Abstract:
A switchboard device and methods of operation of same are disclosed. Embodiments of the invention may provide a flexible means of interconnecting wideband and narrowband communications interfaces, where wideband communications interfaces may transfer wideband data sampled at a higher sampling rate, and narrowband communication interfaces may transfer narrowband data sampled at a lower sampling rate. Data streams sampled at different sampling rates can be combined and the sampling rate of the result adjusted as needed by the destination interface. Methods of operating embodiments of the present invention are included. An additional aspect of the present invention may include machine-readable storage having stored thereon a computer program having a plurality of code sections executable by a machine for causing the machine to perform the foregoing.

Description:
CROSS-REFERENCE TO RELATED APPLICATIONS/INCORPORATION BY REFERENCE  
       [0001]    This application is also related to the following co-pending applications, each of which is herein incorporated by reference:  
                                               Ser. No.   Docket No.   Title   Filed   Inventors                   60/414,059   14057US01   Multiple Data Rate Communication   Sep. 27, 2002   LeBlanc               System       Houghton                       Cheung       60/414,460   14061US01   Dual Rate Single Band Communication   Sep. 27, 2002   LeBlanc               System       Houghton                       Cheung       60/414,491   14063US01   Splitter and Combiner for Multiple Data   Sep. 27, 2002   LeBlanc               Rate Communication System       Houghton                       Cheung       60/414,492   14062US01   Method and System for an Adaptive   Sep. 27, 2002   LeBlanc               Multimode Media Queue       Houghton                       Cheung                  
 
     
    
     
       FEDERALLY SPONSORED RESEARCH OR DEVELOPMENT  
         [0002]    [Not Applicable] 
         MICROFICHE/COPYRIGHT REFERENCE  
         [0003]    [Not Applicable] 
         BACKGROUND OF THE INVENTION  
         [0004]    Traditional voice telephony products are typically band-limited to 4 kHz bandwidth using an 8 kHz sampling rate. These products, sometimes labeled as “narrowband”, include the telephone, data modems, and fax machines. Newer products aiming to achieve higher voice quality have doubled the sampling rate to 16 kHz to encompass a larger 8 kHz bandwidth, which is also known as “wideband” capable. The software implications of using a higher sampling rate are significant. Increasing the sampling rate not only increases the processing cycles needed, but also increases the memory used to store the data. In addition, software for systems supporting wider bandwidths and higher sampling rates must not preclude support for legacy band-limited functionality.  
           [0005]    Increasing memory and processor cycles requirements is expensive because the memory and processing power footprints of digital signal processors (DSPs) are generally small. Implementing support for wider bandwidths thus requires creativeness to optimize memory and processor cycles, and in the means to support a variety of sampling rates.  
           [0006]    In an environment with both narrowband and wideband devices, a voice call between a narrowband terminal and a wideband terminal cannot be accomplished by simply exchanging voice data streams. Voice telephony services such as conferencing require that the voice data streams from devices using different sampling rates be combined, so that each participant may hear the voices of all other participants. Combining the digital audio streams from a narrowband terminal with a lower sampling rate and a wideband terminal with a higher sampling rate requires that adjustments be made to the voice data streams to allow them to be combined. It is also necessary that the resulting combined voice data stream be made available in a form acceptable to each participant&#39;s terminal, whether it uses a lower or higher sampling rate.  
           [0007]    Accordingly, there is a need for switchboard functionality that can support the interconnection of both narrowband devices and wideband devices.  
           [0008]    Further limitations and disadvantages of conventional and traditional approaches will become apparent to one of skill in the art, through comparison of such systems with aspects of the present invention as set forth in the remainder of the present application with reference to the drawings.  
         BRIEF SUMMARY OF THE INVENTION  
         [0009]    The present invention relates to the interconnection of telephony devices in a digital communications networks. More specifically, aspects of the present invention can be seen in a switchboard device used to interconnect voice telephony terminals operating at different sampling rates.  
           [0010]    An embodiment in accordance with the present invention may comprise a first input for receiving a first stream of data sampled at a first sampling rate, a second input for receiving a second stream of data sampled at a second sampling rate, a converter for modifying the first stream of data producing a modified stream of data sampled at the second sampling rate, and a combiner for combining the modified stream of data and the second stream of data, producing a combined stream of data. It may further comprise a converter for modifying the combined stream of data, producing a modified combined stream of data sampled at the first sampling rate. The first and second sampling rates may be different, and the combiner may add the modified stream of data and the second stream of data. The first sampling rate may be approximately 8 kHz, and the second sampling rate may be approximately 16 kHz.  
           [0011]    Another aspect of the present invention relates to a method of operating a switchboard device, the method comprising receiving a first stream of data sampled at a first sampling rate, receiving a second stream of data sampled at a second sampling rate, converting the first stream of data to the second sampling rate, and combining the converted first stream of data and the second stream of data, producing a combined stream of data. The method may further comprise converting the combined stream of data to the second sampling rate, where the first and second sampling rates may be different. The combining may comprise adding. The first sampling rate may be approximately 8 kHz, and the second sampling rate may be approximately 16 kHz.  
