CN114374628A - Model-based audio and video communication low-delay congestion control method - Google Patents
Model-based audio and video communication low-delay congestion control method Download PDFInfo
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- CN114374628A CN114374628A CN202210121673.7A CN202210121673A CN114374628A CN 114374628 A CN114374628 A CN 114374628A CN 202210121673 A CN202210121673 A CN 202210121673A CN 114374628 A CN114374628 A CN 114374628A
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L43/00—Arrangements for monitoring or testing data switching networks
- H04L43/08—Monitoring or testing based on specific metrics, e.g. QoS, energy consumption or environmental parameters
- H04L43/0852—Delays
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L43/00—Arrangements for monitoring or testing data switching networks
- H04L43/08—Monitoring or testing based on specific metrics, e.g. QoS, energy consumption or environmental parameters
- H04L43/0876—Network utilisation, e.g. volume of load or congestion level
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L47/00—Traffic control in data switching networks
- H04L47/10—Flow control; Congestion control
- H04L47/12—Avoiding congestion; Recovering from congestion
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L5/00—Arrangements affording multiple use of the transmission path
- H04L5/003—Arrangements for allocating sub-channels of the transmission path
- H04L5/0053—Allocation of signaling, i.e. of overhead other than pilot signals
- H04L5/0055—Physical resource allocation for ACK/NACK
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
- H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
- H04N21/60—Network structure or processes for video distribution between server and client or between remote clients; Control signalling between clients, server and network components; Transmission of management data between server and client, e.g. sending from server to client commands for recording incoming content stream; Communication details between server and client
- H04N21/63—Control signaling related to video distribution between client, server and network components; Network processes for video distribution between server and clients or between remote clients, e.g. transmitting basic layer and enhancement layers over different transmission paths, setting up a peer-to-peer communication via Internet between remote STB's; Communication protocols; Addressing
- H04N21/647—Control signaling between network components and server or clients; Network processes for video distribution between server and clients, e.g. controlling the quality of the video stream, by dropping packets, protecting content from unauthorised alteration within the network, monitoring of network load, bridging between two different networks, e.g. between IP and wireless
- H04N21/64723—Monitoring of network processes or resources, e.g. monitoring of network load
- H04N21/64738—Monitoring network characteristics, e.g. bandwidth, congestion level
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
- H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
- H04N21/60—Network structure or processes for video distribution between server and client or between remote clients; Control signalling between clients, server and network components; Transmission of management data between server and client, e.g. sending from server to client commands for recording incoming content stream; Communication details between server and client
- H04N21/63—Control signaling related to video distribution between client, server and network components; Network processes for video distribution between server and clients or between remote clients, e.g. transmitting basic layer and enhancement layers over different transmission paths, setting up a peer-to-peer communication via Internet between remote STB's; Communication protocols; Addressing
- H04N21/647—Control signaling between network components and server or clients; Network processes for video distribution between server and clients, e.g. controlling the quality of the video stream, by dropping packets, protecting content from unauthorised alteration within the network, monitoring of network load, bridging between two different networks, e.g. between IP and wireless
- H04N21/64784—Data processing by the network
- H04N21/64792—Controlling the complexity of the content stream, e.g. by dropping packets
Abstract
The invention discloses a model-based audio and video communication low-delay congestion control method, relates to the technical field of audio and video communication, and aims to solve the problem that a buffer area can be quickly filled up and packet loss is caused because the sending rate of an existing sender is fast enough. Two estimation parameters: two of the estimated parameters are BtlBw and RTprop(ii) a RTT round trip delay: the RTT round trip delay sends the time of the interval between the information and the ACK (acknowledgement character) of the received information; bandwidth delay product BDP: the sum of the two estimation parameters and the round trip time delay of the RTT; two control mechanisms are provided: the two control mechanisms are respectivelyTo control the sending rate of packets using paging _ gain and to quickly reduce the amount of data in transmission to a specified value using cwnd according to BtlBw; the control method comprises the following steps: the state machine in the control method comprises four states of Startup, Drain, Probe Bandwidth and Probe RTT. The model-based audio and video communication low-delay congestion control method maximally utilizes the bandwidth of a bottleneck link on a network, and is suitable for being used in a weak network environment with a certain packet loss rate.
Description
Technical Field
The invention relates to the technical field of audio and video communication, in particular to a model-based audio and video communication low-delay congestion control method.
