CA2271669A1 - Method and system for scheduling packets in a telecommunications network - Google Patents
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Abstract
A system and method for scheduling data packets in a telecommunications network whereby the data packets are received and stored in a first buffer means (16) at a node (2) in the network, the packets being representative of one or more sessions.
The system further includes a second buffer means (18) to which data packets are buffered on initiation of a trigger condition to avoid overflow in the first buffer means (16) during a congestion period. A packet switch in the network incorporating the system and method is also described.
The system further includes a second buffer means (18) to which data packets are buffered on initiation of a trigger condition to avoid overflow in the first buffer means (16) during a congestion period. A packet switch in the network incorporating the system and method is also described.
Description
Y
METHOD AND SYSTEM FOR SCHEDULING PACKETS IN A
TELECOMMUNICATIONS NETWORK
FIELD OF THE INVENTION
This invention relates to a method and system for scheduling packets in a telecommunications network and more particularly relates to a method and system for scheduling data packets in a telecommunications network that strives to meet predetermined quality of service objectives set by a service provider operating the telecommunications network.
BACKGROUND OF THE INVENTION
An increasing amount of traffic on packet switched networks is time-critical.
This traffic includes traditional real-time services such as telephony as well as new services including video on demand, multicast video and virtual reality applications.
There are also numerous non-time critical applications in which delay may still affect user perception of service quality, such as Web browsing and file transfer.
For such applications to be successful, networks must provide sufficient Quality of Service (QoS) to user sessions with respect to several quality measures such as packet delay and packet loss rates. Guarantees with respect to QoS can be achieved through a combination of effective admission control and real-time scheduling.
However, the services mentioned above typically have relatively bursty traffic profiles and at session set up, where a customer wishes to transmit data across a network, the customers are not usually able to accurately describe their traffic profile.
Therefore, unless peak bandwidth admission control is employed, it is inevitable that moments of transient congestion will occur in the network.
As discussed in "Hardware Implementation of Fair Queueing Algorithms for Asynchronous Transfer Mode Networks", by A. Varma and D. Stiliadis, IEEE
Communications Magazine, Volume 35, pp.54-68, December 1997, traffic scheduling systems should possess certain desirable characteristics which are summarised as follows:
(a) the system must be able to treat individual data traffic streams separately;
reo:fph:#31281.CAP 7 May 1999 (b) the system should be able to provide local delay guarantees for each traffic stream, thus providing implicit end-to-end delay guarantees.
(c) there must be efficient use of available bandwidth;
(d) the system must be simple to implement in order to enable hardware implementation so as to meet the requirements for high speed links;
(e) the system must be scalable in that it must be able to perform well over a large number of sessions over a wide range of link speeds, and (f) the system must be able to allocate bandwidth amongst traffic streams in a fair and reasonable manner.
Many previous and existing packet scheduling systems have been designed in an attempt to meet the above requirements two of which will be described briefly.
The first system is First In - First Out (FIFO) which is a simple system to implement.
In FIFO systems, ,each packet (or cell in cell-relay based protocols) is placed in an outgoing transmit buffer in the order of arnval of each packet or cell. The size of the transmit buffer determines the maximum delay that a packet will experience at the particular link. If the buffer is full when a packet arrives, the packet is simply discarded. While FIFO systems are simple to implement, they are unsuitable for support of real-time applications as they are unable to isolate flows and treat individual traffic streams separately and are not necessarily fair across all user sessions. The latter is of concern because a single high volume traffic stream may generate such large volumes of traffic as to cause severely degraded service for all other sessions on the link or in the network.
The second scheduling system relates to Fair Queueing (FQ) which was introduced by A. Demers, S. Keshav and S. Shenker in an article entitled "Analysis and Simulation of a Fair Queueing Algorithm", Proc. SIGCOMM'89, 1989. In Fair Queueing, the available bandwidth is shared equally among all competing sessions and therefore users with moderate bandwidth requirements are not penalised because of the excessive demands of other users. As a result of equalizing the bandwidth, these Fair Queueing schemes typically provide satisfactory QoS for sessions whose bandwidth requirements are less than their fair share. The concept of Fair Queueing has since been enhanced to allow for weighted assignment of bandwidth, called reo:fph:#31281.CAP 7 May 1999 Weighted Fair Queueing (WFQ), which allows more general access to available bandwidth by traffic streams. In the WFQ scheme a scheduler may assign different weights to each session. This approach has yielded a vast range of adaptations each seeking to improve on minor perceived shortcomings in fairness, implementation detail or scalability in the original WFQ approach.
However, in all of these variants, fairness is an intrinsic goal of the scheduling discipline and the idea of fairness is applied on a very small time scale. The focus on fairness and small time scales tends to obscure the fact that fairness is an attempt at inferring user QoS from quantities that may be directly measured by the network, rather than an actual measure of the user perceived QoS. Fairness is not a direct measure of user perceived QoS because a user is unable to determine the quality of service received by other users nor the bandwidth received by other users.
Fairness is a measure of how the network assigns resources during periods of congestion, a measure that the user is not able to directly perceive. Fairness is not a guarantee of good service quality, for example, a given link may be carrying two real-time sessions and both simultaneously have a burst at a data rate slightly over half of the link capacity. If each session is allocated half of the available bandwidth, the queues for both sessions will grow until no packets arrive within their delay limits.
Therefore the effective throughput is zero and both users are dissatisfied. However, the system has ample capacity to provide one customer with the required QoS and by doing so may not make the other customer less satisfied than in the "fair" case.
When the system is lightly loaded, the QoS requirements of all sessions can usually be met. However, during periods of congestion, some or all sessions will receive service which does not meet their QoS requirements. Hence, it is necessary to decide which traffic streams should receive a degraded service, either in terms of loss, delay or both. This is a particular problem of real-time bandwidth scheduling.
It is considered that the more relevant measures of QoS are the proportion of customers that are receiving poor service at any time during their session or during a congestion episode because this reflects the number of customers that might complain to the service provider. The service provider will be motivated by minimising the number of complaints they receive from their customers concerning their perceived reo:fph:#31281.CAP 7 May 1999 QoS. Therefore, the network and the scheduling systems it employs should attempt to maximise the number of users that are receiving acceptable service, rather than strive for a fairness of which the customers are unaware. This is analogous to the reasons for implementing network admission controls to protect user QoS for sessions in progress.
However such a scheduling system should provide the mechanisms to enforce fairness as well if the service provider deems it important. In this way, decisions on the desirability of fairness in a network may be left to the service providers as a business decision.
Scheduling systems such as FIFO and FQ do not achieve the abovementioned aims. In FIFO systems a single misbehaving session can result in all traffic streams receiving poor service which is measured by the fraction of packets discarded or excessively delayed per traffic stream. In FQ systems, if all traffic streams burst simultaneously, they are all treated equally and all receive degraded service.
It is considered that fairness on very small time scales is not necessarily a good measure of the performance of a scheduling system for real-time services such as multi media data, motion pictures expert group (MPEG) video streaming and Internet protocol (IP) telephony, nor necessarily for non-real time service such as file transfer (FTP) and Web browsing (HTTP).
SUMMARY OF THE INVENTION
It is an object of the invention to provide a scheduling infrastructure which allows service providers and network operators to select and implement their own QoS objectives. A new approach to the management of transient congestion is presented by the invention. Rather than aiming to maximise fairness and letting good performance follow, the present invention provides an infrastructure that allows selectable QoS measures, including maximising the number of users receiving good service. This aim is achieved by selecting certain sessions to bear the brunt of degraded services during periods of congestion. On short time scales, the approach may be very unfair, giving very poor service to a small number of sessions in order to provide sufficient resources for the remaining sessions. However, the invention provides a packet scheduling system wherein fairness can be obtained over moderate time scales, of the order of seconds, and still provide substantially better overall reo:fph:#31281.CAP 7 May 1999 quality of service than FQ systems. The present invention supports a range of QoS
goals, from traditional concepts of fairness through to maximising the number of users receiving excellent service over their entire session. This is not addressed by any existing scheduling discipline such as FIFO or FQ.
METHOD AND SYSTEM FOR SCHEDULING PACKETS IN A
TELECOMMUNICATIONS NETWORK
FIELD OF THE INVENTION
This invention relates to a method and system for scheduling packets in a telecommunications network and more particularly relates to a method and system for scheduling data packets in a telecommunications network that strives to meet predetermined quality of service objectives set by a service provider operating the telecommunications network.
BACKGROUND OF THE INVENTION
An increasing amount of traffic on packet switched networks is time-critical.
This traffic includes traditional real-time services such as telephony as well as new services including video on demand, multicast video and virtual reality applications.
There are also numerous non-time critical applications in which delay may still affect user perception of service quality, such as Web browsing and file transfer.
For such applications to be successful, networks must provide sufficient Quality of Service (QoS) to user sessions with respect to several quality measures such as packet delay and packet loss rates. Guarantees with respect to QoS can be achieved through a combination of effective admission control and real-time scheduling.
However, the services mentioned above typically have relatively bursty traffic profiles and at session set up, where a customer wishes to transmit data across a network, the customers are not usually able to accurately describe their traffic profile.
Therefore, unless peak bandwidth admission control is employed, it is inevitable that moments of transient congestion will occur in the network.
As discussed in "Hardware Implementation of Fair Queueing Algorithms for Asynchronous Transfer Mode Networks", by A. Varma and D. Stiliadis, IEEE
Communications Magazine, Volume 35, pp.54-68, December 1997, traffic scheduling systems should possess certain desirable characteristics which are summarised as follows:
(a) the system must be able to treat individual data traffic streams separately;
reo:fph:#31281.CAP 7 May 1999 (b) the system should be able to provide local delay guarantees for each traffic stream, thus providing implicit end-to-end delay guarantees.
(c) there must be efficient use of available bandwidth;
(d) the system must be simple to implement in order to enable hardware implementation so as to meet the requirements for high speed links;
(e) the system must be scalable in that it must be able to perform well over a large number of sessions over a wide range of link speeds, and (f) the system must be able to allocate bandwidth amongst traffic streams in a fair and reasonable manner.
Many previous and existing packet scheduling systems have been designed in an attempt to meet the above requirements two of which will be described briefly.
The first system is First In - First Out (FIFO) which is a simple system to implement.
In FIFO systems, ,each packet (or cell in cell-relay based protocols) is placed in an outgoing transmit buffer in the order of arnval of each packet or cell. The size of the transmit buffer determines the maximum delay that a packet will experience at the particular link. If the buffer is full when a packet arrives, the packet is simply discarded. While FIFO systems are simple to implement, they are unsuitable for support of real-time applications as they are unable to isolate flows and treat individual traffic streams separately and are not necessarily fair across all user sessions. The latter is of concern because a single high volume traffic stream may generate such large volumes of traffic as to cause severely degraded service for all other sessions on the link or in the network.
The second scheduling system relates to Fair Queueing (FQ) which was introduced by A. Demers, S. Keshav and S. Shenker in an article entitled "Analysis and Simulation of a Fair Queueing Algorithm", Proc. SIGCOMM'89, 1989. In Fair Queueing, the available bandwidth is shared equally among all competing sessions and therefore users with moderate bandwidth requirements are not penalised because of the excessive demands of other users. As a result of equalizing the bandwidth, these Fair Queueing schemes typically provide satisfactory QoS for sessions whose bandwidth requirements are less than their fair share. The concept of Fair Queueing has since been enhanced to allow for weighted assignment of bandwidth, called reo:fph:#31281.CAP 7 May 1999 Weighted Fair Queueing (WFQ), which allows more general access to available bandwidth by traffic streams. In the WFQ scheme a scheduler may assign different weights to each session. This approach has yielded a vast range of adaptations each seeking to improve on minor perceived shortcomings in fairness, implementation detail or scalability in the original WFQ approach.
However, in all of these variants, fairness is an intrinsic goal of the scheduling discipline and the idea of fairness is applied on a very small time scale. The focus on fairness and small time scales tends to obscure the fact that fairness is an attempt at inferring user QoS from quantities that may be directly measured by the network, rather than an actual measure of the user perceived QoS. Fairness is not a direct measure of user perceived QoS because a user is unable to determine the quality of service received by other users nor the bandwidth received by other users.
Fairness is a measure of how the network assigns resources during periods of congestion, a measure that the user is not able to directly perceive. Fairness is not a guarantee of good service quality, for example, a given link may be carrying two real-time sessions and both simultaneously have a burst at a data rate slightly over half of the link capacity. If each session is allocated half of the available bandwidth, the queues for both sessions will grow until no packets arrive within their delay limits.
Therefore the effective throughput is zero and both users are dissatisfied. However, the system has ample capacity to provide one customer with the required QoS and by doing so may not make the other customer less satisfied than in the "fair" case.
