WO2003065661A2

WO2003065661A2 - Method and system of data packet transmission timing for controlling bandwidth

Info

Publication number: WO2003065661A2
Application number: PCT/IB2003/000993
Authority: WO
Inventors: Jarno Marchetto; Emanuele La Cognata
Original assignee: The Fantastic Corporation
Priority date: 2002-01-31
Filing date: 2003-01-31
Publication date: 2003-08-07
Also published as: US20090262755A1; US7539756B2; NO20043502L; CN1647463B; KR20040081481A; CN1647463A; EP1472838A2; KR100963858B1; JP2005516539A; US20030145100A1; WO2003065661A3; US8175117B2; WO2003065661A9; PL370081A1; JP2009105981A; EP3255848A1; EP1472838B1; CA2513637A1

Abstract

A method and system for determining a wait time (tw) to be used between successive transmissions of packets of a content to achieve a selected target bandwidth BT for the transmission. The wait time between successive packets of a content being transmitted is determined as a function of the selected target bandwidth (BT) to be achieved during the transmission and the size (P) of the packets based on the algorithm formula (I). The invention provides bandwidth control at the source (the sending device) without relying on network Quality of Service (QoS) facilities.

Description

METHOD AND SYSTEM OF DATA PACKET TRANSMISSION TIMING FOR CONTROLLING BANDWIDTH

Field of the Invention

The invention relates to a method and system for timing the

transmission of data packets from a sender device to one or more receiving

devices, for achieving bandwidth control.

Background of the Invention

In performing data transmission using various media such as cable,

telephone line, satellite communication, etc., it is conventional for the data to be

sent from one device to other devices. The data is first fragmented, or segmented,

into packets and then the packets transmitted, or broadcast. The term "packet" is

used to indicate a sequence of bytes and represents the smallest unit of

transmission and reception. Hereafter, a "device" is defined as a hardware element

with a software component capable of receiving and/or transmitting data packets.

Examples of such devices include computers, GPRS mobile phones and network

equipment. The term "content", as used herein, indicates data that is segmented

into a sequence of packets. The data contained within the content could be a file,

or part of a file, part of a data stream, or any collection of data. The content can

be of pure data, or audio and video data streams, or of any combination. The

content is sent in sequentially transmitted packets from a sender device to one or

more receiving devices.

The sending and receiving devices typically run software programs,

whose purpose at the sender device is to fragment the content to be sent into

packets, and at the receiver device to reassemble the packets received into the

original content.

Bandwidth is defined as the amount of data that can be transmitted

over the media in a given time frame, for example, 1 0 Mbps (1 0 million bits per

second). Essentially, this is the speed of the transmission, that is, the speed at

which data is sent from the sending device to the receiving devices. Each

transmitting-receiving system has a certain bandwidth capability that is defined by

various factors, such as the type of media used for the transmission and equipment

used for transmission and reception of the data. For example, a broadband cable

medium has a larger bandwidth capability than a telephone line.

Various types of protocols are used to transmit the data packets

forming the content. Some protocols may be considered to be "unreliable", herein

meaning any transmission protocol that provides best-effort delivery of packets, and in particular, does not perform automatic re-transmission of lost or corrupted

packets. Examples of "unreliable" protocols currently in common use include

unicast and multicast User Datagram Protocol (UDP), ATM Adaption Layer (AAL)

Types 3/4 and 5 in non-assured transmission mode, AppleTalk DDP datagrams, and

unicast and broadcast MPEG-2 transport streams.

Often, data from different files is to be transmitted at the same time

over the same media. For example, if a particular system has a 10 Mbps

bandwidth transmission capability, the packets of two different contents can be

transmitted at the same time in separate streams, for example, each stream of 5

Mbps bandwidth. In various commercial applications, portions of the available

bandwidth of a given media are sold or licensed to different customers for use at

the same time.

Some applications require that packets are to be received in a regular

and timely fashion, for example, for audio and video transmissions in which it is

important that the jitter (i.e., the amount of variation in the end-to-end packet

transit time) is kept as low as possible. Increasing reliability at the receiving device

implies use of a high precision control of the bandwidth at the sending device,

especially for broadband applications.

In general, the bandwidth usage of a broadcast application is selected

according to network load and speed, leased bandwidth capacity, and processing

speed of the receiving device. The bandwidth allocated for a given transmission is hereafter referred to as the "target bandwidth". During the transmission, the actual

bandwidth used can vary relative to the selected target bandwidth.