           [0012]    A further embodiment of the present invention may include machine-readable storage, having stored thereon a computer program having a plurality of code sections executable by a machine for causing the machine to perform the foregoing.  
           [0013]    These and other advantages, aspects, and novel features of the present invention, as well as details of illustrated embodiments, thereof, will be more fully understood from the following description and drawings.  
       
    
    
     BRIEF DESCRIPTION OF SEVERAL VIEWS OF THE DRAWINGS  
       [0014]    [0014]FIG. 1 is a block diagram of an exemplary communication system wherein the present invention can be practiced.  
         [0015]    [0015]FIG. 2 is a data flow diagram for a single-band architecture in accordance with an embodiment of the present invention.  
         [0016]    [0016]FIG. 3 is a system block diagram of a signal processing system operating in a voice mode in accordance with an illustrative embodiment of the present invention.  
         [0017]    [0017]FIG. 4 is a high-level block diagram of a switchboard device in accordance with an embodiment of the present invention.  
         [0018]    [0018]FIG. 5 is a data flow diagram for a switchboard connection between a wideband PXD and two VHDs, one wideband and the other narrowband, in accordance with an embodiment of the present invention.  
         [0019]    [0019]FIG. 6 is a data flow diagram for a switchboard connection between a wideband PXD, a wideband VHD, and a narrowband VHD in accordance with an embodiment of the present invention.  
         [0020]    [0020]FIG. 7 is a block diagram of an exemplary terminal in which aspects of the present invention may be practiced.  
     
    
     DETAILED DESCRIPTION OF THE INVENTION  
       [0021]    Referring now to FIG. 1, there is illustrated a block diagram of an exemplary voice over packet network  100  wherein the present invention can be practiced. The voice over packet network  100  comprises a packet network  105  and a plurality of terminals  110 . The terminals  110  are capable of receiving user input. The user input can comprise, for example, voice, video, or a document for facsimile transmission.  
         [0022]    The terminals  110  are equipped to convert the user input into an electronic signal, digitize the electronic signal, and packetize the digital samples. Additionally, the terminals  110  can selectively address a particular one of the other terminals  110 , a destination terminal for transmission of the packetized digital samples.  
         [0023]    The communication system  100  utilizes single-band data streams that may be sampled at different sampling rates. For example, one or more of terminals  110  may be narrowband terminals with a 4 kHz bandwidth, exchanging digital voice data sampled at an 8 kHz sampling rate. Other terminals  110  may operate in wideband mode using, for example, 8 kHz of bandwidth and exchanging voice data sampled at a 16 kHz sampling rate. Yet other terminals  110  may be capable of operating in either narrowband or wideband mode, or at yet another sampling rate supporting another bandwidth. The choices of the number of sampling rates and bandwidths are arbitrary, and the present invention is not limited thereby. The narrowband and wideband terminals may be similar in implementation except for the sampling rate of the media streams.  
         [0024]    Referring now to FIG. 2, there is illustrated a signal flow diagram of a single-band architecture  200  in accordance with an embodiment of the present invention. The single-band architecture  200  includes a Virtual Hausware Driver (VHD)  205 , a switchboard  210 , a physical device driver (PXD)  215 , an interpolator  220 , and a decimator  225 .  
         [0025]    The VHD  205  is a logical interface to destination terminal  110  via the packet network  105  and performs functions such as dual tone multi-frequency (DTMF) detection and generation, and call discrimination (CDIS). During a communication (e.g., voice, video, fax) between terminals, each terminal  110  associates a VHD  205  with each of the terminal(s)  110  with which it is communicating. For example, during a voice-over-packet (VoP) network call between two terminals  110 , each terminal  110  associates a VHD  205  with the other terminal  110 .  
         [0026]    The switchboard  210  associates the VHD  205  and the PXD  215  in a manner that will be described below.  
         [0027]    The PXD  215  represents an interface for receiving the input signal from the user and performs various functions, such as echo cancellation. The top of the PXD  215  is at the switchboard  210  interface. The bottom of the PXD  215  is at the interpolator  220  and decimator  225  interface. In general, the functions within a wideband PXD  215  would be designed to use 16 kHz sampled data, while functions in a narrowband PXD  215  would expect to process 8 kHz sampled data.  
