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 low-band data and high-band data, and narrowband communication interfaces may transfer low-band data. Low-band data may be combined and sent to a narrowband communications interface or a wideband communications interface. High-band data may be combined and sent to a wideband communications interface. The low-band data may represent audio signals below a predetermined frequency, while the high-band data may represent audio signals above the predetermined frequency. The predetermined frequency may be, for example, approximately 4 kHz. The spectral mask of the low-band data may meet the spectral mask of G.712. 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 
   The present application is a continuation application of co-pending U.S. patent application Ser. No. 10/313,672, filed Dec. 6, 2002, which issued as U.S. Pat. No. 7,333,475 on Feb. 19, 2008, and which makes reference to, claims benefit of, and claims priority to on provisional application Ser. No. 60/414,493, “Switchboard for Multiple Data Rate Communication System”, filed Sep. 27, 2002, the complete subject matter of each of which is hereby incorporated herein by reference in its entirety. With respect to the present application, Applicant hereby rescinds any disclaimer of claim scope made in the parent application or any predecessor or related application. The Examiner is advised that any previous disclaimer of claim scope, if any, and the alleged prior art that it was made to allegedly avoid, may need to be revisited. Nor should a disclaimer of claim scope, if any, in the present application be read back into any predecessor or related application. 
   This application is also related to the following co-pending applications, each of which are herein incorporated by reference: 
   
     
       
             
             
             
             
           
         
             
                 
             
             
               Ser. No. 
               Title 
               Filed 
               Inventors 
             
             
                 
             
           
           
             
               60/414,059 
               Multiple Data Rate Communication 
               Sep. 27, 
               LeBlanc 
             
             
                 
               System 
               2002 
               Houghton 
             
             
                 
                 
                 
               Cheung 
             
             
               60/414,460 
               Dual Rate Single Band 
               Sep. 27, 
               LeBlanc 
             
             
                 
               Communication System 
               2002 
               Houghton 
             
             
                 
                 
                 
               Cheung 
             
             
               60/414,491 
               Splitter and Combiner for Multiple 
               Sep. 27, 
               LeBlanc 
             
             
                 
               Data Rate Communication System 
               2002 
               Houghton 
             
             
                 
                 
                 
               Cheung 
             
             
               60/414,492 
               Method and System for an Adaptive 
               Sep. 27, 
               LeBlanc 
             
             
                 
               Multimode Media Queue 
               2002 
               Houghton 
             
             
                 
                 
                 
               Cheung 
             
             
                 
             
           
        
       
     
   

