Abstract:
In a packet communication network, to compensate for rate mismatches between transmitting and receiving devices, an apparatus and method provides for adjusting the playback sampling rate and for monitoring the buffer. The receiving device will monitor its buffer; relevant buffer data can comprise whether the buffer is approaching capacity or approaching depletion and the speed in which the buffer is approaching capacity of approaching depletion. The receiving device will then trigger an adjustment to the playback sampling rate to attune the rates of the transmitting and receiving devices or to compensate for jitters from any number of network complications. The receiving device may also store the buffer data for later action, for example, to formulate specific adjustment procedures or to compile specific conference profiles. The present invention may function alone or function in conjunction with other known methods in the art.

Description:
RELATED APPLICATION 
   The present application references and incorporates herein a related U.S. application entitled Method and System for Dynamically Adjusting Video Bit Rates, filed on Nov. 13, 2001, and assigned Ser. No. 10/008,100. 
   FIELD OF THE INVENTION 
   The present invention relates to data transmission of streaming data. The invention particularly provides a method and system for controlling the playback rate of real-time audio data received over a network. 
   BACKGROUND OF THE INVENTION 
   A telephony application enables transmission of real-time audio data over a packet-based network. To name a few, applications include voice over private Internet Protocol (IP) backbones, Internet or intranets, messaging, and streaming audio play, such as music or announcements. The most popular application is IP Telephony, that is, any telephony application that enables voice transmission via Internet Protocol (VoIP). This technology allows a device to transmit voice as just another form of data over the same IP network. For the purposes of this patent application, we also consider the audio transmissions in a video conference to be a form of IP Telephony. IP Telephony comprises numerous applications that support connections such as PC-to-PC connections, PC-to-phone connections, and phone-to-phone connections. 
   The crux of VoIP lies in converting an analog signal to digital IP packets (A/D), transmitting the IP packets over a network, and converting the IP packets back into a playable analog signal (D/A). At the transmitting end, a device generally digitizes the signal at a specific sampling rate, encodes that digital data into frames, converts the frames into IP packets, and transmits the IP packets over an IP network. At the receiving end, a device typically receives the packets, extracts the digital data from the packets, and converts the digital data into analog output at the same sampling rate as that used by the transmitter. 
   VoIP has both advantages and disadvantages when compared with traditional (e.g. PSTN) digital telephony systems. As for the advantages, the technology operates on the existing infrastructure, utilizing PSTN switches, customer premises equipment, and Internet connections. IP Telephony also improves the efficiency of bandwidth use for real-time voice transmission. And of particular interest, IP Telephony offers a new line of applications, combining real-time voice communication and data processing. 
   Regarding the disadvantages, VoIP and packet communication introduce issues of “reassembling” the packets, that is, playing the packets as if the packets were the original, continuous analog signal. Playing the IP packets appears simplistic; the receiving station could, upon receiving IP packets, convert the IP packets to an analog signal and immediately play the analog signal. Playing the packets upon reception, however, would resemble an accurate reconstruction only if the sender transmits the packets at uniform intervals, the packets transfer through the network without inconsistent delay, and the packets successfully reach the receiver. Each of these premises are often false. At times, starvation periods exist where the receiver has no packet to play, and at other times, burst periods overwhelm the receiver with too many packets to play. This non-uniformity is generally referred to as “jitter.” 
   Accordingly, to account for this “jitter,” most applications employ a buffer. A buffer loads incoming packets or frames to allow the receiver to retrieve and play the packets or frames at a uniform rate. The number of frames or packets in the buffer can fluctuate up and down with the network jitter. As long as the buffer never empties or overflows, the receiver will be able to play at its uniform rate, without audio disturbances. This buffering technique exists in most real-time media systems that receive audio or video from a network. 