         [0028]    A wideband system may contain a mix of narrowband and wideband VHDs  205  and PXDs  215 . A difference between narrowband and wideband device drivers is their ingress and egress sample buffer interface. A wideband VHD  205  or PXD  215  has wideband data at its sample buffer interface and includes wideband services and functions. A narrowband VHD  205  or PXD  215  has narrowband data at its sample buffer interface and can include narrowband services and functions. The switchboard interfaces with narrowband and wideband VHDs  205  and PXDs  215  through their sample buffer interfaces. The switchboard  210  is incognizant of the wideband or narrowband nature of the device drivers, but is aware of the sampling rate of the data that it reads and writes data through the sample buffer interfaces. To accommodate differences in the sampling rates of data streams, an embodiment of the present invention may upsample data received from narrowband sources and downsample data being sent to narrowband destinations. The sample buffer interfaces may provide data at any arbitrary sampling rate. In an embodiment of the present invention, the narrowband sample buffer interface may provide data sampled at 8 kHz and the wideband sample buffer interface may provide data sampled at 16 kHz. Additionally, a VHD  205  may be dynamically changed between wideband and narrowband and vice versa.  
         [0029]    The VHD  205  and PXD  215  driver structures may include sample rate information to identify the sampling rates of the wideband and narrowband data. The information may be part of the interface structure that the switchboard understands and may contain a buffer pointer and an enumeration constant or the number of samples to indicate the sample rate.  
         [0030]    The single-band architecture  200  is also characterized by an ingress path and an egress path, wherein the ingress path transmits user inputs to the packet network, and wherein the egress path receives packets from the packet network  105 . The ingress path and the egress path can either operate in a wideband support mode or a narrowband support mode. Additionally, the ingress path and the egress path are not required to operate in the same mode. For example, the ingress path can operate in the wideband support mode, while the egress path operates in the narrowband mode. The ingress path comprises the decimator  225 , echo canceller  235 , switchboard  210 , and services including but not limited to DTMF detector  240  and CDIS  245 , and packet voice engine (PVE)  255  comprising an encoder algorithm  260 .  
         [0031]    In the ingress path of a wideband device, the decimator  225  receives the user inputs and provides 16 kHz sampled data, B, for an 8 kHz band-limited signal. The 16 kHz sampled data, B, is transmitted through echo canceller  235  and switchboard  210  to the VHD  205  associated with the destination terminal  110 . In some cases, the DTMF detector  240  may be designed for operation on only narrowband digitized samples, and wideband data, B, is downsampled and passed to DTMF detector  240 . Similarly, where CDIS  245  is designed for operation on only narrowband digitized samples, only the downsampled wideband data is provided to CDIS  245 , which distinguishes a voice call from a facsimile transmission.  
         [0032]    The PVE  255  is responsible for issuing media queue mode change commands consistent with the active encoder and decoder. The media queues can comprise, for example, the media queues described in provisional patent application Ser. No. 60/414,492, “Method and System for an Adaptive Multimode Media Queue”, which is incorporated herein by reference in its entirety. The PVE  255  ingress thread receives raw samples. Depending upon the operating mode of VHD  205 , the raw samples include either narrowband or wideband data. At PVE  255 , encoder  260  packetizes the sampled data for transmission over the packet network  105 . The encoder  260  can comprise, for example, the BroadVoice 32 Encoder made by Broadcom, Inc.  
         [0033]    The egress path comprises decoder  263 , CDIS  266 , DTMF generator  269 , switchboard  210 , echo canceller  235 , and interpolator  220 . The egress queue receives data packets from the packet network  105  at the decoder  263 . The decoder  263  can comprise the BroadVoice 32 Decoder made by Broadcom, Inc. The decoder  263  decodes data packets received from the packet network  105  and provides 16 kHz sampled data. If CDIS  266  and DTMF generator support 16 kHz sampled data, the 16 kHz sampled is provided to CDIS  266  and DTMF generator  269 . Again, in one embodiment, where CDIS  266  and DTMF generator  269  require narrowband digitized samples, the wideband data may be downsampled and used by CDIS  266  and the DTMF generator  269 .  
         [0034]    The DTMF generator  269  generates DTMF tones if detected from the sending terminal  110 . These tones are written to the wideband data, A. The wideband data, A, is received by the switchboard  210 , which provides the data to the PXD  215 . The sampled data is passed through the echo canceller  235  and provided to interpolator  220 . The interpolator  220  provides 16 kHz sampled data.  
         [0035]    The services invoked by the network VHD in the voice mode and the associated PXD are shown schematically in FIG. 3. In the described exemplary embodiment, the PXD  60  provides two-way communication with a telephone or a circuit-switched network, such as a PSTN line (e.g. DS 0 ) carrying a 64 kb/s pulse code modulated (PCM) signal, i.e., digital voice samples.  
         [0036]    The incoming PCM signal  60   a  is initially processed by the PXD  60  to remove far-end echoes that might otherwise be transmitted back to the far-end user. As the name implies, echoes in telephone systems are the return of the talker&#39;s voice resulting from the operation of the hybrid with its two-four wire conversion. If there is low end-to-end delay, echo from the far end is equivalent to side-tone (echo from the near-end), and therefore, not a problem. Side-tone gives users feedback as to how loudly they are talking, and indeed, without side-tone, users tend to talk too loudly. However, far-end echo delays of more than about 10 to 30 msec significantly degrade the voice quality and are a major annoyance to the user.  
