CN100372291C - Method for realizing jitter resistance in jitter precognition network - Google Patents
Method for realizing jitter resistance in jitter precognition network Download PDFInfo
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- CN100372291C CN100372291C CNB031228984A CN03122898A CN100372291C CN 100372291 C CN100372291 C CN 100372291C CN B031228984 A CNB031228984 A CN B031228984A CN 03122898 A CN03122898 A CN 03122898A CN 100372291 C CN100372291 C CN 100372291C
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Abstract
The present invention relates to a method for realizing shake resistance in the field of communication technology. Under the condition of network shake prescience, a message is attached with a frame number field with rough time information at a message sending end, and a message receiving end is provided with a buffer area. The message can be sent by the frame number after entered the buffer area and waited for definite time, and thereby, network shake can be eliminated. The receiving end can find a communication opposite end restarting according to the abrupt change of the frame number, and adopt corresponding synchronous measures to make the communication continuously carry out. The present invention overcomes the defects that the existing AAL2, etc. of load bearing voice/media flow have shake and bandwidth waste, and provides a method for realizing shake resistance under the condition of coarse grain timestamps. Thereby, when the network shake in a range which can be foreseen, the communication quality of voice or other multimedia flow can be enhanced, and the bandwidth waste can be reduced simultaneously.
Description
Technical field
The present invention relates to communication technical field, when relating in particular to a kind of employing, under the predictable situation of network jitter, realize the method for shake opposing as asynchronous transfer mode adaptation layer two (AAL2) equivalent-load voice/Media Stream.
Background technology
At present, in order on the IP network that has bigger shake, to provide phone/multimedia communication service, except guaranteeing by service quality, opposing jitter-buffer function also is provided to provide at IP network termination place, to eliminate the influence that network jitter causes voice/Streaming Media, by means of RTP (RTP, Real-time Transport Protocol) help, the equipment of receiving terminal can recover the timed sending of voice/Media Stream again, just as IP network is an exclusive access with constant time lag characteristic.
In IP network, the jitter range of network is bigger, and stab (precision is 1/8000 second) information by precise time subsidiary in the message, and the due in of message, can know the time-delay that each message is experienced on network, because the shake of IP network, these time-delays can be had nothing in common with each other, by certain algorithm, can be by the network time-delay distribution situation in period the last period, next step time-delay situation of prediction network, after the time-delay that dopes network, the message that just can force follow-up a period of time to arrive is detained the regular hour in a special buffering area that is provided with, the shake of the network that the size of this time equals to dope, at this moment, even if the time-delay difference that each message experienced, but before needs send to message next link processing, can guarantee that abundant message has arrived buffering area, the situation that can occur stopping not, simultaneously, in theory, this delay time is the minimum time-delay that may reach under the equal effect, and after time-delay, message sends logic can be according to the timestamp information of message, message is sent, thereby reach jitter elimination.
In addition, order when IP network itself can not guarantee that message sends according to it arrives destination node, so, destination node need realize the disorderly continuous function of eliminating, in Real-time Transport Protocol, this function is to realize by the sequence number in the Real-time Transport Protocol, at transmitting terminal, stamp sequence number for all RTP messages, and make next sequence number of message equal present sequence of message number to add 1 (mould 65536); At receiving terminal, can recover the message sequence of transmit leg according to this information.
When RTP voice-bearer/Media Stream, the jitter elimination function at network two ends is stabbed information, message by means of correct time in the Real-time Transport Protocol and is arrived information such as moment of processing node and sequence number of message and realize the shake opposing, in transmission, timestamp accounts for 32 bits, sequence number accounts for 16 bits, has taken valuable transmission bandwidth.
In asynchronous transfer mode adaptation layer two (AAL2) carrying, the situation that does not have packet out-ordering, but, because the difference (representative value is between 1-5ms) of multiplexing stand-by period, so the shake be objective reality, unless do not adopt multiplexing, still, do not adopt the multiplexing waste on the bandwidth that brought, the original intention of this and AAL2 design of protocol is inconsistent.In order to eliminate the shake of AAL2 equivalent-load, it obviously is worthless directly adopting the way of Real-time Transport Protocol, this is because the expense of Real-time Transport Protocol for AAL2 (the average head consumption of AAL2 is 4 bytes, and RTP is 12 bytes) bigger than normal really, has been wasted bandwidth equally.