   The buffer, however, cannot account for inconsistent sender transmission rate and receiver playback rate (or buffer output rate). In traditional digital telephony systems, a master clock synchronizes end points to ensure that the D/A and A/D converters at both ends operate at identical sampling rates. Identical sampling rates ensure that, on average, the data transmission rate will equal the receiver output rate. In contrast, in IP Telephony, no master clock exists to synchronize the sampling rates. In VoIP systems, it is common to employ personal computers, or similar hardware, with sound cards that have inaccurate sampling rates. Sound cards set at 8000 samples per second, for example, can actually have sampling rates that vary between 7948 and 8130 samples per second. For PC-based VoIP and videoconferencing systems, the clocks are not necessarily accurate enough to guarantee identical sampling rates. As a result, a receiver that operates at a slightly higher sampling rate will playback data faster than the sender transmits the data, ultimately emptying the buffer and requiring the receiver to play periods of “silence.” A receiver that operates at a slightly lower sampling rate will play data slower than the sender transmits the data. With the receiver steadily falling behind, the data will ultimately overwhelm the buffer, requiring the receiver to “discard” periods of playback data (frames or packets). Increasing the buffer size fails to remedy the problem because the concomitant delay between transmission and actual playback becomes unacceptable for real-time audio transmission. 
   A common solution is to insert “silent” periods when the buffer approaches depletion and to remove “silent” periods when the buffer approaches capacity. This solution has numerous flaws. From a hardware perspective, problems include detecting periods of silence and handling the requisite additional processing. From a user perspective, any inserting or deleting “silent” periods degrades the conversation, as no true periods of silence exist in VoIP applications. Therein lies the rub: the inherent difference between the human eye and ear. While a video frame may be left on display a split second longer than the next frame without human detection, a tone cannot simply be left playing. Accordingly, the prior art focuses on inserting sound periods or removing sound periods, seemingly the only suitable way to manipulate the flow rate of audio data in a real-time environment. See, e.g., U.S. Pat. No. 6,658,027 (“Jitter Buffer Management”). 
   The forgoing illustrates that during real-time audio transmission over a network a need exists to continually monitor the buffer and adjust the playback rate of a receiver to account for variances in sampling rates among transmitters and receivers. 
   SUMMARY OF INVENTION 
   The present invention provides a method and system for data transmission of streaming data. More specifically, the invention provides a method and system for controlling a receiver&#39;s playback sampling rate when playing data that was sent over a network. In an exemplary embodiment, a transmitter converts analog data to digital data at a transmitter&#39;s sampling rate, places the data in packets, and sends the packets over a packet-based network. The receiver receives the packets, forwards the packets to a buffer, monitors the buffer, and converts the packets for playback at the receiver&#39;s playback sampling rate. In this exemplary embodiment, as with many telephony applications, the sender and receiver apparatuses utilize separate clocking mechanisms for analog to digital or digital to analog conversion. Imperfections in hardware create variations in these sampling rates, and thus, ultimately create variations in transmission and playback rates. The present invention solves the above problem by providing a system and method for monitoring a receiver&#39;s buffer and adjusting the receiver&#39;s playback sampling rate to maintain an adequate number of packets in the buffer; this accounts for sampling rate variations among the apparatuses. 
   In one aspect, an exemplary embodiment is a receiver apparatus that comprises an interface for receiving packets from a packet-based network, a buffer for temporarily storing the data packets, a buffer monitor, a digital to analog converter for converting the digital data to an analog signal, and a clocking mechanism operable to provide the digital to analog converter with different frequencies. The interface can employ any means to communicate over any type of packet-based network. The present invention can serve as a supplement to current buffering techniques or can operate independently. Additionally, techniques of communication and data compression have no effect on the present invention, and the present invention can incorporate all such techniques, such as utilizing frames and encoding schemes. Those of ordinary skill in the art will also appreciate that the present invention can be implemented over any network. 
   Turning back to the exemplary receiver, the buffer monitor queries the buffer to determine the buffer&#39;s activity. Generally, querying the buffer&#39;s activity entails determining the number of packets in the buffer, but might also entail determining other activity such as rates at which the buffer&#39;s capacity changes. In accord with this exemplary embodiment, if the buffer approaches capacity or depletion, the buffer monitor can trigger changes in the playback sampling rate of the receiver. Typically, a clocking mechanism provides a frequency to the digital to analog converter, and adjusting that frequency adjusts the playback sampling rate. The buffer monitor triggers adjustments to the playback rate to, in effect, synchronize the playback rate to the transmission rate. The degree of adjustment and the number of possible adjustments that can be made are endless. Typically, small adjustments are made that do not effect the sound quality, namely, between 0 and 4 Hz. 
