JPS6365181B2 - - Google Patents

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Publication number
JPS6365181B2
JPS6365181B2 JP1547282A JP1547282A JPS6365181B2 JP S6365181 B2 JPS6365181 B2 JP S6365181B2 JP 1547282 A JP1547282 A JP 1547282A JP 1547282 A JP1547282 A JP 1547282A JP S6365181 B2 JPS6365181 B2 JP S6365181B2
Authority
JP
Japan
Prior art keywords
delay time
voice
packets
audio
packet
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Expired
Application number
JP1547282A
Other languages
Japanese (ja)
Other versions
JPS58134557A (en
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed filed Critical
Priority to JP57015472A priority Critical patent/JPS58134557A/en
Publication of JPS58134557A publication Critical patent/JPS58134557A/en
Publication of JPS6365181B2 publication Critical patent/JPS6365181B2/ja
Granted legal-status Critical Current

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Classifications

    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L12/00Data switching networks
    • H04L12/64Hybrid switching systems

Landscapes

  • Engineering & Computer Science (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Signal Processing (AREA)
  • Data Exchanges In Wide-Area Networks (AREA)
  • Use Of Switch Circuits For Exchanges And Methods Of Control Of Multiplex Exchanges (AREA)

Description

【発明の詳細な説明】 本発明は音声パケツト交換システムにおける音
声パケツトの伝送遅延時間の変動を吸収する音声
パケツト伝送遅延制御方式に関するものである。
DETAILED DESCRIPTION OF THE INVENTION The present invention relates to a voice packet transmission delay control system for absorbing variations in the transmission delay time of voice packets in a voice packet switching system.

近年、音声をPCM方式等によりデイジタル化
し、更にパケツト化してパケツト交換装置を介し
て送受する音声パケツト交換システムが実現され
ている。第1図は音声パケツト交換システムの概
略図であり、1と2は電話機、3は電話機1から
のアナログ信号をデイジタル信号に変換するA/
D変換装置、4は有音無音検出装置、5はパケツ
ト組立て装置、71,72,…,7nはパケツト
交換装置、6はパケツト交換装置71〜7nから
成るパケツト交換網、8はパケツト分解装置、9
はデイジタル信号をアナログ信号に変換するD/
A変換装置、10は遅延時間の変動吸収装置であ
り、電話機1、A/D変換装置3、有音無音検出
装置4、パケツト組立装置5は送信側を、遅延時
間の変動吸収装置10、パケツト分解装置8、
D/A変換装置9、電話機2は受信側を示す。電
話機1からの音声信号はA/D変換装置3により
デイジタル化され、更に有音無音検出装置4によ
り転送単位毎に分割され音声の有無が判定され、
有音のみパケツト組立て装置5によりヘツダ情報
(アドレス情報等)の付加等の処理がなされてパ
ケツト化され、パケツト交換網6に送出される。
パケツト交換網6において、中継するパケツト交
換装置の数は送信側から受信側への呼の設定によ
つて異り、音声パケツトはパケツト交換装置71
〜7nを中継し受信側に伝送される。受信側にお
いて音声パケツトはパケツト分解装置8により分
解され、更にD/A変換装置9により音声信号に
復号化される。
In recent years, a voice packet switching system has been realized in which voice is digitized using a PCM method or the like, and further converted into packets and sent and received via a packet switching device. Figure 1 is a schematic diagram of a voice packet switching system, where 1 and 2 are telephones, and 3 is an A/C converter that converts analog signals from telephone 1 into digital signals.
D conversion device, 4 is a voice/silence detection device, 5 is a packet assembling device, 71, 72, . 9
is a D/ that converts digital signals to analog signals.
The A converter 10 is a delay time fluctuation absorber, and the telephone 1, A/D converter 3, utterance/silence detector 4, and packet assembly device 5 are on the transmitting side, and the delay time fluctuation absorber 10 and packet Decomposition device 8,
A D/A converter 9 and a telephone 2 represent the receiving side. The voice signal from the telephone 1 is digitized by the A/D converter 3, and further divided into transmission units by the voice/silence detector 4 to determine the presence or absence of voice.
The sound-only packet assembler 5 performs processing such as adding header information (address information, etc.), converts it into a packet, and sends it to the packet switching network 6.
In the packet switching network 6, the number of relaying packet switching devices varies depending on the setting of the call from the sending side to the receiving side, and voice packets are transferred to the packet switching device 71.
~7n is relayed and transmitted to the receiving side. On the receiving side, the audio packets are decomposed by a packet decomposer 8 and further decoded into audio signals by a D/A converter 9.