As the broadcast takes place, the bandwidth varies and can be

measured at any time. "The instantaneous" bandwidth is the bandwidth measured

in the shortest measurable time. For example, if the bandwidth usage can be

regularly checked at most once per second, the instantaneous bandwidth is

computed by dividing the quantity of data transferred in that time frame (in this

case 1 second) by that time interval.

The broadcast of the content has an "average (or mean) bandwidth".

This is the total amount of data transmitted during a transmission divided by the

transmission duration. The terms "mean bandwidth" and "average bandwidth" are

used interchangeably herein.

Since the content is transmitted in packets, there is a pause, or wait

time, between the start of transmission of successive packets. The wait time

between packets is related to the bandwidth used during transmission of a content.

For example, increasing the wait time between packets while maintaining a fixed

packet size results in less data transmitted during a given time, and vice versa.

If data is sent without a precise bandwidth control, various problems

can arise. Basically, there are three different scenarios that can cause trouble:

(1 ) the average bandwidth is too high relative to the target bandwidth

value; (2) the average bandwidth is too low relative to the target bandwidth

value;

(3) the average bandwidth is equal, or very close, to the target

bandwidth, but the instantaneous bandwidth values measured during the

transmission are different from the target bandwidth. This type of transmission is

hereafter referred to as heterogeneous, i.e., contains peaks.

The problems caused by the different scenarios described above can

adversely affect different components and structures of the system that receive the

data, such as routing and switching devices and the destination receiver devices.

This is described below.

(1 ) Sending data packets too fast. Broadcasting of the data at a

speed above the target bandwidth usually results in substantial packet losses. Loss

of packets reduces the integrity of the data transmission. Also, too high a

transmission speed can cause congestion if routing devices are not able to buffer

the incoming data stream. The same problem can affect encapsulators if data is

sent to a satellite uplink. Here, entire series of packets can be lost. At the

receiving device side, packet reassembly can be difficult if the hardware or

application program processing the received data is not designed for an incoming

data stream speed higher than the bandwidth mean (average) value. Moreover, a

local area network (LAN) to which the receiving devices are connected can experience congestion if no traffic buffering mechanism is provided between the

data transmitter and the receiving devices.

(2) Sending data packets too slow. If the packets are transmitted

at a speed lower than the specified target bandwidth, no losses of data packets

should be experienced. However, a resource waste is caused, especially if the

connection between the data transmitter and receiving devices is purchased on a

bandwidth usage basis instead of on a transmitted data amount basis. Also, audio

and video streaming quality can be affected if the receiving data rate is too low.

That is, the received sound and video will be intermittent, garbled, or otherwise

distorted

(3) Peaks in bandwidth usage. Even when the average output

transmission bandwidth is close to the target bandwidth does not always guarantee

that the data stream will be free of problems. For example, there can be bandwidth

usage peaks at any instant during transmission. Also, if the packets are not

homogeneously distributed during the transmission, i.e., a fairly constant time

between the successive packets, packet clustering occurs that causes an increase

or decrease of the bandwidth used at a given instant, this being the instantaneous

bandwidth. As a result, the same problems described above for transmissions with

mean bandwidth higher or lower than the target bandwidth can arise, although the

adverse affects are experienced to a lesser degree. Accordingly, it would be desirable to be able to better control the

bandwidth usage for the transmission of packets containing any type of content.

Prior attempts to solve the various problems discussed above typically enforce a

maximum bandwidth. That is, the transmission protocol is such so as to intervene

when a maximum value bandwidth is exceeded, but do not intervene when the

bandwidth usage is lower than the maximum. Other existing solutions correct

existing traffic streams. That is, a protocol is used to solve congestion problems

caused by senders without inbuilt traffic control. Still other existing solutions

discard packets when buffer space is exceeded. Some of the solutions of this type

discard packets using an algorithm, such as the "leaky bucket" algorithm

(Tanenbaum, Computer Networks, 3^rd Edition, p. 380, Prentice Hall). Other

solutions permit a certain amount of burstiness in the output stream using an

algorithm such as the "token bucket" algorithm (Tanenbaum, p. 381 ) .

Brief Description of the Invention

In accordance with the invention, it has been discovered that the

bandwidth can be held more closely to a desired target bandwidth value if the

pause (wait time) between the successive packets is properly selected and the

packets are transmitted using the selected wait time during the transmission of the

content. A novel method and system are provided that determines and maintains the wait time between packets in response to selected input parameters of desired

target bandwidth and size of the packets.