When the system is lightly loaded, the QoS requirements of all sessions can usually be met. However, during periods of congestion, some or all sessions will receive service which does not meet their QoS requirements. Hence, it is necessary to decide which traffic streams should receive a degraded service, either in terms of loss, delay or both. This is a particular problem of real-time bandwidth scheduling.
It is considered that the more relevant measures of QoS are the proportion of customers that are receiving poor service at any time during their session or during a congestion episode because this reflects the number of customers that might complain to the service provider. The service provider will be motivated by minimising the number of complaints they receive from their customers concerning their perceived reo:fph:#31281.CAP 7 May 1999 QoS. Therefore, the network and the scheduling systems it employs should attempt to maximise the number of users that are receiving acceptable service, rather than strive for a fairness of which the customers are unaware. This is analogous to the reasons for implementing network admission controls to protect user QoS for sessions in progress.
However such a scheduling system should provide the mechanisms to enforce fairness as well if the service provider deems it important. In this way, decisions on the desirability of fairness in a network may be left to the service providers as a business decision.
Scheduling systems such as FIFO and FQ do not achieve the abovementioned aims. In FIFO systems a single misbehaving session can result in all traffic streams receiving poor service which is measured by the fraction of packets discarded or excessively delayed per traffic stream. In FQ systems, if all traffic streams burst simultaneously, they are all treated equally and all receive degraded service.
It is considered that fairness on very small time scales is not necessarily a good measure of the performance of a scheduling system for real-time services such as multi media data, motion pictures expert group (MPEG) video streaming and Internet protocol (IP) telephony, nor necessarily for non-real time service such as file transfer (FTP) and Web browsing (HTTP).
SUMMARY OF THE INVENTION
It is an object of the invention to provide a scheduling infrastructure which allows service providers and network operators to select and implement their own QoS objectives. A new approach to the management of transient congestion is presented by the invention. Rather than aiming to maximise fairness and letting good performance follow, the present invention provides an infrastructure that allows selectable QoS measures, including maximising the number of users receiving good service. This aim is achieved by selecting certain sessions to bear the brunt of degraded services during periods of congestion. On short time scales, the approach may be very unfair, giving very poor service to a small number of sessions in order to provide sufficient resources for the remaining sessions. However, the invention provides a packet scheduling system wherein fairness can be obtained over moderate time scales, of the order of seconds, and still provide substantially better overall reo:fph:#31281.CAP 7 May 1999 quality of service than FQ systems. The present invention supports a range of QoS
goals, from traditional concepts of fairness through to maximising the number of users receiving excellent service over their entire session. This is not addressed by any existing scheduling discipline such as FIFO or FQ.
5 Accordingly, there is provided in a first aspect of the invention a method for scheduling data packets in a telecommunications network, said method comprising the steps of receiving and storing in a first buffer means at a node in the network packets of data representative of one or more sessions;
monitoring the used capacity or occupancy of packets stored in said first buffer means;
buffering data packets in a second buffer means at said node on initiation of a trigger condition, so as to avoid the first buffer means from overflowing during a congestion period.
The trigger condition may be a derivative-based threshold such that when the rate of increase of the amount of data stored in the first buffer means exceeds said derivative-based threshold, data packets are buffered in the second buffer means.
Alternatively, the trigger condition may be the crossing of a first or subsequent capacity threshold in the first buffer means due to the arrival of a data packet in the first buffer means. The first data packet in the first buffer means that crosses the first or subsequent capacity threshold may be buffered in the second buffer means and any subsequently arnving packets of a session to which the first data packet belongs may also be buffered in the second buffer means.
The invention fizrther provides in a second aspect a system for scheduling data packets in a telecommunications network, said system comprising;
first buffer means located at a node in the network for receiving data packets representative of one or more sessions;
a second buffer means located at said node in the network;
wherein data packets are buffered in the second buffer means on initiation of a trigger condition so as to avoid overflow in the first buffer means during a congestion period.
reo:fph:#31281.CAP 7 May 1999 According to a third aspect of the invention there is provided a data packet scheduling system for scheduling packets in a telecommunications network, in accordance with a quality of service standard set by a telecommunications service provider wherein said packets are representative of one or more sessions, said system comprising;
a first buffer means for receiving and temporarily storing data packets of one or more sessions transmitted in the telecommunications network;
a second buffer means wherein data packets are buffered in said second buffer means upon initiation of a trigger condition set in the first buffer means due to the arrival of a data packet in said first buffer means such that the first buffer means is prevented from overflowing during a congestion period in the network.
BRIEF DESCRIPTION OF THE DRAWINGS
Preferred embodiments will hereinafter be described, by way of example only, with reference to the following drawings wherein;
Figure 1 is a schematic diagram of a packet switch or node of a telecommunications network implementing the system and method of the present invention according to a first embodiment;
Figure 2 is a schematic diagram showing the buffering of data packets in a second buffer means intended for a first buffer means;
Figure 3 is a schematic diagram showing the transfer of data packets from a second buffer means to the first buffer means for transmission on an output link from the first buffer means;
Figure 4 is a schematic diagram of a packet switch or node of a telecommunications network implementing the system and method of the present invention according to a second embodiment;
Figure 5 is a schematic diagram showing a first, second and third buffer means and the treatment of data packets arriving at the node of Figure 4;
Figure 6 hows a block diagram of a system used to simulate a number of traffic sources being input into a node having first buffer means and second buffer means;
Figure 7 s a series of plots showing the cumulative probability distributions of packet delay of a number of sessions using the FIFO, DRRexp and DQ
simulations;
reo:fph:#31281.CAP 7 May 1999 Figure 8a and b show a series of plots of the highest packet delay per second against the simulation time for a number of sessions with an increased link transmission rate using the FIFO, DRRexp and DQ simulations;
Figure 9 is a plot of a number of sessions receiving "good service" in a one second period for each of the FIFO, DRRexp and DQ simulations;
Figure 10 shows a plot for each of the DRRexp and DQ simulations of degraded service experienced by all sessions;
Figure 11 shows a plot for each of the DRRexp and DQ simulations showing the number of degraded seconds for each of the sessions shown in Figure 3 with a reduced line transmission rate;
Figure 12 is a plot similar to Figure 9 but with a still further reduced line transmission rate;
Figure 13 is similar to Figures 9 and 10 wherein degraded seconds are shown for a set of different sessions at a low line transmission rate;
Figure 14 is a plot showing the degraded seconds for a further set of different sessions using the DRRexp and DQ simulations at an increased line transmission rate;
Figure 15 is a plot of the overall percentage of degraded packets for each of the FIFO, DRR, DRRexp and DQ algorithms for different average offered loads that are normalised to the link transmission rate, 2 o Figure 16 is a flow diagram showing the processes involved on receipt of a packet of a session on a link associated with a node, Figure 17a and b are flow diagrams having the processes involved when a packet is transferred from the second buffer means to the first buffer means, and Figure 18 a and b are flow diagrams showing the processes involved in treating 2 5 one or more packets arriving at the third buffer means of Figure 5.
DETAILED DESCRIPTION
Shown in Figure 1 there is a packet switch or node 2 which forms part of a telecommunications network, such as a packet switching network. The switch 2 consists 30 of a number of input ports 4 and a number of output ports 6 and a switching fabric 8 that is connected between the input ports 4 and the output ports 6. A
routing processor 10 is linked to the switching fabric 8 over link 12 which is responsible for the routing or switching of data packets between any one of the input ports 4 to any one of the output ports 6. Data packets that are input to the switch 2 are received by any one of the input ports 4 on input links 14. The data packets are then switched by the switching fabric 8 to one of the output ports 6. The data packets are then either received by a first buffer means 16 or a second buffer means 18 located in each one of the output ports 6 depending on how those input packets are to be treated, which will be hereinafter described. All data packets, apart from discarded packets, leave the output ports 6 from each of the first buffer means 16 on links 15.
Alternatively, the first buffer means 16 and second buffer means 18 may be located in each of the input ports 4.
Shown in Figure 2 there is a more detailed view of the first buffer means 16 and the second buffer means 18. When the system or network, or even a switch of the network is not congested, all sessions (a session being a series of packets that are dedicated to one particular user for one application) are received by the first buffer means 16 which is preferably in the form of a single short FIFO queue, and which is denoted as the a-queue. The length of the a-queue is selected such that any packet passing straight through the a- queue will meet the tightest delay requirements of any particular session. The length of the a-queue corresponds to the longest delay that can be considered "good service" in terms of delay. At the onset of congestion, when the level of the a- queue rises, due to the arrival of a packet from a particular session, and exceeds a capacity threshold T o,~et,l and it is decided that the packets from this session contribute to the possibility of the a-queue overflowing, this packet and subsequent packets from the session are time stamped and buffered in the second buffer means 18, hereinafter referred to as the 13-queue. The exceeding of the capacity threshold Tor~ec,l represents a trigger condition under which a fraction of the packets intended to be received in the a-queue are buffered in the 13-queue. The 13-queue may be significantly longer than the a-queue. As subsequent capacity thresholds, T
o~et~, j >1, are crossed then further sessions are also buffered in the 13-queue. The length of the reo:fph:#31281.CAP 7 May 1999 a-queue, and the settings of each threshold To,~et~~j>1 are constituted by memory size, in bytes, such that as each threshold is crossed a predetermined amount of memory in the a-queue is used by the data packets stored therein ready for transmission on an output link.
There is a wide range of possible algorithms that may be used for selecting those sessions or packets of sessions that are to be transferred either from the a-queue to the 13-queue or buffered in the 13-queue before arriving in the a-queue depending on the properties desired. Such algorithms include the following:
(a) Random:
A session is chosen randomly from those currently in progress. Such a selection attempts to spread performance degradation across all sessions over their lifetime, but not instantaneously across all sessions. The packets of the sessions may be chosen randomly in accordance with, for example, random numbers generated in the system that match a specific count number associated with the packets.
(b) Fair:
The session with the most packets in the a-queue at the time the threshold is crossed is chosen. Such a choice attempts to penalise those sessions whose instantaneous bandwidth requirement is higher than the fair share. It attempts to be instantaneously fair in deciding which sessions to penalise but has the benefit of minimizing the number of sessions simultaneously affected. In this case, each packet of each session stored in the a-queue is identified and recorded by the a-queue and subsequently counted.
(c) Next Packet:
The session of the packet which first breaks the first or subsequent capacity thresholds which is not already in the 13-queue or the session of the packet which breaks the first capacity threshold is chosen. Such a choice is a mix between being fair (the next packet arriving is likely to be from a session with the highest instantaneous bandwidth) and randomising sessions to be penalised.
It has the added advantage of being the simplest algorithm to implement.
reo:fph:#31281.CAP 7 May 1999 (d) Historical:
A session that has been in the 13-queue at some stage in the past, but is now in the a-queue is selected. Such a decision attempts to keep penalising sessions that have already been penalised. It minimises the amount of inconvenience 5 experienced by other sessions during the entire duration and thus attempts to maximise the number of users receiving good service over their entire session.
The above algorithms may be implemented in either software or hardware.
Shown in Figure 3 are a number of sessions that have been stored in the second buffer 18 or the 13-queue. A time-out threshold, teXp~.e, is set and all packets that have 10 been in the 13-queue longer than te,~ue are discarded. For example in the 13-queue of Figure 3, the oldest packets stored therein are discarded in the order that they were input to the 13-queue. If the 13-queue overflows, due to the packets stored therein exceeding the memory capacity of the 13-queue, the system may either drop newly arriving packets from various sessions or push out or discard the oldest packet in the 13- queue. The latter approach attempts to minimise the age of the packets in the queue and therefore maximise the chance that the transmitted packets will meet their end to end delay requirements.
A number of time-out thresholds, te,~,;re, ~, may be set depending on the types of data traffic stored in the second buffer means 18. For example, if the data traffic consists of video or voice packets, a small teXpre will be set, because a delayed packet in a real time stream is of very limited value and will only serve to delay subsequent packets. At the other extreme, packets may be representative of a file transfer from a server on the Internet. In this case, the texpire may be larger as the information need not be delivered in real time. Therefore, a slight delay is more acceptable than losing a packet, which would then require retransmission.