Background
The application scenes of low-delay real-time audio and video technologies are continuously expanded, from live broadcasting in a show field, PK live broadcasting in a microphone to online education, video conferences and real-time command and scheduling, the floor application of the audio and video technologies develops towards higher definition and lower delay, TCP is adopted to initiate messages at multiple ends at present, network congestion occurs when the actual bandwidth cannot reach the maximum bandwidth which can be provided, more and more TCP initiating retries exist at the moment, the congestion situation is further aggravated, along with the large outbreak of the internet, the number of various applications, particularly audio and video applications, is greatly increased, and a TCP congestion control algorithm cannot meet the requirements of the current internet application on high real-time performance, high bandwidth utilization rate and high throughput of network transmission.
Congestion control algorithms based on packet loss, such as Reno and CUBIC, have been widely used in the internet, and congestion control is divided into four parts: slow start, congestion avoidance, fast retransmission, fast recovery, its meaning is when the bottleneck bandwidth of the link is unknown, with the lower sending rate of the beginning, with every RTT, the speed of two times increases the sending rate fast, until reaching a threshold, after the threshold, enter the stage of improving the sending rate linearly, the stage is called congestion avoidance, until losing packet, after losing packet, the sending rate drops by a wide margin, use fast retransmission algorithm to resend and send to the packet loss, use fast recovery algorithm to raise sending rate as smooth as possible at the same time, if the buffer memory of the bottleneck router is especially big, this kind of congestion algorithm taking packet loss as the detection foundation will lead to the serious problem: the long RTT on the TCP link becomes large but the throughput remains unchanged. The algorithm based on packet loss can not reach the theoretical optimal solution of time delay and bandwidth, and the algorithm works well for a long time, and the good performance is that the buffer areas of the network switch and the router are quite adaptive to the current network bandwidth, and once the sending rate of a sender is fast enough, the buffer areas can be quickly filled, so that packet loss is caused.
Disclosure of Invention
The invention aims to provide a model-based audio and video communication low-delay congestion control method to solve the problem that the transmission rate of a sender is high enough, a buffer area is quickly filled, and packet loss is caused in the background technology.
In order to achieve the purpose, the invention provides the following technical scheme: a model-based audio and video communication low-delay congestion control method comprises the following steps:
two estimation parameters: two of the estimated parameters are BtlBw and RTprop；
RTT round trip delay: the RTT round trip delay sends the time of the interval between the information and the ACK (acknowledgement character) of the received information;
bandwidth delay product BDP: the sum of the two estimation parameters and the round trip time delay of the RTT;
two control mechanisms are provided: the two control mechanisms respectively control the sending rate of the packets by using the paging _ gain and rapidly reduce the data volume in transmission to a specified value by using the cwnd according to the BtlBw, wherein the paging _ gain is more than 1, the inter-packet time interval is reduced, the data volume in transmission is increased, the paging _ gain is less than 1, the opposite effect is achieved, the inter-packet time interval is increased, the data volume in transmission is reduced, and the paging _ gain is dynamic gain and is used for adapting to the bottleneck bandwidth; cwnd is a congestion window, controls the maximum data volume allowed to be transmitted at any time in the network, continuously adjusts cwnd by using a network path model, and determines how to detect the path by using a state machine;
the control method comprises the following steps: the state machine in the control method comprises four states of Startup, Drain, Probe Bandwidth and Probe RTT.
By adopting the technical scheme: according to the invention, the expected value and the standard deviation are considered at the same time, and if the standard deviation which is four times of the normal distribution is extremely rare, the packet loss can be considered to occur.
RTT＝α×RTT+(1-α)×RTTcurrent
Deviation＝α×Deviation+(1-α)×|RTTcurrent-RTT|
Timeout＝RTT+4×Deviation
Further, the estimated bottleneck bandwidth of the transmission channel of BtlBw estimates the maximum transmission rate sample of the self-sliding window, and the bottleneck transmission rate determines the maximum transmission rate of the link.
By adopting the technical scheme: the control method of the present invention attempts to bring the sending rate close to the bottleneck rate to seek "rate balancing", i.e. the arrival rate of the packets is equal to BtlBw. The bottleneck rate varies continuously throughout the lifetime of the link, so the control method of the present invention uses the most recent transmission rate samples to continuously estimate BtlBw.
Further, the RT ispropEstimated two-way round-trip propagation delay of the path, estimated minimum round-trip delay sample of self-sliding window, the RTpropDetermines the minimum time that the connection needs to maintain the BtlBw rate.