         [0037]    An echo canceller  70  is used to remove echoes from far-end speech present on the incoming PCM signal  60   a  before routing the incoming PCM signal  60   a  back to the far-end user. The echo canceller  70  samples an outgoing PCM signal  60   b  from the far-end user, filters it, and combines it with the incoming PCM signal  60   a . Preferably, the echo canceller  70  is followed by a non-linear processor (NLP)  72  which may mute the digital voice samples when far-end speech is detected in the absence of near-end speech. The echo canceller  70  may also inject comfort noise which in the absence of near-end speech may be roughly at the same level as the true background noise or at a fixed level.  
         [0038]    After echo cancellation, the power level of the digital voice samples is normalized by an automatic gain control (AGC)  74  to ensure that the conversation is of an acceptable loudness. Alternatively, the AGC can be performed before the echo canceller  70 . However, this approach would entail a more complex design because the gain would also have to be applied to the sampled outgoing PCM signal  60   b . In the described exemplary embodiment, the AGC  74  is designed to adapt slowly, although it should adapt fairly quickly if overflow or clipping is detected. The AGC adaptation should be held fixed if the NLP  72  is activated.  
         [0039]    After AGC, the digital voice samples are placed in the media queue  66  in the network VHD  62  via the switchboard  32 ′. In the voice mode, the network VHD  62  invokes three services, namely call discrimination, packet voice exchange, and packet tone exchange. The call discriminator  68  analyzes the digital voice samples from the media queue to determine whether a 2100 Hz tone, a 1100 Hz tone or V.21 modulated HDLC flags are present. If either tone or HDLC flags are detected, the voice mode services are terminated and the appropriate service for fax or modem operation is initiated. In the absence of a 2100 Hz tone, a 1100 Hz tone, or HDLC flags, the digital voice samples are coupled to the encoder system which includes a voice encoder  82 , a voice activity detector (VAD)  80 , a comfort noise estimator  81 , a DTMF detector  76 , a call progress tone detector  77  and a packetization engine  78 .  
         [0040]    Typical telephone conversations have as much as sixty percent silence or inactive content. Therefore, high bandwidth gains can be realized if digital voice samples are suppressed during these periods. A VAD  80 , operating under the packet voice exchange, is used to accomplish this function. The VAD  80  attempts to detect digital voice samples that do not contain active speech. During periods of inactive speech, the comfort noise estimator  81  couples silence identifier (SID) packets to a packetization engine  78 . The SID packets contain voice parameters that allow the reconstruction of the background noise at the far end.  
         [0041]    From a system point of view, the VAD  80  may be sensitive to the change in the NLP  72 . For example, when the NLP  72  is activated, the VAD  80  may immediately declare that voice is inactive. In that instance, the VAD  80  may have problems tracking the true background noise level. If the echo canceller  70  generates comfort noise during periods of inactive speech, it may have a different spectral characteristic from the true background noise. The VAD  80  may detect a change in noise character when the NLP  72  is activated (or deactivated) and declare the comfort noise as active speech. For these reasons, the VAD  80  should generally be disabled when the NLP  72  is activated. This is accomplished by a “NLP on” message  72   a  passed from the NLP  72  to the VAD  80 .  
         [0042]    The voice encoder  82 , operating under the packet voice exchange, can be a straight 16-bit PCM encoder or any voice encoder which supports one or more of the standards promulgated by ITU. The encoded digital voice samples are formatted into a voice packet (or packets) by the packetization engine  78 . These voice packets are formatted according to an applications protocol and sent to the host (not shown). The voice encoder  82  is invoked only when digital voice samples with speech are detected by the VAD  80 . Since the packetization interval may be a multiple of an encoding interval, both the VAD  80  and the packetization engine  78  should cooperate to decide whether or not the voice encoder  82  is invoked. For example, if the packetization interval is 10 msec and the encoder interval is 5 msec (a frame of digital voice samples is 5 ms), then a frame containing active speech should cause the subsequent frame to be placed in the 10 ms packet regardless of the VAD state during that subsequent frame. This interaction can be accomplished by the VAD  80  passing an “active” flag  80   a  to the packetization engine  78 , and the packetization engine  78  controlling whether or not the voice encoder  82  is invoked.  
         [0043]    In the described exemplary embodiment, the VAD  80  is applied after the AGC  74 . This approach provides optimal flexibility because both the VAD  80  and the voice encoder  82  are integrated into some speech compression schemes such as those promulgated in ITU Recommendations G.729 with Annex B VAD (March 1996)—Coding of Speech at 8 kbits/s Using Conjugate-Structure Algebraic-Code-Exited Linear Prediction (CS-ACELP), and G.723.1 with Annex A VAD (March 1996)—Dual Rate Coder for Multimedia Communications Transmitting at 5.3 and 6.3 kbit/s, the contents of which is hereby incorporated herein by reference as though set forth in full herein.  