At present, in typical A AL2 uses in real time, or do not adopt multiplexing, force smaller (less than 1 millisecond) of multiplexing stand-by period setting, thereby jitter limits in a very little scope, is reduced to shake the influence to the real-time media current mass, but wasted bandwidth like this.
Summary of the invention
Technical problem to be solved by this invention is: the deficiency that has shake and bandwidth waste when overcoming existing AAL2 equivalent-load voice/Media Stream, a kind of method of the realization shake opposing under coarseness timestamp condition is provided, thereby make network jitter in the time can predicting in the scope for one, improve the communication quality of voice or other media streams, reduce the waste of bandwidth simultaneously.
The present invention solves the problems of the technologies described above the technical scheme that is adopted to be:
This method that realizes the shake opposing in the network of shake precognition is characterized in that it may further comprise the steps:
At message source, subsidiary 4 bit frame sequence number field on message with rough temporal information, the numerical value in described frame number territory increases 1 or subtract 1 every the time of a speech frame, and 16 deliverys are upgraded;
At message sink buffering area is set, message is waited for the corresponding time according to the jitter conditions of network after entering buffering area, and postponing to dash in the district by frame number sends, thereby eliminates network jitter.
Aging attribute is set on message, the attribute that will wear out when message has just entered buffering area is changed to 0, the time of the every fixed interval attribute that should wear out increases 1, if should aging attribute greater than enough threshold values of predefined stand-by period, illustrate that then message waited for the sufficiently long time in buffering area; If should continue to increase by aging property value,, the overlong time that message is waited for is described then in buffering area greater than the threshold value of predefined waits for too long.
System repeated buffering area is detected with the same time interval.
The initial start sign of having waited for sufficiently long time and buffering area when first message in the buffering area in buffering area is during set, perhaps, not set of initial start sign when buffering area, but when the frame number of first message equals the frame number of buffering area expectation at present, upgrade the frame number of buffering area expectation, and upgrade the type of message of the previous transmission of buffering area according to type of message, remove the initial start sign, and send this message.
If the initial start flag set of buffering area, then the frame number of the expectation of described buffering area increases 1 and 16 deliverys are upgraded by this message frame number; If the not set of initial start sign of buffering area, then the frame number of the expectation of described buffering area is by increasing 1 and 16 deliverys are upgraded.
If buffering area is empty, or first message is not waited for sufficiently long time and buffering area in buffering area initial start sign if the type of message of described previous transmission is a voice message, then sends wrong frame indication message during set; If the type of message of described previous transmission is quiet message, then send discontinuous message transmission.
If the not set of initial start sign of buffering area, and the frame number of first message is not equal to the buffering area frame number of expectation at present, then when overlong time that message is waited for, enforceable buffering area is emptied, and put the initial start sign, make the next message that arrives be considered to first message of this session; If message is not waited for the long time, the frame number of the expectation of buffering area upgrades by increasing 1 mould 16, and sends the bad frame indication.
Beneficial effect of the present invention is: the characteristics when the present invention is directed to AAL2 equivalent-load voice/Media Stream are simplified Real-time Transport Protocol, by subsidiary " frame number " territory on message with rough temporal information, under the cooperation of two ends shake opposing function, in transmission, can realize respond well shake opposing equally, can reach satisfied effect equally, thereby effectively utilized the mechanism of Real-time Transport Protocol, when network jitter can be predicted, under the situation that increases very little bandwidth, utilize the rough temporal information of data message, realization is to the elimination of bigger shake, with respect to adopting the required timestamp of Real-time Transport Protocol to account for 32 bits, sequence number accounts for 16 bits, valuable bandwidth resources have been saved, the deficiency that has shake and bandwidth waste when having overcome existing AAL2 equivalent-load voice/Media Stream, the communication quality of raising voice or other media streams.
Because frame number had both had rough temporal information, the characteristic that has general sequence mechanism simultaneously is so receiving terminal can use the sudden change of frame number to find that Correspondent Node restarts among the present invention, and take the corresponding synchronous measure, make communication to proceed.