When the length or occupancy by packets of the a-queue drops to an abatement threshold, TabateW'herein the memory presently being used in the a-queue drops to a specified size, then packets from the 13-queue are moved back to the a-queue. Packets from a session cease being redirected to the 13-queue when there are no packets from that session left in the 13-queue. All packets moved from the 13-queue to reo:fph:#31281.CAP 7 May 1999 the a-queue are marked as "delayed". There are a number of algorithms for deciding which packets are moved first from the 13-queue to the a-queue, these being:
(a) FIFO:
This is the simplest 13-queue algorithm to implement and simply moves the packets to the a-queue in order of arrival in the 13-queue when the length of the a-queue drops to Tabate- The number of packets from each session in the 13-queue must be recorded so that the a-queue will know when to stop redirecting a session. For example, in session 20 which is the most recent session stored in the 13- queue there are 3 packets so that when the last or third packet of session 20 is transferred to the a-queue, the a-queue or the first buffer means 16 will be notified that it is the last packet in that session. Therefore any subsequent packets for session 20 that are input to the a-queue will only be transmitted after the first 3 packets of session 20 have been transmitted or output from the a-queue. This recordal also applies to the algorithms described under (b), (c) and (d) below.
(b) FIFO - FIFO:
This is where each session transferred to or buffered in the 13-queue is maintained separately. When the a-queue drops to below Tabate~ packets from the first session placed in the 13-queue are moved to the a-queue, that is, sessions are handled in a first in - first out order. The packets from this session are given priority over other sessions contained in the 13-queue. Within a session, packets are also transferred in first in - first out order to preserve sequencing within a session.
(c) LIFO - FIFO:
Each session transferred to or buffered in the 13-queue is maintained separately.
When the a-queue drops to below Tabate~ packets from the session most recently placed in the 13-queue are moved to the a-queue, that is, sessions are handled in a last in - first out fashion. The packets from this session are given priority over other sessions contained in the 13-queue. Within a session however, packets are still transferred in first in - first out order to preserve reo:fph:#31281.CAP 7 May 1999 sequencing within a session. Such an algorithm attempts to maximise the number of sessions receiving acceptable delay performance at any time.
(d) Fair:
Packets are transferred from the 13-queue to the a-queue using a Fair Queueing approach when the length of the a-queue is less than or equal to Tabale~ Thus, the sessions in the 13-queue use the residual bandwidths left over by the sessions still using the a-queue. Such a policy attempts to be fair to those sessions currently being penalised.
It is to be noted that all packets leave the output port 6 via the a-queue.
Packets that have gone through the second buffer means 18 are marked to allow for identification by other nodes or switches in the network. At subsequent nodes these packets may be the first choice for buffering in the second buffer means 18 during a congestion period. In this manner, the network can co-ordinate which sessions are penalised if congestion occurs simultaneously at more than one point in the network. For example, if a session has been buffered in the second buffer means 18 at one node, then the first packet or subsequent packets transmitted from that session to another node or nodes in the network will have an indication, in its header, that the session to which it belongs has been buffered previously. Thus other nodes may choose to continue buffering that session.
Another embodiment of the present invention is shown in Figures 4 and 5. The input ports 4, output ports 6 and switching fabric 8 are represented as in Figure 1. The first buffer means 16 is in the form of an output queue and receives data packets representative of one or more sessions from either the second buffer means 18 or a third buffer means 19. Each of the second buffer means 18 and third buffer means 19, in this alternative embodiment, comprise a series of per-session queues 21.
That is data packets from each session are received and temporarily stored in individual queues 21. It is to be noted that the second buffer means 18 and third buffer means 19 are simply subsets of a single collection of per-session queues 21. These queues 21 can be dynamically reclassified as either part of the second buffer means or the third buffer means 19. Generally the packets stored in the per-session queues of third buffer means 19 are transferred to the first buffer means 16 according to the scheduling reo:fph:#31281.CAP 7 May 1999 discipline chosen, such as FIFO or Fair Queueing as discussed previously. The packets in the per-session queues of the buffer means 19 are polled or read at a rate significantly higher than the rate at which packets are read from the first buffer means 16 onto an output link 15. Packets in the first buffer means 16 are then transmitted in FIFO order onto output link 15. The second and third buffer means 18, 19 may be implemented as logical queues. As with the first embodiment, at the onset of congestion, when the level of the first buffer means rises due to the arrival of packets from the third buffer means one or more capacity thresholds To,~et> ; J > 1 may be exceeded. If a particular packet arriving in buffer means 16 causes the capacity of the buffer means 16 to exceed a first threshold To,~et,l, this represents a trigger condition under which the session belonging to that packet is subsequently buffered in the second buffer means 18 into a respective queue in the buffer means 18. This will assist in preventing the first buffer means 16 from overflowing. As subsequent capacity thresholds are crossed, further sessions are buffered in the second buffer means 18. The buffering of data packets from various sessions is done according to any one of the algorithms mentioned earlier, that is either random, fair, next packet or historical algorithms.
A time-out threshold teXp;re is set for those packets that have been buffered in the second buffer means so that any packets buffered longer than texpue are discarded.
Different time-out thresholds teXp;res may be set depending on the type of data stream buffered, such as voice, video or web-browsing as with the previous embodiment.
When the length of the first buffer means drops to an abatement threshold, Tabate~ packets are moved from the second buffer means 18 to the first buffer means 16.
This may be performed by the previous algorithms mentioned with respect to the first embodiment, such as FIFO, FIFO-FIFO or LIFO-FIFO but preferably the LIFO-FIFO
algorithm is used whereby the most recent session is transferred from buffer means 18 to buffer means 16. Each packet within that most recent session is transferred in the order in which they entered the respective per-session queue so that sequencing is maintained.
In a filrther embodiment, the third buffer means may act as the first buffer means wherein packets of sessions would be output onto output link 15 directly from reo:fph:#31281.CAP 7 May 1999 the per-session queues 21. In this embodiment there is no need for an output buffer as with the previous embodiment. The capacity thresholds, To~e~, would be based on the current logical contents of the third buffer means, that is, the sum of the capacity taken up by individual sessions in each of the per-session queues 21. Therefore any packets of particular sessions that cross the thresholds would be treated according to any one of the previously mentioned situations.
Some simulation experiments have been performed and the results of those experiments will be hereinafter described. The simulation set up is shown in Figure 6 where a number of traffic sources 1 to N, identified by numeral 22, generating MPEG
data streams originating from video servers are multiplexed together by a node or switch 24 onto a single data link 26 to be transmitted to sink 28. The sink 28 acts as a discarder for packets in the simulation. Any other type of server from which higher volume traffic may be multiplexed may also be used in this example.
The frame sizes from real MPEG traces generated by O. Rose in the publication "Statistical Properties of MPEG Video Traffic and their Impact on Traffic Modelling in ATM Systems", in Proceedings of the 20th annual conference on local computer networks, 1995, pp. 397-406 (MPEG traces at ftp: //ftp-info3.
informatik.
uni-wuerzburg.de/pub/MPEG), are used for sources in the simulation. Table 1 summarises the characteristics of the MPEG sources. All sources have a peak rate of 4 Mbps at 25 frames per second and are 1600 seconds in duration (except for the newsl trace which is 1260 seconds). They are randomly started over a 1600 second interval and measurements are taken between 1600 seconds and 3200 seconds.
When an MPEG trace comes to the end it is cycled. The frames are broken into 1500 byte packets for transmission with a 40 byte minimum packet size.
reo:fph:#31281.CAP 7 May 1999 No. Name Mean bits/frame Peak/Mean Frame Number Size of Frames 1 asterix 22,349 6.6 40000 2 apt 21,890 8.7 40000 3 bond 24,308 10.1 40000 4 dino 13,078 9.1 40000 S lambs 7,312 18.4 40000 6 movie 14,288 12.1 40000 7 mrbean 17,647 13.0 40000 8 mtv 1 24,604 9.3 40000 9 mtv 2 19,780 12.7 40000 10 newsl 20,664 9.4 31515 11 news2 15,358 12.3 40000 12 race 30,749 6.6 40000 13 sbowl 23,506 6.0 40000 14 simpsons 18,576 12.9 40000 15 soccerl 27,129 6.9 40000 16 soccer2 25,110 7.6 40000 17 starwars 9,313 13.4 40000 18 talkl 14,537 7.3 40000 19 talk2 17,914 7.4 40000 terminator 10,905 7.3 40000 reo:fph:#31281.CAP 7 May 1999 The following measurements are collected for each session:
- End to end delay for each packet - Actual packets lost S - Degraded packets (packets delayed > SOms) - Degraded frames (frames with degraded or lost packets) - Degraded seconds (one second intervals with degraded frames).
A session is receiving "good service" during any particular second if no frames of the session have been degraded. Any session not receiving "good service" is deemed to be receiving "degraded service".
The method of the present invention is implemented in these simulation experiments to provide as many sessions as possible with good service during a particular transient congestion period. The second buffer means 18 or the 13-queue follows the LIFO - FIFO principle earlier described. Packets from the f3-queue are transferred to the a-queue only when the a- queue is empty, in other words when Tabate equals 0. The thresholds for redirecting packets to the 13-queue, To~et~ are set at L-L/(3 +j-1) for j>_ 1 where L is the length of the a-queue in the simulation.
Sources are removed from the redirection list when there are no packets from that source (or session) in the 13-queue. In a real network with a changing number of sessions in progress, the threshold assignment can be made dynamic by simply replacing the number of sessions in the simulation with the number of sessions in progress.
The packet time out threshold texpire is set to 400ms.
In this set of experiments the simplest method of selecting which sessions are redirected to the 13-queue is chosen. That method is the next packet method where the session of the first packet to break To"Set~ is chosen to be redirected to the 13- queue.
The 13-queue uses an oldest packet push out mechanism on overflow of the f3-queue.
In Table 2 there is listed a set of simulation experiments used to compare the performance of the algorithm or method of the present invention, termed dual queue (DQ), with the existing FIFO and Fair Queueing methods. A deficit round robin (DRR) technique is used for the Fair Queueing comparison, but is enhanced to include old packet discard (DRRexp) to make a fairer comparison with DQ.
reo:fph:#31281.CAP 7 May 1999 ExperimentNumber Offered MPEG Buffer Transmission Number of load (Mbps)Sources Space rate (Mbps) Sources used (Bytes) 1 20 9.5 all 20 500x103 8,9,10 & 12 2 4 1.6 1,2,3 & 100x103 1.3,1.65, 5 &
3 40 19 all twice 1x106 16, 20 & 24 4 20 30.7 race 1.5x106 24,30 &
A DRR technique was introduced by M. Shreedhar and G. Varghese in a document titled "Efficient Fair Queueing Using Deficit Round Robin", IEEE/ACM
Trans. Networking, Volume 4 No. 3, 1996, pp. 375-385.
As mentioned previously a packet is degraded if it is delayed more than SOms.
This is used as an estimate of the maximum delay allowed before a frame cannot be displayed to a viewer or user. Packets that have been queued for longer than te,~pue =
400ms are discarded in the DRRexp and DQ algorithms. However, an MPEG frame that arrives too late for display may still be useful for the decoding of future frames which may not be so badly delayed.
All queues are given the same total buffer size and kept constant for the bit rates used in each simulation experiment. The a-queue of the DQ implementation has a size that is changed whenever the link transmission speed is changed so as to give a maximum packet delay of 50 ms. It should also be noted that the worst case service delay overhead in DRR introduced by the round robin scheduling is less than the good service delay threshold of SOms.
In experiment 1 shown in Table 2 all 20 heterogeneous MPEG sources are multiplexed onto link 26. Four representative sessions, being numbers 3, 10, 14 and have arbitrarily been selected to give a detailed comparison between the FIFO, 20 DRRexp and DQ scheduling algorithms.
Figure 7 shows graphs of the cumulative probability distributions of packet delay for session 3 (represented by graph 30), session 10 (represented by graph 32), session 14 (represented by graph 34) and session 20 (represented by graph 36) when reo:fph:#31281.CAP 7 May 1999 the link transmission rate is lOMbps (compared with the total average offered rate of 9.SMbps). Dropped packets are considered to have infinite delay in Figure 5. A
highly loaded case has been chosen in order to exaggerate the difference between the algorithms during congestion. A line is drawn at the 50 ms good service cut off point.
It is seen that the DQ algorithm provides the highest fraction of packets receiving a delay of less than SOms. This is achieved by giving some packets a delay much greater than SOms. Given that the service provided to such packets is already considered degraded, the amount by which the delay exceeds SOms is considered to be of little relevance. Note that DRRexp gives better service to session number 20 than the other sessions as it is usually operating at less than its fair share of the bandwidth as expected from a Fair Queueing based discipline. Note that this is also true of the DQ algorithm.