   Exemplary receiver and transmitter apparatuses may exist as a personal computer, laptop, phone, cellular phone, or any other device that includes a buffer, buffer monitor, digital to analog converter, and an interface to the incoming data. The components of the apparatus (buffer, buffer monitor, etc.) can be separate modules or exist in combination. An exemplary implementation, for example, can be on sound cards in conjunction with a personal computer that has an interface, either directly or indirectly, to a packet-based network. 
   In another aspect, a method provides for real-time audio communication sessions where a transmitter sends audio digital data; a receiver receives the digital data, monitors its buffer, and optionally adjusts the playback rate; and the receiver plays the audio data at the receiver&#39;s playback rate. In this exemplary embodiment, with each incoming packet, the receiver can query the buffer to determine the number of packets in the buffer, update a variable representing the sum of the queries, and update a variable representing the number of incoming packets (number of queries here). Accordingly, at any point, the buffer monitor can calculate the average number of packets in the buffer with these two variables. The buffer monitor can then adjust the playback rate. Alternately, in another aspect, a transmitter sends audio digital data in any digital format, and the receiver or an interface can format the digital data for buffering in accordance with the present invention. 
   In an exemplary embodiment, the buffer monitor allows a ten second initiation period to elapse before monitoring the buffer. Then, the buffer monitor calculates the average number of packets in the buffer every 20 seconds, and adjusts the playback rate if the average is too high or too low. In this exemplary embodiment, the buffer monitor adjusts the playback rate more dramatically if the average is dangerously high or low, adjusts the playback rate less dramatically if the average is near satisfactory conditions, and does not adjust the playback rate if the average falls in a satisfactory zone. 
   Accordingly, by monitoring the buffer and adjusting the playback sampling rate, the present invention remedies the problem of varying sampling rates among devices communicating audio data over a network. 

   
     BRIEF DESCRIPTION OF THE DRAWINGS 
       FIG. 1  depicts a network in which a transmitter and receiver communicate via real-time audio data transmission in accord with an exemplary embodiment of the invention. 
       FIG. 2  illustrates a transmitter and receiver operable to communicate in real-time via voice over Internet transmission in accord with an exemplary embodiment of the invention. 
       FIG. 3  represents a personal computer which can function as a receiver or transmitter in accord with an exemplary embodiment of the invention. 
       FIG. 4  depicts the flow of data through a receiver apparatus in accord with an exemplary embodiment of the invention. 
       FIG. 5  is a flowchart of monitoring the buffer and adjusting the playback sampling rate in accord an exemplary embodiment of the invention. 
       FIG. 6  is a flowchart of monitoring the buffer and adjusting the playback sampling rate according to an exemplary embodiment of the invention. 
   

   DETAILED DESCRIPTION 
   The present invention entails real-time transmission of audio data over a network.  FIG. 1  illustrates an exemplary environment  1  for operation of the present invention. More specifically,  FIG. 1  illustrates a packet-based network  50  in which a transmitter  20  and receiver  100  communicate via real-time audio data transmission. While the present invention can operate over any network, for clarity, the following description of the exemplary embodiments of the invention will focus on packet-based networks, such as the Internet network. Similarly, the transmitter  20  can operate as a receiver, and the receiver  100  can operate as a transmitter. Again, the following description also addresses systems with a single, direct voice terminal for convenience, but one can implement the invention with multiple, indirect voice terminals. 
   Referring to  FIG. 1 , live audio data  10  feeds into a transmitter  20 , which digitizes the analog signal. The transmitter  20  digitizes the signal at sampling rate  32  according to a frequency originating from a local clock  30 . The transmitter sends the digital data in digital packets  57  over the packet-based network  50  to the receiver  100 . The receiver  100  converts the digital signal into an analog signal for playback  175  at playback sampling rate  152  according to a frequency originating from a local clock  150 . The receiver  100  is able to increase or decrease the playback sampling rate  152 . The two sampling rates  32  and  152  originate from different clocks that have different local frequency references,  30  and  150  respectively. And as the Background of the Invention explains, the sampling rates of transmitter  20  and receiver  100  may vary due to inherent hardware imperfections. 