   FEDERALLY SPONSORED RESEARCH OR DEVELOPMENT 
   [Not Applicable] 
   MICROFICHE/COPYRIGHT REFERENCE 
   [Not Applicable] 
   BACKGROUND OF THE INVENTION 
   Traditional voice telephony products are band-limited to 4 kHz bandwidth with 8 kHz sampling. These products 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 doubling the sampling rate are significant. Doubling the sampling rate not only requires doubling the processing cycles, but nearly doubling the memory used to store the data. In addition, software supporting wideband capabilities must not preclude support for legacy 4 kHz band-limited functionality. 
   Doubling memory and processor cycles requirements is expensive because the memory and processing power footprints of digital signal processors (DSPs) are generally small. Implementing wideband support thus requires creativeness to optimize both memory and cycles. 
   Additionally, much of the software providing various functions and services, such as echo cancellation, dual-tone multi-frequency (DTMF) detection and generation, and call discrimination (between voice and facsimile transmission, for example), are written for only narrowband signals. Either new software must be written for wideband signals, or the wideband signal must be down-sampled. Where the software is modified, the software should also be capable of integration with preexisting narrowband devices. Providing software for operation with both narrowband and wideband devices is complex and costly. 
   Accordingly, there is a need for switchboard functionality that manages a device&#39;s connections with both narrowband devices and wideband devices. 
   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 
   Seamless wideband support is afforded by utilizing band-split data streams. In an illustrative embodiment of the present invention, the 8 kHz bandwidth is divided into a low band, with approximately 0-4 kHz bandwidth, and a high band, with approximately 4-8 kHz bandwidth. Narrowband functions and services operate on the low band, while wideband functions and services operate on both low and high bands. Switchboard functionality provides the low-band and high-band connections between inputs and outputs for both narrowband and wideband data streams, and, where necessary, sums input data streams to form output data streams. 
   An embodiment according to the present invention may include at least one high-band port for receiving high-band data from at least one wideband communications interface and at least one low-band port for receiving low-band data from at least one narrowband communications interface and the at least one wideband communications interface. It may include a mixer for combining the low-band data received at the at least one low-band port and for combining high-band data received at the at least one high-band port. In addition, it may comprise at least one low-band port for transmitting the combined low-band data to one of the at least one wideband communications interface. The mixer may add the low-band data to produce summed low-band data, and add the high-band data to produce summed high-band data. Low-band data of the at least one wideband communications interface may represent spectral components less than a predetermined frequency, and high-band data of the at least one wideband communications interface may represent spectral components greater than the predetermined frequency, where the predetermined frequency may be, for example, approximately 4 kHz. In addition, the spectral mask of the low-band data of the at least one wideband communications interface may meet the spectral mask of G.712. 
   A method for operating a switchboard device according to one embodiment of the present invention is also disclosed, the method comprising receiving wideband data from at least one wideband communications interface, the received wideband data comprising low-band data and high-band data; receiving narrowband data from at least one narrowband communications interface, the received narrowband data comprising low-band data; combining low-band data from at least one of the at least one wideband communications interface and low-band data from at least one of the at least one narrowband communications interface to produce combined low-band data to be sent to at least one of a designated narrowband communications interface or designated wideband communications interface; combining high-band data from at least one of the at least one wideband communications interface to produce combined high-band data to be sent to a designated wideband communications interface; sending the combined low-band data to at least one of the designated narrowband communications interface or designated wideband communications interface, and sending the combined high-band data to the designated wideband communications interface. The act of combining may comprise adding the low-band data to produce summed low-band data, and adding the high-band data to produce summed high-band data. In an exemplary embodiment, the low-band data of the at least one wideband communications interface may represent the spectral components less than a predetermined frequency, and the high-band data of the at least one wideband communications interface may represent the spectral components greater than the predetermined frequency, where the predetermined frequency may be, for example, approximately 4 kHz. In addition, the spectral mask of the low-band data may meet the spectral mask of G.712. 
   Another aspect of the present invention relates to a method for transmitting audible signals between a first terminal and a second terminal. The method may comprise receiving high-band data and low-band data representing the audible signals from an interface of the first terminal, wherein the high-band data and the low-band data are received separately. The low-band data may be transmitted to an interface of the second terminal. The method may also include transmitting the high-band data to the interface of the second terminal, if the second terminal is a wide band device. In addition, the method may further comprise receiving low-band data from the second terminal, and receiving high-band data from the second terminal if the second terminal is a wideband terminal. In such an embodiment, the low-band data may represent spectral components less than a predetermined frequency, and the high-band data may represent spectral components greater than the predetermined frequency. The predetermined frequency may be, for example, approximately 4 kHz. The spectral mask of the low-band data may also meet the spectral mask of G.712. 
   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. 
   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 
       FIG. 1  is a block diagram of an exemplary communication system wherein the present invention can be practiced. 
       FIG. 2  is a data flow diagram for a split-band architecture in accordance with an embodiment of the present invention. 
       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. 
       FIG. 4  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. 
       FIG. 5  is a data flow diagram for a switchboard connection between a wideband PXD and two wideband VHDs, in accordance with an embodiment of the present invention. 
       FIG. 6  is a block diagram of an exemplary terminal in which aspects of the present invention may be practiced. 
   