In Figure 8 there is shown a plot of the highest packet delays per second for each of the sources or sessions 3, 10, 14 and 20 against the simulation time where the transmission rate on link 26 has been increased to 12 Mbps. Plots 38, 40, 42 and 44 are for the respective sessions 3, 10, 14 and 20. A line is drawn at SOms mark to indicate the good service cut off point. The time range as shown is chosen so as to include a number of transient congestion periods for comparison between the different algorithms. The FIFO is fair at the packet level and all sessions' packets, on average, experience the same delay passing through the node 24 on a FIFO basis. The DRRexp performs a little better but fails to deliver good service when the bandwidth has to be shared in such a way that many of the sessions receive less than their required service.
The priority nature of the FQ can cause longer delays to a burst of packets than encountered in a FIFO scheme. Therefore these extra delays can cause packets from a particular session to become degraded even when the link is not congested as is illustrated in the top trace 38 of source 3 in Figure 8. The DQ discipline selected tries to maximise the instantaneous number of sessions receiving good service so we see that one of the sessions receives very poor delay performance, so that the other sessions are spared service degradation, at times of 1925 seconds and 1975 seconds.
A better illustration of the level of QoS by all of the 20 sessions is illustrated in Figure 9 which shows a plot 46 of the percentage of all sessions simultaneously reo:fph:#31281.CAP 7 May 1999 receiving good service during similar one second periods. It is to be noted that the FIFO and DRRexp algorithms present degraded service to many sessions at times of 1855 seconds, 1925 seconds and 1975 seconds. These QoS figures can be represented in binary form where service is either good or degraded. A binary representation of service per session is generally all that is required for comparisons between each of the scheduling disciplines. Figure 10 shows such a representation 48 for the DRRexp algorithm and a representation 50 for the DQ algorithm over the full measurement time scale for each session at a link transmission rate of l2Mbps. A vertical bar is drawn for every degraded second experienced by any particular session. The more white space on the graph, the better the service a session is receiving. All the results that are shown in subsequent figures will use this method of comparison. From Figure 8 we see that the DQ approach results in fewer sessions being affected by congestion than DRRexp at any particular time.
As the transmission rate is reduced to lOMbps, shown in Figure 1 l, and 8Mbps in Figure 12 the contrast between DRRexp and DQ becomes even more evident. The plots 52 and 54 are respectively for the DRRexp and DQ algorithms in Figure 11 and the plots 56 and 58 are respectively for the DRRexp and DQ algorithms in Figure 12.
It is apparent that the DQ approach provides good service to most of the sessions even under extreme overload conditions as compared to the DRRexp algorithm.
Although these experiments or simulations have provided the simplest possible implementation of the DQ approach, it is possible to provide significantly better performance for real-time services than can DRRexp.
Refernng to Table 2, experiments 2 and 3 use 4 and 40 sources respectively and are conducted to determine the scalability of DQ algorithm. Experiment 2 has the added complexity that the peak bit rates of any of the sessions is higher than the link transmission rate and therefore cause the a-queue to begin to fill. In Figure 13 four sources are used and it is evident that even when presented with an unmanageable load, the DQ algorithm depicted in plot 60 performs as well as the DRRexp algorithm depicted in plot 62, if not slightly better especially for sources 3 and 4. In experiment 3 both scheduling disciplines provided good performance almost all of the time and figures have not been provided in this case. For DRRexp, 13 sessions experienced reo:fph:#31281.CAP 7 May 1999 degraded service during a single congestion event whereas only one session experienced degraded service using the DQ algorithm during the equivalent period. In the DQ algorithm, there was only one second in which sessions had degraded service, whereas there was one other second in which degraded service occurred using the 5 DRRexp. It is therefore concluded that the DQ approach scales reasonably well when compared to DRR Exp.
To examine fairness over moderate time scales, 20 homogeneous sources were used in experiment 4. The sessions consisted of 20 copies of the race trace which started at random times over the period 0 to 1600 seconds. Figure 14 shows the 10 degraded seconds plots 64 and 66 respectively for the DRRexp and DQ
algorithms for each session where it is seen that DRRexp provides fair, although very degraded service to each session. Qualitatively, the DQ algorithm also provides fair service to each session over the entire length of the simulation rather than instantaneously. As seen earlier, the overall QoS of each session is significantly better in DQ
than that 15 provided by DRRexp.
Overall it is seen that when there is no congestion all of the schemes compared provide good service. However as the periods of congestion increase, a greater contrast can be seen between the 3 algorithms. In Figure 15 there is shown the plot 68 of the percentage of degraded packets for the FIFO, DRR, DRRexp and DQ
20 algorithms for different average offered loads (normalised to the link transmission rate). Apart from achieving its aim of giving good service to as many sessions as possible during transient congestion periods, the DQ algorithm also minimises the overall proportion of degraded packets. It is to be noted that the percentage of degraded packets even under the simple implementation of the DQ approach is close to the excess offered load above one Erlang, and hence is near optimal. While this would also be true for a short FIFO queue, the short FIFO queue has the disadvantage of resulting in unnecessary loss during transient congestion periods when the offered load is close to, but less than the transmission link speed. The short FIFO
queue also loses packets randomly rather than selectively as in DQ.
During transient congestion episodes, packets entering a normal FIFO queue all experience similar degraded service. If the congestion is caused by a small number of reo:fph:#31281.CAP 7 May 1999 sessions, Fair Queueing provides relief for the other sessions. However, results presented hereinbefore indicate that this is not always the case as transient congestion is often caused by many sessions whose combined traffic fills the queue. Fair Queueing divides bandwidth equally and therefore degrading the service of a large number of sessions and therefore a large number of packets. The DQ algorithm outperforms Fair Queueing because it degrades the service of only enough sessions so as to obtain good service for the rest of the sessions during the congestion period and therefore also increases throughput.
As mentioned previously, the service provider may wish to maximise the number of users receiving "perfect" service, that is, showing no sign of congestion so as to minimise the number of complaints received by users when they make a connection and receive unsatisfactory QoS. This can be achieved by recording in a "target" list or file the connections which have been redirected to the 13-queue at some stage, and increasing the probability that they will be redirected again the next time the 1 S system is congested. Therefore the degradation is not spread evenly among users, even over considerable time scales, increasing the probability of a connection remaining unaffected by the congestion. The probability of redirecting connections in the target list or file may be increased to one so that no new connection will be redirected until all connections which have previously been redirected are currently being redirected. Another alternative is to consider several factors including the current data rate as well as the presence of the connections in the target list or file.
If the aim of the service provider is to maximise the number of the satisfied customers at any moment in time, the optimal strategy is to redirect the highest rate sessions. If the statistics of a given connection change significantly with time, the above approach of repeatedly targeting a particular user for the entire connection may not be appropriate. Instead, sessions could be removed from the target list or file after some time, t~get. This would result in bursts of bad service on a time scale of tt~.get, but may improve the total number of customer minutes of high QoS.
Alternatively a telecommunications regulatory body may require a stricter guarantee of fairness over moderate time scales than that provided by the hereinbefore described DQ implementation. This is also possible again by recording those reo:fph:#31281.CAP 7 May 1999 connections or sessions which have been redirected. This time, these sessions will be less likely to be redirected again. Once again there is a trade-off with the total number of customer minutes of high QoS which can be achieved.
Even greater flexibility is possible if other service disciplines are considered for S the 13-queue rather than only the LIFO-FIFO system hereinbefore mentioned.
For example, the 13-queue could use Fair Queueing. Let L denote the set of low-rate users who require less than their fair share of the bandwidth. In pure Fair Queueing, sessions in L are allocated their full bandwidth and the residual bandwidth is allocated to the remaining sessions. In the hybrid DQ/FQ system, the set of sessions which have been selected to receive good services at time t, G(t), takes the role of L so that the residual bandwidth after serving users in G(t) is fairly divided. If G(t) = L for all t, or if the size of the a-queue is set to zero, then the hybrid approach reduces to Fair Queueing. Conversely, if G(t) is the set of all users, FIFO results. However, it is often possible to have G(t) a strict superset of L whose total instantaneous bandwidth is still less than the output link capacity, increasing the number of satisfied customers without penalising any well behaved users.
As an alternative to using threshold crossing based schemes for initiating the redirection of sessions from the a-queue to the 13-queue, there is a wide variety of possible trigger conditions. Some examples include the use of derivative based schemes such that when the rate of increase of the amount of data stored in the a-queue exceeds the derivative-based threshold, the data packets are redirected to the second buffer means. Fuzzy logic based approaches may also be used. Thus whenever a predetermined rate of increase of the amount of packets stored in the a-queue is reached, packets of a session or sessions are redirected to the f3-queue.
Shown in Figure 16 is a flow diagram 100 illustrating the steps or processes involved on receipt of a packet representative of a session on an output link to the output port 6 or node 24. At step 102 the received packet is time stamped to enable the buffers 16 and 1$ to place the packet in a particular sequence based on time. At step 104 a check is made to see if other packets associated with that session are in the 13-queue redirection list. If there are no previous packets in the 13-queue then at step 106 a decision is made as to whether or not the received packet will cross a capacity reo:fph:#31281.CAP 7 May 1999 threshold if it was placed in the a-queue. If the packet does not cross the capacity threshold it is placed in the a-queue at step 108 or if it will cross the capacity threshold a stream identifier or session identifier is placed in the I3-queue redirection list at step 110 and at step 112 the next capacity threshold is then used.
Thereafter, the packet is placed in the 13-queue at step 114. Going back to step 104, if there are previous packets from that session already in the 13-queue redirection list then the received packet is immediately placed in the 13-queue again at step 114.
In Figure 17 there is shown a flow diagram 200 illustrating the processes and steps involved when a packet is transferred from the 13-queue (or the second buffer means 18) to the a-queue (or the first buffer means 16). At step 202 a determination is made as to whether there are packets in the 13-queue and if there are none, then the process does nothing at step 204. If there are packets queued then at step 206 the oldest packet is retrieved from the most recent session redirected to the 13-queue (LIFO-FIFO). A determination is made at step 208 as to whether the packet is too old and has gone beyond te,~,ue. If this is the case then the packet is discarded at step 220 and a decision is made at step 222 as to whether that discarded packet was the last packet from that session. If it was the last packet, then at step 224 the session owning that packet is removed from the 13-queue redirection list and at step 226 the previous a-queue abatement threshold is then implemented. The process then reverts back to step 202. If the discarded packet was not the last packet from that session in the 13-queue, then the process reverts again to step 202. Referring back to step 208, if the packet is not too old and is within the time limits set by teXpue, the packet is then placed in the a-queue at step 210 and a determination is made at step 212 as to whether that placed packet is the last packet from the session in the 13-queue. If it is not the last packet then the process does nothing at step 214. If it was the last packet in the 13-queue belonging to that session, then at step 216 the session is removed from the 13-queue redirection list and at step 218 the previous a-queue abatement threshold is implemented.
Shown in Figure 18 is a flow chart 300 showing processes involved in placing packets of sessions from a third buffer means into a first buffer means and the steps reo:fph:#31281.CAP 7 May 1999 involved for dealing with such packets or sessions at the onset of congestion in the first buffer means (refer to Figures 4 and 5). Incoming packets are received at one of the output ports 6 in the third buffer means 19. At step 302 a check is made as to whether or not packets are temporarily stored in the third buffer means in any one of the per-session queues 21. If there are packets stored in any one of the per-session queues 21, the packet is selected according to the chosen scheduling discipline from one of these queues at step 304 and then placed in the first buffer means 16, or output queue at step 306. At step 308 if the selected packet crosses a current capacity threshold in the first buffer means 16 then at step 310 the session that that particular packet belongs to is identified and is buffered in one of the per-session queues in the second buffer means 18. Again at step 308 if the packet has not crossed a current capacity threshold then the process reverts to step 302. At step 312 the next capacity threshold that has been set in the first buffer means 16 is used and assessed against for further packets that are input to the first buffer means 16. In this regard step 308 is repeated to test whether incoming packets cross subsequent capacity thresholds.
If at step 302, there are no packets queued in the third buffer means 19 the process moves to step 314 where a determination is made as to whether the capacity of the first buffer means has dropped below the abatement threshold, Tabate~ If it has then at step 316 a determination is made as to whether there are any packets temporarily stored in the second buffer means in any one of the per-session queues. If the capacity in the first buffer means has not dropped below Tabate at step 314, the process returns to step 302. At step 316 if there are packets queued in the second buffer means 18 the packet from the most recent stream or session that is identified as part of the logical queue forming the second buffer means is retrieved at step 318. If there are no packets queued in the second buffer means 18, the process reverts to step 302. At Step 320 a determination is made as to whether the retrieved packet from the second buffer means 18 is too old or has exceeded its time-out threshold, teXpire~ If a packet has been in the second buffer means too long and beyond its time-out threshold the packet is discarded at step 322 or if it has not exceeded its time out threshold the packet is placed in the first buffer means 16 at step 324. A determination is then made at step 326 as to whether or not the most recently retrieved packet is the last packet in that reo:fph:#31281.CAP 7 May 1999 particular stream or session stored in any one of the per-session queues 21 of the second buffer means 18. If this is the case the stream is then identified as part of a stream originating from the third buffer means 19 at step 328 and at step 330 the previous capacity threshold is then used or implemented. If at step 326 the most 5 recently retrieved packet is not the last packet in that particular stream or session then the process returns to step 302. At step 330 after the previous capacity threshold has been implemented then the process also returns to step 302.