         [0044]    Operating under the packet tone exchange, a DTMF detector  76  determines whether or not there is a DTMF signal present at the near end. The DTMF detector  76  also provides a pre-detection flag  76   a  which indicates whether or not it is likely that the digital voice sample might be a portion of a DTMF signal. If so, the pre-detection flag  76   a  is relayed to the packetization engine  78  instructing it to begin holding voice packets. If the DTMF detector  76  ultimately detects a DTMF signal, the voice packets are discarded, and the DTMF signal is coupled to the packetization engine  78 . Otherwise the voice packets are ultimately released from the packetization engine  78  to the host (not shown). The benefit of this method is that there is only a temporary impact on voice packet delay when a DTMF signal is pre-detected in error, and not a constant buffering delay. Whether voice packets are held while the pre-detection flag  76   a  is active could be adaptively controlled by the user application layer.  
         [0045]    Similarly, a call progress tone detector  77  also operates under the packet tone exchange to determine whether a precise signaling tone is present at the near end. Call progress tones are those which indicate what is happening to dialed phone calls. Conditions like busy line, ringing called party, bad number, and others each have distinctive tone frequencies and cadences assigned them. The call progress tone detector  77  monitors the call progress state, and forwards a call progress tone signal to the packetization engine to be packetized and transmitted across the packet based network. The call progress tone detector may also provide information regarding the near end hook status which is relevant to the signal processing tasks. If the hook status is on hook, the VAD should preferably mark all frames as inactive, DTMF detection should be disabled, and SID packets should only be transferred if they are required to keep the connection alive.  
         [0046]    The decoding system of the network VHD  62  essentially performs the inverse operation of the encoding system. The decoding system of the network VHD  62  comprises a de-packetizing engine  84 , a voice queue  86 , a DTMF queue  88 , a precision tone queue  87 , a voice synchronizer  90 , a DTMF synchronizer  102 , a precision tone synchronizer  103 , a voice decoder  96 , a VAD  98 , a comfort noise estimator  100 , a comfort noise generator  92 , a lost packet recovery engine  94 , a tone generator  104 , and a precision tone generator  105 .  
         [0047]    The de-packetizing engine  84  identifies the type of packets received from the host (i.e., voice packet, DTMF packet, call progress tone packet, SID packet), transforms them into frames which are protocol independent. The de-packetizing engine  84  then transfers the voice frames (or voice parameters in the case of SID packets) into the voice queue  86 , transfers the DTMF frames into the DTMF queue  88  and transfers the call progress tones into the call progress tone queue  87 . In this manner, the remaining tasks are, by and large, protocol independent.  
         [0048]    A jitter buffer is utilized to compensate for network impairments such as delay jitter caused by packets not arriving with the same relative timing in which they were transmitted. In addition, the jitter buffer compensates for lost packets that occur on occasion when the network is heavily congested. In the described exemplary embodiment, the jitter buffer for voice includes a voice synchronizer  90  that operates in conjunction with a voice queue  86  to provide an isochronous stream of voice frames to the voice decoder  96 .  
         [0049]    Sequence numbers embedded into the voice packets at the far end can be used to detect lost packets, packets arriving out of order, and short silence periods. The voice synchronizer  90  can analyze the sequence numbers, enabling the comfort noise generator  92  during short silence periods and performing voice frame repeats via the lost packet recovery engine  94  when voice packets are lost. SID packets can also be used as an indicator of silent periods causing the voice synchronizer  90  to enable the comfort noise generator  92 . Otherwise, during far-end active speech, the voice synchronizer  90  couples voice frames from the voice queue  86  in an isochronous stream to the voice decoder  96 . The voice decoder  96  decodes the voice frames into digital voice samples suitable for transmission on a circuit switched network, such as a 64 kb/s PCM signal for a PSTN line. The output of the voice decoder  96  (or the comfort noise generator  92  or lost packet recovery engine  94  if enabled) is written into a media queue  106  for transmission to the PXD  60 .  
         [0050]    The comfort noise generator  92  provides background noise to the near-end user during silent periods. If the protocol supports SID packets, (and these are supported for VTOA, FRF-11, and VoIP), the comfort noise estimator at the far-end encoding system should transmit SID packets. Then, the background noise can be reconstructed by the near-end comfort noise generator  92  from the voice parameters in the SID packets buffered in the voice queue  86 . However, for some protocols, namely, FRF-11, the SID packets are optional, and other far-end users may not support SID packets at all. In these systems, the voice synchronizer  90  continues to operate properly. In the absence of SID packets, the voice parameters of the background noise at the far end can be determined by running the VAD  98  at the voice decoder  96  in series with a comfort noise estimator  100 .  