パケツト交換システムにおいては、送信側から
送出された音声パケツトが受信側に到達するまで
には、主として送信側のパケツト組立て装置5に
おけるパケツト組立、受信側のパケツト分解装置
8におけるパケツトの分解ならびに伝送に要する
固定の遅延時間と、中継するパケツト交換装置7
1〜7nでの待ち合せによつて遅延され、後者の
遅延時間は一定でない。音声パケツト伝送におけ
る遅延時間の変動は音声の劣化を招くのみなら
ず、極端な場合には会話を不可能にする場合もあ
るので、音声パケツトの受信に際しては遅延時間
の変動を吸収することが必要不可欠であり、従来
においても遅延時間の変動は遅延時間変動吸収装
置10により処理されていた。
In a packet switching system, before a voice packet sent from the transmitting side reaches the receiving side, it mainly undergoes packet assembly in the packet assembling device 5 on the transmitting side, and disassembly and transmission in the packet disassembling device 8 on the receiving side. Required fixed delay time and relaying packet switching device 7
1 to 7n, and the latter delay time is not constant. Fluctuations in delay time in voice packet transmission not only cause audio deterioration, but in extreme cases may even make conversation impossible, so it is necessary to absorb fluctuations in delay time when receiving voice packets. This is indispensable, and in the past, variations in delay time have been handled by the delay time variation absorbing device 10.