   DETAILED DESCRIPTION OF THE INVENTION 
   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. 
   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. 
   The communication system  100  utilizes band-split data streams. In one embodiment, an 8 kHz bandwidth is divided into two bands: a G.712 compliant low band and a high band. The low band is stored as 8 kHz sampled data, while the high band is stored as 16 kHz sampled data. Both bands are not stored symmetrically as 8 kHz sampled data because the 8 kHz bandwidth is not split symmetrically down the center. This design incurs a memory cost in return for voice quality and G.712 compliance. In an alternative embodiment where aliasing may be ignored, the 8 kHz bandwidth may be split symmetrically with both low and high bands stored as 8 kHz sampled data. This alternative avoids the increased memory requirement but at the cost of voice quality. Both symmetric and asymmetric split-band architectures are similar in implementation except for the sampling rate of the media streams. In some designs, one may be more desirable. In other designs, the reverse may be true. The optimal choice depends on an acceptable memory versus performance trade-off. 
   The split-band approach enables straightforward support for narrowband and wideband services because narrowband services are incognizant of the wideband support. Narrowband services operate on the 8 kHz-sampled stream of data (i.e., the low-band data). Generally, only wideband services understand and operate on both bands. 
   Referring now to  FIG. 2 , there is illustrated a signal flow diagram of a split-band architecture  200  in accordance with an embodiment of the present invention. The split-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 . 
   The PXD  215  represents an interface for receiving the input signal from the user and performs various functions, such as echo cancellation. The order of the PXD  215  functions maintains continuity and consistency of the data flow. 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. For wideband operation, the band splitter/combiner PXD  215  function may be located as follows. On the switchboard  210  side of this PXD  215  function is split-band data. On the other side is single-band data. PXD  215  functions that operate on single-band data, like side-tone or high-pass PXD  215  functions, are ordered below the band splitter/combiner PXD  215  function. Other PXD  215  functions that operate on split-band data are ordered above it. 
   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 (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 . The switchboard  210  associates the VHD  205  and the PXD  215  in a manner that will be described below. 
   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 useful data at its high and low-band sample buffer interfaces and can include both narrowband and wideband services and functions. A narrowband VHD  205  or PXD  215  has useful data at its low-band sample buffer interface and no data at its high-band sample buffer interface. The switchboard interfaces with narrowband and wideband VHDs  205  and PXDs  215  through their high and low-band sample buffer interfaces. The switchboard  210  is incognizant of the wideband or narrowband nature of the device drivers. The switchboard  210  reads and writes data through the sample buffer interfaces. The high and low-band sample buffer interfaces may provide data at any arbitrary sampling rate. In an embodiment of the present invention, the low-band sample buffer interface provides data sampled at 8 kHz and the high-band sample buffer interface provides data sampled at 16 kHz. Additionally, a VHD  205  can be dynamically changed between wideband and narrowband and vice versa. 
   The VHD  205  and PXD  215  driver structures may include sample rate information to identify the sampling rates of the high and low-band 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. 
   The split-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 mode. Additionally, although the illustrated ingress path and egress path are both operating in the wideband support mode, 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 , band splitter  230 , echo canceller  235 , switchboard  210 , and services including but not limited to DTMF detector  240 , CDIS  245 , and packet voice engine (PVE)  255  comprising a combiner  250  and an encoder algorithm  260 . 
   In a wideband device, the decimator  225  receives the user inputs and provides 16 kHz sampled data for an 8 kHz band-limited signal. The 16 kHz sampled data is received by the band splitter  230 . The band splitter  230  splits the 8 kHz bandwidth into low-band data (L) and high-band data (H). The low-band data, L, and high-band data, H, are transmitted through the echo canceller  235 , and switchboard  210  to the VHD  205  associated with the destination terminal  110 . The band splitter  230  can comprise, for example, the band splitter described in provisional patent application Ser. No. 60/414,491, “Splitter and Combiner for Multiple Data Rate Communication System”, which is incorporated herein by reference in its entirety. 