It will also be appreciated that various modifications and alterations may be made to the preferred embodiments above, without departing from the scope and spirit 10 of the present invention.
reo:fph:#31281.CAP 7 May 1999
monitoring the used capacity or occupancy of packets stored in said first buffer means;
buffering data packets in a second buffer means at said node on initiation of a trigger condition, so as to avoid the first buffer means from overflowing during a congestion period.
The trigger condition may be a derivative-based threshold such that when the rate of increase of the amount of data stored in the first buffer means exceeds said derivative-based threshold, data packets are buffered in the second buffer means.
Alternatively, the trigger condition may be the crossing of a first or subsequent capacity threshold in the first buffer means due to the arrival of a data packet in the first buffer means. The first data packet in the first buffer means that crosses the first or subsequent capacity threshold may be buffered in the second buffer means and any subsequently arnving packets of a session to which the first data packet belongs may also be buffered in the second buffer means.
The invention fizrther provides in a second aspect a system for scheduling data packets in a telecommunications network, said system comprising;
first buffer means located at a node in the network for receiving data packets representative of one or more sessions;
a second buffer means located at said node in the network;
wherein data packets are buffered in the second buffer means on initiation of a trigger condition so as to avoid overflow in the first buffer means during a congestion period.
reo:fph:#31281.CAP 7 May 1999 According to a third aspect of the invention there is provided a data packet scheduling system for scheduling packets in a telecommunications network, in accordance with a quality of service standard set by a telecommunications service provider wherein said packets are representative of one or more sessions, said system comprising;
a first buffer means for receiving and temporarily storing data packets of one or more sessions transmitted in the telecommunications network;
a second buffer means wherein data packets are buffered in said second buffer means upon initiation of a trigger condition set in the first buffer means due to the arrival of a data packet in said first buffer means such that the first buffer means is prevented from overflowing during a congestion period in the network.
BRIEF DESCRIPTION OF THE DRAWINGS
Preferred embodiments will hereinafter be described, by way of example only, with reference to the following drawings wherein;
Figure 1 is a schematic diagram of a packet switch or node of a telecommunications network implementing the system and method of the present invention according to a first embodiment;
Figure 2 is a schematic diagram showing the buffering of data packets in a second buffer means intended for a first buffer means;
Figure 3 is a schematic diagram showing the transfer of data packets from a second buffer means to the first buffer means for transmission on an output link from the first buffer means;
Figure 4 is a schematic diagram of a packet switch or node of a telecommunications network implementing the system and method of the present invention according to a second embodiment;
Figure 5 is a schematic diagram showing a first, second and third buffer means and the treatment of data packets arriving at the node of Figure 4;
Figure 6 hows a block diagram of a system used to simulate a number of traffic sources being input into a node having first buffer means and second buffer means;
Figure 7 s a series of plots showing the cumulative probability distributions of packet delay of a number of sessions using the FIFO, DRRexp and DQ
simulations;
reo:fph:#31281.CAP 7 May 1999 Figure 8a and b show a series of plots of the highest packet delay per second against the simulation time for a number of sessions with an increased link transmission rate using the FIFO, DRRexp and DQ simulations;
Figure 9 is a plot of a number of sessions receiving "good service" in a one second period for each of the FIFO, DRRexp and DQ simulations;
Figure 10 shows a plot for each of the DRRexp and DQ simulations of degraded service experienced by all sessions;
Figure 11 shows a plot for each of the DRRexp and DQ simulations showing the number of degraded seconds for each of the sessions shown in Figure 3 with a reduced line transmission rate;
Figure 12 is a plot similar to Figure 9 but with a still further reduced line transmission rate;
Figure 13 is similar to Figures 9 and 10 wherein degraded seconds are shown for a set of different sessions at a low line transmission rate;
Figure 14 is a plot showing the degraded seconds for a further set of different sessions using the DRRexp and DQ simulations at an increased line transmission rate;
Figure 15 is a plot of the overall percentage of degraded packets for each of the FIFO, DRR, DRRexp and DQ algorithms for different average offered loads that are normalised to the link transmission rate, 2 o Figure 16 is a flow diagram showing the processes involved on receipt of a packet of a session on a link associated with a node, Figure 17a and b are flow diagrams having the processes involved when a packet is transferred from the second buffer means to the first buffer means, and Figure 18 a and b are flow diagrams showing the processes involved in treating 2 5 one or more packets arriving at the third buffer means of Figure 5.
DETAILED DESCRIPTION
Shown in Figure 1 there is a packet switch or node 2 which forms part of a telecommunications network, such as a packet switching network. The switch 2 consists 30 of a number of input ports 4 and a number of output ports 6 and a switching fabric 8 that is connected between the input ports 4 and the output ports 6. A
routing processor 10 is linked to the switching fabric 8 over link 12 which is responsible for the routing or switching of data packets between any one of the input ports 4 to any one of the output ports 6. Data packets that are input to the switch 2 are received by any one of the input ports 4 on input links 14. The data packets are then switched by the switching fabric 8 to one of the output ports 6. The data packets are then either received by a first buffer means 16 or a second buffer means 18 located in each one of the output ports 6 depending on how those input packets are to be treated, which will be hereinafter described. All data packets, apart from discarded packets, leave the output ports 6 from each of the first buffer means 16 on links 15.
Alternatively, the first buffer means 16 and second buffer means 18 may be located in each of the input ports 4.
Shown in Figure 2 there is a more detailed view of the first buffer means 16 and the second buffer means 18. When the system or network, or even a switch of the network is not congested, all sessions (a session being a series of packets that are dedicated to one particular user for one application) are received by the first buffer means 16 which is preferably in the form of a single short FIFO queue, and which is denoted as the a-queue. The length of the a-queue is selected such that any packet passing straight through the a- queue will meet the tightest delay requirements of any particular session. The length of the a-queue corresponds to the longest delay that can be considered "good service" in terms of delay. At the onset of congestion, when the level of the a- queue rises, due to the arrival of a packet from a particular session, and exceeds a capacity threshold T o,~et,l and it is decided that the packets from this session contribute to the possibility of the a-queue overflowing, this packet and subsequent packets from the session are time stamped and buffered in the second buffer means 18, hereinafter referred to as the 13-queue. The exceeding of the capacity threshold Tor~ec,l represents a trigger condition under which a fraction of the packets intended to be received in the a-queue are buffered in the 13-queue. The 13-queue may be significantly longer than the a-queue. As subsequent capacity thresholds, T
o~et~, j >1, are crossed then further sessions are also buffered in the 13-queue. The length of the reo:fph:#31281.CAP 7 May 1999 a-queue, and the settings of each threshold To,~et~~j>1 are constituted by memory size, in bytes, such that as each threshold is crossed a predetermined amount of memory in the a-queue is used by the data packets stored therein ready for transmission on an output link.
There is a wide range of possible algorithms that may be used for selecting those sessions or packets of sessions that are to be transferred either from the a-queue to the 13-queue or buffered in the 13-queue before arriving in the a-queue depending on the properties desired. Such algorithms include the following:
(a) Random:
A session is chosen randomly from those currently in progress. Such a selection attempts to spread performance degradation across all sessions over their lifetime, but not instantaneously across all sessions. The packets of the sessions may be chosen randomly in accordance with, for example, random numbers generated in the system that match a specific count number associated with the packets.
(b) Fair:
The session with the most packets in the a-queue at the time the threshold is crossed is chosen. Such a choice attempts to penalise those sessions whose instantaneous bandwidth requirement is higher than the fair share. It attempts to be instantaneously fair in deciding which sessions to penalise but has the benefit of minimizing the number of sessions simultaneously affected. In this case, each packet of each session stored in the a-queue is identified and recorded by the a-queue and subsequently counted.
(c) Next Packet:
The session of the packet which first breaks the first or subsequent capacity thresholds which is not already in the 13-queue or the session of the packet which breaks the first capacity threshold is chosen. Such a choice is a mix between being fair (the next packet arriving is likely to be from a session with the highest instantaneous bandwidth) and randomising sessions to be penalised.
It has the added advantage of being the simplest algorithm to implement.
reo:fph:#31281.CAP 7 May 1999 (d) Historical:
A session that has been in the 13-queue at some stage in the past, but is now in the a-queue is selected. Such a decision attempts to keep penalising sessions that have already been penalised. It minimises the amount of inconvenience 5 experienced by other sessions during the entire duration and thus attempts to maximise the number of users receiving good service over their entire session.
The above algorithms may be implemented in either software or hardware.
Shown in Figure 3 are a number of sessions that have been stored in the second buffer 18 or the 13-queue. A time-out threshold, teXp~.e, is set and all packets that have 10 been in the 13-queue longer than te,~ue are discarded. For example in the 13-queue of Figure 3, the oldest packets stored therein are discarded in the order that they were input to the 13-queue. If the 13-queue overflows, due to the packets stored therein exceeding the memory capacity of the 13-queue, the system may either drop newly arriving packets from various sessions or push out or discard the oldest packet in the 13- queue. The latter approach attempts to minimise the age of the packets in the queue and therefore maximise the chance that the transmitted packets will meet their end to end delay requirements.
A number of time-out thresholds, te,~,;re, ~, may be set depending on the types of data traffic stored in the second buffer means 18. For example, if the data traffic consists of video or voice packets, a small teXpre will be set, because a delayed packet in a real time stream is of very limited value and will only serve to delay subsequent packets. At the other extreme, packets may be representative of a file transfer from a server on the Internet. In this case, the texpire may be larger as the information need not be delivered in real time. Therefore, a slight delay is more acceptable than losing a packet, which would then require retransmission.
When the length or occupancy by packets of the a-queue drops to an abatement threshold, TabateW'herein the memory presently being used in the a-queue drops to a specified size, then packets from the 13-queue are moved back to the a-queue. Packets from a session cease being redirected to the 13-queue when there are no packets from that session left in the 13-queue. All packets moved from the 13-queue to reo:fph:#31281.CAP 7 May 1999 the a-queue are marked as "delayed". There are a number of algorithms for deciding which packets are moved first from the 13-queue to the a-queue, these being:
(a) FIFO:
This is the simplest 13-queue algorithm to implement and simply moves the packets to the a-queue in order of arrival in the 13-queue when the length of the a-queue drops to Tabate- The number of packets from each session in the 13-queue must be recorded so that the a-queue will know when to stop redirecting a session. For example, in session 20 which is the most recent session stored in the 13- queue there are 3 packets so that when the last or third packet of session 20 is transferred to the a-queue, the a-queue or the first buffer means 16 will be notified that it is the last packet in that session. Therefore any subsequent packets for session 20 that are input to the a-queue will only be transmitted after the first 3 packets of session 20 have been transmitted or output from the a-queue. This recordal also applies to the algorithms described under (b), (c) and (d) below.
(b) FIFO - FIFO:
This is where each session transferred to or buffered in the 13-queue is maintained separately. When the a-queue drops to below Tabate~ packets from the first session placed in the 13-queue are moved to the a-queue, that is, sessions are handled in a first in - first out order. The packets from this session are given priority over other sessions contained in the 13-queue. Within a session, packets are also transferred in first in - first out order to preserve sequencing within a session.
(c) LIFO - FIFO:
Each session transferred to or buffered in the 13-queue is maintained separately.
When the a-queue drops to below Tabate~ packets from the session most recently placed in the 13-queue are moved to the a-queue, that is, sessions are handled in a last in - first out fashion. The packets from this session are given priority over other sessions contained in the 13-queue. Within a session however, packets are still transferred in first in - first out order to preserve reo:fph:#31281.CAP 7 May 1999 sequencing within a session. Such an algorithm attempts to maximise the number of sessions receiving acceptable delay performance at any time.
(d) Fair:
Packets are transferred from the 13-queue to the a-queue using a Fair Queueing approach when the length of the a-queue is less than or equal to Tabale~ Thus, the sessions in the 13-queue use the residual bandwidths left over by the sessions still using the a-queue. Such a policy attempts to be fair to those sessions currently being penalised.
It is to be noted that all packets leave the output port 6 via the a-queue.