     FIG. 2  illustrates the components of exemplary environment  1  in greater detail. More specifically,  FIG. 2  illustrates exemplary transmitter  20  and receiver  100  operable to communicate in real-time via audio data transmission over the Internet network  55  in accord with one embodiment of the invention. Referring to  FIG. 2 , the receiver  100  accounts for the potential difference between the sampling rate  32  of the transmitter  20  and sampling rate  152  of the receiver  100  by monitoring the buffer  120  of the receiver  100  and adjusting the playback sampling rate  152  of the receiver  100 . Transmitter  20  receives an analog audio signal  10 . The transmitter  20  comprises hardware to digitize the analog signal  10  for packet transmission. Transmitter  20  can have an analog to digital converter  22 , such as a CODEC, and can have a clocking mechanism  34  that provides a frequency to the analog to digital converter via port  65 . Port  65  can be any means for providing a clocking frequency to the analog to digital converter. The Transmitter can comprise compressor/encoder hardware or software  24  to perform such functions as compressing the data and framing the data. Common voice coding techniques include G.711, G.726, G.728, G.729, and G.723.1. Accordingly, the data, in one exemplary embodiment, can travel from the A/D converter  22  as a PCM signal (Pulse Code Modulated)  23 , and travel from the compressor/encoder  24  to the packetizer/depacketizer  26  as digital frames  25 . The packetizer  26  ultimately structures the data into packets in accordance with a known IP protocol for transmission over the IP network  55 . The Transmitter  20  comprises an interface  28  to the IP network. The interface  28  can communicate with the receiver  100  according to any communication method  102  and can comprise any attendant hardware or software to implement the communication method  102 . A software interface  28 , for example, may initiate a socket connection with the receiver  100 . 
   Again referring to  FIG. 2 , the receiver  100  comprises a buffer  120 , buffer monitor  140 , and a clocking mechanism  154  that operates independent from the transmitter&#39;s clocking mechanism  34 . Communication ports  142  and  151 , respectively, couple the buffer monitor  140  to the buffer  120  and the clocking mechanism  154 . The receiver  100  receives the packets over the IP network  55 ; the receiver  100  can implement any type of interface  28  to receive the packets. The packetizer/depacketizer  110  can unpack the IP packets into frames or simply forward the packets to the buffer  120 . The digital data  112  can thus exist as a known format of frames, a proprietary format, or any form of packets. The term packet will herein incorporate all such formats for clarity. 
   Packets arrive non-uniformly due to jittering from the network  55 . A jitter buffer is well know in the art, and the present invention can supplement all such buffering techniques. The buffer monitor  140  monitors the activity of the buffer. Typically, monitoring the buffer&#39;s activity entails querying the buffer  120  to determine the number of packets in the buffer  120 , but can also entail determining the rate at which the buffer  120  is filling or emptying, the rate at which packets are entering the buffer  120 , or any other activity regarding the packets in relation to the buffer  120 . The buffer monitor  140  is operable to trigger an adjustment to the playback sampling rate  152  when the buffer monitor  140  determines the buffer  120  satisfies certain criteria. The buffer monitor can query the buffer through port  142 , which may be any physical means for monitoring the buffer, including software and hardware-only implementations. When the monitor  140  determines the buffer  120  satisfies said criteria, the monitor  140  communicates with the clocking mechanism  154  through port  151 , directing the clocking mechanism  154  to adjust the playback sampling rate  152 . Exemplary clocking mechanism  154  is operable to adjust the playback sampling rate in relatively small intervals. For example, the buffer monitor  140  preferably can trigger an 8 Hz increase in the playback sampling rate, and the receiver  100  then preferably can increase the playback rate in an increment of approximately 8 Hz. Playback devices vary with respect to their accuracy in altering their playback sampling rates. When the buffer monitor  140  triggers an increase or decrease in playback sampling rate, the actual adjustment to the playback sampling rate may not be identical to the adjustment that the buffer monitor  140  triggers. Exemplary clocking mechanism  154  can send clocking frequencies through port  156  to the digital to analog converter  160 . 