The method and system of the invention operate to transmit the data

in a manner to control the bandwidth for the transmission of content by controlling

the pause, or wait time, between the packets of the data stream. As part of the

method and system, a novel algorithm has been developed and is implemented that

relates the desired size (P) of the packets of a content to be transmitted to a

desired target bandwidth (BT) by selecting and controlling the wait, or pause, time

pause (tw), between the transmission of the packets. The invention can be

implemented in software and/or hardware.

In a preferred embodiment of the invention, the precision of the

bandwidth control is maximized by the use of the highest resolution clock or other

timing device available at the sending device. This provides the best compensation

for wait time rounding errors.

In a preferred embodiment of the invention, the wait time between

packets is calculated based on a fixed packet size and target bandwidth.

The invention does not require or use feedback from the receiver

devices. Therefore, it can be used for unicast, multicast and broadcast

transmissions and with simplex, half-duplex and full-duplex media.

As compared to the existing transmission problem solutions discussed

above, the invention causes data flow to tend towards target bandwidth, decreasing or increasing transmission speed as necessary. Also, the invention

provides traffic control at the source so that the outgoing traffic is smoothly

distributed and tends towards the target bandwidth. Further, the invention

prevents excess packets by introducing pauses between packets, blocking the

sender until the next packet is to be transmitted and never discards packets. Also,

the present invention maintains a constant bandwidth usage with minimal

burstiness.

Brief Description of the Drawings

Other objects and advantages of the present invention will become

more apparent upon reference to the following specification and annexed drawings

in which:

Fig. 1 is a schematic diagram of a system implementation of the

invention;

Fig. 2 is a flow chart of the process;

Fig. 3 is a flow chart showing the generation of various timing

functions;

Fig. 4 is a diagram showing percentage error relative to a specified

target bandwidth for an exemplary transmission bandwidth of up to 1 0 Mbps;

Fig. 5 is a diagram showing the percentage error relative to a specified

target bandwidth for an exemplary transmission bandwidth of above 1 00 Mbps; Fig. 6 is a diagram showing the percentage error relative to a specified

target bandwidth for an exemplary transmission bandwidth of up to 1 Gbps;

Fig. 7 is a diagram showing various time relationships between

packets;

Fig. 8 is a flow chart showing an alternate function for producing wait

time of a fixed value between the start of sending of successive packets; and

Fig. 9 is a modification of the function of Fig. 8 introducing an error

compensation mechanism.

Detailed Description of the Invention

In the transmission of data packets, bandwidth is usually defined as

the amount of data transmitted in a given time frame. Bandwidth control refers to

the capability to control the bandwidth, i.e., the rate of transmission for a given

amount of data. A significant value to evaluate bandwidth control quality is the

mean (or average) bandwidth BM, that is, the arithmetic mean value about which

the actual transmission rate centers during the entire transmission. The target

bandwidth BT is defined as the needed (selected) bandwidth for a given content

transmission. In some cases, BT is an amount of bandwidth that is sold or leased to

a content producer by a broadcast facility.

If BT is known, the mean bandwidth BM can be defined by the

following equation: (BT - ε) < BM < (BT + ε) (Equation 1 -1 )

where

BT = the target bandwidth

BM = mean bandwidth

ε = error or deviation of BM from BT

The main goal of bandwidth control is to minimize the error ε between BM and BT.

To consider it from another perspective, E is the percentage error of the mean of

the bandwidth actually transmitted relative to the desired target bandwidth BT,

expressed in percentage terms as:

E = -100% (Equation 1 -2)

It is also important to consider the amount of data transmitted in a

fixed time window smaller than the time needed for the transmission of a complete

content, that is, the instantaneous bandwidth, and to compare this value during the

entire content transmission with the target bandwidth. This is necessary because if

the instantaneous bandwidth values are largely different from the target bandwidth,

the data transfer rate is oscillating, that is, the transmission is heterogeneous. This

can happen even if the mean bandwidth BM is equal or very near to the target

bandwidth BT.

This heterogeneity in the bandwidth usage can be very difficult to

manage at the receiving end, for routers and other routing devices, especially where the allowed target bandwidth BT is fixed and cannot be exceeded, e.g., in the case

of transmission of data by satellite transmissions.