従来、遅延時間変動吸収装置10における音声
パケツトの伝送遅延時間の変動を吸収する方式と
しては、受信した音声パケツトを順次バツフアメ
モリに蓄積し、各音声パケツトの遅延時間を一定
の固定値に補正するか、音声時間のトークスパー
ト(有音部)ごとに遅延時間を変えて補正する方
式が行われていた。前者の方式、すなわち受信し
た各音声パケツトの遅延時間を一定の固定値Dに
補正する方式では、伝送遅延時間D1で受信側の
遅延時間変動吸収装置10に入力した音声パケツ
トを前記バツフアメモリに(D―D1)時間滞在
させ、同様に伝送遅延時間Doで受信した音声パ
ケツトは前記バツフアメモリに(D―Do)時間
滞在させることにより各音声パケツトの遅延時間
を一定値Dに補正することにより伝送遅延時間の
変動を吸収し、伝送遅延時間がD以上の音声パケ
ツトは音声品質の劣化の要因となる為、廃棄する
ものである。しかし音声パケツトのパケツト交換
網での遅延時間の分布は、中継パケツト交換装置
当りの平均伝送遅延時間を等しいものとすると
き、第2図に示すガンマ分布となる。第2図にお
いて、横軸は伝送遅延時間(t)を縦軸は遅延時
間の累積分布(%)を示し、図中の数字1,2,
3,4は各々中継パケツト交換装置数に対応す
る。同図において中継パケツト交換装置数、各々
に対応して2つの曲線が示されているが、これは
各中継パケツト交換装置数における平均伝送遅延
時間の大小による違いを示す。1中継パケツト交
換装置当りの平均伝送遅延時間が等しいとき、同
じ伝送遅延時間に対して比較した場合、累積分布
は中継パケツト交換装置数が小さい場合の方が大
きい場合に比べて高く、因つて中継パケツト交換
装置の数が小さい場合の伝送遅延時間の変動は小
さく、逆に中継パケツト交換装置の数が大きい場
合の伝送遅延時間の変動は大きいことが明らかと
なる。従つてこの方式では中継パケツト交換装置
数による伝送遅延時間の変動に対処できず、パケ
ツト交換網6での伝送遅延時間の変動が小さい時
は音声パケツトを前記バツフアメモリに蓄積する
時間を不必要に長くすることになり、又逆に伝送
遅延時間の変動が大きい時はパケツト紛失率(音
声パケツトをバツフアメモリから取り出すべき時
刻に音声パケツトがバツフアメモリに無い確率)
が高く、音声品質に支障をきたすなどの欠点があ
つた。
Conventionally, as a method for absorbing variations in the transmission delay time of voice packets in the delay time variation absorbing device 10, received voice packets are sequentially stored in a buffer memory and the delay time of each voice packet is corrected to a certain fixed value. , a method was used to compensate by changing the delay time for each talk spurt (sound part) of the audio time. In the former method, that is, a method in which the delay time of each received audio packet is corrected to a constant fixed value D, the audio packet input to the delay time fluctuation absorbing device 10 on the receiving side at a transmission delay time D 1 is stored in the buffer memory ( Similarly , audio packets received with transmission delay time Do are allowed to stay in the buffer memory for (D- D o ) time, thereby correcting the delay time of each audio packet to a constant value D. This absorbs fluctuations in transmission delay time, and audio packets with a transmission delay time of D or more are discarded because they cause deterioration in audio quality. However, the distribution of delay times of voice packets in a packet switching network becomes a gamma distribution as shown in FIG. 2, assuming that the average transmission delay time per relay packet switching device is equal. In Figure 2, the horizontal axis shows the transmission delay time (t) and the vertical axis shows the cumulative distribution of delay time (%).
3 and 4 correspond to the number of relay packet switching devices, respectively. In the same figure, two curves are shown corresponding to the number of relay packet switching devices, and these curves indicate differences depending on the size of the average transmission delay time for each number of relay packet switching devices. When the average transmission delay time per relay packet switching device is equal, when comparing the same transmission delay time, the cumulative distribution is higher when the number of relay packet switching devices is small than when it is large; It is clear that the variation in transmission delay time is small when the number of packet switching devices is small, and on the contrary, the variation in transmission delay time is large when the number of relay packet switching devices is large. Therefore, this method cannot cope with variations in transmission delay time due to the number of relay packet switching devices, and when the variation in transmission delay time in packet switching network 6 is small, the time required to store voice packets in the buffer memory becomes unnecessarily long. On the other hand, when the transmission delay time fluctuates greatly, the packet loss rate (the probability that an audio packet is not in the buffer memory at the time when it should be taken out) increases.
However, there were drawbacks such as high audio quality and poor audio quality.

