Abstract:
An internet telephone system uses a Public Switched Telephone Network (PSTN). An internet network is coupled to the PSTN. At least two telephone stations are part of the system wherein one of the at least two telephone stations is a DSL telephone station which is coupled to the PSTN. The DSL telephone station has a Plain Old Telephone Set (POTS) splitter coupled to the PSTN for directing low frequency signals to a first line and DSL signals to a second line. A DSL line interface is coupled to the POTS splitter for driving and terminating the second line. A DSL transmitter is coupled to the DSL line interface. A DSL receiver is also coupled to the DSL line interface. A Digital Signal Processor (DSP) is coupled to the DSL transmitter and the DSL receiver. A telephone interface is coupled to the first line. An audio transceiver device is coupled to the telephone interface. A CODEC circuit is also coupled to the telephone interface. A ring and hook detect/control circuitry is coupled to the DSP.

Description:
RELATED APPLICATIONS 
     This patent application is related to pending U.S. patent application entitled “INTERNET TELEPHONE SYSTEM &amp; METHOD THEREFOR,” having a Ser. No. 09/192,761 and a filing data of Nov. 16, 1998, in the name of Yuan-Neng Fan as inventor. The disclosure of the above referenced application is hereby incorporated by reference into this application. 
    
    
     BACKGROUND OF THE INVENTION 
     1. Field of the Invention 
     This invention relates to telephony products and methods therefor and, more specifically, to a quick connect internet telephone and method therefor. The quick connect internet telephone allows voice communication to other internet telephones via a DSL connection to an internet network. 
     2. Description of the Prior Art 
     Various types of internet telephony systems or methods exist today. These systems or methods generally fall into one of the following categories. 
     1. PC to PC Call 
     This system or method uses a personal computer (PC) to establish communications with a second PC. The communication is established using add-on software and hardware to allow the PCs to convey the user&#39;s voices via an internet provider (IP) connection. While this system and method does work, it has numerous drawbacks. First, the two parties wishing to communicate must prearrange the date and time for the internet telephone call to take place. Both users must then establish an IP connection, which further requires the users to have previously exchanged correct internet IP address information. 
     2. PC to Plain Old Telephone Set (POTS) Call 
     A lessor known or used method is a PC to POTS call method. A user places a modem telephone call to their internet service provider (ISP) with an internet telephone software equipped PC. An internet link is then established to a second ISP, or IT (Internet Telephone) gateway, located proximate to the area to which it is desired to place a phone call. The second ISP/IT gateway is then used to place a phone call using the conventional phone systems to the desired local number. While this method does work, it also has several drawbacks. An IT gateway must be available in the same local calling area as the called party. Otherwise, the user has to pay toll charges from the IT gateway to the called party. Furthermore, the IT gateway typically charges a fee for the time connected which further increases the cost involved with this method. A further problem with this method is that the user has to sign up for IT services with an IT gateway for each geographic area the user desires to place a call. This IT gateway sign up typically includes a monthly subscription fee that must be paid regardless of use or non-use. 
     3. IT Phone Call to IT Phone Call 
     Another alternative method is the IT phone call to IT phone call. This method is very similar to the previous method except that regular phones are used at both ends of the call. A user places a regular telephone call to a first local IT gateway with a regular telephone. An internet link is then established from the first IT gateway to a second IT gateway located proximate to the area to which it is desired to place a phone call. The second IT gateway is then used by the user to place a phone call using conventional phone systems to the desired local number. This method also has several drawbacks. An IT gateway must be available at both ends, locally and in the area to which it is desired to call. If an IT gateway is not available at either end, the user may actually have to pay double toll charges, one toll charge at the originating end and another toll charge at the receiving end. Furthermore, each IT gateway typically charges a fee for the time connected. Finally, the user has to sign up for IT gateway services with every IT gateway service provider for each geographic area in which he desires to place calls. This IT sign up typically includes a monthly subscription fee that must be paid regardless of use or non-use. It can be seen that this last method can be both cumbersome and could actually be quite expensive. 
     SUMMARY OF THE INVENTION 
     In accordance with one embodiment of the present invention, it is an object of the present invention to provide an improved internet telephone system. 
     It is another object of the present invention to provide an improved internet telephone system which allows an internet telephone to quickly make a call via an internet network. 
