CN106789427A - A kind of transmission volume computational methods - Google Patents
A kind of transmission volume computational methods Download PDFInfo
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- CN106789427A CN106789427A CN201611203333.XA CN201611203333A CN106789427A CN 106789427 A CN106789427 A CN 106789427A CN 201611203333 A CN201611203333 A CN 201611203333A CN 106789427 A CN106789427 A CN 106789427A
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- network
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- speech frame
- error correction
- packet loss
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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
- H04L43/0894—Packet rate
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L1/00—Arrangements for detecting or preventing errors in the information received
- H04L1/0001—Systems modifying transmission characteristics according to link quality, e.g. power backoff
- H04L1/0002—Systems modifying transmission characteristics according to link quality, e.g. power backoff by adapting the transmission rate
-
- 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
- H04L47/26—Flow control; Congestion control using explicit feedback to the source, e.g. choke packets
- H04L47/263—Rate modification at the source after receiving feedback
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- Engineering & Computer Science (AREA)
- Computer Networks & Wireless Communication (AREA)
- Signal Processing (AREA)
- Quality & Reliability (AREA)
- Environmental & Geological Engineering (AREA)
- Data Exchanges In Wide-Area Networks (AREA)
Abstract
The present invention relates to a kind of transmission volume computational methods, comprising four steps, the first step, packet loss is adjusted by adjusting transmission rate, second step, build the appraising model of the network bandwidth, 3rd step, frame per second to transmitting is adjusted, packet to network transmission is sampled, 4th step, during transmission, carry out error correction, present invention employs the method that probe method is combined with modelling, carry out the calculating of transmission volume, the network condition for remaining transmitting realtime stream media is swift in response, it is friendly, it is easy to be received by fire wall, the advantages of being easy to implement with safeguarding, simultaneously breach transmission rate cannot adaptive network available bandwidth fluctuation, retransmit the delay jitter and the obstacle such as out of order for easily increasing media bag.
Description
Technical field
The present invention relates to field of network transmission, it is related to a kind of transmission volume computational methods.
Background technology
With developing rapidly for network and multimedia technology, the dependence of the work and life of people to network communication services
With require more and more higher, at any time, everywhere, by any way, more rapidly, it is less expensive multimedia communication is carried out by network, oneself
Common requirements as people, this cause to interconnect transfers on network content oneself be gradually changed into by simple text and image and include
Text, picture, audio, the multi-medium data of video, particularly, under will being based on the real time flow medium of digital real-time video
The prevailing transmission object of generation network, is calculated as primary task to transmission volume.
The content of the invention
In view of this, the present invention provides a kind of transmission volume computational methods for solving or partly solving the above problems.
To reach the effect of above-mentioned technical proposal, the technical scheme is that：A kind of transmission volume computational methods, bag
Containing following steps：
1) two ends of network transmission are respectively divided into transmitting terminal and receiving terminal, feedback cycle, the time of feedback cycle is set
Be 1 millisecond, transmitting terminal periodically to the real-time packet loss of transmitting terminal feedback network, when transmitting terminal finds that real-time packet loss is less than
During the packet loss of minimum media play requirement, the threshold value and minimum transmission rate threshold value of maximum transmission rate are set, change hair
Transmission rate to 1.5 times of original transmission rate, reexamine real-time packet loss less than the requirement of minimum media play packet loss whether
Set up, if still setting up, continue to change transmission rate, if reaching the threshold value of maximum transmission rate, stop changing；
2) bandwidth of network is estimated using model, the appraising model of the network bandwidth is：
Wherein, TmaxBe the size of MTU in network, RTT be packet from transmitting terminal to receiving terminal again to hair
One time back and forth of sending end, PDIt is the probability of packet loss, RTO is the time-out time being transmitted again；
3) frame per second transmitted is adjusted, the packet to network transmission is sampled, and the packet after sampling is first
Being stored in buffering area carries out framing processing, in order to encoding and transmitting, the packet after sampling is classified, and is divided into speech data
Bag and video data bag, for VoP, sampled point is continuous, is divided into speech frame, and each speech frame includes 10
Individual continuous sampled point, for video data bag, is converted into two field picture, per second that 10 samplings of video are extracted from two field picture
Point, the encoder for sending into video is encoded；
4) during transmission, error correction is carried out, the mode of error correction is divided into voice error correction and video error correction；Voice error correction
It is divided into two steps, the first step increases redundant data in speech frame, the key message of speech frame described in redundant data, to connect
Receiving end error correction, recovery, reduction VoP, second step deposits the redundant data of next speech frame in speech frame, next
The redundant data of individual speech frame deposits the key message of next speech frame, so as to the VoP that receiving terminal is received from oneself
Middle recovery；Video error correction adds extra block code area information, block code area information storage Video coding in each two field picture
Feature, coding characteristic is extracted by video data bag grouping feature and obtained, therefore when there is partial video data packet loss, can be with root
The reconstruct of the video data bag lost according to coding characteristic.