又後者の方式すなわち音声信号のトークスパー
ト(有音部)ごとに遅延時間を変えて伝送遅延時
間の変動を補正する方式では、パケツト交換網6
での伝送遅延時間の分布が第3図に示す指数分布
に従うものとしてこの関係を利用するものであ
る。すなわち、第3図は第2図と同様に横軸を伝
送遅延時間(t)、縦軸を累積分布(%)として
音声パケツトの伝送遅延時間(t)に対する累積
分布の関係を示す外、縦軸を更に累積分布の目盛
りとは逆にパケツトの紛失率0〜100%(累積分
布100%に対してパケツト紛失率0%、累積分布
0%に対してパケツト紛失率100%)とし、音声
信号のトークスパート(有音部)ごとに平均伝送
遅延時間を求めることにより伝送遅延時間分布の
曲線を求め、あらかじめ決定しておいたパケツト
紛失率に対応する伝送遅延時間を求め(第3図に
おけるa→b→c)、該伝送遅延時間を補正すべ
き遅延時間Dとし、次のトークスパートの音声パ
ケツトの伝送遅延時間の変動を吸収する時に前者
の方式にて説明した方法で補正を行うものであ
る。この方式は伝送遅延時間の分布に従い、パケ
ツト紛失率が一定になるように遅延時間を制御す
るので、前者の方式に比較してより良い通話品質
を得ることができるが、第2図にて説明したよう
に、パケツト交換網での伝送遅延時間の分布は中
継パケツト交換装置数と平均伝送遅延時間に依存
するガンマ分布になる為、第3図に示す指数分布
で近似した場合、実際の遅延時間分布との差が大
きく、遅延時間を求める際に誤差が大きくなりパ
ケツト紛失率を一定にできず、通話品質を保持す
ることは困難であつた。
In addition, in the latter method, that is, a method in which the delay time is changed for each talk spurt (sound part) of the audio signal to compensate for variations in transmission delay time, the packet switching network 6
This relationship is used assuming that the distribution of transmission delay times in 2 follows the exponential distribution shown in FIG. In other words, as in Figure 2, Figure 3 shows the relationship between the cumulative distribution and the transmission delay time (t) of voice packets, with the horizontal axis as the transmission delay time (t) and the vertical axis as the cumulative distribution (%). Further, the axis is set opposite to the scale of the cumulative distribution, and the packet loss rate is 0 to 100% (packet loss rate is 0% for the cumulative distribution of 100%, packet loss rate is 100% for the cumulative distribution of 0%), and the audio signal is The transmission delay time distribution curve is determined by determining the average transmission delay time for each talk spurt (sound part), and the transmission delay time corresponding to the predetermined packet loss rate is determined (a in Figure 3). →b→c), the transmission delay time is set as the delay time D to be corrected, and the correction is performed using the method explained in the former method when absorbing the variation in the transmission delay time of the voice packet of the next talk spurt. be. This method controls the delay time so that the packet loss rate is constant according to the transmission delay time distribution, so it can obtain better call quality than the former method, but this is explained in Figure 2. As shown above, the distribution of transmission delay time in a packet switching network is a gamma distribution that depends on the number of relay packet switching devices and the average transmission delay time, so when approximated by the exponential distribution shown in Figure 3, the actual delay time The difference in the distribution was large, and the error in determining the delay time was large, making it impossible to keep the packet loss rate constant and making it difficult to maintain call quality.

本発明の目的はこれらの欠点を除去する為に、
パケツト交換網での変動のある伝送遅延時間の分
布をガンマ分布に従うものとし、受信側にて伝送
遅延時間を測定し、前記伝送遅延時間の分布に応
じて遅延時間を制御してパケツト紛失率を一定に
し、通話品質を保持するようにしたものであり、
以下詳細に説明する。
The purpose of the present invention is to eliminate these drawbacks.
Assuming that the distribution of transmission delay time, which fluctuates in a packet switching network, follows a gamma distribution, the transmission delay time is measured on the receiving side, and the packet loss rate is reduced by controlling the delay time according to the transmission delay time distribution. It is designed to maintain the call quality by keeping it constant.
This will be explained in detail below.