A "constant" bandwidth can be theoretically obtained if the difference

between the instantaneous bandwidth (measured in a time frame Δt), and the target

bandwidth BT tends to zero in a time t during the transmission. That is:

(BT - ε) < Bit,t+Δ i < (Bτ + ε) (Equation 1 -3)

with ε and Δt tending to zero, and

tstan < t <tend, and where

tstart : transmission start time;

tend : transmission end time;

Δt : measurement time frame, which must be less than the transmission duration;

Biu+Δti : instantaneous bandwidth B measured in the interval included between

t and t + Δt.

The above is only a theoretical situation, not reproducible in the real world.

Nevertheless, the bandwidth control described here is designed to approach as near

as possible to this ideal solution, using the maximum precision supported by the

sending device.

Another parameter that can be considered is the transmission

burstiness, defined as:

Burstiness = -^- (Equation 1 -4)

B. M w^*here BM is the mean bandwidth of a transmission content, and Bpeak is the highest

of the absolute instantaneous bandwidth values measured during a transmission.

That is:

Bpea = max(Bit,t+Δti) (Equation 1 -5)

Ideally, the burstiness should tend to one. That is, the bandwidth being used

should be centered at BT. Note also that under the same conditions of burstiness,

the transmission heterogeneity could be different, since no information is provided

in this value about the recurrence of Bpeak. In fact, the burstiness value says how

large the peak is relative to the mean transmission bandwidth BM, but not how

many times the peak is repeated during a transmission. That is, a transmission can

experience 1 or 1000 peaks, but the value Bpeak is the same in most or all cases.

Also, no information is provided about the heterogeneity of the transmission, that

is, about the peak frequency.

Application programs making use of unreliable network protocols such

as the User Datagram Protocol (UDP) for transmissions typically are not able to

specify a target bandwidth BT for a given transmission. As a consequence of the

absence of direct control, data is sent at the maximum bandwidth allowed by the

network interface. When the program communicates the instruction of sending a

content to the network card, that content is sent at the maximum speed

(bandwidth) allowed by the network card. For example, if the network card can handle 10 Mbps bandwidth, the content is fragmented into packets and these are

sent at 10 Mbps on the network; there is no application level control.

In accordance with the invention, a wait, or paresidual time is imposed

between successive packets of the transmission in order to control bandwidth

usage and to achieve the selected target bandwidth BT. The method and system of

the invention achieves bandwidth control by introducing this pause, in order that

the average transmission bandwidth BM is as near as possible to the target

bandwidth BT. Also, all of the "instantaneous" bandwidth values measured during

the transmission of the content are made to occur as closely as possible to the

target bandwidth BT. For this reason, a high precision is needed in timing the time

between packets, and the timing preferably should be as precise as possible. The

needed precision needed for the wait time depends on the target transmission

bandwidth BT. AS described below, the higher the value of the target bandwidth

BT, the greater the control of the precision of the pause between packets should be

in order to maintain the deviation E from the target bandwidth to acceptable values

for practical use.

Consider the following example of dividing the data of a content to be

transmitted into packets of e.g. 81 92 Bytes, and needing to transmit a fixed

amount of data at a target bandwidth BT = 10 Mbps. This computes as the wait

time, or pause, between every transmitted packet being 6.5536 milliseconds (81 92

Bytes = 81 92*8 = 65536 bits. 1 0 Mbps = 1 0000000 (ten millions) bits per second. Dividing 65536 by 1 0000000 = > wait time (between packets) =

0.0065536 seconds = 6.5536 milliseconds). If the precision of the time

measuring system for determining the wait time between packets is one

millisecond, e.g., a one millisecond clock, then this time value has to be rounded

down or up from the computed 6.5536 millisecond value so that the wait time will

be 6 or 7 milliseconds. The_^ rounding to 6 or 7 milliseconds will produce respective

errors of -6.38% or 9.23% in average bandwidth usage from the specified BT =

1 0Mbps. This example illustrates the need for a precise computation of the wait

time intervals between packets during broadcasting. To ensure that the average

bandwidth BM is equal to the target bandwidth BT, theoretically the wait time must

not be rounded. In practice, in some cases, this value must be rounded, due to

software and hardware limitations. As described below, it is preferred that the

highest precision supported by the sender device be used, and therefore that the

achieved bandwidth control is optimized for the software and hardware

configuration employed.