     It is still another object of the present invention to provide an improved internet telephone system which allows an internet telephone to quickly make a call via an internet network without requiring a dial-up MODEM. 
     BRIEF DESCRIPTION OF THE PREFERRED EMBODIMENTS 
     In accordance with one embodiment of the present invention, a quick connect internet telephone station is disclosed. The quick connect internet telephone station has a Plain Old Telephone Set (POTS) splitter coupled to a public switched telephone network (PSTN) for directing low frequency signals to a first line and DSL signals to a second line. A DSL line interface is coupled to the POTS splitter for driving and terminating the second line. A DSL transmitter is coupled to the DSL line interface. A DSL receiver is also coupled to the DSL line interface. A Digital Signal Processor (DSP) is coupled to the DSL transmitter and the DSL receiver. A telephone interface is coupled to the first line. An audio transceiver device is coupled to the telephone interface. A CODEC circuit is also coupled to the telephone interface. A ring and hook detect/control circuitry is coupled to the DSP. 
     In accordance with another embodiment of the present invention, an internet telephone system is disclosed. The internet telephone system uses a Public Switched Telephone Network (PSTN). An internet network is coupled to the PSTN. At least two telephone stations are part of the system wherein one of the at least two telephone stations is a DSL telephone station which is coupled to the PSTN. 
     In accordance with another embodiment of the present invention, a method for using a first DSL telephone for a making telephone call is disclosed. The method comprises the steps of: dialing a desired telephone number from the first DSL telephone via a telephone network; determining the type of telephone set at the desired telephone number; routing the telephone call through an internet network when the first DSL telephone recognizes the telephone set of the desired telephone number is one of an internet telephone or a second DSL telephone; and routing the telephone call through the telephone network when the first DSL telephone recognizes the telephone set of the desired telephone number is a POTS telephone. 
     The foregoing and other objects, features, and advantages of the invention will be apparent from the following, more particular, description of the preferred embodiments of the invention, as illustrated in the accompanying drawing. 
    
    
     BRIEF DESCRIPTION OF THE DRAWINGS 
     FIG. 1 is a simplified functional block diagram of the internet telephone system of the present invention. 
     FIG. 2 is a simplified functional block diagram of the quick connect internet telephone station of the present invention. 
     FIG. 3 is a functional block diagram of the firmware module used in the DSP of the quick connect internet telephone station depicted in FIG.  2 . 
    
    
     DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT 
     High-bit-rate Digital Subscriber Loop (HDSL) and Asymmetrical Digital Subscriber Loop (ADSL) was originally developed to replace T1 carrier and video on demand services, respectively. These digital subscriber loops allow a Customer Premise Equipment (CPE) to have a permanent data connection to a Public Switched Telephone Network (PSTN). Thus, a dial-up connection like an analog MODEM is not required. 
     ADSL has been developed for making Plain Old Telephone Set (POTS) telephone calls and allowing a higher speed data transmission on the same telephone line simultaneously. A POTS telephone call utilizes the lowest frequency band (approximately 0 to 4 kHz). There are currently many proposals for ADSL technologies. The most widely accepted and approved by standards technology committees is called Discrete Multiple Tone (DMT). The frequency band for DMT application is between 25 kHz and 1.1 MHz. In theory, up to an 8 Mbits/sec data string could be transmitted downstream by up to 256 separate 4.3125 KhZ tones in the frequency band of 138 kHz to 1.1 MHz. Up to 800 Kbits/s of data could be transmitted upstream in 32 tones between 25 kHz and 138 kHz. 
     Referring to FIG. 1, a simplified block diagram of the operating environment of the quick connect internet telephone system  10  (hereinafter system  10 ) of the present invention is shown. The system has at least one ADSL internet telephone  12 . The ADSL internet telephone  12  is coupled to a Public Switched Telephone Network (PSTN)  16  via an ADSL line  14 . A second telephone  18  is also coupled to the PSTN  16 . The second telephone may be a second ADSL telephone  18 A, an internet telephone  18 B, or a non-internet telephone  18 C (i.e. Plain Old Telephone Set (POTS)). If the second telephone  18  is an ADSL telephone  18 A, the ADSL telephone  18 A would be coupled to the PSTN by an ADSL line  20 . If the second telephone  18  is an internet telephone  18 B or a non-internet telephone  18 C, the second telephone  18  is coupled to the PSTN  16  via lines  22  and  24  respectively. 