Packets that have gone through the second buffer means 18 are marked to allow for identification by other nodes or switches in the network. At subsequent nodes these packets may be the first choice for buffering in the second buffer means 18 during a congestion period. In this manner, the network can co-ordinate which sessions are penalised if congestion occurs simultaneously at more than one point in the network. For example, if a session has been buffered in the second buffer means 18 at one node, then the first packet or subsequent packets transmitted from that session to another node or nodes in the network will have an indication, in its header, that the session to which it belongs has been buffered previously. Thus other nodes may choose to continue buffering that session.
Another embodiment of the present invention is shown in Figures 4 and 5. The input ports 4, output ports 6 and switching fabric 8 are represented as in Figure 1. The first buffer means 16 is in the form of an output queue and receives data packets representative of one or more sessions from either the second buffer means 18 or a third buffer means 19. Each of the second buffer means 18 and third buffer means 19, in this alternative embodiment, comprise a series of per-session queues 21.
That is data packets from each session are received and temporarily stored in individual queues 21. It is to be noted that the second buffer means 18 and third buffer means 19 are simply subsets of a single collection of per-session queues 21. These queues 21 can be dynamically reclassified as either part of the second buffer means or the third buffer means 19. Generally the packets stored in the per-session queues of third buffer means 19 are transferred to the first buffer means 16 according to the scheduling reo:fph:#31281.CAP 7 May 1999 discipline chosen, such as FIFO or Fair Queueing as discussed previously. The packets in the per-session queues of the buffer means 19 are polled or read at a rate significantly higher than the rate at which packets are read from the first buffer means 16 onto an output link 15. Packets in the first buffer means 16 are then transmitted in FIFO order onto output link 15. The second and third buffer means 18, 19 may be implemented as logical queues. As with the first embodiment, at the onset of congestion, when the level of the first buffer means rises due to the arrival of packets from the third buffer means one or more capacity thresholds To,~et> ; J > 1 may be exceeded. If a particular packet arriving in buffer means 16 causes the capacity of the buffer means 16 to exceed a first threshold To,~et,l, this represents a trigger condition under which the session belonging to that packet is subsequently buffered in the second buffer means 18 into a respective queue in the buffer means 18. This will assist in preventing the first buffer means 16 from overflowing. As subsequent capacity thresholds are crossed, further sessions are buffered in the second buffer means 18. The buffering of data packets from various sessions is done according to any one of the algorithms mentioned earlier, that is either random, fair, next packet or historical algorithms.
A time-out threshold teXp;re is set for those packets that have been buffered in the second buffer means so that any packets buffered longer than texpue are discarded.
Different time-out thresholds teXp;res may be set depending on the type of data stream buffered, such as voice, video or web-browsing as with the previous embodiment.
When the length of the first buffer means drops to an abatement threshold, Tabate~ packets are moved from the second buffer means 18 to the first buffer means 16.
This may be performed by the previous algorithms mentioned with respect to the first embodiment, such as FIFO, FIFO-FIFO or LIFO-FIFO but preferably the LIFO-FIFO
algorithm is used whereby the most recent session is transferred from buffer means 18 to buffer means 16. Each packet within that most recent session is transferred in the order in which they entered the respective per-session queue so that sequencing is maintained.
In a filrther embodiment, the third buffer means may act as the first buffer means wherein packets of sessions would be output onto output link 15 directly from reo:fph:#31281.CAP 7 May 1999 the per-session queues 21. In this embodiment there is no need for an output buffer as with the previous embodiment. The capacity thresholds, To~e~, would be based on the current logical contents of the third buffer means, that is, the sum of the capacity taken up by individual sessions in each of the per-session queues 21. Therefore any packets of particular sessions that cross the thresholds would be treated according to any one of the previously mentioned situations.
Some simulation experiments have been performed and the results of those experiments will be hereinafter described. The simulation set up is shown in Figure 6 where a number of traffic sources 1 to N, identified by numeral 22, generating MPEG
data streams originating from video servers are multiplexed together by a node or switch 24 onto a single data link 26 to be transmitted to sink 28. The sink 28 acts as a discarder for packets in the simulation. Any other type of server from which higher volume traffic may be multiplexed may also be used in this example.
The frame sizes from real MPEG traces generated by O. Rose in the publication "Statistical Properties of MPEG Video Traffic and their Impact on Traffic Modelling in ATM Systems", in Proceedings of the 20th annual conference on local computer networks, 1995, pp. 397-406 (MPEG traces at ftp: //ftp-info3.
informatik.
uni-wuerzburg.de/pub/MPEG), are used for sources in the simulation. Table 1 summarises the characteristics of the MPEG sources. All sources have a peak rate of 4 Mbps at 25 frames per second and are 1600 seconds in duration (except for the newsl trace which is 1260 seconds). They are randomly started over a 1600 second interval and measurements are taken between 1600 seconds and 3200 seconds.
When an MPEG trace comes to the end it is cycled. The frames are broken into 1500 byte packets for transmission with a 40 byte minimum packet size.
reo:fph:#31281.CAP 7 May 1999 No. Name Mean bits/frame Peak/Mean Frame Number Size of Frames 1 asterix 22,349 6.6 40000 2 apt 21,890 8.7 40000 3 bond 24,308 10.1 40000 4 dino 13,078 9.1 40000 S lambs 7,312 18.4 40000 6 movie 14,288 12.1 40000 7 mrbean 17,647 13.0 40000 8 mtv 1 24,604 9.3 40000 9 mtv 2 19,780 12.7 40000 10 newsl 20,664 9.4 31515 11 news2 15,358 12.3 40000 12 race 30,749 6.6 40000 13 sbowl 23,506 6.0 40000 14 simpsons 18,576 12.9 40000 15 soccerl 27,129 6.9 40000 16 soccer2 25,110 7.6 40000 17 starwars 9,313 13.4 40000 18 talkl 14,537 7.3 40000 19 talk2 17,914 7.4 40000 terminator 10,905 7.3 40000 reo:fph:#31281.CAP 7 May 1999 The following measurements are collected for each session:
- End to end delay for each packet - Actual packets lost S - Degraded packets (packets delayed > SOms) - Degraded frames (frames with degraded or lost packets) - Degraded seconds (one second intervals with degraded frames).
A session is receiving "good service" during any particular second if no frames of the session have been degraded. Any session not receiving "good service" is deemed to be receiving "degraded service".
The method of the present invention is implemented in these simulation experiments to provide as many sessions as possible with good service during a particular transient congestion period. The second buffer means 18 or the 13-queue follows the LIFO - FIFO principle earlier described. Packets from the f3-queue are transferred to the a-queue only when the a- queue is empty, in other words when Tabate equals 0. The thresholds for redirecting packets to the 13-queue, To~et~ are set at L-L/(3 +j-1) for j>_ 1 where L is the length of the a-queue in the simulation.
Sources are removed from the redirection list when there are no packets from that source (or session) in the 13-queue. In a real network with a changing number of sessions in progress, the threshold assignment can be made dynamic by simply replacing the number of sessions in the simulation with the number of sessions in progress.
The packet time out threshold texpire is set to 400ms.
In this set of experiments the simplest method of selecting which sessions are redirected to the 13-queue is chosen. That method is the next packet method where the session of the first packet to break To"Set~ is chosen to be redirected to the 13- queue.
The 13-queue uses an oldest packet push out mechanism on overflow of the f3-queue.
In Table 2 there is listed a set of simulation experiments used to compare the performance of the algorithm or method of the present invention, termed dual queue (DQ), with the existing FIFO and Fair Queueing methods. A deficit round robin (DRR) technique is used for the Fair Queueing comparison, but is enhanced to include old packet discard (DRRexp) to make a fairer comparison with DQ.
reo:fph:#31281.CAP 7 May 1999 ExperimentNumber Offered MPEG Buffer Transmission Number of load (Mbps)Sources Space rate (Mbps) Sources used (Bytes) 1 20 9.5 all 20 500x103 8,9,10 & 12 2 4 1.6 1,2,3 & 100x103 1.3,1.65, 5 &
3 40 19 all twice 1x106 16, 20 & 24 4 20 30.7 race 1.5x106 24,30 &
A DRR technique was introduced by M. Shreedhar and G. Varghese in a document titled "Efficient Fair Queueing Using Deficit Round Robin", IEEE/ACM
Trans. Networking, Volume 4 No. 3, 1996, pp. 375-385.
As mentioned previously a packet is degraded if it is delayed more than SOms.
This is used as an estimate of the maximum delay allowed before a frame cannot be displayed to a viewer or user. Packets that have been queued for longer than te,~pue =
400ms are discarded in the DRRexp and DQ algorithms. However, an MPEG frame that arrives too late for display may still be useful for the decoding of future frames which may not be so badly delayed.
All queues are given the same total buffer size and kept constant for the bit rates used in each simulation experiment. The a-queue of the DQ implementation has a size that is changed whenever the link transmission speed is changed so as to give a maximum packet delay of 50 ms. It should also be noted that the worst case service delay overhead in DRR introduced by the round robin scheduling is less than the good service delay threshold of SOms.
In experiment 1 shown in Table 2 all 20 heterogeneous MPEG sources are multiplexed onto link 26. Four representative sessions, being numbers 3, 10, 14 and have arbitrarily been selected to give a detailed comparison between the FIFO, 20 DRRexp and DQ scheduling algorithms.
Figure 7 shows graphs of the cumulative probability distributions of packet delay for session 3 (represented by graph 30), session 10 (represented by graph 32), session 14 (represented by graph 34) and session 20 (represented by graph 36) when reo:fph:#31281.CAP 7 May 1999 the link transmission rate is lOMbps (compared with the total average offered rate of 9.SMbps). Dropped packets are considered to have infinite delay in Figure 5. A
highly loaded case has been chosen in order to exaggerate the difference between the algorithms during congestion. A line is drawn at the 50 ms good service cut off point.
It is seen that the DQ algorithm provides the highest fraction of packets receiving a delay of less than SOms. This is achieved by giving some packets a delay much greater than SOms. Given that the service provided to such packets is already considered degraded, the amount by which the delay exceeds SOms is considered to be of little relevance. Note that DRRexp gives better service to session number 20 than the other sessions as it is usually operating at less than its fair share of the bandwidth as expected from a Fair Queueing based discipline. Note that this is also true of the DQ algorithm.
In Figure 8 there is shown a plot of the highest packet delays per second for each of the sources or sessions 3, 10, 14 and 20 against the simulation time where the transmission rate on link 26 has been increased to 12 Mbps. Plots 38, 40, 42 and 44 are for the respective sessions 3, 10, 14 and 20. A line is drawn at SOms mark to indicate the good service cut off point. The time range as shown is chosen so as to include a number of transient congestion periods for comparison between the different algorithms. The FIFO is fair at the packet level and all sessions' packets, on average, experience the same delay passing through the node 24 on a FIFO basis. The DRRexp performs a little better but fails to deliver good service when the bandwidth has to be shared in such a way that many of the sessions receive less than their required service.
The priority nature of the FQ can cause longer delays to a burst of packets than encountered in a FIFO scheme. Therefore these extra delays can cause packets from a particular session to become degraded even when the link is not congested as is illustrated in the top trace 38 of source 3 in Figure 8. The DQ discipline selected tries to maximise the instantaneous number of sessions receiving good service so we see that one of the sessions receives very poor delay performance, so that the other sessions are spared service degradation, at times of 1925 seconds and 1975 seconds.
A better illustration of the level of QoS by all of the 20 sessions is illustrated in Figure 9 which shows a plot 46 of the percentage of all sessions simultaneously reo:fph:#31281.CAP 7 May 1999 receiving good service during similar one second periods. It is to be noted that the FIFO and DRRexp algorithms present degraded service to many sessions at times of 1855 seconds, 1925 seconds and 1975 seconds. These QoS figures can be represented in binary form where service is either good or degraded. A binary representation of service per session is generally all that is required for comparisons between each of the scheduling disciplines. Figure 10 shows such a representation 48 for the DRRexp algorithm and a representation 50 for the DQ algorithm over the full measurement time scale for each session at a link transmission rate of l2Mbps. A vertical bar is drawn for every degraded second experienced by any particular session. The more white space on the graph, the better the service a session is receiving. All the results that are shown in subsequent figures will use this method of comparison. From Figure 8 we see that the DQ approach results in fewer sessions being affected by congestion than DRRexp at any particular time.
As the transmission rate is reduced to lOMbps, shown in Figure 1 l, and 8Mbps in Figure 12 the contrast between DRRexp and DQ becomes even more evident. The plots 52 and 54 are respectively for the DRRexp and DQ algorithms in Figure 11 and the plots 56 and 58 are respectively for the DRRexp and DQ algorithms in Figure 12.
It is apparent that the DQ approach provides good service to most of the sessions even under extreme overload conditions as compared to the DRRexp algorithm.