As described above, in order to provide high precision control, an

important aspect to be considered is the time granularity (precision) for the

measurement of the wait time between the successive transmitted packets. As an

example, Fig. 4 shows the absolute percentage error E (logarithmic scale) on the

specified (target) bandwidth BT for a given transmission at bandwidth up to 1 0

Mbps, with a packet size of 8 KB. The upper curve shows the error with rounding of the wait time to the millisecond and the lower curve shows the error using

rounding of the wait time to the microsecond. It can be seen that when the

rounding is to the microsecond (higher precision) the error is less.

Fig. 5 shows, as an example, the absolute percentage error E

(logarithmic scale) on the specified target bandwidth BT for a given transmission at

a bandwidth up to 1 00 Mbps, with a packet size of 8 KB. The upper curve shows

the error with rounding of the wait time to the millisecond and the lower section

shows rounding of the wait time to the microsecond. It can be seen that the

percentage error E in actual bandwidth usage never exceeds 0.1 %. By rounding

the wait time to the millisecond the resulting error is significantly greater; the actual

bandwidth usage would be 45-50% higher than the target bandwidth at a BT of 50

Mbps and 35 % lower at BT = 1 00 Mbps.

Fig. 6 shows the percentage error E on the specified bandwidth BT for

a given exemplary transmission (up to 1 Gbps), when rounding the wait time

between the packets of the transmission to the microsecond, where the packet size

= 8 KB. The percentage error obtained when rounding to the millisecond is not

shown on this graph. By rounding the wait time to the microsecond, the

percentage error on the target bandwidth BT up to 1 Gbps remains under 0.8% in

the worst case. This means that the difference between mean bandwidth BM and

target bandwidth BT would be under 8 Mbps. This might be considered to be

acceptable for transmission speeds up to 1 Gbps. But using the 1 microsecond granularity would yield much higher and unacceptable errors for target bandwidths

even significantly lower than 1 Gbps.

The packet size (P) plays an important role during transmission. In

general, the larger the packet size, the smaller the percentage error of the target

bandwidth BT relative to the mean bandwidth BM, when there is rounding to at least

the precision of a microsecond. This is due to the fact that if the packet size is

increased, the number of packets for a given transmission content decreases and,

consequently, the number of wait time occurrences between the packets of a given

transmission decreases, causing the rounding error sum to decrease.

In the invention, the bandwidth in a transmission is controlled by

selecting and placing wait time intervals tw of a predetermined value between

packets. It is preferred that the wait time interval tw be computed with the highest

precision possible on the sender device by using the highest resolution timing

device available. As demonstrated above, at least microsecond granularity is

preferable to guarantee an acceptable bandwidth control for transmission speeds in

the Mbps range.

The algorithm applied to achieve the bandwidth control is as follows:

given the target bandwidth BT and the packet size P, the wait time

interval tw between every packet is:

= ~ (Equation 2-1 )

DT In operation, both BT and P are provided as input parameters to the

computer that controls the transmission. The target bandwidth Bτ oan be set by

the person who wants to send a transmission through a user interface. For

example, if one wants to send a content to a third person, a program can be made

available where the "speed" at which the file is to be transmitted is entered by the

user. That is, the target bandwidth BT is given as an input parameter.

As to the packet size P parameter, typically the packet size is set

when a program is installed to broadcast the transmissions, and then packet size is

kept the same for all transmissions. The packet size can be, for example, contained

in a configuration file. It also can be input as a separate parameter. In a typical

use, the content to be sent and the target transmission bandwidth BT are selected.

Then the broadcast application reads the needed packet size P from the

configuration settings, which was previously set by the broadcast application or

was input. At this point, the broadcast application "fragments" the file to be

transmitted into smaller packets. Typically, the header of these packets will specify

the address of the receiver and information about the packet content. The

broadcast application already knows the packet size, since it was read from the

configurations or from the input before fragmenting the content to be transmitted

into packets. In use, the target bandwidth BT selected for use with a broadcast

application program is largely based on the transmission media to be used and other

factors, such as the amount of bandwidth leased or purchased for use.

During the transmission of the packets a loop operation is repeated for

every packet transmission as follows:

(1 ) obtain the actual time value ti (as start of packet transmission);

(2) send the packet;

(3) obtain the actual time value t₂ (after packet transmission);

(4) compute the time used to send the packet tused, where

tused = t₂ - ti (Equation 2-2)

(5) compute a value t (residual time)

t = tw - tuse (Equation 2-3)

(6) Wait for t;

(7) go to step 1 .