By adopting the technical scheme: the round-trip propagation delay will vary over the lifetime of a connection, so the control method of the invention continues to estimate the RT from the most recent round-trip delay sampleprop。
According to the formula RTT-RTprop+η，RTpropDetermined by the physical device and η represents noise, including switch queue delay, ACK (acknowledgement character) processing delay, etc. According to this formula, RT can be estimated using the minimum RTT over a period of timeprop。
Further, the RTT round trip delay is a basis for retransmitting timeout.
By adopting the technical scheme: and judging whether to retransmit timeout or not based on the RTT round trip delay.
Further, the paging _ gain is a dynamic gain, and the cwnd is a congestion window.
By adopting the technical scheme: cwnd controls the maximum amount of data that can be transmitted at any time in the network, and is constantly adjusted using a network path model, with a state machine determining how to probe the path.
Compared with the prior art, the invention has the beneficial effects that:
1. the model-based audio and video communication low-delay congestion control method maximizes the utilization of the bandwidth of a bottleneck link on a network, is particularly suitable for being used in a weak network environment with a certain packet loss rate, and can effectively use the available bandwidth to achieve high throughput.
Drawings
FIG. 1 is a state diagram of a control method according to the present invention.
Detailed Description
The technical solutions in the embodiments of the present invention will be clearly and completely described below with reference to the drawings in the embodiments of the present invention, and the described embodiments are only a part of the embodiments of the present invention, but not all of the embodiments. All other embodiments, which can be derived by a person skilled in the art from the embodiments given herein without making any creative effort, shall fall within the protection scope of the present invention.
A method for controlling low-delay congestion of audio and video communication based on a model comprises the following steps: the method comprises the following steps:
two estimation parameters: two of the estimated parameters are BtlBw and RTprop；
RTT round trip delay: the RTT round trip delay sends the time of the interval between the information and the ACK (acknowledgement character) of the received information;
bandwidth delay product BDP: the sum of the two estimation parameters and the round trip time delay of the RTT;
two control mechanisms are provided: the two control mechanisms respectively use the paging _ gain to control the sending rate of the packet according to BtlBw and the cwnd to quickly reduce the data volume in transmission to a specified value;
the control method comprises the following steps: the state machine in the control method comprises four states of Startup, Drain, Probe Bandwidth and Probe RTT.
According to the invention, the expected value and the standard deviation are considered at the same time, and if the standard deviation which is four times of the normal distribution is extremely rare, the packet loss can be considered to occur.
RTT＝α×RTT+(1-α)×RTTcurrent
Deviation＝α×Deviation+(1-α)×|RTTcurrent-RTT|
Timeout＝RTT+4×Deviation
And estimating the maximum transmission rate sample of the self-sliding window by the bottleneck bandwidth of the transmission channel estimated by BtlBw, wherein the bottleneck transmission rate determines the maximum transmission rate of the link.
The control method of the present invention attempts to bring the sending rate close to the bottleneck rate to seek "rate balancing", i.e. the arrival rate of the packets is equal to BtlBw. The bottleneck rate varies continuously throughout the lifetime of the link, so the control method of the present invention uses the most recent transmission rate samples to continuously estimate BtlBw.
The RT ispropEstimated two-way round-trip propagation delay of the path, estimated minimum round-trip delay sample of self-sliding window, the RTpropDetermines the minimum time that the connection needs to maintain the BtlBw rate.
The round-trip propagation delay will vary over the lifetime of a connection, so the control method of the invention continues to estimate the RT from the most recent round-trip delay sampleprop。
According to the formula RTT-RTprop+η，RTpropDetermined by the physical device and η represents noise, including switch queue delay, ACK (acknowledgement character) processing delay, etc. According to this formula, RT can be estimated using the minimum RTT over a period of timeprop。
The RTT round trip delay is the basis for retransmitting timeout.
The paging _ gain is a dynamic gain, and the cwnd is a congestion window.
cwnd controls the maximum amount of data that can be transmitted at any time in the network, and is constantly adjusted using a network path model, with a state machine determining how to probe the path.