The common encoding and decoding speech that has the silence compression function is gone up in voice service (Voice over Packet) based on packet switch, the invention provides more deep analyzing and processing, thereby under the situation of network report lost, for the entities such as digital signal processing logic in downstream send " bad frame indication " or " discontinuous transmission " message, thereby provide better support for realizing that the mistake of audio coder ﹠ decoder (codec) under error situation compensates, and further improved voice quality thus.
Description of drawings
Fig. 1 realizes shake opposing flow chart for the present invention.
Embodiment
With embodiment the present invention is described in further detail with reference to the accompanying drawings below:
Under the situation to protocol networkings such as AAL2, the shake that underlying protocol causes almost can be ignored, such as, the shake that ATM causes has only several microseconds, and the shake of can not ignore derives from the setting of multiplexing stand-by period, if the multiplexing stand-by period is 5 milliseconds, then the maximum jitter of network is exactly 5 milliseconds, so, can think that the shake of this network can predict (＜=5 milliseconds), under the predictable situation of network jitter, to reduce bandwidth waste when realizing the shake opposing, need special shake opposing function.
Different with IP network, during AAL2 equivalent-load voice/Media Stream, it is disorderly continuous message can not occur, so sequence number information not be used between two end points and transmits, thereby IP network is saved certain bandwidth relatively, and, under the predictable situation of network jitter, utilize some other information of all knowing of both sides in advance of voice flow/Media Stream, further compression time stabs.
The present invention is under the situation of network jitter precognition, utilize on the message existingly or increase " frame number " territory have rough temporal information, at receiving terminal buffering area is set, after message enters buffering area, according to the situation of network jitter, message sends after waiting for the corresponding time.Under the cooperation of two ends shake opposing function, utilize the present invention in transmission, can realize respond well shake opposing equally, can reach satisfied effect equally.
" frame number " territory increases 1 by the time every a speech frame, and this pairing greatest measure delivery in " frame number " territory upgraded, take 4 bits as frame number, then this " frame number " territory is just upgraded by " time every a speech frame increases 1, and to 16 deliverys ".Under situation such as ATM(Asynchronous Transfer Mode)/asynchronous transfer mode adaptation layer two network jitters such as (AAL2) precognition, the temporal information of this granularity is just enough, and this moment, this frame number has only taken the bandwidth of 4 bits, with respect to adopting the required timestamp of Real-time Transport Protocol to account for 32 bits, sequence number accounts for 16 bits, has saved valuable bandwidth resources.
Similar with the processing of jitter elimination among the RTP, when message arrives, with its afterbody of inserting buffering area (when just beginning, buffering area is empty, the initial start flag set of buffering area), the attributes such as frame number of preserving message use for the processing of back, system with fixing interval (as 1 millisecond, 5 milliseconds etc.) message that enters buffering area worn out (each message is provided with an old attribute, be changed to 0 when just entering buffering area, during each wearing out, this old attribute is increased 1), size has at interval determined the precision of time-delay control, more little at interval, precision is high more, but the disposal ability that needs is also high more, and system repeats buffering area is detected with same interval, be illustrated in figure 1 as the present invention and realize shake opposing flow chart, flow process is as follows:
If 1 buffering area is empty, to send " wrong frame indication " or " discontinuous transmission frame " message according to certain condition, and finish this processing, the condition of sending " wrong frame indication " or " discontinuous transmission frame " message is as follows:
(1), then sends " wrong frame indication " if the previous message that sends is a voice message; Wrong frame indication is a kind of sign of introducing in the algorithm, be used for showing that to digital signal processor (DSP) present needs send to the not arrival of message of DSP, perhaps wrong (such as checksum error, so be dropped), when receiving this message, DSP carries out mistake bag compensation (ErrorCancellation).
(2), then send " discontinuous transmission frame " if the previous message that sends is quiet message; Discontinuous transmission (DTX) frame is meant during silence compression, digital signal processor (DSP) can be exported the DTX frame, but do not need to pass to partner, with conserve bandwidth (benefit of silence compression that Here it is), and the recipient, when discovery is in during the silence compression, send the DTX frame to DSP equally, to guarantee passing to message of DSP in each frame period.