Referring to Fig. 7, this shows the various times. As seen, tw is the

time between the start (ti) of two successive packets, while t is the time between

the end (t₂) of one packet and the start (ti) of the next packet. The residual time

value t must be respectively based on the selected packet size P and target

bandwidth BT.

The operation to send a packet of data is time consuming. That is,

time tused will be used to physically take a packet and send it on the network. To ensure that there is sufficient time for the wait time tw to be achieved, the sender

device needs to wait a residual time t after the send packet operation is completed.

The computed wait time tw between packets ensures that information will be sent

at the needed target bandwidth. That is, the residual time t is the difference

between tw (wait between successive packets) and tused (to send a packet).

Fig. 1 is a diagram that explains the transmission of the data packets.

At the sending station there is a computer 1 0 that has a broadcast application

program installed. The broadcast application program operates on a content 1 2

that is to be transmitted in data packets. The parameters of the transmission

target bandwidth BT and packet size P are input to the computer 10. As explained

above, the value P of the packet size might already be known in the computer. It

can be set if required. The computer computes the wait time tw from the algorithm

of Equation 2-1 and operates to control the transmission 1 6 of packets 1 8 (black

bars), shown as having a controlled wait time tw between the successive packets.

The packets 1 8 are received at a computer 20 that has a receiving

application program installed. This program reassembles the received packets into

a file that corresponds to the content file 1 2 that was sent. This is conventional.

Fig. 2 is a flow chart of the overall process. Here, in S1 and S2 the

target bandwidth BT and packet size P are input to the computer. In S3, the

computer computes tw in accordance with the algorithm of equation 2-1 . The

computed value tw is used as a control parameter in S4 that is input to the broadcast application process S5. The tw value is applied to the file to be sent as

input at S6 to the broadcast process S5. The broadcast process S5 knowing the

packet size from S2 and the wait time tw from S3 transmits the S7 packets.

Fig. 3 shows the operation of the computer in performing the steps of

the timing loop to obtain the residual time t as referred to above. At S1 01 , the

time ti at the start of transmission of a packet is determined and stored in S1 02.

In S 1 03, the packet to be sent is made available and is sent in S 1 04. In S 1 05, the

end time t₂ of the transmission of the packet is determined and is stored in S106.

In S1 08, the time quantity tused in transmitting the packet is computed (S106 value

minus S1 02 value) and is made available in S1 10.

In S1 1 1 , the computed value of tw (see S4) is used in S1 1 2 with tused

(from S 1 1 0) to compute the residual time t (see Equation 2-3) . The computed

value of the residual time t is available in S1 1 4 as a time to be satisfied in S1 1 6

before transmission of the next packet is started at S 1 01 and S 1 02. The loop of

Fig. 3 is repeated for each packet sent.

Using the algorithm of Equation 2-1 it is preferred that the facilities

used to get the current time values ti and t₂ of Equation 2-2 and to wait for the

residual time t of Equation 2-3 have at least microsecond granularity (precision or

resolution) . Making the granularity more precise decreases the error E of the actual

mean bandwidth BM relative to the target bandwidth BT The most direct method to

obtain the ti and t₂ values would be to get the actual time from the operating system, such as from its timing clock. However, this would imply that the

operating system has a timer of microsecond precision. In the absence of this

facility, a high precision hardware counter could be used. A counter is a general

term used in programming to refer to an incrementing variable. Some systems

include a high-resolution performance counter that provides high-resolution elapsed

times. The counter values are managed by the computer central processing unit

(CPU) and read by the operating system. Usually, the counter must be hardware

controlled in a CPU-sharing operating system if other tasks, in addition to time

computation, are carried out. If the increasing frequency of the counter, i.e., how

many times in a second the count is increased, is known, it is possible to compute

a time interval such as tused by obtaining the counter value at two different times

and dividing this difference by the counter frequency. As seen, it is desirable to

use a high precision counter. Any reliable counter can be used, however, the

highest resolution counter available on the device should be used in order to

maximize the precision with which bandwidth is controlled.

Another key issue is how to suspend the program that sends the

packets for a given residual time interval t. As shown above, after a packet is

transmitted, the program must wait for the residual time t = tw - tused to have

elapsed before sending another packet. If the broadcasting device does not have

the necessary hardware and/or software facilities to accomplish the task of

determining the residual time with the desired granularity, but provides at least a method to know the current time with the desired granularity, then other

approaches can be used.