   As  FIG. 2  illustrates, the receiver  100  continuously converts the incoming data via an optional decompressor/decoder  130  and digital to analog converter  160  at sampling rate  152 . The receiver  100  can implement any techniques of encoding or jitter buffering in accordance with the present invention. Techniques, therefore, can manipulate the data  114  leaving the buffer  120  via the decompressor/decoder  130 , or can manipulate the data as the data  116  leaves the decompressor/decoder  130 . Those of ordinary skill in the art will appreciate the modules above may exist as separate modules or may exist as one module which can remove any need of separate ports  65 ,  142 ,  151 , and  156 . 
     FIG. 3  illustrates a conventional personal computer  200  suitable for functioning as a receiver  100  or transmitter  20  in accord with an exemplary embodiment of the invention. Any device, however, that comprises a buffer, buffer monitor, and variable clocking mechanism can implement the present invention. Examples include laptops, phones, cellular phones, and handheld devices. Referring to  FIG. 3 , the exemplary personal computer  200  can operate in a network environment, including local area networks  290  and wide area networks  50 . The exemplary personal computer  200  comprises a processing unit  202 , such as “PENTIUM” microprocessors, manufactured by Intel Corporation. The exemplary personal computer  220  also includes system memory  210 , including read only memory (ROM)  212  and random access memory (RAM)  216 , which is connected to the processor  202  by a system bus  18 . The exemplary personal computer  200  utilizes a BIOS  214 , which is stored in ROM  212 . Those skilled in the art will recognize that the BIOS  214  is a set of basic routines that helps to transfer information between elements within the exemplary personal computer  200 . Those skilled in the art will also appreciate that the present invention may be implemented on computers having other architectures, such as computers that do not use a BIOS, and those that utilize other microprocessors. 
   Within the exemplary personal computer  200 , a hard disk drive interface  231  connects the local hard disk drive  230  to the system bus  18 . A floppy disk drive interface  232  and CD-ROM/DVD interface  234  can connect floppy disk drives (not shown) and CD-ROM devices (not shown) to the system bus  18 , such as an Industry Standard Architecture bus (ISA). A user enters commands and information into the exemplary personal computer  200  by using input devices, such as a keyboard  264  and/or pointing device, such as a mouse  262 , which are connected to the system bus  18  via a serial port interface  260 . Other types of pointing devices (not shown in  FIG. 1 ) include track pads, track balls, pens, head trackers, data gloves and other devices suitable for positioning a cursor on a computer monitor  206 . The monitor  206  or other kind of display device can connect to the system bus  18  via a video adapter  204 . Although other internal components of the personal computer  200  are not shown, those of ordinary skill in the art will appreciate that such components and the interconnection between them are well known. Those of ordinary skill in the art also will appreciate the modules and hardware in  FIG. 3  can exist as separate modules and hardware pieces or can exist in many different forms in which certain modules and hardware couple together as single modules or hardware pieces. 
   Additional details regarding the internal construction of the exemplary personal computer  200  focus on aspects pertinent to the present invention. Referring to  FIG. 3 , the exemplary personal computer  200  includes a sound card  250  that comprises a digital to analog converter, such as a CODEC  252 , and an encoder  254 . The buffer monitor  140  can exist as a computer program module  220  residing on the hard drive  230  that utilizes the RAM  216  to implement its functioning. The buffer monitor program  220  can access the soundcard via ISA bus  18 . The sound card  250  can connect to the personal computer  200  via a serial port interface  260 , connect via the ISA bus  18 , or connect via direct incorporation on the motherboard. A clock  268  forms part of the clocking mechanism  154 . 
   The exemplary personal computer  200  can connect to networks via a network interface  280 , such as local area networks  290 , which can provide indirect connection to wide area networks. The exemplary personal computer  200  also can comprise a modem  270  for direct communication over packet networks. In the case of an exemplary transmitter  20 , the real-time audio signal  10  preferably transmits to the sound card  250  via a microphone or other device (not shown). The sound card  250  converts the data to digital packets which the sound card  250  feeds to the ISA  18  (the packets may directly trace on the mother board if the sound chip has a direct connection to the motherboard). 