     ADSL technology will allow an ADSL internet telephone  12  to have a continuous on-line data connection to the PSTN  16  and then to an internet network  26  via an Internet Service Provider (ISP)  28 . With the present system  10 , all toll charges may be avoided when calling a compatible ADSL internet telephone  18 A or an internet telephone  18 B by establishing a voice communication channel through the internet network  26  via each parties&#39; ISP  28 . Furthermore, the call set-up time for the ADSL telephone  12  is usually much shorter than that of a normal DTMF dialing through the PSTN, since a dial-up connection through an analog modem is not required. 
     The ADSL internet telephone  12  will preferably have a keypad for entering the name, PSTN number, E-Mail/IP address, telephone type, and other information of frequently called, and/or most recently called individuals. This information is generally saved in the memory of the ADSL internet telephone  12 . 
     When making a telephone call, the ADSL internet telephone  12  will detect a hook switch off hook signal. The user of the ADSL internet telephone  12  will then enter a desired PSTN number. The ADSL internet telephone  12  will search for the PSTN number in the data base. If the PSTN number is not found in the data base of the ADSL internet telephone  12 , the ADSL internet telephone  12  will send an off-hook signal followed by the PSTN number of the called number in DTMF digits to the PSTN  16 . A procedure for information exchange then begins. 
     When the ADSL internet telephone  12  is calling a compatible ADSL internet telephone  18 A or an internet telephone  18 B, the call is automatically answered on the first ring by the corresponding telephone of the called party. A direct modem transmission is set up between the calling ADSL internet telephone  12  and the called party&#39;s telephone (i.e., compatible ADSL internet telephone  18 A or an internet telephone  18 B). Information such as each parties&#39; telephone number, name, telephone type, and IP address is exchanged between the ADSL internet telephone  12  and the called party&#39;s telephone (i.e., compatible ADSL internet telephone  18 A or an internet telephone  18 B). This information is then saved in each telephone&#39;s data base. 
     If the called party has a compatible ADSL internet telephone  18 A, the ADSL internet telephone  12  sends a START_VOICE_COMMUNICATION protocol and user information to the IP address of the called party via an ADSL link to its ISP  28 . The ADSL internet telephone  12  also sends an off-hook signal and number to the PSTN  16 . Upon receipt of READY_VOICE_COMMUNICATION protocol from the called party&#39;s ADSL internet telephone  18 A, the ADSL internet telephone  12  sends an off-hook signal to the telephone company to terminate the PSTN call. A voice communication through the internet network  26  is established via the ADSL lines and each parties ISP  28 . If a READY_VOICE_COMMUNICATION is not received by the ADSL internet telephone  12 , the ADSL internet telephone  12  waits to establish a POTS telephone call via the PSTN  16 . 
     Upon detection of a hook switch on-hook signal, the calling ADSL telephone  12  will stop sending voice packets. The calling ADSL internet telephone  12  will send an END_VOICE_COMMUNICATION protocol to the IP address of the called ADSL internet telephone  18 A. When the ADSL internet telephone  12  receives a REQUEST_END_VOICE protocol from the called party&#39;s ADSL internet telephone  18 A, or after a period of time without reception of voice packets, the ADSL internet telephone  12  will apply background noise to the speaker informing the user that the called party has ended the telephone conversation. 