One approach, is shown in Fig. 8. Here the residual time t is supplied

at step S201 and the determined value of tstart (start of the tw function) and tnow

(present time) are provided at S203 and S205. A time teia sed is computed in S207

as:

teiapsed = tnow - tstart (Equation 3-1 )

A function (i.e., a software process) is implemented which, periodically, in a loop,

compares in S209 the values of the difference between teiapsed and the residual

time t to wait before the next packet is sent. When the value of teiapsed is equal or

greater than the desired residual time t, the looping process ends and lets the

program continue, sending a new packet. This approach provides high precision if

the time measurements have high precision and the appropriate values, described

below, are compared at frequent intervals.

The method of Fig. 8 can be described in an algorithm as follows:

S201 Get the desired residual time t (i.e., the how long the sending of

packets must be suspended);

S203 Get the start time tstart (when the wait process is called);

Continue performing the following loop as often as possible:

S205 Get the current time tnow;

S207 Compute tnow - tstart = teiapsed. When teiapsed is greater than or equal to t (S209), then exit the loop and

let the program continue running.

When the process exits the loop, it means that (tnow - tstart) is greater or

equal to the residual time t. Otherwise, the loop keeps on running. In some cases,

it can happen that teiapsed (tnow - tstan ) is greater than the residual time t, that is, the

process waited longer than the residual time t.

An improvement of the process of Fig. 8 is shown in Fig. 9, in which

the same steps used in the Fig. 8 process are shown with the same reference

numerals. In the process of Fig. 9, the small difference δ which can occur

between teiapsed and the residual time t is considered in each call of the process.

First, the value of δ is set to zero in S200. Step S21 1 is added to the Fig. 8

process in which δ is computed. That is:

δ = teiapsed - t (Equation 3-2)

The value δ is stored in S21 3 for the next process call and subtracted in S21 0 from

the next residual time t (passed as input to the process at the next call of starting

the next tw cycle) .

The process of Fig. 9 dynamically compensates the rounding error by

modifying the residual time t with the error δ. In this way, the time spent between

sending one packet and the next tends toward tw on the average, notwithstanding

errors caused by rounding. The errors introduced in one cycle are compensated for

in the subsequent cycles. With this correction, if the wait time tw between packets (computed from P and BT) , which was in the process of Fig. 8 held constant during

the entire transmission, changes dynamically in order to compensate for previous

errors. During this period of error compensation the instantaneous bandwidth

usage can exceed the target bandwidth BT.

This process of error correction process reduces the precision of the

instantaneous bandwidth used in order to increase the precision of the mean

bandwidth BM used. The period for which the instantaneous bandwidth used

exceeds the target bandwidth BT depends on the rounding errors described above.

The use of a high resolution clock or counter will limit this period. For this reason,

this correction should be implemented only if it is possible and acceptable to exceed

the target bandwidth for brief periods. This completes the bandwidth control

process.

Considering the burstiness situation, it should be understood that if the

timing rounding errors are not compensated, then the lowest possible limit to the

burstiness is limited using the timing facilities provided by the sending device, but

the average bandwidth may be lower than the target bandwidth BT. If the timing

rounding errors are compensated, then the average bandwidth is as close as is

possible to the target bandwidth using the timing facilities provided by the sending

device, but the burstiness is greater than if error correction was not used.

Reliable protocols such as Transmission Control Protocol (TCP) send

additional packets for acknowledgement of receipt, and retransmission of lost or corrupted data packets. The sender device cannot typically determine the number

or size of additional packets sent, and so cannot control the bandwidth used. For

this reason the invention is of significant benefit principally when used with

unreliable packet based protocols.

The bandwidth control method and system of the invention can be

applied to contents of whatever size. It is not necessary for the sender to know

content size a priori. The only knowledge required up front is the size of the

packets hereafter designated as P, and the target bandwidth. The content size only

affects the transmission duration, and not the average bandwidth, if the process

and/or device used to wait for the paresidual time between the packets is precise

enough.

The method and system of the invention can be used in conjunction

with existing traffic shaping solutions. - Consider a network with many sender

devices, many receiver devices and a traffic shaper between senders and receivers.

