Telecommunications – Radiotelephone system – Including private cordless extension system
Reexamination Certificate
1999-06-14
2002-02-19
Vo, Nguyen T. (Department: 2682)
Telecommunications
Radiotelephone system
Including private cordless extension system
C455S462000, C704S270000
Reexamination Certificate
active
06349213
ABSTRACT:
BACKGROUND OF THE INVENTION
1. Field of the Invention
This invention relates to cordless telephones and, more particularly, to a cordless telephone having a speech encoding/decoding switching system for use within a multiple handset environment.
2. Description of the Related Art
Cordless telephones have proven to be popular in domestic, business and industrial environments due to the unrestricted freedom of movement they offer users. In fact, in 1997, for the first time ever, sales of cordless telephones exceeded sales of corded telephones with total cordless units sold being in excess of 28 million. Furthermore, total sales for 1998 are expected to have increased over 1997 sales by at least 25 percent.
Standard cordless telephones route incoming and outgoing telephone signals through a base station which is hard wire connected to a telephone line. The base station communicates with the battery-operated handset using a wireless signal transmitted over a distance. That is, the physical hard wire connection between a conventional handset and telephone base is replaced by a radio frequency (RF) link, which can range from the 46 and 49 MHz bands to the more recent 900 MHz bands. The spoken voice is usually communicated between the base and handset by first converting the user's voice into an analog electrical signal and modulating the signal using an RF carrier for radio transmission to the receiver, typically through the use of a Narrow-Band Frequency Modulation (NBFM) technique. At the receiver, the modulated analog voice signal is demodulated and directed to a speaker through which the voice is heard. Outgoing telephone signals follow a reverse direction through generation at the handset, transmission to the base, and then routing to the outgoing phone line. The wireless transmission of the telephone signals between the handset and the base can occur over a range of wavelengths and to varying distances.
The quality of the wireless signal is of paramount importance to the user of a cordless telephone. The quality of a transmission between handset and base is limited by the size of the components and frequency of the signal. A more powerful transmitter results in a more powerful signal which can travel longer distances. However, such power comes with the attendant negative factors of bulky handsets and bases and shortened battery life. In addition, the bandwidth within which cordless telephones are limited is subject to interference from numerous electromagnetic sources.
Cordless telephone systems must, therefore, meet a basic standard of speech quality, often called “toll quality” speech. Toll quality speech transmission standards comprise the minimum bandwidth needed to assure recognition of the speaker by the receiver at the other end of the link in combination with at least 98% understandability of the speech in context. Originally, in telephone signals, the minimum bandwidth was 300 Hz to 3400 Hz, which resulted in 4 kHz frequency spacing for single sideband (SSB) cable and radio transmission. These standards have been preserved in digital transmission, using pulse code modulation (PCM), and are perpetuated in the increasingly common ISDN standard.
To provide toll quality speech transmission between the handset and the base, some cordless telephone systems rely on digital transmission of the analog telephone signal. This requires that a digital to analog coder/decoder chip (“codec”) be placed in both the handset and the base. According to information theory, when PCM was discovered the sampling rate of an analog signal was set at 2 W for perfect recovery of signals having a bandwidth of less than W. In order to prevent foldover intermodulation distortion, the speech spectrum had to be strictly limited to less than 4 kHz. Thus, the sampling rate for voice telecommunications was set at 8 K samples/sec.
CCITT Recommendation G.726-1990 specifies how a digital telephone signal is to be compressed before transmission and how a received digital signal is to be expanded after reception using ADPCM. ADPCM is a technique for converting sound or analog information to binary information by taking frequent samples of the sound and expressing the value of the sampled sound modulation in binary terms. The G.726 standard specifics the functionality that is required for the receive (ADPCM decoder) and transmit (ADPCM encoder) signal processing functions. G.726 allows for the conversion of a 64 kilobit-per-second (kbps) pulse code modulation (PCM) channel to and from a 40, 32, 24, or 16 kbps ADPCM channel. G.726 incorporates the previously-existing G.721 (32 kbps) and G.723 (24 kbps) standards. In addition to cordless handsets, ADPCM is used to encode data on CD-ROMS and data transmitted over fiber-optic transmission lines.
While 8 Kb/s transmission rates meet a minimum level for speech comprehension, it is by no means an ideal transmission rate. An 8 Kb/s transmission rate transmits all vowels very well. However, transmission of consonants, which have main speech energies concentrated between 7 kHz to 8 kHz, is rudimentary at best. Generally, speech taken in context provides sufficient clues for good understandability, although unexpected words and names typically must be spelled in order to circumvent the lack of bandwidth in toll quality telephone connections. Thus, in general, telephone systems having a higher-fidelity transmission became desirable.
Current cordless telephone systems use an improved ADPCM which is capable of much higher quality transmissions. The ADPCM signal conversion device is conventionally known which compresses data and converts that data into PCM signals and further converts from the PCM signals into ADPCM signals. Transmitter side voice signals are compressed and coded in the form of ADPCM signals and then transmitted, and in which on the receiver side the ADPCM signals are expanded and demodulated into voice signals. ADPCM allows analog voice conversation to be carried within a 32 Kb/s digital channel. The sampling rate is 8,000 times per second and three or four bits are used to described each sample. At current transmission values, ADPCM provides a high quality transmission signal between a cordless telephone's handset and base.
Traditionally, to provide multiple handset use in telephone systems utilizing a single subscriber line, multiple sets of bases and handsets must be plugged into that line. Other solutions have involved using multiple hardwired telephones as described in U.S. Pat. No. 5,367,570 (Hector D. Figueroa). To reduce the amount of equipment necessary in such a case, multiple handset cordless telephones are currently available. Such systems allow multiple handsets to transmit and receive signals from a single base station. Such systems are highly desirable in cases in which there is a limited number of incoming lines or there is limited space for the base stations. Multiple handset cordless telephones operate in one of two ways when in use. Either a single handset is used and the other handsets are rendered inactive or any handset may be used to allow multiple participants to use the system. The latter case is preferable in that it mimics a standard telephone system in which multiple persons can speak on the same subscriber line using different telephones. For the concept of a multiple handset telephone to be commercially and practically useful, the system must support simultaneous transmission of signals from the multiple handsets.
Current multiple handset cordless systems which support the use of simultaneous off-hook handsets transmit speech between the handsets and base by using speech coding instead of ADPCM. The use of speech coding at a lower bit rate than ADPCM allows multiple communications signals to share the same frequency bandwidth. Speech coding is normally accomplished by a speech codec chip. A speech codec allows analog voice conversation to be carried within a 4, 8 or 16 Kb/s digital channel. This reduced transmission rate allows for multiple transmissions within a limited bandwidth, for example allowing 8, 4, or 2 han
Iyengar Vasu
Ubowski Richard M.
Agere Systems Guardian Corp.
Dickstein , Shapiro, Morin & Oshinsky, LLP
Nguyen Duc
Vo Nguyen T.
LandOfFree
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