     If the called party has an internet telephone  18 B, the ADSL internet telephone  12  will detect only a single ringing tone. The ADSL internet telephone  12  will mute both the voice transmitter and receiver. The ADSL internet telephone  12  will then send its IP address and user information by a direct modem signal to the internet telephone  18 B. If a data packet including the called internet telephone&#39;s  18 B IP address and user information is not received, both voice transmitter and receiver of the ADSL internet telephone  12  are unmuted to establish a POTS telephone call via the PSTN  16 . If a data packet including the internet telephone&#39;s  18 B IP address and user information is received, the ADSL internet telephone  12  saves the information in its data base. This information may further be displayed. The ADSL internet telephone  12  will then go on-hook for a short duration (approximately 2 seconds). The ADSL internet telephone  12  will dial the telephone number of its ISP  28 . Upon connection to its ISP  28 , the ADSL internet telephone  12  will send a beginning-of-conversation protocol to the IP address of the internet telephone  18 B. Upon detection of a hook switch off-hook signal, the ADSL Internet telephone  12  will send packetized voice signals to the IP address of the internet telephone  18 B. Upon receipt of packetized data from the called internet telephone  18 B, the ADSL internet telephone  12  will convert the voice data into analog signals to establish a voice communication path via the internet network  26 . When the ADSL internet telephone  12  detects a hook switch on-hook signal, it stops sending voice packets and sends an end-of-conversation protocol to the IP address of the internet telephone  18 B. When the ADSL internet telephone  12  receives an end-of-conversation protocol from the called party&#39;s internet telephone  18 B, or after a period of time without reception of voice packets, the ADSL internet telephone  12  will apply background noise to the speaker informing the user that the called party has ended the telephone conversation. 
     If the called party only has a non-internet telephone  18 C, the non-internet telephone  18 C will not automatically answered the telephone call at the first ring tone as in the previous two examples above (i.e., compatible ADSL internet telephone  18 A or an internet telephone  18 B). Thus, the ADSL internet telephone  12  will know to process the call as a regular telephone call through the PSTN  16 . Even if the called party answers the non-internet telephone  18 C after only a single ring tone, the non-internet telephone  18 C will not be able to receive the calling party&#39;s user information and reply with its own information. Thus, the call will be processed as a regular POTS call via the PSTN  16 . 
     Referring to FIG. 2, a simplified function block diagram of the ADSL internet telephone  12  and  18 A (hereinafter ADSL internet telephone  12 ) is shown. The ADSL internet telephone  12  is coupled to an ADSL line  30 . The ADSL line  30  is a regular external telephone line which has ADSL capabilities. A POTS splitter  32  is coupled to the ADSL line  30 . The POTS splitter  32  is generally comprised of both a low pass filter  32 A and a high pass filter  32 B. The low pass filter  32 A is used to provide the pass band for voice frequency signals, dial tone, ringing, and on/off hook signals. The high pass filter  32 B is used for the ADSL signals. An ADSL interface  34  is coupled to the POTS splitter  32 . The ADSL interface  34  is used to drive and terminate the ADSL line. 
     An ADSL receiver  36  is coupled to an output of the ADSL line interface  34 . The ADSL receiver first converts the analog signal into a digital signal. The digital signal is then passed through a time domain equalizer into serial data. The serial data is converted into multiple channels. Data in each channel is converted into frequency domain signals by a Fast Fourier Transform (FFT) algorithm. The frequency domain signals pass through a frequency domain equalizer, then to a symbol decision, bit decision, and invert parsing functions. The output bit stream from the ADSL receiver  36  is then sent to a Digital Signal Processor (DSP)  38 . 
     An ADSL transmitter  40  is coupled to both the DSP  38  and the ADSL line interface  34 . The ADSL transmitter  40  receives bit streams from the DSP  38 . The ADSL transmitter  40  will convert the serial bit stream into parallel data. The parallel data is mapped into multibit subchannels according to a bit allocation algorithm. Each multibit subchannel data is converted into time domain signals by Inverse Fast Fourier Transform (IFFT). The parallel time domain signals are then converted into serial signals and then converted into analog signals to be outputted by the ADSL transmitter  40 . 
     The ADSL internet telephone  12  will further comprise a Digital-Analog-Analog (DAA) telephone interface  42 , a ring detector  44 , an off-hook detector  46 , a hook switch  48 , and a ringer  50  all of which have input terminals coupled to the ADSL line  30 . The DAA telephone interface  42  is used to convert signals from the ADSL line  30  to a four wire interface. The DAA telephone interface  42  is further used to send signals from the DSP  38 , which have been converted to analog signals by the CODEC  52 , to the ADSL line  30 . The ring detector  44  is a circuit which monitors when an incoming telephone call is made to the ADSL internet telephone  12 . If an incoming call is placed to the ADSL internet telephone  12 , the DSP  38  will energize the ringer  50  to signal that a calling is being placed to the ADSL internet telephone  12 . The off-hook detector  46  will monitor when the headset of the ADSL internet telephone  12  is lifted thereby allowing dialing and transmission but prohibiting incoming calls from being answered. The hook switch  48  is a switch that closes a circuit when the headset of the ADSL internet telephone  12  is lifted thereby allowing dialing and transmission but prohibiting incoming calls from being answered. 