     FIG. 3  represents only one exemplary embodiment of the present invention. All the requisite components of the current invention may reside on the soundcard or may be spread out through the exemplary personal computer  200  or other device.  FIG. 4  depicts the flow of data through an exemplary receiver  100  in accord with one embodiment of the present invention. The playback device  420  comprises the necessary hardware to convert the packets to an analog signal. Packets  57  enter the receiver  100  through interface  102  and then flow to the buffer  120  through a pathway  405 . The buffer monitor  140  monitors the activity of the buffer  120  through port  142 ; this monitoring can be querying the number of packets  430  in the buffer  120 . The playback device  420  continuously samples the data at sampling rate  152 , and the data flows from the buffer  120  to the playback device  420  along pathway  435  at the rate in which the playback device  420  plays the data. When the activity of the packets  430  in the buffer  120  satisfy certain criteria, the buffer monitor  140  directs the clocking mechanism  154  through port  151  to adjust the playback sampling rate using frequency controller  440 . The clocking mechanism  154  can send a clocking frequency to the playback device through port  156 . 
   Port  151  from the buffer monitor  140  to the clocking mechanism controller  154  can be through any physical means, and the components of the buffer monitor and clocking mechanism can actually reside in a single module. Likewise, the port  142  from the buffer monitor to the buffer  120  can be through any means that allows the buffer monitor  140  to monitor the activity of the buffer  120 , and the components of the buffer monitor  140  and the buffer  120  can form a single module. Finally, port  156  from the clocking mechanism  154  to the playback device  420  can also assume any form to provide a frequency to the playback device  420 , and the clocking mechanism  154  may be part of the playback device module  420 . 
     FIG. 5  illustrates an exemplary process  500  for monitoring the buffer and adjusting the playback sampling rate process in accord with an exemplary embodiment of the invention. The process begins at the initialize procedure in step  505 , whether automatic triggering per a communication initiation, automatic triggering per an independent program monitoring the performance of the communication, or manual triggering. The buffer monitor  140  determines whether the monitor trigger is set in step  510 . If the monitoring trigger is set, the buffer monitoring program module  220  queries the buffer  120  in step  520 . When the buffer monitoring program module  220  queries the buffer  120 , the buffer monitoring program module  220  can determine the number of packets in the buffer  120 , determine the rate at which the buffer is filling or emptying, or use any other monitoring method to determine the buffer&#39;s activity. In step  530 , the buffer monitoring program module  220  decides whether the playback rate  152  should be adjusted. If an adjustment is not made, the process  500  loops back to the step of determining whether the monitor trigger is set in step  510 . If the buffer monitoring program module  220  decides to adjust the playback rate  152 , it sends an communication to the clocking mechanism  154 . 
     FIG. 6  illustrates exemplary process  600  for monitoring the buffer and adjusting the playback sampling rate according to the preferred embodiment of the present invention. The variables have the following definitions. “streamTime” represents the total time that the data stream has been running. The invention can idle for this period of time after initiation to account for typical sporadic variations that occur as the transmitter and receiver establish a connection. This period approximates 10 seconds in exemplary process  600 . “sInt” represents the running time from when the last decision was made to determine whether to adjust the playback rate. The preferable period for this variable is 20 seconds in exemplary process  600 . “sReceived” represents the number of instances of receiving a packet and querying the buffer. “buffFullAvg” represents the average number of packets in the buffer over the last sInt interval of time. 
   Referring to  FIG. 6 , the exemplary process  600  starts with the buffer monitor  140  initializing the variables in step  605 , and exemplary process  600  can trigger according to any number of events. The receiver  100  receives a packet in step  610  and places the packet in the buffer  120 . An initial loop between steps  610  and  620  then occurs until the streamTime elapses. After streamTime elapses at step  620 , exemplary process  600  loops through steps  610 ,  620 , and  630  until sInt time elapses at step  640 . At step  630 , the buffer monitor  140  queries the buffer&#39;s activity  120 , tallying the number of packets in the buffer and tallying the number of packets received. At step  640 , the process will loop back to step  610  unless sInt has elapsed. 
