Abstract:
A telephone conferencing system interface is provided which is operable at high bandwidths via a VoIP telephone. A teleconference system interface module is connected to a the VoIP telephone via the telephone&#39;s headset connector and to a teleconference system via conferencing input circuitry, conferencing output circuitry and interface module connection circuitry. The teleconference interface module includes ground isolation transformers selected for high bandwidth operation and includes a privacy switch which can disconnect a teleconference from the VoIP telephone.

Description:
FIELD OF THE INVENTION 
       [0001]    The present invention relates to telecommunications and, more particularly to a system and method for interfacing a wideband telephone with a teleconferencing system. 
       BACKGROUND OF THE INVENTION 
       [0002]    In modern implementations, the majority of audio signal transmission over the public switched telephone PSTN is digital; however, signal transmission over the PSTN was originally analog in nature. Disadvantageously, analog circuits introduce random variation (i.e. noise) into the signal, and this noise increases in proportion to the distance the signal has traveled through the analog circuit. To mitigate this problem, the analog PSTN uses band pass filters to eliminate all frequencies outside the voice frequency range of about 300-3,400 Hz. This frequency range was chosen because most of the energy for intelligible speech occurs between about 0-4,000 Hz, and while the human voice can produce frequencies in the range of about 30-14,000 Hz, early telecommunications engineers determined that frequencies in the range of about 0-300 Hz and about 3,400-14,000 Hz were not necessary to understand transmitted voice signals. Advantageously, band pass filters remove signal transmission noise generated as the analog signal is amplified by network repeaters during transmission between the transmitting and receiving points, and help maintain a beneficial signal to noise ratio. This historical implementation of band pass filters that only transmit the voice frequency was maintained as the analog PSTN was converted to the pulsed code modulation (PCM) system of the digital PSTN. 
         [0003]    The ability to accurately discern speech is influenced by the range of vocal frequencies of the human voice and the auditory range of the human ear. While the human voice can produce frequencies in the range of about 30-14,000 Hz, the human ear can typically hear frequencies in the range of about 15-20,000 Hz. Frequency spectrum analysis indicates that the fundamental frequency of the average human voice has a range between about 80-400 Hz. Disadvantageously, much of this fundamental frequency range is not included in the voice frequency transmitted over the PSTN. Generally, enough of the harmonic series in a given speech pattern will be present in the voice frequency to give the listener at the receiving point the impression of actually hearing the fundamental frequencies, even though they have been removed by the band pass filter. 
         [0004]    The harmonic series in a speech pattern are made up of concentrations of acoustic energy around a particular frequency in a speech wave. These concentrations usually occur at 1,000 Hz intervals, and play an important role in enabling listeners to discriminate between certain consonant sounds, for example, “s and f” or “p and t” or “m and n.” Disadvantageously, band pass filters on the PSTN remove some of these frequencies which are important in allowing a listener to correctly distinguish many consonant sounds. 
         [0005]    As digital signal processing power has increased, the ability to implement more advanced voice-compression algorithms has also increased. Consequently, telecommunications have trended toward a wideband environment, for example the use of voice over internet protocol (VoIP) in which the historical voice frequency range has been expanded. The current implementation standard for wideband voice quality is the G.722 codec, which uses an adaptive differential PCM to double the audio content within a typical 64 kbps audio data stream. While the original PCM used a 64 kbps audio data stream with a sampling rate of about 8,000 Hz, the G.722 codec doubles the sampling rate to about 16,000 Hz. Since the sampling rate corresponds to about double the highest frequency in the audio data stream, the G.722 codec expands the wideband voice frequency range to about 50-7,000 Hz. 
         [0006]    An important aspect of telephony is that users be able to hear themselves in the earpiece of the telephone when they are speaking. This has the advantage of providing a positive feedback signal so that the user knows the telephone is working. This is usually accomplished by diverting a low level of the user&#39;s audio transmission back into the user&#39;s earpiece, and is known in the industry as sidetone. In modern telephones, sidetone is created by electronic circuitry within the phone. Disadvantageously, sidetone creates audio feedback in teleconferencing systems that leads to acoustic echo if it is not attenuated. 