     A (Coder/Decoder) CODEC  52  is coupled to the DAA telephone interface  42 . The CODEC  52  receives analog signals from the DAA telephone interface  42 . The CODEC  52  will convert the analog signals to digital signals and send the digital signals to the DSP  38  for processing. A microphone  54  and a speaker  56  may also be coupled to the DAA telephone interface  42 . The microphone  54  is used to convert sound waves into electronic signals. The speaker  56  will convert electronic impulses to sound waves of sufficient volume to be heard. 
     A keypad  56  is coupled to the DSP  38 . The keypad  58  is used to enter user information of parties who are frequently called. Such information may include, but is not limited to, a party&#39;s name, telephone number, telephone type, and ISP/IP address. This information is generally stored in a data base in the ADSL internet telephone  12 . The data base is generally a memory module. In the embodiment depicted in FIG. 2, the memory module is comprised of Random Access Memory (RAM)  58  and FLASH memory  60 . The FLASH memory  60  is generally used to store firmware programs and information entered by the user via the keypad  56  (i.e., information of parties who are frequently called such as, a party&#39;s name, telephone number, telephone type, and ISP/IP address). The RAM  58  is generally used as a scratch pad during program execution. 
     A display mechanism  54  is also coupled to the DSP  38 . In the embodiment depicted in FIG. 2, the display mechanism  54  is a Liquid Crystal Display (LCD)  54 . The LCD  54  is used for displaying information that the user entered via the keypad  56  such as the party&#39;s name, telephone number, telephone type, and ISP/IP address. The LCD  54  may also function to display information concerning an incoming call. Thus, the LCD  54  may function like a Caller ID unit. 
     Referring to FIG. 3, a block diagram of the firmware features implemented in the DSP  38  is shown. Hardware components for these features are commercially available. Firmware implementation of these features is used as an example. The firmware features generally include: a signal selector  62 , packet assembler  64 , DTMF generator  66 , MODEM transmitter  68 , data compressor  70 , packet disassembler  72 , Call Progress (CP) detector, MODEM receiver  76 , and data decompressor  78 . The functions of these firmware features of the DSP  38  will be discussed below. 
     Referring now to FIGS. 1-3, the operation of the system  10  will be discussed. It should be noted that the internet telephone  18 B is similar in design to that of the ADSL internet telephones  12  and  18 A shown in FIGS. 2 and 3 except that the specific ADSL function blocks are not included. Thus, the operation of the internet telephone  18 B will be described in reference to FIGS. 2 and 3. 
     The ADSL internet telephone  12  dials a telephone number by sending Dual Tone Multiple Frequencies (DTMF) to the PSTN  16 . It should be noted that DTMF dialing is a traditional hardware function. It is implemented in this embodiment of the ADSL internet telephone  12  as a firmware function. The DTMF generator  66 , a firmware function of the DSP  38 , sends DTMF signals to the signal selector  62 . This signal is converted into an analog signal by the CODEC  23  and sent to the ADSL line  30  through the DAA telephone interface  42 . A telephone company will make a connection to the dialed number via the PSTN  16  and will send a ringing signal to the dialed number. 
     If the called number is a non-internet telephone  18 C, the non-internet telephone  18 C will ring continuously till an end user answers the call. If the called number is a compatible internet telephone  18 A or an internet telephone  18 B (hereinafter called telephone  18 A), the ringing detector  44  detects incoming ringing signals and informs the DSP  38  of the incoming ringing signals. Upon receipt of a first ringing signal, the DSP  38  disables the ringer  50  to stop the called party from lifting the headset of the called telephone  18 A. The DSP  38  activates the electronic hook switch  48  to close the loop and answer the incoming telephone call. 
     At the calling ADSL internet telephone  12 , incoming signals on the ADSL line  30  are converted to a four wire interface by the DAA telephone interface  42 . The incoming signals are digitized by the CODEC  52  and sent to the CP detector  74 , a firmware module within the DSP  38 . 