Although these experiments or simulations have provided the simplest possible implementation of the DQ approach, it is possible to provide significantly better performance for real-time services than can DRRexp.
Refernng to Table 2, experiments 2 and 3 use 4 and 40 sources respectively and are conducted to determine the scalability of DQ algorithm. Experiment 2 has the added complexity that the peak bit rates of any of the sessions is higher than the link transmission rate and therefore cause the a-queue to begin to fill. In Figure 13 four sources are used and it is evident that even when presented with an unmanageable load, the DQ algorithm depicted in plot 60 performs as well as the DRRexp algorithm depicted in plot 62, if not slightly better especially for sources 3 and 4. In experiment 3 both scheduling disciplines provided good performance almost all of the time and figures have not been provided in this case. For DRRexp, 13 sessions experienced reo:fph:#31281.CAP 7 May 1999 degraded service during a single congestion event whereas only one session experienced degraded service using the DQ algorithm during the equivalent period. In the DQ algorithm, there was only one second in which sessions had degraded service, whereas there was one other second in which degraded service occurred using the 5 DRRexp. It is therefore concluded that the DQ approach scales reasonably well when compared to DRR Exp.
To examine fairness over moderate time scales, 20 homogeneous sources were used in experiment 4. The sessions consisted of 20 copies of the race trace which started at random times over the period 0 to 1600 seconds. Figure 14 shows the 10 degraded seconds plots 64 and 66 respectively for the DRRexp and DQ
algorithms for each session where it is seen that DRRexp provides fair, although very degraded service to each session. Qualitatively, the DQ algorithm also provides fair service to each session over the entire length of the simulation rather than instantaneously. As seen earlier, the overall QoS of each session is significantly better in DQ
than that 15 provided by DRRexp.
Overall it is seen that when there is no congestion all of the schemes compared provide good service. However as the periods of congestion increase, a greater contrast can be seen between the 3 algorithms. In Figure 15 there is shown the plot 68 of the percentage of degraded packets for the FIFO, DRR, DRRexp and DQ
20 algorithms for different average offered loads (normalised to the link transmission rate). Apart from achieving its aim of giving good service to as many sessions as possible during transient congestion periods, the DQ algorithm also minimises the overall proportion of degraded packets. It is to be noted that the percentage of degraded packets even under the simple implementation of the DQ approach is close to the excess offered load above one Erlang, and hence is near optimal. While this would also be true for a short FIFO queue, the short FIFO queue has the disadvantage of resulting in unnecessary loss during transient congestion periods when the offered load is close to, but less than the transmission link speed. The short FIFO
queue also loses packets randomly rather than selectively as in DQ.
During transient congestion episodes, packets entering a normal FIFO queue all experience similar degraded service. If the congestion is caused by a small number of reo:fph:#31281.CAP 7 May 1999 sessions, Fair Queueing provides relief for the other sessions. However, results presented hereinbefore indicate that this is not always the case as transient congestion is often caused by many sessions whose combined traffic fills the queue. Fair Queueing divides bandwidth equally and therefore degrading the service of a large number of sessions and therefore a large number of packets. The DQ algorithm outperforms Fair Queueing because it degrades the service of only enough sessions so as to obtain good service for the rest of the sessions during the congestion period and therefore also increases throughput.
As mentioned previously, the service provider may wish to maximise the number of users receiving "perfect" service, that is, showing no sign of congestion so as to minimise the number of complaints received by users when they make a connection and receive unsatisfactory QoS. This can be achieved by recording in a "target" list or file the connections which have been redirected to the 13-queue at some stage, and increasing the probability that they will be redirected again the next time the 1 S system is congested. Therefore the degradation is not spread evenly among users, even over considerable time scales, increasing the probability of a connection remaining unaffected by the congestion. The probability of redirecting connections in the target list or file may be increased to one so that no new connection will be redirected until all connections which have previously been redirected are currently being redirected. Another alternative is to consider several factors including the current data rate as well as the presence of the connections in the target list or file.
If the aim of the service provider is to maximise the number of the satisfied customers at any moment in time, the optimal strategy is to redirect the highest rate sessions. If the statistics of a given connection change significantly with time, the above approach of repeatedly targeting a particular user for the entire connection may not be appropriate. Instead, sessions could be removed from the target list or file after some time, t~get. This would result in bursts of bad service on a time scale of tt~.get, but may improve the total number of customer minutes of high QoS.
Alternatively a telecommunications regulatory body may require a stricter guarantee of fairness over moderate time scales than that provided by the hereinbefore described DQ implementation. This is also possible again by recording those reo:fph:#31281.CAP 7 May 1999 connections or sessions which have been redirected. This time, these sessions will be less likely to be redirected again. Once again there is a trade-off with the total number of customer minutes of high QoS which can be achieved.
Even greater flexibility is possible if other service disciplines are considered for S the 13-queue rather than only the LIFO-FIFO system hereinbefore mentioned.
For example, the 13-queue could use Fair Queueing. Let L denote the set of low-rate users who require less than their fair share of the bandwidth. In pure Fair Queueing, sessions in L are allocated their full bandwidth and the residual bandwidth is allocated to the remaining sessions. In the hybrid DQ/FQ system, the set of sessions which have been selected to receive good services at time t, G(t), takes the role of L so that the residual bandwidth after serving users in G(t) is fairly divided. If G(t) = L for all t, or if the size of the a-queue is set to zero, then the hybrid approach reduces to Fair Queueing. Conversely, if G(t) is the set of all users, FIFO results. However, it is often possible to have G(t) a strict superset of L whose total instantaneous bandwidth is still less than the output link capacity, increasing the number of satisfied customers without penalising any well behaved users.
As an alternative to using threshold crossing based schemes for initiating the redirection of sessions from the a-queue to the 13-queue, there is a wide variety of possible trigger conditions. Some examples include the use of derivative based schemes such that when the rate of increase of the amount of data stored in the a-queue exceeds the derivative-based threshold, the data packets are redirected to the second buffer means. Fuzzy logic based approaches may also be used. Thus whenever a predetermined rate of increase of the amount of packets stored in the a-queue is reached, packets of a session or sessions are redirected to the f3-queue.
Shown in Figure 16 is a flow diagram 100 illustrating the steps or processes involved on receipt of a packet representative of a session on an output link to the output port 6 or node 24. At step 102 the received packet is time stamped to enable the buffers 16 and 1$ to place the packet in a particular sequence based on time. At step 104 a check is made to see if other packets associated with that session are in the 13-queue redirection list. If there are no previous packets in the 13-queue then at step 106 a decision is made as to whether or not the received packet will cross a capacity reo:fph:#31281.CAP 7 May 1999 threshold if it was placed in the a-queue. If the packet does not cross the capacity threshold it is placed in the a-queue at step 108 or if it will cross the capacity threshold a stream identifier or session identifier is placed in the I3-queue redirection list at step 110 and at step 112 the next capacity threshold is then used.
Thereafter, the packet is placed in the 13-queue at step 114. Going back to step 104, if there are previous packets from that session already in the 13-queue redirection list then the received packet is immediately placed in the 13-queue again at step 114.
In Figure 17 there is shown a flow diagram 200 illustrating the processes and steps involved when a packet is transferred from the 13-queue (or the second buffer means 18) to the a-queue (or the first buffer means 16). At step 202 a determination is made as to whether there are packets in the 13-queue and if there are none, then the process does nothing at step 204. If there are packets queued then at step 206 the oldest packet is retrieved from the most recent session redirected to the 13-queue (LIFO-FIFO). A determination is made at step 208 as to whether the packet is too old and has gone beyond te,~,ue. If this is the case then the packet is discarded at step 220 and a decision is made at step 222 as to whether that discarded packet was the last packet from that session. If it was the last packet, then at step 224 the session owning that packet is removed from the 13-queue redirection list and at step 226 the previous a-queue abatement threshold is then implemented. The process then reverts back to step 202. If the discarded packet was not the last packet from that session in the 13-queue, then the process reverts again to step 202. Referring back to step 208, if the packet is not too old and is within the time limits set by teXpue, the packet is then placed in the a-queue at step 210 and a determination is made at step 212 as to whether that placed packet is the last packet from the session in the 13-queue. If it is not the last packet then the process does nothing at step 214. If it was the last packet in the 13-queue belonging to that session, then at step 216 the session is removed from the 13-queue redirection list and at step 218 the previous a-queue abatement threshold is implemented.
Shown in Figure 18 is a flow chart 300 showing processes involved in placing packets of sessions from a third buffer means into a first buffer means and the steps reo:fph:#31281.CAP 7 May 1999 involved for dealing with such packets or sessions at the onset of congestion in the first buffer means (refer to Figures 4 and 5). Incoming packets are received at one of the output ports 6 in the third buffer means 19. At step 302 a check is made as to whether or not packets are temporarily stored in the third buffer means in any one of the per-session queues 21. If there are packets stored in any one of the per-session queues 21, the packet is selected according to the chosen scheduling discipline from one of these queues at step 304 and then placed in the first buffer means 16, or output queue at step 306. At step 308 if the selected packet crosses a current capacity threshold in the first buffer means 16 then at step 310 the session that that particular packet belongs to is identified and is buffered in one of the per-session queues in the second buffer means 18. Again at step 308 if the packet has not crossed a current capacity threshold then the process reverts to step 302. At step 312 the next capacity threshold that has been set in the first buffer means 16 is used and assessed against for further packets that are input to the first buffer means 16. In this regard step 308 is repeated to test whether incoming packets cross subsequent capacity thresholds.
If at step 302, there are no packets queued in the third buffer means 19 the process moves to step 314 where a determination is made as to whether the capacity of the first buffer means has dropped below the abatement threshold, Tabate~ If it has then at step 316 a determination is made as to whether there are any packets temporarily stored in the second buffer means in any one of the per-session queues. If the capacity in the first buffer means has not dropped below Tabate at step 314, the process returns to step 302. At step 316 if there are packets queued in the second buffer means 18 the packet from the most recent stream or session that is identified as part of the logical queue forming the second buffer means is retrieved at step 318. If there are no packets queued in the second buffer means 18, the process reverts to step 302. At Step 320 a determination is made as to whether the retrieved packet from the second buffer means 18 is too old or has exceeded its time-out threshold, teXpire~ If a packet has been in the second buffer means too long and beyond its time-out threshold the packet is discarded at step 322 or if it has not exceeded its time out threshold the packet is placed in the first buffer means 16 at step 324. A determination is then made at step 326 as to whether or not the most recently retrieved packet is the last packet in that reo:fph:#31281.CAP 7 May 1999 particular stream or session stored in any one of the per-session queues 21 of the second buffer means 18. If this is the case the stream is then identified as part of a stream originating from the third buffer means 19 at step 328 and at step 330 the previous capacity threshold is then used or implemented. If at step 326 the most 5 recently retrieved packet is not the last packet in that particular stream or session then the process returns to step 302. At step 330 after the previous capacity threshold has been implemented then the process also returns to step 302.
It will also be appreciated that various modifications and alterations may be made to the preferred embodiments above, without departing from the scope and spirit 10 of the present invention.
reo:fph:#31281.CAP 7 May 1999
Claims (58)
1. A method for scheduling data packets in a telecommunications network, said method comprising the steps of:
receiving and storing in a first buffer means at a node in the network packets of data representative of one or more sessions;
monitoring the used capacity or occupancy of packets in said first buffer means;
buffering data packets in a second buffer means at said node on initiation of a trigger condition, so as to avoid the first buffer means from overflowing during a congestion period.
receiving and storing in a first buffer means at a node in the network packets of data representative of one or more sessions;
monitoring the used capacity or occupancy of packets in said first buffer means;
buffering data packets in a second buffer means at said node on initiation of a trigger condition, so as to avoid the first buffer means from overflowing during a congestion period.
2. A method according to claim 1 wherein the data packets that are stored in either the first buffer means or second buffer means exit from the first buffer means on a link associated with the node.
3. A method according to claim 2 wherein said link is an output link to said node.
4. A method according to claim 1 further comprising the step of receiving and storing said packets in a third buffer means prior to transferring the received packets to the first buffer means.
5. A method according to claim 4 further comprising storing the packets according to the session to which each packet belongs and transferring the packets from the third buffer means to the first buffer means according to a scheduling discipline.
6. A method according to claim 4 wherein each session has respective packets stored in a per-session queue in either the second buffer means or the third buffer means.
7. A method according to claim 1 wherein the trigger condition is a derivative-based threshold such that when the rate of increase of the amount of data stored in the first buffer means exceeds said derivative-based threshold, data packets are buffered in said second buffer means.