         [0051]    Preferably, the voice synchronizer  90  is not dependent upon sequence numbers embedded in the voice packet. The voice synchronizer  90  can invoke a number of mechanisms to compensate for delay jitter in these systems. For example, the voice synchronizer  90  can assume that the voice queue  86  is in an underflow condition due to excess jitter and perform packet repeats by enabling the lost frame recovery engine  94 . Alternatively, the VAD  98  at the voice decoder  96  can be used to estimate whether or not the underflow of the voice queue  86  was due to the onset of a silence period or due to packet loss. In this instance, the spectrum and/or the energy of the digital voice samples can be estimated and the result  98   a  fed back to the voice synchronizer  90 . The voice synchronizer  90  can then invoke the lost packet recovery engine  94  during voice packet losses and the comfort noise generator  92  during silent periods.  
         [0052]    When DTMF packets arrive, they are de-packetized by the de-packetizing engine  84 . DTMF frames at the output of the de-packetizing engine  84  are written into the DTMF queue  88 . The DTMF synchronizer  102  couples the DTMF frames from the DTMF queue  88  to the tone generator  104 . Much like the voice synchronizer, the DTMF synchronizer  102  is employed to provide an isochronous stream of DTMF frames to the tone generator  104 . Generally speaking, when DTMF packets are being transferred, voice frames should be suppressed. To some extent, this is protocol dependent. However, the capability to flush the voice queue  86  to ensure that the voice frames do not interfere with DTMF generation is desirable. Essentially, old voice frames which may be queued are discarded when DTMF packets arrive. This will ensure that there is a significant gap before DTMF tones are generated. This is achieved by a “tone present” message  88   a  passed between the DTMF queue and the voice synchronizer  90 .  
         [0053]    The tone generator  104  converts the DTMF signals into a DTMF tone suitable for a standard digital or analog telephone. The tone generator  104  overwrites the media queue  106  to prevent leakage through the voice path and to ensure that the DTMF tones are not too noisy.  
         [0054]    There is also a possibility that DTMF tone may be fed back as an echo into the DTMF detector  76 . To prevent false detection, the DTMF detector  76  can be disabled entirely (or disabled only for the digit being generated) during DTMF tone generation. This is achieved by a “tone on” message  104   a  passed between the tone generator  104  and the DTMF detector  76 . Alternatively, the NLP  72  can be activated while generating DTMF tones.  
         [0055]    When call progress tone packets arrive, they are de-packetized by the de-packetizing engine  84 . Call progress tone frames at the output of the de-packetizing engine  84  are written into the call progress tone queue  87 . The call progress tone synchronizer  103  couples the call progress tone frames from the call progress tone queue  87  to a call progress tone generator  105 . Much like the DTMF synchronizer, the call progress tone synchronizer  103  is employed to provide an isochronous stream of call progress tone frames to the call progress tone generator  105 . And much like the DTMF tone generator, when call progress tone packets are being transferred, voice frames should be suppressed. To some extent, this is protocol dependent. However, the capability to flush the voice queue  86  to ensure that the voice frames do not interfere with call progress tone generation is desirable. Essentially, old voice frames which may be queued are discarded when call progress tone packets arrive to ensure that there is a significant inter-digit gap before call progress tones are generated. This is achieved by a “tone present” message  87   a  passed between the call progress tone queue  87  and the voice synchronizer  90 .  
         [0056]    The call progress tone generator  105  converts the call progress tone signals into a call progress tone suitable for a standard digital or analog telephone. The call progress tone generator  105  overwrites the media queue  106  to prevent leakage through the voice path and to ensure that the call progress tones are not too noisy.  
         [0057]    The outgoing PCM signal in the media queue  106  is coupled to the PXD  60  via the switchboard  32 ′. The outgoing PCM signal is coupled to an amplifier  108  before being outputted on the PCM output line  60   b.    
         [0058]    An exemplary embodiment according to the present invention is shown in FIG. 4. As shown in FIG. 4, the switchboard  400  is responsible for establishing connections between input ports and output ports and when necessary, combining input data streams to form output data streams. In addition, the switchboard  400  may provide for the upsampling of data received from narrowband sources to be sent to wideband destinations, and for the downsampling of data received from wideband sources to be sent to narrowband destinations. It may also pass data unchanged. In order to combine input data streams sampled at different sampling rates, switchboard  400  may resample such input data streams to a common sampling rate. Similarly, in order to provide one or more output data streams each with a sampling rate appropriate to its designated destination, switchboard  400  may resample a data stream to the sampling rate that may be required by a specific device or service.  
