to modulation. Channel coding involves representing a certain number of data bits using a certain number of code bits. Thus, for example, it is common to refer to convolutional codes by their code rates, e.g., ½ and ⅓, wherein the lower code rates provide greater error protection but lower user bit rates for a given channel bit rate.
Conventionally, each of the techniques which impacted the user bit rate were fixed for any given radiocommunication system, or at least for the duration of a connection established by a radiocommunication system. That is, each system established connections that operated with one type of speech coding, one type of channel coding, one type of modulation and one resource allocation. More recently, however, dynamic adaptation of these techniques has become a popular method for optimizing system performance in the face of the numerous parameters which may vary rapidly over time, e.g., the radio propagation characteristics of radiocommunication channels, the loading of the system, the user's bit rate requirements, etc. This type of dynamic adaptation of coding techniques has been referred to in the GSM standard as adaptive multirate (AMR) communications. AMR techniques and the like are likely to be used in next generation radiocommunication systems, e.g., Universal Mobile Telecommunication Systems (UMTS).
In GSM systems, AMR techniques have been traditionally been coordinated by the base station controller (BSC). For context, consider
FIG. 1
, which depicts various nodes in a hybrid UMTS/GSM communication system
10
. This Figure will also be referred to below in describing aspects of the present invention. The system
10
is designed as a hierarchical network with multiple levels for managing calls. Using a set of uplink and downlink frequencies, mobile stations
12
operating within the system
10
participate in calls using time slots allocated to them on these frequencies. At an upper hierarchical level, a group of Mobile Switching Centers (MSCs)
14
a
-
14
c
are responsible for the routing of calls from an originator to a destination. In particular, these entities are responsible for setup, control and termination of calls. One of the MSCs
14
a
, known as the gateway MSC, handles communication with a Public Switched Telephone Network (PSTN)
18
, or other public and private networks.
At a lower hierarchical level, each of the MSCs
14
a
-
14
c
are connected to a group of BSCs
16
a-b
(although the BSCs are only depicted for MSC
14
b
to simplify the figure) using, for example, PCM (pulse code modulated) links. Under the GSM standard, the BSCs
16
a
-
16
b
communicate with MSCs
14
a
-
14
c
under a standard interface known as the A-interface, which is based on the Mobile Application Part (MAP) of CCITT Signali
Hodges Phil
Stüempert Martin
D'Agosta Stephen
Telefonaktiebolaget LM Ericsson
Trost William
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