   Once sInt elapses at step  640 , the buffer monitor  140  calculates the average number of packets in the buffer for that sInt period and re-initializes the variables at step  660 . The process then turns to steps  670  to  686  to determine whether to adjust the playback sampling rate. At step  670 , if buffFullAvg&gt;4.5, the buffer monitor  140  instructs the frequency controller  440  to increase the playback rate by 4 Hz at step  680 . If not, proceeding to step  672 , if buffFullAvg&gt;4.0, the buffer monitor  140  increases the playback rate by 2 Hz at step  682 . If not, proceeding to step  674 , if buffFullAvg&lt;0.5, the buffer monitor  140  decreases the playback rate by 4 Hz at step  682 . If not, proceeding to step  676 , if buffFullAvg&lt;1.5, the buffer monitor  140  decreases the playback rate by 2 Hz at step  682 . Whether or not an adjustment is made, the buffer monitor  140  reinitializes buffFullAvg at step  650  and returns to step  610 . 
     FIG. 6  illustrates the ability to adjust the playback sampling rate to a greater degree when the buffer approaches extreme danger areas (example, less than 0.5 packets full or more than 4.0 packets full, on average). The exemplary process  600  adjusts the rate twice as many Hz as the first adjustment upon detecting a danger area. The invention can entail a greater number of variant adjustments and a manifold range of adjustment. Likewise, one can easily change the range of no action, i.e., where no adjustment is made, in  FIG. 6  between 1.5 and 4.0 Hz. 
   As an illustration, taking sound cards capable of adjusting their playback sampling rate in increments of 2 Hz, a nominal 22050 Hz sampled stream typically will playback at anywhere from 22048 to 22056 Hz. This error range implies a possible 8 Hz variation between the sender and the receiver. Assuming a typical 5-packet buffer, and assuming typical packets that each represent about 60 mSec of actual time, a positive 8 Hz sampling error would result in the receiver playing each packet in about 59.98 mSec (error of 0.02 mSec with each packet the transmitter sends and the receiver plays). Thus, after receiving 3000 packets (three minutes), the receiver would gain a whole packet&#39;s worth of time (3000 packets*0.02 mSec), that is, the receiver would play the 3000 packets in the time it took the sender to send 2999 packets. Were the receiver to start with 3 packets in its buffer, the above error indicates that about every 9 minutes the buffer would empty. The emptying causes a “blank spot” in the audio on the receiving end. Thereafter, a “blank spot” or interruption would accompany practically every packet, because no buffer remains to cushion the 0.02 mSec error. The receiver would finish playing a packet 0.02 mSec before the next packet arrives. In practice, a 0.02 mSec “blank spot” may be a short interval that test subjects fail to notice. After 1000 packets (60 seconds), however, this error would accumulate to about 20 mSec, a “blank spot” that would prove quite noticeable. 
   In the converse case, where the receiver plays 8 Hz too slowly, the buffer progressively would fill. Were the buffer to have no size limitation, the buffer would accumulate a packet (60 mSec of data) every 3 minutes. After 30 minutes, the buffer would accumulate 10 packets (600 mSec of data), which represents more than a half second of delay. This delay would prove burdensome and annoying in strictly real-time voice communication. In a live media environment, with concurrent transmission of video and audio signals, this delay would prove disastrous because synchronization of the signals is of critical import. 
   The buffer monitoring program module  220  can compensate for these variations by making adjustments to the playback sampling rate  152 . This can be done in an exemplary embodiment of the invention where the receiver  100  typically makes one or two frequency adjustments within the first minute of operation, settles on a playback rate  152  between 22048 and 22056 Hz, and remains at single playback rate  152  for 10 hours or more. 
   The above embodiments are merely demonstrative of the scope of the present invention. Factors that will alter the above variables include the jitter buffer size, how often rate adjustments should be made, and how much disruption the adjustment creates for an individual user. While the foregoing embodiments discuss voice communication over a packet network as an example, the teachings described herein can also be applied to other instances where real-time audio data is transmitted over a network.