   The VHD  205  receives the low-band data, L, and high-band data, H. In some cases, the DTMF detector  240  may be designed for operation on only narrowband digitized samples, and only the low-band data is passed to DTMF detector  240 . Similarly, where CDIS  245  is designed for operation on only narrowband digitized samples, only the low-band data is provided to CDIS  245 , which distinguishes a voice call from a facsimile transmission. The low-band data, L, and high-band data, H, are combined at a combiner  250  in PVE  255 . The combiner  250  can comprise, for example, the combiner described in provisional patent application Ser. No. 60/414,491, “Splitter and Combiner for Multiple Data Rate Communication System”, which is incorporated herein by reference in its entirety. 
   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. The raw samples include both low and high-band data. However, to save memory only low-band data is forwarded when the VHD  205  is operating in narrowband mode. Both low and high-band data are combined and forwarded when operating in wideband mode. 
   At PVE  255 , encoder  260  packetizes the combined signal for transmission over the packet network  105 . The encoder  260  can comprise, for example, the BroadVoice 32 Encoder made by Broadcom, Inc. 
   The egress path comprises decoder  263 , band splitter  264 , CDIS  266 , DTMF generator  269 , switchboard  210 , echo canceller  235 , band combiner  272 , 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. The 16 kHz sampled data is provided to band splitter  264  which separates low-band data, L 1 , from high-band data, H 1 . Again, in one embodiment, where CDIS  266  and DTMF generator  269  require narrowband digitized samples, only the low-band data is used by CDIS  266  and the DTMF generator  269 . 
   The DTMF generator  269  generates DTMF tones if detected from the sending terminal  110 . These tones are written to the low-band data, L 1 . The low-band data, L 1 , and high-band data, H 1 , are received by the switchboard  210 . The switchboard  210  provides the low-band data, L 1 , and high-band data, H 1 , to the PXD  215 . The low-band data, L 1 , and high-band data, H 1 , are passed through the echo canceller  235  and provided to the band combiner  272  which combines the low-band data, L 1 , and high-band data, H 1 . The combined low-band data, L 1 , and high-band data, H 1 , are provided to interpolator  220 . The interpolator  220  provides 16 kHz sampled data. 
   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. 
   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. 
   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. 
   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. 
   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 . 
   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. 
   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 . 
   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. 
   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 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. 
   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. 
   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. 
   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 . 
   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. 
   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 . 
   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 . 
   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 . 
   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. 
   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 . 
   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. 
   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. 
   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 . 
   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. 
   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.    
   Referring again to  FIG. 2 , the switchboard  210  is responsible for managing connections between inputs and outputs, and when necessary, summing input data streams to form output data streams. In the split-band model, wideband switchboard connections comprise both low and high-band switchboard connections. Switchboard connection routines create additional switchboard connections for the high-band data, and the high-band data streams are additional ports for the switchboard to sum. 
   The switchboard module comprises a mixer and a connection control. The mixer is responsible for summing input data streams and writing the results to an output data stream. The connection control is responsible for creating and destroying connections between input and output data streams. A switchboard connection determines how data flows. 
   The switchboard understands and operates on source and destination ports. A port may be a PXD or a VHD, and in an asymmetric rate system 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. 
   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, is changed to the following:
 