         [0007]    Existing teleconferencing systems use line echo cancellation (LEC) circuits that attenuate sidetone. A problem with prior art echo cancellation systems in existing teleconferencing systems is that they are designed for the standard 300-3,400 Hz voice frequency range associated with the PSTN, and such systems have not been developed for audio data streams that transmit in the expanded frequency range of about 50-7,000 Hz. 
         [0008]    Another problem with prior art solutions is that traditionally a third party control method is required to initiate and control the receive level and muting functions of the microphone during a call. Disadvantageously, such third party audio equipment solutions typically mount in an equipment rack that is not accessible to the user, thereby preventing the user from easily controlling the equipment. 
       SUMMARY OF THE INVENTION 
       [0009]    The present invention provides a telephone conferencing system interface operable at high bandwidths via a VoIP telephone. A teleconference system interface module is connected to the VoIP telephone via the telephone&#39;s headset connector and to a teleconference system via conferencing input circuitry, conferencing output circuitry and interface module connection circuitry. The teleconference interface module includes ground isolation transformers selected for high bandwidth operation and includes a privacy switch which can disconnect a teleconference from the VoIP telephone. 
         [0010]    In an illustrative embodiment, digital signal processing software removes side tone or electrical echo present on the connections. The software also provides automatic level or gain control to optimize the level. Embodiments of the invention directly connect and integrate standard corporate telephone hardware with conferencing hardware to allow user control of the conferencing hardware via the standard telephone hardware. 
     
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
         [0011]    The features and advantages of the present invention will be better understood when reading the following detailed description, taken together with the following drawings in which: 
           [0012]      FIG. 1A  is an illustration of a teleconferencing system interface to a VoIP telephone in accordance with an illustrative embodiment of the present invention. 
           [0013]      FIG. 1B  is an illustration of a VoIP telephone including an integrated headset port. 
           [0014]      FIG. 1C  illustrates cable interconnection types for the VoIP and teleconferencing system interface according to the invention. 
           [0015]      FIG. 2  is a schematic diagram of an interface module according to an illustrative embodiment of the invention; 
           [0016]      FIG. 3  is a schematic diagram of interface module input circuitry in accordance with an illustrative embodiment of the invention; 
           [0017]      FIG. 4  is a schematic diagram of conferencing output circuitry in accordance with an illustrative embodiment of the present invention; and 
           [0018]      FIG. 5  is a functional block diagram of software steps for controlling mute control via in interface module in accordance with an illustrative embodiment of the present invention. 
       
    
    
     DETAILED DESCRIPTION OF THE INVENTION 
       [0019]    An illustrative embodiment of the invention is described with reference to  FIG. 1A  which illustrates a teleconference system interface to a broadband telephone such as a VoIP telephone. A teleconference system  100  such as a SimphoniX phone interface is connected to an interface module  102  via a first cable  104 . An illustrative embodiment of the interface module  102  is referred to as the HSET module. Illustratively, the SimphoniX includes processor circuitry running line echo cancellation software. The interface module  102  is connected to a VoIP telephone  106  via a second cable  108 . The VoIP telephone  106  may be connected to a VoIP network  105  via a third cable  107 . 
         [0020]    The teleconference system  100  is typically configured to drive a speaker  110  and receive audio signals from one or more microphones  112 .  FIG. 1B  illustrates a standard integrated headset port  114  of the VoIP telephone  106 . 
         [0021]    An illustrative embodiment of the interface module  102  is described with reference to the schematic diagram of  FIG. 2 . The interface module includes a first jack  200  adapted for connection to a headset port of a VoIP telephone such as the headset port  114  of  FIG. 1B . A second jack  202  is adapted for connection to the teleconference system. Connections are made according to the cable types illustrated in  FIG. 1C . 
         [0022]    Ground isolation circuitry is electrically connected between the first jack and the second jack to prevent a direct ground connection between the first jack and the second jack. The ground isolation circuitry includes a first transformer  204  and a second transformer  206 . The first jack  200  includes a first conductor pair  206  connected across a first winding  208  of the first transformer  204  and a second conductor pair  210  connected across a first winding  212  of the second transformer  206 . The second jack  202  includes a third conductor pair  214  connected across a second winding  216  of the first transformer  204  and a fourth conductor pair connected across a second winding  220  of the second transformer  206 . 