Useful achievement of the invention is：Present invention employs the method that probe method is combined with modelling, network biography is carried out
The calculating of throughput rate, the network condition for remaining transmitting realtime stream media is swift in response, it is friendly, be easy to be received by fire wall, just
In realize with safeguard the advantages of, while breach transmission rate cannot adaptive network available bandwidth fluctuation, retransmit easily increase
The delay jitter of media bag and the obstacle such as out of order.
Specific embodiment
In order that the technical problems to be solved by the invention, technical scheme and beneficial effect become more apparent, below tie
Embodiment is closed, the present invention will be described in detail.It should be noted that specific embodiment described herein is only used to explain
The present invention, is not intended to limit the present invention, and can realize that the product of said function belongs to equivalent and improvement, is all contained in this hair
Within bright protection domain.Specific method is as follows：
Embodiment 1：Congestion and best effort model have close ties, due to the network user cannot cooperation share resource,
Therefore when multiple users file a request to consolidated network resource, so that it may congestion can be caused.When offered load is smaller, handling capacity
Linear relationship is kept substantially with offered load, and network delay is slowly increased after offered load exceedes key point, network throughput
Growth slows down, and network delay increases to be accelerated after offered load exceedes key point, and network throughput drastically declines, and network delay is fast
Speed increases.The purpose of congestion control is so that the application program of end main frame can rationally and effectively utilize Internet resources, obtains most
Good transmission performance, congestion control can be reduced or avoided network and enter congestion state, make the network operation in high-throughput, low latency
In the state of.Real time flow medium relative to tradition, for, there are requirements at the higher level to packet real-time,
Therefore real time flow medium application program has to carry out congestion control, ensures application program to changes in network resources
Adaptability, to make full use of the network bandwidth, reduces postpone and packet loss to greatest extent, it is ensured that the adaptability of Streaming Media.
The real time streaming congestion control being based on can be divided into the congestion control of unicast and the congestion control of multicast, due to
Agreement only supports the application of unicast in itself, and does not support multicast, therefore the present embodiment only discusses the congestion control of unicast.Unicast
The main method of Streaming Media congestion control is the sending rate control of end main frame.The purpose of speed control is by controlling Streaming Media
The change of the transmission rate matching network available bandwidth of source, reaches the possibility for reducing congestion, the mesh for reducing delay and packet loss
's.Rate control mechanism is by detecting the feedback such as network and packet loss situation, play quality, buffer size in conjunction with receiving terminal
Information, the coding parameter of streaming media, data packet format, sampling make corresponding adjustment into frame rate etc. so that transmitting terminal
Transmission rate can be effectively matched the available bandwidth of network, reach the purpose that congestion is reduced or avoided.
Embodiment 2：The main thought of probe method is actually the speed control that agreement is imitated on the basis of using agreement
Strategy adjusts the speed for sending data flow by the response message in transmitting terminal monitored data bag, at steady state, when not having
Linearly increasing transmission rate when having packet loss, and when data-bag lost is detected, that is, think congestion, speed will be sent immediately
Rate halves.
Modelling directly estimates current available network bandwidth by Mathematical Modeling, by the use of available bandwidth calculated value as
Constraints periodically adjusts transmission rate.
The appraising model of the network bandwidth in the present embodiment has been initially set up when it is entirely response,triple that packet loss is indicated
Throughput model.Then derived using similar method while thering is what two kinds of packet losses of response,triple and retransmission time out indicated to handle up
Amount model.Finally derive the throughput model when congestion window size is restricted.
The appraising model of the network bandwidth is based on following point and assumes：
1) prevailing tcp data stream is TCP Reno in network, therefore should be based on for the analysis of handling capacity
TCP Reno algorithms.
2) analysis of the slow start stage after the quick Restoration stage and time-out in TCP congestion avoidance algorithms to handling capacity
Result influence is little, can ignore.
3) the most routers in network use Queue Management Algorithm, and packet loss is presented broken hair.Once that is, certain number
According to packet loss, then the ensuing packet being in the same transmission cycle with it will all be dropped.
The error correction of transmission, congestion control attempts to prevent the generation of packet loss by matching for Media Stream and available bandwidth, but
In fact the packet loss gone up is inevitable, and it also has tremendous influence for stream media quality, therefore in the case where there is packet drop
How media are maximized, be exactly the main purpose of Streaming Media Error Control, mainly there is forward error correction, re-transmission, fault-tolerant volume at present
Code and four kinds of methods of error concealment.
The preferred embodiments of the invention is the foregoing is only, claims of the invention are not limited to.