     As stated above, if more than one ringing tone is detected, the called party is a non-internet telephone  18 C. Ringing will continue till the non-internet telephone  18 C is answered by the called party or the calling party discontinues the telephone call. The call will be routed through the PSTN  16 . No attempt is made to reroute the call through the internet network  26 . 
     If the calling ADSL internet telephone  12  detects only a single ringing tone, this call is answered by the called telephone  18 A. An exchange of user information will begin. The DSP  38  will send a signal causing the LED  55  to begin to flash indicating that an exchange of user information has begun. 
     The ADSL internet telephone  12  will transfer its user information in the following manner. The DSP  38  will retrieve the user information (i.e., name, telephone number, telephone type, and ISP/IP address) of the ADSL internet telephone  12  from the FLASH memory  29 . The packet assembler  64  assembles the information into data packets which is converted into a MODEM signal in a digitized format by the MODEM transmitter  68 . The digitized MODEM signal is sent to the CODEC  52  through the signal selector  62 . The CODEC  52  converts the digitized MODEM signal into an analog signal. The analog signal is sent to the ADSL line  30  through the DAA telephone interface  42  to the called telephone  18 A. The called telephone  18 A receives the MODEM signal in its analog form at its DAA telephone interface  42 . The analog signal is converted to a digital signal by the CODEC  52  and is sent to the MODEM receiver  76  via the signal selector  62 . The DSP  38  disassembles the data packet from the MODEM receiver  76 , displays the calling party&#39;s telephone number and name on the LCD  54 . The DSP  38  will also store the calling party&#39;s information in its FLASH memory  60 . 
     The DSP  38  of both the calling ADSL internet telephone  12  and the called telephone  18 A deactivate their respective electronic hook switch  48  to go on-hook. After approximately two seconds of delay, each DSP  38  activates its respective electronic hook switch  48  to go to an off-hook state. 
     The DTMF generator  66  of the calling ADSL internet telephone  12  sends the telephone number of its ISP  28  through the signal selector  62 . The digital signal is converted into an analog signal by the CODEC  52 . The DTMF signal in analog format is sent to the ADSL line  30  via the DAA telephone interface  42 . Upon receipt of the DTMF signal, the PSTN  16  connects this call to the ISP  28 . When the CP detector  74  detects the end of a ringing tone, it informs the MODEM transmitter  68  to begin transmission of a MODEM signal training sequence at a speed of 14.4 kbps. After the connection of 14.4 kbps modem to the ISP  28 , the CODEC  52  begins to receive voice signals through the microphone  54  via the DAA telephone interface  42 . The CODEC digitizes the voice signal at 8 k bytes per second or 64 kbps. The digitized voice signal is sent to the data compressor  70  where the voice signal is compressed using the GSM algorithm to 13 kbps or the ITU G 723.1 algorithm to about 6 kbps. The data packet assembler  64  assembles the compressed data with the IP address of the called telephone  18 A. The data packet is sent to the MODEM transmitter  68 , to the signal selector  62 , to the CODEC  52 , through the DAA telephone interface  42 , then to the ADSL line  30 . 
     At the same time, the called telephone  18 A calls its ISP  28  and begins transmission of packetized voice signals with the IP address of the calling ADSL internet telephone  12 . A voice transmission path is now established via the internet network  26 . 
     The ADSL internet telephone  12  receives the MODEM signal from the ADSL line  30 . The MODEM signal is sent through the DAA telephone interface  42 , to the CODEC  52 , to the MODEM receiver  76  where the signal is converted to data packets. The data packets are sent to the data decompressor  78  where the IP address and other overhead bytes are removed to obtain the compressed data. The compressed data is converted into 64 kbps data. This data is converted into an analog signal which is sent to the speaker  56  via the CODEC  52  and the DAA telephone interface  42 . A voice reception path is now established through the internet network  26 . Thus, instead of making a call through the PSTN  16  and paying toll charges, the call is routed through the internet network  26 . Toll charges can now be saved. 
     While the invention has been particularly shown and described with reference to preferred embodiments thereof, it will be understood by those skilled in the art that the foregoing and other changes in form and details may be made therein without departing from the spirit and scope of the invention.