         [0023]    In the interface module  102 , the ground isolation circuitry may further include a first resistor  222  in series with the second conductor pair  210 . The first resistor is carefully selected to in accordance with the VoIP telephone or other device connected to the first jack. In the illustrative embodiment the first transformer  204  and second transformer  206  are type TY-145P transformers manufactured by Triad Magnetics of Corona, Calif. Persons having ordinary skill in the art should understand that other transformers that are substantially equivalent to type TY-145P transformers may be used within the scope of the present invention, and that modification of the transformers may widen the frequency response of the interface module  102 . 
         [0024]    The interface module  102  may further include switching means  224  having a first pair of switch terminals  226  in series with the first conductor pair  206  and a second pair of switch terminals  228  in series with the fourth conductor pair  218 . The switching means  224  are illustratively adapted to simultaneously open or close the first conductor pair  206  connection to the first winding  208  of the first transformer  204  and the fourth conductor pair  218  connection to the second winding  220  of the second transformer  206 . In an illustrative embodiment of the invention, the switching means can be a double pole single throw switch, however persons having ordinary skill in the art should understand that numerous other types of switches, relays or other circuitry may provide switching means  224  to connect and disconnect the respective windings and conductor pairs. 
         [0025]    The interface module  102  may include a third jack  230  adapted for connection to a cell phone. The third jack  230  includes a fifth conductor pair  232  connected in parallel with the third conductor pair  214  and a first conductor  234  connected in series with a second resistor  236  to a conductor of the fourth conductor pair  218 . In the illustrative embodiment, both the first resistor  222  and second resistor  236  are 1K ohm resistors. Persons having ordinary skill in the art should appreciate that various other resistors may be added or substituted for the 1K ohm resistors to adjust isolation characteristics of the interface module  102  within the scope of the present invention. 
         [0026]      FIG. 3  is a schematic diagram of interface module input circuitry in accordance with an illustrative embodiment of the invention. The interface module interface circuitry  300  is illustratively located inside the teleconference system  100  ( FIG. 1 ) and connected to the interface module  102  ( FIG. 1 ) via connector  301 . Impedance of the input circuitry  300  and the second transformer  206  are selected to provide a wideband frequency response. The interface module interface circuitry  300  includes a radio frequency [RF] filter portion  302  which filters radio frequency components from the interface module  102 . Band pass filter circuits  304  filter out low-frequency bands and high frequency bands to improve the signal to noise ratio of the signal from the interface module  102 . The signal is amplified through operational amplifiers  306  and  308  and its gain is controlled by a programmable gain amplifier [PGA] via logic pins  310 . After RF filtering, band-pass filtering and amplification in the PGA circuitry, the signal from the interface module  102  is sent to analog to digital conversion (ADC) circuitry via ADC output  312  for digital sampling. 
         [0027]      FIG. 4  is a schematic diagram of conferencing output circuitry  400  in accordance with an illustrative embodiment of the present invention. The conference output circuitry  400  converts a digital interface module output signal into an analog signal in ADC circuitry  402 . The then passes the signal through band pass filter circuitry  404  to improve the signal to noise ratio and then through amplifier buffering circuitry  406 . The buffering  406  circuitry is configured to expand frequency response of the second transformer  206 . 
         [0028]    The teleconferencing system according to the invention includes echo cancellation and automatic gain control software as illustrated in  FIG. 5 . In a Dual-Tone Multiple-Frequency (DTMF) detection stop  502 , a digital signal processor [DSP] in the teleconference system  100  performs DTMF detection to check if certain period of tones (for example, “*” or “#” or Beep tone) is received from the interface module input. Based on DTMF tone detection, a Mute” flag is set or reset. The “Mute” flag is used as a control signal  504 ,  506  which controls the software running on the DSP to mute or pass interface module input signals  508  and/or output signals  510 . In an illustrative embodiment of the invention, the mute function can be controlled by the tone detection software steps illustrated in  FIG. 5  and/or by a hardware switch such as switching means  224  of  FIG. 2 , for example. 
         [0029]    Although the disclosure hereof has been stated by way of example of illustrative embodiments, it will be evident that other adaptations and modifications may be employed without departing from the spirit and scope thereof. The terms and expressions employed herein have been used as terms of description and not of limitation; and thus, there is no intent of excluding equivalents, but on the contrary it is intended to cover any and all equivalents that may be employed without departing from the spirit and scope of the invention set forth in the claims.