Simultaneously it is described above, for those skilled in the technology concerned it would be appreciated that and implement, therefore other be based on institute of the present invention
The equivalent change that disclosure is completed, should be included in the covering scope of the claims.
Useful achievement of the invention is：Present invention employs the method that probe method is combined with modelling, network biography is carried out
The calculating of throughput rate, the network condition for remaining transmitting realtime stream media is swift in response, it is friendly, be easy to be received by fire wall, just
In realize with safeguard the advantages of, while breach transmission rate cannot adaptive network available bandwidth fluctuation, retransmit easily increase
The delay jitter of media bag and the obstacle such as out of order.
Claims (1)
1. a kind of transmission volume computational methods, it is characterised in that comprise the steps of：
1) two ends of network transmission are respectively defined as transmitting terminal and receiving terminal, feedback cycle is set, the feedback cycle is 1 milli
Second, the transmitting terminal periodically to the real-time packet loss of the receiving terminal feedback network, when the transmitting terminal finds the net
When the real-time packet loss of network is less than the packet loss that minimum media play is required, the threshold value of maximum transmission rate is set, increase is described
The transmission rate of transmitting terminal reexamines the real-time packet loss of the network less than minimum media to 1.5 times of original transmission rate
Whether the packet loss of broadcast request is still set up, if still setting up, continues to increase the transmission rate of the transmitting terminal, if reached
The threshold value of the maximum transmission rate, then stop increase；
2) bandwidth of network is estimated using model, the appraising model of the bandwidth of network is：
Wherein, TmaxBe the size of MTU in network, RTT be packet from the transmitting terminal to the receiving terminal again
To a time back and forth of the transmitting terminal, PDIt is the probability of packet loss, RTO is if re-starting the time-out time of transmission；
3) frame per second transmitted is adjusted, the packet to network transmission is sampled, and the packet after sampling is divided
Class, is divided into VoP and video data bag, and for the VoP, sampled point is continuous, by the voice
Packet is divided into speech frame, i.e., each described speech frame includes 10 continuous sampled points, for the video data
Bag, two field picture is converted into by the speech frame, per second that 10 discrete sampled points are extracted from the two field picture, and sends into video
Encoder encoded；
4) during transmission, error correction is carried out, the mode of error correction is divided into voice error correction and video error correction；The voice error correction
It is divided into two steps, the first step increases redundant data in the speech frame, the key of the speech frame described in the redundant data
Information, so as to receiving terminal energy error correction, recovers, reduces the VoP, second step, under being deposited in the speech frame
One redundant data of the speech frame, the redundant data of next speech frame deposits the key of next speech frame
Information, so that the receiving terminal recovers data from the VoP that oneself receives；The video error correction is in each institute
Extra block code area information of add in two field picture is stated, the coding characteristic of storage video, described to regard in described piece of code area information
The coding characteristic of frequency carries out feature extraction and obtains again after being grouped by the video data bag, therefore ought have the part video data
During packet loss, the reconstruct of the video data bag that can be lost according to the coding characteristic of the video.
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| CN201611203333.XA CN106789427A (en) | 2016-12-23 | 2016-12-23 | A kind of transmission volume computational methods |
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| CN201611203333.XA CN106789427A (en) | 2016-12-23 | 2016-12-23 | A kind of transmission volume computational methods |
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Cited By (6)
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| CN107483296A (en) * | 2017-09-26 | 2017-12-15 | 北京润科通用技术有限公司 | A kind of fiber-optic transmission rate method of testing and system |
| CN108390799A (en) * | 2018-03-23 | 2018-08-10 | 西安电子科技大学 | The measurement method of quantum key distribution network end-to-end available key rate |
| CN110572333A (en) * | 2019-10-25 | 2019-12-13 | 中南大学 | transmission rate adjusting method and congestion control algorithm in bandwidth detection stage |
| CN111327864A (en) * | 2018-12-17 | 2020-06-23 | 中移（杭州）信息技术有限公司 | Video call control method and device, terminal equipment and storage medium |
| CN111654871A (en) * | 2020-07-03 | 2020-09-11 | 张胜雄 | Network operation cost planning method based on operator |
| CN116260802A (en) * | 2023-05-15 | 2023-06-13 | 世优（北京）科技有限公司 | Data transmission method, device and system |
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| CN107483296A (en) * | 2017-09-26 | 2017-12-15 | 北京润科通用技术有限公司 | A kind of fiber-optic transmission rate method of testing and system |
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| CN111654871A (en) * | 2020-07-03 | 2020-09-11 | 张胜雄 | Network operation cost planning method based on operator |
| CN111654871B (en) * | 2020-07-03 | 2023-03-24 | 张胜雄 | Network operation cost planning method based on operator |
| CN116260802A (en) * | 2023-05-15 | 2023-06-13 | 世优（北京）科技有限公司 | Data transmission method, device and system |
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