         [0059]    As shown in the exemplary embodiment illustrated in FIG. 4, the switchboard module  400  comprises combiner  450 , with converters  430  and  440  at its inputs, and converters  460  and  470  at its outputs. Converters  430 ,  440 ,  460  and  470  may be designed to upsample, downsample, or to pass unchanged the sampled data received at their inputs. As illustrated in FIG. 4, narrowband data  410  is provided to input converter  430 , which may upsample the narrowband data to the sampling rate of the wideband data  420 . It then provides the upsampled data to combiner  450 . At the same time, wideband data  420  is provided to converter  440  which may pass the wideband data unchanged to combiner  450 . The combiner is responsible for combining the input data streams where combining may include but is not limited to, for example, adding inputs, subtracting inputs, passing inputs unchanged, or any combination of these operations. Although FIG. 4 shows combiner  450  as having two inputs and two outputs, one of skill in the art will recognize that embodiments with greater or fewer inputs and outputs do not depart from the spirit of the invention. The output of combiner  450  may be provided to one or more converters such as converter  460  and converter  470 . As shown in the illustration, converter  460  may be used to downsample the combined data from combiner  450  to form narrowband data  480 . In addition, converter  470  may be used to pass the combined data from combiner  450  unchanged to form wideband data  490 .  
         [0060]    The switchboard understands and operates on source and destination ports. As shown in the illustration, switchboard  400  may have a number of input ports for streams of narrowband data  410  and wideband data  420 , and may have a number of output ports for streams of narrowband data  480  and wideband data  490 . A port may be a PXD or a VHD, and in a system utilizing multiple sampling rates a port&#39;s identity may indicate its sampling rate. To embed sample rate information into the switchboard ports, the PXD and VHD structures may contain a switchboard port structure that not only provides a pointer to the data buffer, but also sample rate information in either the number of samples or an enumeration type.  
         [0061]    The switchboard ports are used in a switchboard connection list to manage input and output media ports. In an exemplary case, a switchboard port type, SWB_Port, may be the following:  
         [0062]    typedef MediaPort*SWB_Port;  
         [0063]    The switchboard port may be a pointer to a media port structure, which in an exemplary case may be defined as:  
                                                                           typedef struct           {                SINT16   *bufp;                MediaRateShift sampleRateShift;           } MediaPort;                      
 
         [0064]    The media port structure may contain a data buffer pointer and the buffer&#39;s sample rate information, and the sample rate information may be stored as a left shift value. The switchboard may operate on a fixed block rate in milliseconds. The sample block size depends on the sampling rate, and the left shift value provides an efficient means to convert block rate (in sampling rate frequency) to block size (in samples). In an exemplary case,  
                                                                       typedef enum           {                Media8kHzSampleShift = 0;           Media16kHzSampleShift = 1;                } MediaRateShift;                      
 
         [0065]    Referring now to FIG. 5, there is illustrated a signal flow diagram for a switchboard connection between a wideband PXD  530 , a narrowband VHD  520 , and a wideband VHD  510 , in accordance with an illustrative embodiment of the present invention. The connections depicted in FIG. 5 as switchboard  500  are implemented by the switchboard  210  functionality illustrated in FIG. 2, also shown in FIG. 3 as switchboard  32 ′ and in FIG. 4 as switchboard  400 . Such connections may exist, for example, when the device is a party to a conference call. Wideband VHD  510  is associated with a wideband destination device, while narrowband VHD  520  is associated with a narrowband destination device. In the illustrative embodiment, narrowband VHD  520  transmits and receives narrowband data, N, while wideband VHD  510  and wideband PXD  530  transmit and receive wideband data, W.  
         [0066]    On the egress side, converter  540  upsamples the narrowband data from narrowband VHD  520  and provides the upsampled data to summer  580  in combiner  570 . Converter  550  passes the wideband data from wideband VHD  510  unchanged to summer  580 . The summed output of summer  580  is then provided to converter  560 , which passes the combined data unchanged to wideband PXD  530 . On the ingress side, converter  565  of switchboard  500  passes the wideband data from PXD  530  unchanged to summer  575 . Wideband data from wideband VHD  510  is passed unchanged by converter  550  to summer  575 , where it is added to the wideband data from PXD  530 . The resulting wideband data is then downsampled by converter  545  and provided to narrowband VHD  520 . The unmodified wideband data from PXD  530  is also provided to summer  585 , where it is summed with the upsampled narrowband data of narrowband VHD  520  provided by converter  540 . The resulting sum is then passed unchanged by converter  555  to wideband VHD  510 . Although this exemplary embodiment shows switchboard  500  providing service to one wideband VHD, one narrowband VHD, and one wideband PXD, this does not represent a limitation of the present invention. The spirit of the present invention extends to embodiments with a greater number of VHDs and PXDs, and a greater variety of sampling rates.  
         [0067]    [0067]FIG. 6 shows a further embodiment according to the present invention, in which is illustrated a signal flow diagram for a switchboard connection between wideband PXD  630 , wideband VHD  610 , and narrowband VHD  620 . The connections depicted in FIG. 6 as switchboard  600  may also be implemented by the switchboard functionality  210  illustrated in FIG. 2, switchboard  32 ′ of FIG. 3, or switchboard  400  of FIG. 4. The connections shown in FIG. 6 may be created when the user terminal associated with wideband PXD  630  establishes a conference call with the wideband communication devices associated with wideband VHD  610  and narrowband VHD  620 . Wideband VHD  610  and wideband PXD  630  transmit and receive wideband data, W. Narrowband VHD  620  transmits and receives narrowband data, N.  