If 2 buffering areas are not empty, first message in the buffering area has been treated the sufficiently long time (if the old attribute of message greater than certain preset threshold, then illustrates the time that it " has waited for long enough ") in buffering area, then:
(1) if the initial start sign set of buffering area, " frame number of expectation " that then upgrade buffering area increases 1 (mould 16) for this message frame number, according to type of message (speech frame/quiet frame), " type of message of previous transmission " that upgrade buffering area is speech frame or quiet frame, remove " initial start sign ", and the transmission message, this processing finishes.
(2) if the initial start sign of buffering area does not have to be provided with (be that buffering area is not to be in the state that has just started, show also to be sent out away without any the message that receives at present), then:
If the frame number of first message in the A buffering area equals buffering area " frame number of expectation " at present, then " frame number of expectation " with buffering area increases 1 (mould 16), according to type of message (speech frame/quiet frame), " type of message of previous transmission " that upgrade buffering area is speech frame or quiet frame, remove " initial start sign ", and sending this speech frame, this processing finishes.
If the frame number of first message in the B buffering area is not equal to buffering area " frame number of expectation " at present, then:
A, if the overlong time in buffering area, treated of first message in the buffering area, as surpassed certain threshold value, then enforceable buffering area is emptied, and put the initial start sign, (the next message that arrives can be considered to first message of this session, thereby restart automatically), and send " bad frame indication " this processing and finish.
B otherwise, " frame number of expectation " of buffering area increased 1 (mould 16), send " bad frame indication ", this processing finishes.
3. if buffering area is not empty, first message in the buffering area do not treat the sufficiently long time in buffering area, then:
(1) as if the set of initial start sign, then sends " wrong frame indication " or " discontinuous transmission " message, and finish this processing according to the previous message that sends.
(2) if the initial start sign does not have set, then:
If the frame number of first message in the A buffering area equals buffering area " frame number of expectation " at present, then " frame number of expectation " with buffering area increases 1 (mould 16), according to type of message (speech frame/quiet frame), " type of message of previous transmission " that upgrade buffering area is speech frame or quiet frame, remove " initial start sign ", and sending this speech frame, this processing finishes.
If the frame number of first message in the B buffering area is not equal to buffering area " frame number of expectation " at present, then:
A, if the overlong time in buffering area, treated of first message in the buffering area, as surpassed certain threshold value, then enforceable buffering area is emptied, and put the initial start sign, (the next message that arrives can be considered to first message of this session, thereby restart automatically), and send " bad frame indication " this processing and finish.
B otherwise, " frame number of expectation " of buffering area increased 1 (mould 16), send " bad frame indication ", this processing finishes.
Among the present invention, because frame number had both had rough temporal information, the characteristic that has general sequence mechanism simultaneously, so, if the not set of initial start sign of buffering area, and the frame number of first message in the buffering area is not equal to buffering area " frame number of expectation " at present, and the overlong time for the treatment of in buffering area then enforceablely empties buffering area, and puts the initial start sign, the next like this message that arrives can be considered to first message of this session, thereby restart automatically, like this, receiving terminal finds that according to the sudden change of frame number Correspondent Node restarts, and take the corresponding synchronous measure, make communication to proceed.
The common encoding and decoding speech that has the silence compression function is gone up in voice service (Voice over Packet) based on packet switch, utilize the present invention under the situation of network report lost, for the entities such as digital signal processing logic in downstream send " bad frame indication " or " discontinuous transmission " message, thereby provide better support for realizing that the mistake of audio coder ﹠ decoder (codec) under error situation compensates, and further improved voice quality thus.
Certainly, if the frame number territory is upgraded by " time every a speech frame subtracts 1, and to 16 deliverys ", correspondingly, buffering area " frame number of expectation " can be realized the present invention equally also by successively decreasing and 16 deliverys being upgraded.Those skilled in the art do not depart from the scope of the present invention and spirit, can have the various deformation scheme to realize the present invention, and appended claim comprises these distortion.
Claims (7)
1. in the network of shake precognition, realize the method that shake is resisted for one kind, it is characterized in that it may further comprise the steps:
At message source, subsidiary 4 bit frame sequence number field on message with rough temporal information, the numerical value in described frame number territory increases 1 or subtract 1 every the time of a speech frame, and 16 deliverys are upgraded;
At message sink buffering area is set, message is waited for the corresponding time according to the jitter conditions of network after entering buffering area, and postponing to dash in the district by frame number sends, thereby eliminates network jitter.