第4図は本発明の実施例であつて第1図の遅延
時間変動吸収装置10に相当するものであり、1
1は中継パケツト交換装置数入力端子、12は中
継パケツト交換装置数受信回路、13は処理装
置、14は音声パケツト入力端子、15は遅延時
間変動吸収用バツフアメモリ、16は音声パケツ
ト受信時刻測定回路、17はタイマ回路、18は
遅延時間記憶部、19は制御回路、20はタイマ
回路、21は音声パケツト出力端子である。これ
を動作させるには、まず呼の設定時に中継パケツ
ト交換装置数入力端子11に中継パケツト交換装
置数を入力し、中継パケツト交換装置数受信回路
12は該中継パケツト交換装置数を受信して処理
装置13に与える。本実施例においては1つの呼
について同一の経路で音声パケツトが転送される
為、中継パケツト交換装置数は共通線信号方式等
種々の方式において受信局にて容易に知ることが
できる。音声パケツト入力端子14より入力され
た音声パケツトは順次遅延時間変動吸収用バツフ
アメモリ15に入力され蓄積される。音声パケツ
ト受信時刻測定回路16はタイマ回路17を用い
音声パケツトを受信するごとに受信時刻を測定
し、処理装置13に与える。処理装置13は前記
中継パケツトの受信時刻を用い、音声のトークス
パート(有音部)において連続する2つの音声パ
ケツトの受信時刻の差の平均値を求め、更に誤差
の平均値と前記中継パケツト交換装置数により、
第5図に示す受信時刻の差の平均値と中継パケツ
ト交換装置数に対応した遅延時間を記憶するリー
ドオンリメモリで構成された遅延時間記憶部18
を参照し、該記憶内容に従つて遅延時間値を制御
回路19に与える。尚第5図は上記のように第4
図の遅延時間記憶部18の記憶内容を示したもの
であり中継パケツト交換装置数1,2,…,Nと
前記トークスパートにおいて連続する2つの音声
パケツトの受信時刻の差の平均値E1,E2,…,
EMに対応した遅延時間D11〜DNMを記憶している。
制御回路19は遅延時間変動吸収用バツフアメモ
リ15に音声パケツトが蓄積されているか否かを
調べ、蓄積されていない時処理装置13より与え
られた遅延時間を新しい遅延時間として設定す
る。又制御回路19はタイマ回路20を用い、設
定更新された後最初に受信された音声パケツトを
前記設定更新された遅延時間だけ遅延時間変動吸
収用バツフアメモリに蓄積した後、該蓄積された
音声パケツトをFirst―In―First―Out方式で音
声パケツト出力端子21に出力し、個々の音声パ
ケツトの伝送遅延時間の変動を吸収する。
FIG. 4 shows an embodiment of the present invention, which corresponds to the delay time fluctuation absorbing device 10 in FIG.
1 is a relay packet switching device number input terminal, 12 is a relay packet switching device number receiving circuit, 13 is a processing device, 14 is a voice packet input terminal, 15 is a buffer memory for absorbing delay time fluctuations, 16 is a voice packet reception time measuring circuit, 17 is a timer circuit, 18 is a delay time storage section, 19 is a control circuit, 20 is a timer circuit, and 21 is an audio packet output terminal. To operate this, first, when setting up a call, the number of relay packet switching devices is input to the number of relay packet switching devices input terminal 11, and the number of relay packet switching devices receiving circuit 12 receives and processes the number of relay packet switching devices. device 13. In this embodiment, since voice packets are transferred through the same route for one call, the number of relay packet switching devices can be easily known at the receiving station using various systems such as the common line signaling system. Audio packets input from the audio packet input terminal 14 are sequentially input to a buffer memory 15 for absorbing delay time fluctuations and stored therein. The voice packet reception time measuring circuit 16 uses a timer circuit 17 to measure the reception time every time a voice packet is received, and provides it to the processing device 13. Using the reception time of the relay packet, the processing device 13 calculates the average value of the difference between the reception times of two consecutive voice packets in the talk spurt (sound part) of the voice, and further calculates the average value of the difference between the reception time of the relay packet and the relay packet exchange. Depending on the number of devices,
A delay time storage unit 18 consisting of a read-only memory that stores the average value of the difference in reception times and the delay time corresponding to the number of relay packet switching devices shown in FIG.
, and provides a delay time value to the control circuit 19 according to the stored contents. In addition, Figure 5 shows the fourth
This figure shows the stored contents of the delay time storage unit 18, and the number of relay packet switching devices 1, 2, ..., N and the average value E 1 of the difference in reception time of two consecutive voice packets in the talk spurt, E 2 ,…,
Delay times D 11 to D NM corresponding to EM are stored.
The control circuit 19 checks whether or not audio packets are stored in the buffer memory 15 for absorbing delay time variations, and if not, sets the delay time given by the processing device 13 as a new delay time. The control circuit 19 also uses a timer circuit 20 to store the first received audio packet after the updated settings in a buffer memory for absorbing delay time fluctuations by the updated delay time, and then stores the stored audio packets in a buffer memory for absorbing delay time fluctuations. The audio packets are output to the audio packet output terminal 21 using the first-in-first-out method, and variations in the transmission delay time of individual audio packets are absorbed.