The working principle is as follows: when a transport stream starts, it will execute a first continuity detection process, the network bandwidth ranges from several bps to 100Gbps, in order to rapidly acquire BtlBw in such a large range, the transmission rate is exponentially increased in the Startup state, the transmission rate is doubled every turn, in order to achieve rapid detection in a smooth manner, when entering the Startup state, the paging _ gain and cwnd _ gain are set as
I.e., the minimum gain value that allows doubling the transmission rate per round, when the transmission rate is increased rapidly, it takes higher transmission rate samples, BtlBw increases, and both the paging _ rate and cwnd need to be increased smoothly and proportionally. Once the pipe is full, the queue has been formed, the cwnd _ gain limit queue is (cwnd _ gain-1) × BDP, then immediately enters the Drain state to quickly Drain the queue, during the Drain phase, the control method of the present invention will reduce the paging _ gain below 1.0, to quickly Drain the queue generated during the Start phase, specifically, using the inverse of the starting phase's paging _ gain as the new paging _ gain, a continuous round of discharge is performed, in the Drain phase, when the amount of data in transmission matches the estimated BDP, i.e., it is estimated that the queue has been completely drained but the pipe is still full, the Drain state is exited and the ProbeBW stage is entered, which takes most of the time, the ProbeBW uses a cyclic gain approach to detect the bandwidth, the method can realize high throughput, low queue delay and fair bandwidth sharing, and the gain cycle has a serialized paging _ gain: 5/4,3/4,1,1,1,1,1,1,each stage will typically last about RTpropUsing 5/4 as a paging _ gain to detect higher bandwidth during phase 0 of the gain cycle, this operation may gradually increase the amount of data in the transmission. This phase will continue for at least one RTpropOr until the amount of data in the transmission reaches 5/4 × estimated _ BDP (when RT is reached)propAt lower times, it may take more time than RTprop) Or a packet loss occurs (it is possible that the network cannot accommodate 5/4 × estimatted _ BDP data size), stage 1 uses 3/4 as the paging _ gain to drain the queue in the bottleneck, the reason for choosing this value is that 1-3/4 equals 5/4-1, and this stage continues with RTpropOr until the amount of data in transmission falls below the estimated _ BDP (queues may be drained ahead due to application constraints), and finally, phases 2 through 7 use a paging _ gain value of 1.0 to achieve a short queue, fully utilized cruise mode, each for RTpropThe average gain of all stages is 1.0, since the purpose of ProbeBW is to make the average paging rate equal to the estimated available bandwidth BtlBw, it is necessary to maintain a small, well-bounded queue of high utilization, in order to detect RTpropThe control method of the invention will periodically enter into the ProbeRTT state, discharge the bottleneck queue, when in the non-ProbeRTT state, if RT is in the statepropIf the ProbeRTT is not updated within 10 seconds, the ProbeRTT phase is entered, and cwnd is limited to a very small value of 4 and lasts at least 200ms, so as to ensure that the flow does not cause packet accumulation on the intermediate device.
It will be evident to those skilled in the art that the invention is not limited to the details of the foregoing illustrative embodiments, and that the present invention may be embodied in other specific forms without departing from the spirit or essential attributes thereof. The present embodiments are therefore to be considered in all respects as illustrative and not restrictive, the scope of the invention being indicated by the appended claims rather than by the foregoing description, and all changes which come within the meaning and range of equivalency of the claims are therefore intended to be embraced therein. Any reference sign in a claim should not be construed as limiting the claim concerned.
Claims (5)
1. A low-delay congestion control method for audio and video communication based on a model is characterized by comprising the following steps:
two estimation parameters: two of the estimated parameters are BtlBw and RTprop；
RTT round trip delay: the RTT round trip delay sends the time of the interval between the information and the ACK (acknowledgement character) of the received information;
bandwidth delay product BDP: the sum of the two estimation parameters and the round trip time delay of the RTT;
two control mechanisms are provided: the two control mechanisms respectively use the paging _ gain to control the sending rate of the packet according to BtlBw and the cwnd to quickly reduce the data volume in transmission to a specified value;
the control method comprises the following steps: the state machine in the control method comprises four states of Startup, Drain, Probe Bandwidth and Probe RTT.
2. The model-based audio-video communication low-delay congestion control method according to claim 1, characterized in that: and estimating the maximum transmission rate sample of the self-sliding window by the bottleneck bandwidth of the transmission channel estimated by BtlBw, wherein the bottleneck transmission rate determines the maximum transmission rate of the link.
3. The model-based audio-video communication low-delay congestion control method according to claim 1, characterized in that: the RT ispropEstimated two-way round-trip propagation delay of the path, estimated minimum round-trip delay sample of self-sliding window, the RTpropDetermines the minimum time that the connection needs to maintain the BtlBw rate.
4. The model-based audio-video communication low-delay congestion control method according to claim 1, characterized in that: the RTT round trip delay is the basis for retransmitting timeout.
5. The model-based audio-video communication low-delay congestion control method according to claim 1, characterized in that: the paging _ gain is a dynamic gain, and the cwnd is a congestion window.
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