8. A method according to claim 1 wherein the trigger condition is the crossing of a first or subsequent capacity threshold in the first buffer means due to the arrival of a data packet in said first buffer means.
9. A method according to claim 8 further comprising buffering in said second buffer means the first data packet in the first buffer means that crosses said first or subsequent capacity threshold.
10. A method according to claim 9 further comprising buffering in the second buffer means subsequently arriving packets of a session to which said first data packet belongs.
11. A method according to claim 8 further comprising buffering in the second buffer means the session having the most data packets in the first buffer means when said one or more capacity thresholds is crossed due to the arrival of a data packet in the first buffer means.
12. A method according to claim 11 further comprising the step of transferring to said second buffer means subsequent data packets for the session having the most data packets stored in the first buffer means such that the transferred data packets are temporarily stored in the second buffer means in sequence.
13. A method according to claim 8 further comprising the step of buffering in the second buffer means the session having a packet that exceeds said first or subsequent capacity thresholds, which session is not temporarily stored in said second buffer means, and buffering subsequent packets arriving from that session in the second buffer means.
14. A method according to claim 8 further comprising buffering in said second buffer means a session that has previously been temporarily stored or buffered in the second buffer means and has data packets in the first buffer means at the time when said first or subsequent capacity threshold is crossed.
15. A method according to claim 1 further comprising setting a time-out threshold in said second buffer means such that all packets that have been stored in the second buffer means longer than the time-out threshold are discarded.
16. A method according to claim 15 further comprising the step of discarding newly arriving data packets to the second buffer means at the onset of the second buffer means overflowing.
17. A method according to claim 1 further comprising setting a time-out threshold in said second buffer means according to the type of stream the data packets stored in the second buffer means represent, such that all packets that have been stored in the second buffer means longer than their respective time-out threshold are discarded.
18. A method according to claim 15 further comprising discarding from the second buffer means the oldest packets stored in the second buffer means at the onset of overflowing of the second buffer means.
19. A method according to claim 1 wherein the method further comprises transferring data packets held in the second buffer means to the first buffer means when the occupancy of the first buffer means is less than or equal to an abatement threshold.
20. A method according to claim 19 further comprising storing in the first buffer means subsequently arriving packets from a particular session when there are no packets from that particular session stored in the second buffer means.
21. A method according to claim 19 further comprising transferring packets from the second buffer means to the first buffer means on a first-in first-out (FIFO) basis such that packets are transferred in the order in which they arrived in the second buffer means.
22. A method according to claim 19 further comprising transferring the packets from each session from the second buffer means to the first buffer means on a FIFO-FIFO basis whereby sessions that were first to be stored in the second buffer means are first to be transferred to the first buffer means and within each session transferred, packets are transferred in the order in which they arrived in the second buffer means..
23. A method according to claim 19 further comprising transferring the packets from each session from the second buffer means to the first buffer means on a LIFO-FIFO basis whereby sessions that were last into the second buffer means are transferred first and within each session packets are transferred in the order in which they arrived in the second buffer means.
24. A method according to claim 19 further comprising transferring packets using a Fair Queueing system such that sessions of data packets that are stored in the second buffer means use residual bandwidth left by sessions stored in the first buffer means.
25. A method according to claim 21 further comprising recording the number of packets from each session so transferred.
26. A method according to claim 1 further comprising time stamping each data packet stored in and buffered in the second buffer means.
27. A method according to claim 1 further comprising providing one or more packets belonging to a session that has been stored in the second buffer means with an identification label to acknowledge to other nodes in the network receiving said one or more packets that such a session has been so stored.
28. A system for scheduling data packets in a telecommunications network, said system comprising;
first buffer means located at a node in the network for receiving data packets representative of one or more sessions;
a second buffer means located at said node in the network;
wherein data packets are buffered in the second buffer means on initiation of a trigger condition so as to avoid overflow in the first buffer means during a congestion period.
first buffer means located at a node in the network for receiving data packets representative of one or more sessions;
a second buffer means located at said node in the network;
wherein data packets are buffered in the second buffer means on initiation of a trigger condition so as to avoid overflow in the first buffer means during a congestion period.
29. A system according to claim 28 wherein the data packets in either the first buffer means or the second buffer means exit from the first buffer means on a link associated with the node.
30. A system according to claim 29 wherein the link is an output link to the said node.
31. A system according to claim 28 further comprising a third buffer means for receiving said data packets prior to transferring said data packets to said first buffer means.
32. A system according to claim 31 wherein said data packets are stored in said third buffer means according to the session to which each packet belongs and are transferred from the third buffer means to the first buffer means according to a scheduling discipline.
33. A system according to claim 31 wherein each session has respective packets stored in a per-session queue in either the second buffer means or the third buffer means.
34. A system according to claim 28 wherein the trigger condition is a derivative-based threshold such that when the rate of increase of the amount of data stored in the first buffer means exceeds said derivative-based threshold, data packets are buffered in said second buffer means.
35. A system according to claim 28 wherein the trigger condition is the crossing of a first or subsequent capacity threshold in the first buffer means due to the arrival of a data packet in said first buffer means.
36. A system according to claim 35 wherein the first data packet in the first buffer means that crosses said first or subsequent capacity threshold is buffered in said second buffer means.
37. A system according to claim 36 wherein subsequently arriving packets of a session to which said first data packet belongs are buffered in the second buffer means.
38. A system according to claim 35 wherein the session having the most data packets in the first buffer means is buffered in the second buffer means when said one or more capacity thresholds is exceeded due to the arrival of a data packet in the first buffer means.
39. A system according to claim 38 wherein subsequent data packets belonging to the session having the most data packets in the first buffer means are transferred to the second buffer means to be temporarily stored in the second buffer means in sequence.
40. A system according to claim 35 wherein the session having a packet that exceeds said first or subsequent capacity thresholds, which session is not temporarily stored in said second buffer means, is buffered in the second buffer means and subsequent packets arriving from said session at the first buffer means are also buffered in the second buffer means.
41. A system according to claim 35 wherein a session that has previously been temporarily stored in the second buffer means and has data packets in the first buffer means is buffered in the second buffer means at the time when said first or subsequent capacity threshold is crossed.
42. A system according to claim 28 wherein a time-out threshold is set in said second buffer means such that all packets that have been stored in the second buffer means longer than the time-out threshold are discarded.
43. A system according to claim 42 wherein at the onset of the second buffer means overflowing, newly arriving data packets to the second buffer means are discarded.
44. A system according to claim 28 wherein a time-out threshold is set in said second buffer means according to the type of stream the data packets stored in the second buffer means represent, such that all packets that have been stored in the second buffer means longer than their respective time-out threshold are discarded.
45. A system according to claim 42 wherein, at the onset of overflowing of the second buffer means, the oldest packets stored in the second buffer means are discarded from the second buffer means.
46. A system according to claim 28 wherein data packets that are stored in the second buffer means are transferred to the first buffer means when the occupancy of the first buffer means is less than or equal to an abatement threshold.
47. A system according to claim 46 wherein when there are no packets from a particular session stored in the second buffer means, subsequently arriving packets from that session are stored in the first buffer means.
48. A system according to claim 46 wherein packets are transferred from the second buffer means to the first buffer means on a first-in first-out (FIFO) basis such that packets are transferred in the order in which they arrived in the second buffer means.
49. A system according to claim 46 wherein packets from each session stored in the second buffer means are transferred to the first buffer means on a FIFO-FIFO basis whereby sessions that were first to be stored in the second buffer means are first to be transferred to the first buffer means and within each session transferred, packets are transferred in the order in which they arrived in the second buffer means.
50. A system according to claim 46 wherein packets from each session stored in the second buffer means are transferred to the first buffer means on a LIFO-FIFO
basis whereby sessions that were last into the second buffer means are transferred first to the first buffer means and within each session transferred, packets are transferred in the order in which they arrived in the second buffer means.
basis whereby sessions that were last into the second buffer means are transferred first to the first buffer means and within each session transferred, packets are transferred in the order in which they arrived in the second buffer means.
51. A system according to claim 46 wherein packets are transferred using a Fair Queueing system such that sessions of data packets that are stored in the second buffer means use residual bandwidth left by sessions stored in the first buffer means.
52. A system according to claim 48 wherein the number of packets from each session transferred is recorded.
53. A system according to claim 28 wherein each data packet stored in and transferred to the second buffer means are time stamped.
54. A system according to claim 28 wherein one or more packets belonging to a session that has been stored in the second buffer means is provided with an identification label to acknowledge to other nodes in the network receiving said one or more packets that such a session has been so stored.
55. A system according to claim 28 wherein the first buffer means is a FIFO
buffer.
buffer.
56. A system according to claim 28 wherein the first buffer means and the second buffer means comprise one or more per-session queues and the trigger condition is the crossing of a first or subsequent capacity threshold in the first buffer means such that the sum of the capacity of individual sessions stored in the queues of the first buffer means exceeds a predetermined threshold.
57. A packet switch of a packet switching network, said packet switch having a system for scheduling data packets according to claim 28 and comprising:
one or more ports for receiving data packets transmitted from respective information sources;
said one or more ports forwarding said packets to another node of said network;
wherein said one or more ports each contain said first buffer means and said second buffer means.
one or more ports for receiving data packets transmitted from respective information sources;
said one or more ports forwarding said packets to another node of said network;
wherein said one or more ports each contain said first buffer means and said second buffer means.
58. A data packet scheduling system for scheduling packets in a telecommunications network, in accordance with a quality of service standard set by a telecommunications service provider wherein said packets are representative of one or more sessions, said system comprising;
a first buffer means for receiving and temporarily storing data packets of one or more sessions transmitted in the telecommunications network;
a second buffer means wherein data packets are buffered in said second buffer means upon initiation of a trigger condition set in the first buffer means due to the arrival of a data packet in said first buffer means such that the first buffer means is prevented from overflowing during a congestion period in the network.
a first buffer means for receiving and temporarily storing data packets of one or more sessions transmitted in the telecommunications network;
a second buffer means wherein data packets are buffered in said second buffer means upon initiation of a trigger condition set in the first buffer means due to the arrival of a data packet in said first buffer means such that the first buffer means is prevented from overflowing during a congestion period in the network.
Applications Claiming Priority (2)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| AU92405/98 | 1998-11-16 | ||
| AU92405/98A AU9240598A (en) | 1998-11-16 | 1998-11-16 | Method and system for scheduling packets in a telecommunications network |
Publications (1)
| Publication Number | Publication Date |
|---|---|
| CA2271669A1 true CA2271669A1 (en) | 2000-05-16 |
Family
ID=3764064
Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| CA 2271669 Abandoned CA2271669A1 (en) | 1998-11-16 | 1999-05-14 | Method and system for scheduling packets in a telecommunications network |
Country Status (2)
| Country | Link |
|---|---|
| AU (1) | AU9240598A (en) |
| CA (1) | CA2271669A1 (en) |
Cited By (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| WO2012112772A1 (en) * | 2011-02-18 | 2012-08-23 | Ab Initio Technology Llc | Managing buffer overflow conditions |
| US9003084B2 (en) | 2011-02-18 | 2015-04-07 | Ab Initio Technology Llc | Sorting |
| US11202116B2 (en) * | 2018-12-13 | 2021-12-14 | Sling Media Pvt Ltd | Systems, methods, and devices for optimizing streaming bitrate based on variations in processor load |
-
1998
- 1998-11-16 AU AU92405/98A patent/AU9240598A/en not_active Abandoned
-
1999
- 1999-05-14 CA CA 2271669 patent/CA2271669A1/en not_active Abandoned
Cited By (7)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| WO2012112772A1 (en) * | 2011-02-18 | 2012-08-23 | Ab Initio Technology Llc | Managing buffer overflow conditions |
| US8447901B2 (en) | 2011-02-18 | 2013-05-21 | Ab Initio Technology Llc | Managing buffer conditions through sorting |
| US8762604B2 (en) | 2011-02-18 | 2014-06-24 | Ab Initio Technology Llc | Managing buffer conditions through sorting |
| US9003084B2 (en) | 2011-02-18 | 2015-04-07 | Ab Initio Technology Llc | Sorting |
| US9128686B2 (en) | 2011-02-18 | 2015-09-08 | Ab Initio Technology Llc | Sorting |
| US11202116B2 (en) * | 2018-12-13 | 2021-12-14 | Sling Media Pvt Ltd | Systems, methods, and devices for optimizing streaming bitrate based on variations in processor load |
| US11722719B2 (en) | 2018-12-13 | 2023-08-08 | Dish Network Technologies India Private Limited | Systems, methods, and devices for optimizing streaming bitrate based on variations in processor load |
Also Published As
| Publication number | Publication date |
|---|---|
| AU9240598A (en) | 2000-05-18 |
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