次に遅延時間の制御方法を詳細に説明する。第
6図は音声信号の伝送状態を示す説明図であり、
矩形波の高い部分はトークスパート(有音部)
を、矩形波の低い部分はポーズ(無音部)を示
し、各トークスパートにおける上向きの矢印は
各々音声パケツトを表わす。同図において、受信
側で第n番目のトークスパートの音声パケツトを
受信し、後述の方法で遅延時間を求め、更に第n
番目のポーズ時に遅延時間変動吸収用バツフアメ
モリ15に音声パケツトが蓄積されていないこと
を確認して遅延時間を更新し、次の第n+1番目
のトークスパートの音声パケツトは新しく更新さ
れた遅延時間によつて変動が吸収される。又第n
番目のポーズが短く、遅延時間変動吸収用バツフ
ア15が空にならない場合は、遅延時間を更新せ
ず第n+1番目のトークスパートの音声パケツト
の伝送遅延時間の変動を吸収する。以下、処理装
置13が中継パケツト交換装置数と音声パケツト
受信時刻よりパケツト交換網での遅延時間の分布
を求め、さらに該遅延時間分布より遅延時間を得
る方法を説明する。Tiをi番目の音声パケツトが
送信された時刻、Riを受信された時刻とすると、
Riは下記の(1)式で表わされる。
Next, a method for controlling the delay time will be explained in detail. FIG. 6 is an explanatory diagram showing the transmission state of the audio signal,
The high part of the square wave is the talk spurt (sound part)
, the low parts of the square wave indicate pauses (silence), and the upward arrows in each talk spurt each represent a voice packet. In the figure, the receiving side receives the audio packet of the nth talk spurt, calculates the delay time using the method described later, and then
At the time of the pause, it is confirmed that no audio packets are accumulated in the buffer memory 15 for absorbing delay time variation, and the delay time is updated, and the audio packets of the next (n+1)th talk spurt are updated according to the newly updated delay time. fluctuations are absorbed. Also nth
If the th pause is short and the delay time variation absorbing buffer 15 is not empty, the delay time is not updated and the variation in the transmission delay time of the voice packet of the (n+1)th talk spurt is absorbed. Hereinafter, a method will be described in which the processing device 13 determines the delay time distribution in the packet switching network from the number of relay packet switching devices and the voice packet reception time, and further obtains the delay time from the delay time distribution. Let T i be the time when the i-th voice packet was sent, and R i be the time it was received.
R i is expressed by the following equation (1).

Ri=Ti+Dc+Qi(i=1、2、3、…) ……(1) ここでDcはパケツト交換網における固定量の
遅延時間、Qiは音声パケツト毎に異なる変動のあ
る遅延時間を表わす。(1)式よりQi=Ri−Ti−Dc
してQiの分布を求めれば良いが、受信側では音声
パケツトの送信時刻Tiを知り得ず、Qiを直接求め
られない。そこで測定可能な受信時刻Riから|Qi
−Qi-1|=|Ri−Ri-1−T|(ここでi=2、3、
…とし、Tは音声パケツトの送信時間間隔を表わ
す)として|Qi−Qi-1の分布の平均値を求め、そ
れよりQiの分布を求める。Qiの分布関数をF(X)
とすると、F(X)は中継パケツト交換装置数K
に依存し、中継パケツト交換装置当たりの平均伝
送遅延時間が等しいとき一般に(2)式のガンマ分布
で表わされることが知られている。
R i = T i + D c + Q i (i = 1, 2, 3,...) ...(1) Here, D c is the fixed amount of delay time in the packet switching network, and Q i is the variable delay time that differs for each voice packet. Represents a certain delay time. The distribution of Q i can be found from equation (1) by setting Q i = R i −T i −D c , but the receiving side cannot know the transmission time T i of the voice packet and cannot directly determine Q i . From the measurable reception time R i |Q i
−Q i-1 |=|R i −R i-1 −T| (where i=2, 3,
..., and T represents the transmission time interval of voice packets), find the average value of the distribution of |Q i -Q i-1 , and then find the distribution of Q i . The distribution function of Q i is F(X)
Then, F(X) is the number of relay packet switching devices K
It is known that when the average transmission delay time per relay packet switching device is equal, it is generally expressed by the gamma distribution of equation (2).

F(X)=λK・XK-1・e-X/(K−1)〓 ……(2) ここでX>0、λ>0、Kは中継パケツト交換
装置数であり、又K/λは分布の平均値Efを表わす。
F(X)=λ KX K- 1・e - 〓 K/λ represents the mean value E f of the distribution.