         [0068]    As illustrated in FIG. 6, on the egress side converter  640  provides upsampled narrowband data from narrowband VHD  620  to summer  680  in combiner  670 . Converter  650  passes the wideband data from wideband VHD  610  unchanged to summer  680 . The summed output of summer  680  is then provided to converter  660 , which passes the combined data unchanged to wideband PXD  630 . On the ingress side, converter  665  of switchboard  600  passes the wideband data from PXD  630  unchanged to converter  645  and converter  655 . Converter  645  downsamples the wideband data from wideband PXD  630  and provides narrowband data to narrowband VHD  620 , while converter  655  passes the wideband data from wideband PXD  630  unchanged to wideband VHD  610 . Again, although the embodiment illustrated shows switchboard  600  providing service to one wideband VHD, one narrowband VHD, and one wideband PXD, this does not represent a limitation of the present invention. The spirit of the present invention extends to embodiments with a greater number of VHDs and PXDs, and a greater variety of sampling rates.  
         [0069]    Referring now to FIG. 7, there is illustrated a block diagram of an exemplary terminal  758 , corresponding to terminal  110  as depicted in FIG. 1, in which an embodiment of the present invention may be practiced. A processor  760  is interconnected via system bus  762  to random access memory (RAM)  764 , read-only memory (ROM)  766 , an input/output adapter  768 , a user interface adapter  772 , a communications adapter  784 , and a display adapter  786 . The input/output adapter  768  connects peripheral devices such as hard disc drive  740 , floppy disc drives  741  for reading removable floppy discs  742 , and optical disc drives  743  for reading removable optical disc  744 . The user interface adapter  772  connects devices such as a keyboard  774 , a speaker  778 , and microphone  782  to the bus  762 . The microphone  782  generates audio signals that are digitized by the user interface adapter  772 . The speaker  778  receives audio signals that are converted from digital samples to analog signals by the user interface adapter  772 . The display adapter  786  connects a display  788  to the bus  762 . Embodiments of the present invention may also be practiced in other types of terminals as well, including but not limited to, a telephone without a hard disk drive  740 , a floppy disk drive  741 , or optical disk drive  743 , including those in which the program instructions may be stored in ROM  766 , or downloaded over communications adapter  784  and stored in RAM  764 . An embodiment may also be practiced in, for example, a portable hand-held terminal with little or no display capability, in a consumer home entertainment system, or even in a multi-media game system console.  
         [0070]    An embodiment of the present invention can be implemented as sets of instructions resident in the RAM  764  or ROM  766  of one or more terminals  758  configured generally as described in FIG. 7. Until required by the terminal  758 , the set of instructions may be stored in another memory readable by the processor  760 , such as hard disc drive  740 , floppy disc  742 , or optical disc  744 . One skilled in the art would appreciate that the physical storage of the sets of instructions physically changes the medium upon which it is stored electrically, magnetically, or chemically so that the medium carries information readable by a processor.  
         [0071]    Accordingly, the present invention may be realized in hardware, software, or a combination of hardware and software. The present invention may be realized in a centralized fashion in one computer system, or in a distributed fashion where different elements are spread across several interconnected computer systems. Any kind of computer system or other apparatus adapted for carrying out the methods described herein is suited. A typical combination of hardware and software may be a general-purpose computer system with a computer program that, when being loaded and executed, controls the computer system such that it carries out the methods described herein.  
         [0072]    The present invention also may be embedded in a computer program product, which comprises all the features enabling the implementation of the methods described herein, and which when loaded in a computer system is able to carry out these methods. Computer program in the present context means any expression, in any language, code or notation, of a set of instructions intended to cause a system having an information processing capability to perform a particular function either directly or after either or both of the following: a) conversion to another language, code or notation; b) reproduction in a different material form.  
         [0073]    Notwithstanding, the invention and its inventive arrangements disclosed herein may be embodied in other forms without departing from the spirit or essential attributes thereof. Accordingly, reference should be made to the following claims, rather than to the foregoing specification, as indicating the scope of the invention. In this regard, the description above is intended by way of example only and is not intended to limit the present invention in any way, except as set forth in the following claims.  
         [0074]    While the present invention has been described with reference to certain embodiments, it will be understood by those skilled in the art that various changes may be made and equivalents may be substituted without departing from the scope of the present invention. In addition, many modifications may be made to adapt a particular situation or material to the teachings of the present invention without departing from its scope. Therefore, it is intended that the present invention not be limited to the particular embodiment disclosed, but that the present invention will include all embodiments falling within the scope of the appended claims.