typedef MediaPort*SWB_Port;
 
   The switchboard port is a pointer to a media port structure, which in an exemplary case can be defined as: 
   
     
       
             
             
           
         
             
                 
                 
             
           
           
             
                 
               typedef struct 
             
             
                 
               { 
             
             
                 
               SINT16   *bufp; 
             
             
                 
               MediaRateShift sampleRateShift; 
             
             
                 
               } MediaPort; 
             
             
                 
                 
             
           
        
       
     
   
   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; 
             
             
                 
                 
             
           
        
       
     
   
   Referring now to  FIG. 4 , there is illustrated a signal flow diagram for a switchboard connection between a wideband PXD  430  and two VHDs, one narrowband, VHD  420 , and one wideband, VHD  410 , in accordance with an illustrative embodiment of the present invention. The connections depicted in  FIG. 4  as switchboard  400  are implemented by the switchboard  210  functionality depicted in  FIG. 2 . Such connections may exist, for example, when the device is a party to a conference call. Wideband VHD  410  is associated with a wideband destination device, while narrowband VHD  420  is associated with a narrowband destination device. Narrowband VHD  420  transmits only low-band data, L, while wideband VHD  410  and wideband PXD  430  transmit both low-band data, L, and high-band data, H. 
   On the ingress side, the switchboard  400  provides the low-band data from wideband PXD  430  to both VHDs  410  and  420 . However, the switchboard  400  provides the high-band data, H, only to the wideband VHD  410  because the narrowband VHD  420  does not support wideband signaling. On the egress side, the switchboard  400  receives the low-band data, L, from the VHDs  410  and  420 . The switchboard  400  sums the low-band data, L, from VHD  410  and VHD  420 , and provides the summed low-band data to the wideband PXD  430 , sums the low-band data from VHD  410  and PXD  430 , and provides the summed low-band data to VHD  420 , and sums the low-band data from VHD  420  and PXD  430 , and provides the summed low-band data to VHD  410 . The switchboard  400  receives the high-band data, H, from wideband VHD  410  and provides the high-band data, H, only to the wideband PXD  430 . It also receives the high-band data, H, from wideband PXD  430  and provides the high-band data, H, only to wideband VHD  410 . 
     FIG. 5  shows a further embodiment according to the present invention, in which is illustrated a signal flow diagram for a switchboard connection between a wideband PXD  530  and two wideband VHDs,  510  and  520 . The connections depicted in  FIG. 5  as switchboard  500  may be implemented by the switchboard functionality  210  depicted in  FIG. 2 . The connections shown in  FIG. 5  may be created when the user terminal associated with wideband PXD  530  establishes a conference call with the wideband communication devices associated with wideband VHD  510  and wideband VHD  520 . Wideband VHDs  510  and  520 , and wideband PXD  530  transmit both low-band data, L, and high-band data, H. 
   As illustrated in  FIG. 5 , switchboard  500  sums the low-band data from wideband PXD  530  and wideband VHD  520  and provides the resulting low-band data to wideband VHD  510 . It also sums the high-band data from wideband PXD  530  and wideband VHD  520 , and provides the resulting high-band data to wideband VHD  510 . In a similar fashion, switchboard  500  sums the low-band data from wideband VHDs  510  and  520 , and provides the resulting low-band data to wideband PXD  530 . It also sums the high-band data from VHD  510  and VHD  520 , and sends the resulting high-band data to wideband PXD  530 . In addition, the switchboard  500  sums the low-band data from wideband PXD  530  and wideband VHD  510 , and provides the resulting low-band data to wideband VHD  520 . It also sums the high-band data from PXD  530  and VHD  510 , and sends the resulting high-band data to wideband VHD  520 . 
   Referring now to  FIG. 6 , there is illustrated a block diagram of an exemplary terminal  658 , corresponding to terminal  110  as depicted in  FIG. 1 . A processor  660  is interconnected via system bus  662  to random access memory (RAM)  664 , read only memory (ROM)  666 , an input/output adapter  668 , a user interface adapter  672 , a communications adapter  684 , and a display adapter  686 . The input/output adapter  668  connects peripheral devices such as hard disc drive  640 , floppy disc drives  641  for reading removable floppy discs  642 , and optical disc drives  643  for reading removable optical disc  644 . The user interface adapter  672  connects devices such as a keyboard  674 , a speaker  678 , and microphone  682  to the bus  662 . The microphone  682  generates audio signals which are digitized by the user interface adapter  672 . The speaker  678  receives audio signals which are converted from digital samples to analog signals by the user interface adapter  672 . The display adapter  686  connects a display  688  to the bus  662 . 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  640 , a floppy disk drive  641 , nor optical disk drive  643 , in which the program instructions may be stored in ROM  666 , or downloaded over communications adapter  684  and stored in RAM  664 . 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. 
   An embodiment of the present invention can be implemented as sets of instructions resident in the RAM  664  or ROM  666  of one or more terminals  658  configured generally as described in  FIG. 6 . Until required by the terminal  658 , the set of instructions may be stored in another memory readable by the processor  660 , such as hard disc drive  640 , floppy disc  642 , or optical disc  644 . 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. 
   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. 
   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. 
   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. 
   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.