そしてF(X)は平均値Efを求めることにより一
義的に決まる。一方、|Qi−Qi-1|の分布の平均
値をEsとするとEsは(3)式で表わされる。
F(X) is uniquely determined by finding the average value E f . On the other hand, if E s is the average value of the distribution of |Q i −Q i-1 |, E s is expressed by equation (3).

Es=1/λ2k-2i=k-1iCK-1(2K−1−i)/2i} ……(3) 従つてEfとEsの関係は(4)式となる。 E s = 1/λ 2k-2i=k-1 { i C K-1 (2K-1-i)/2 i } ...(3) Therefore, the relationship between E f and E s is (4) The formula becomes

Ef=K・Es2k-2i=k-1iCK-1(2K−1−i)/2i}〕-1 ……(4) すなわち受信側で|Qi−Qi-1|の分布の平均値
Esを測定することによりEfが(4)式の計算により求
まり、更に(2)式を用いQiの分布関数F(X)すな
わち伝送遅延時間の分布が求まる。
E f = K・E s [ 2k-2i=k-1 { i C K-1 (2K-1-i)/2 i }] -1 ...(4) That is, on the receiving side |Q i − The mean value of the distribution of Q i-1 |
By measuring E s , E f is found by calculating equation (4), and furthermore, using equation (2), the distribution function F(X) of Q i , that is, the distribution of transmission delay time is found.

次にQiの分布関数F(X)より遅延時間を求め
る方法を以下に示す。(2)式より遅延時間がD以上
の確率Pは(5)式で表わされる。
Next, a method for determining the delay time from the distribution function F(X) of Q i will be shown below. From equation (2), the probability P that the delay time is greater than or equal to D is expressed by equation (5).

P=e-D k-1r=0 (λD)r/r〓 ……(5) 又(5)式のPはパケツト紛失率を表わす為、通話
品質を一定に保つためには(5)式においてPが一定
になるようにλ(=K/Ef)に対応して遅延時間
Dを求めれば良いことになる。すなわち、受信側
で音声パケツト受信時刻を順次測定し、連続する
音声パケツトの受信時刻の差の平均値Esを計算
し、(4)式を用いてパケツト交換網における平均伝
送遅延時間EfすなわちK/λを求める。さらに(5)
式を用い、前記λ、中継パケツト交換装置数Kお
よびパケツト紛失率Pより遅延時間Dを求める。
P=e -D k-1r=0 (λD) r / r〓 ...(5) Also, since P in equation (5) represents the packet loss rate, in order to keep the call quality constant, ( In formula 5), it is sufficient to find the delay time D corresponding to λ (=K/E f ) so that P is constant. That is, the reception side sequentially measures the voice packet reception times, calculates the average value E s of the difference between the reception times of consecutive voice packets, and uses equation (4) to calculate the average transmission delay time E f in the packet switching network, that is, Find K/λ. Furthermore(5)
Using the formula, the delay time D is calculated from the above λ, the number K of relay packet switching devices, and the packet loss rate P.

第5図に示す遅延時間記憶18の記憶テーブル
は複数の音声パケツトの各々連続する2つの音声
パケツトの受信時刻の差の平均値Es(=E1、E2
…、EM)と中継パケツト交換装置数K(=1、
2、…、N)をパラメータとして、前述の方法に
より求めた遅延時間DNHを記憶している。
The memory table of the delay time memory 18 shown in FIG. 5 stores the average value E s (=E 1 , E 2 ,
..., E M ) and the number of relay packet switching devices K (=1,
2 , .