If one or more of the sender devices produce bursty traffic, the traffic shaper will

intervene to contain the resulting congestion by discarding packets. The

introduction of the invention at one or more of the sender devices will reduce the

burstiness of the output streams and so reduce the likelihood of congestion and

packet loss.

Specific features of the invention are shown in one or more of the

drawings for convenience only, as each feature may be combined with other features in accordance with the invention. Alternative embodiments will be

recognized by those skilled in the art and are intended to be included within the

scope of the claims.

Claims

WE CLAIM:

1 . A method of using the wait time (tw) between transmission of

successive packets of known packet size (P) of a content to be transmitted to

achieve a target bandwidth (BT) during the transmission comprising the steps of:

selecting a target bandwidth (BT) sought to be achieved during the

transmission;

computing a wait time (tw) between successive packets of the

transmission using the algorithm

^v = - _ ^{■ and} controlling the transmission of the packets using the wait time.

2. The method as claimed in claim 1 wherein the computed wait

time tw that is used is rounded to a time unit.

3. The method as claimed in claim 2 wherein the rounding to the

time unit is accomplished by a counter.

4. The method as claimed in claim 1 further comprising the step

of:

determining the start time ti of transmission of a packet;

determining the end time t₂ of transmission of the packet, and

determining the time used tused in transmitting the packet as t₂ - ti .

5. The method as claimed in claim 4 further comprising the steps

of:

(a) determining the time used (tused) in the transmission of a packet;

(b) determining a residual time (t) as tused - tw;

(c) waiting the time t between the end of transmission of one

packet to the start of transmission of the next packet.

6. The method as claimed in claim 5 further comprising the step of

repeating steps (a), (b) and (c) for each packet transmitted.

7. A method as in claim 5 wherein the residual time t is controlled

by:

determining a value of start time tstan, of sending a packet a current

time tnow;

performing a loop operation of: (a) Computing a time teiapsed = tnow - tstart,

(b) comparing teiapsed to the residual time t and transmitting the next

packet when the value of teiapsed > t.

8. The method as claimed in claim 7 further comprising the steps

of computing an error value δ = teiapse - t and subtracting the value δ from a later

supplied value of t.

9. A method as in claim 5 wherein the computed wait time tw that

is used is rounded to a time unit.

1 0. A method as in claim 9 wherein the rounding to the time unit is

accomplished by a counter.

1 1 . The method of claim 1 , including the additional step of selecting

the known packet size (P) of the packets to be transmitted.

1 2. The method of claim 1 wherein the known packet size (P) is

provided by an application.

1 3. Apparatus for using the wait time (tw) between transmission of

successive packets of a content to be transmitted to achieve a target bandwidth BT

during the transmission comprising:

a computer including

a program to control transmission of a content in packets of data;

means to input and receive parameters of the size (P) of the packets to

be transmitted and of the desired target bandwidth (BT) ;

processing means to calculate a wait time (tw) between successive

packets of the transmission using the algorithm

tw = -£- and

BT

control means to successively transmit the packets with the wait time

tw between the packets.

1 4. Apparatus as in claim 1 3 wherein said computer further

comprises:

means for determining the start time (ti) of transmission of a packet;

means for determining the end time (t₂) of transmission of the packet,

and

means for determining the time used (tused) in transmitting the packet

1 5. Apparatus as in claim 14 wherein said computer further

comprises:

first means for determining the time used (tused) in the transmission of a

packet;

second means for determining a residual time t as tw - tused; and

wherein said control means operates to wait the residual time t

between the star of transmission of one packet to the start of transmission of the

next packet.

1 6. Apparatus as in claim 1 5 wherein said first and second

determining means operates to determine the residual time t for each packet

transmitted and said control means operates to wait the residual time t between the

start of transmission of one packet to the start of transmission of the next packet.

1 7. Apparatus as in claim 1 6 further comprising means for controlling

the residual time t by

determining a value of start time tstart, and a current time tnow

performing a loop operation of:

(a) computing a time teiapsed = tnow - tstart, and (b) comparing teiapsed to the residual time t and transmitting the next

packet when the value of teiapsed > t.

1 8. Apparatus as in claim 1 7 further comprising means for

computing an error value δ = teiapsed - t and subtracting the value δ from a later

supplied value of t.

1 9. Apparatus as in claim 1 3 wherein said control means further

comprises a counter that operates on a periodic basis to measure the wait time tw.

20. Apparatus as in claim 1 3 wherein said computer operates said

control means to compute the wait time tw based on other measured times.