2. the method that in the network of shake precognition, realizes the shake opposing according to claim 1, it is characterized in that: aging attribute is set on message, the attribute that will wear out when message has just entered buffering area is changed to 0, the time of the every fixed interval attribute that should wear out increases 1, if this aging attribute, then illustrates message greater than enough threshold values of predefined stand-by period and waited for the sufficiently long time in buffering area; If should continue to increase by aging property value,, the overlong time that message is waited for is described then in buffering area greater than the threshold value of predefined waits for too long.
3. the method that realizes the shake opposing in the network of shake precognition according to claim 2, it is characterized in that: system repeated buffering area is detected with the same time interval.
4. the method that in the network of shake precognition, realizes the shake opposing according to claim 3, it is characterized in that: the initial start sign of having waited for sufficiently long time and buffering area when first message in the buffering area in buffering area is during set, perhaps, not set of initial start sign when buffering area, but when the frame number of first message equals the frame number of buffering area expectation at present, upgrade the frame number of buffering area expectation, and upgrade the type of message of the previous transmission of buffering area according to type of message, remove the initial start sign, and send this message.
5. the method that in the network of shake precognition, realizes the shake opposing according to claim 4, it is characterized in that: if the initial start flag set of buffering area, then the frame number of the expectation of described buffering area increases 1 and 16 deliverys are upgraded by this message frame number; If the not set of initial start sign of buffering area, then the frame number of the expectation of described buffering area is by increasing 1 and 16 deliverys are upgraded.
6. the method that in the network of shake precognition, realizes the shake opposing according to claim 3, it is characterized in that: if buffering area is for empty, or first message is not waited for sufficiently long time and buffering area in buffering area initial start sign is during set, if the type of message of described previous transmission is a voice message, then send wrong frame indication message; If the type of message of described previous transmission is quiet message, then send discontinuous message transmission.
7. the method that in the network of shake precognition, realizes the shake opposing according to claim 3, it is characterized in that: if the not set of initial start sign of buffering area, and the frame number of first message is not equal to the buffering area frame number of expectation at present, then when overlong time that message is waited for, enforceable buffering area is emptied, and put the initial start sign, make the next message that arrives be considered to first message of this session; If message is not waited for the long time, the frame number of the expectation of buffering area upgrades by increasing 1 mould 16, and sends the bad frame indication.
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| CNB031228984A CN100372291C (en) | 2003-05-07 | 2003-05-07 | Method for realizing jitter resistance in jitter precognition network |
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Families Citing this family (6)
| Publication number | Priority date | Publication date | Assignee | Title |
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| CN102484601B (en) * | 2009-06-26 | 2014-12-10 | 瑞典爱立信有限公司 | Detection of jitter in a communication network |
| CN103795649B (en) * | 2013-11-06 | 2017-05-17 | 桂林电子科技大学 | Network delay jitter smoothing method |
| WO2021003707A1 (en) * | 2019-07-10 | 2021-01-14 | 海能达通信股份有限公司 | Synchronization method for voice information and communication system |
| CN112787877B (en) * | 2019-11-07 | 2022-08-26 | 华为技术有限公司 | Network delay detection method and related equipment |
| CN112822502B (en) * | 2020-12-28 | 2022-06-07 | 阿里巴巴（中国）有限公司 | Live broadcast jitter removal intelligent caching and live broadcast method, equipment and storage medium |
| CN114501114A (en) * | 2022-02-11 | 2022-05-13 | 福建星网智慧科技有限公司 | Audio network self-adaption method and storage device |
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| CN1453949A (en) * | 2002-04-26 | 2003-11-05 | 东方通信科技发展有限公司 | Method of eliminating business flow jitter and slip code in CDMA communication system |
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| CN1321396A (en) * | 1999-05-14 | 2001-11-07 | 皇家菲利浦电子有限公司 | Method of converting packetized stream of information signals into stream of information signals with time stamps and vice versa |
| CN1453949A (en) * | 2002-04-26 | 2003-11-05 | 东方通信科技发展有限公司 | Method of eliminating business flow jitter and slip code in CDMA communication system |
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