従つて第6図にて上述したように、第n番目の
トークスパートにおいて各々連続する2つの音声
パケツトの受信時刻の差の平均値Esと中継パケツ
ト交換装置数Kをパラメータとして、遅延時間記
憶部18から遅延時間Dを読み出し、更に第n番
目のポーズ時に遅延時間変動吸収用バツフアメモ
リ15に音声パケツトが蓄積されていないことを
確認して遅延時間を前記遅延時間Dに更新し、次
の第n+1番目のトークスパートの音声パケツト
は新しく更新された遅延時間Dによつて伝送遅延
時間の変動が吸収される。そして第n+1番目の
トークスパートにおいては、最初に受信した音声
パケツトを遅延時間Dだけ遅延時間変動吸収用バ
ツフアメモリ15に蓄積させた後、該順次蓄積さ
れた音声パケツトをFirst―In―First―Out方式
で音声パケツト出力端子21に出力して個々の音
声パケツトの伝送遅延時間の変動を吸収する。
Therefore, as described above with reference to FIG. 6, the delay time is stored using the average value Es of the difference in reception times of two consecutive voice packets in the n-th talk spurt and the number K of relay packet switching devices as parameters. The delay time D is read out from the unit 18, and furthermore, at the nth pause, it is confirmed that no audio packets are stored in the buffer memory 15 for absorbing delay time variation, and the delay time is updated to the delay time D. For the audio packet of the (n+1)th talk spurt, the newly updated delay time D absorbs the variation in transmission delay time. In the (n+1)th talk spurt, the first received audio packet is stored in the buffer memory 15 for absorbing delay time variation by the delay time D, and then the sequentially stored audio packets are stored in the first-in-first-out method. The audio packets are output to the audio packet output terminal 21 to absorb variations in transmission delay time of individual audio packets.

以上説明したように第1の実施例では音声パケ
ツトの伝送遅延時間の変動を補正する為の遅延時
間を、パケツト交換網での伝送遅延の分布に従つ
て制御することにより、伝送遅延時間の変動の大
小にかかわらず通話の品質を保持することが可能
であり、又伝送遅延時間の変動が小さい時は遅延
量を小さくできるので、相手の音声が遅れて聞こ
え会話がやりづらくなるという欠点を除去できる
利点がある。
As explained above, in the first embodiment, the delay time for correcting the variation in the transmission delay time of voice packets is controlled in accordance with the transmission delay distribution in the packet switching network, thereby reducing the variation in the transmission delay time. It is possible to maintain the quality of the call regardless of the size of the transmission delay, and when the variation in transmission delay time is small, the amount of delay can be reduced, eliminating the drawback that the other party's voice is delayed and it becomes difficult to have a conversation. There are advantages that can be achieved.

第1の実施例では遅延時間記憶部18に1つの
トークスパートにおける複数の音声パケツトの連
続する各々2つの音声パケツトの受信時刻の差の
平均値Esと中継パケツト交換装置数Kに対応する
遅延時間DNMを記憶して、遅延時間を制御するこ
とにより個々の音声パケツトの伝送遅延時間の変
動を補正していたが、遅延時間記憶部18に遅延
時間の代りに各トークスパートの最初に受信され
た音声パケツトからの音声パケツト受信個数を記
憶させることによつても同様の効果を得ることが
できる。
In the first embodiment, the delay time storage unit 18 stores the average value E s of the difference in reception time of each of two consecutive voice packets of a plurality of voice packets in one talk spurt and the delay corresponding to the number K of relay packet switching devices. By storing the time D NM and controlling the delay time, fluctuations in the transmission delay time of individual voice packets were corrected. A similar effect can be obtained by storing the number of voice packets received from the received voice packets.

すなわち、第5図に示す記憶テーブルに前記第
1の実施例における遅延時間DNMに対応する音声
パケツトの受信個数XNM(≒遅延時間DNM/音声パ
ケツト送信時間間隔)を前記平均値Esと中継パケ
ツト交換装置数Kに対応させて記憶させておき、
第1の実施例と同じく第6図の第n番目のトーク
スパートにおいて前記平均値Esと中継パケツト交
換装置数Kをパラメータとして遅延時間記憶部1
8から受信個数Xを読み出す。次に第n番目のポ
ーズ時に遅延時間変動吸収用バツフアメモリ15
に音声パケツトが蓄積されていないことを確認し
て受信個数を前記受信個数Xに更新し、次の第n
+1番目のトークスパートは新しく更新された受
信個数Xによつて伝送遅延時間の変動が吸収され
る。そして第n+1番目のトークスパートにおい
ては、最初に受信した音声パケツトを、該最初の
音声パケツトから前記受信個数Xだけ後続する音
声パケツトが受信されるまで遅延時間変動吸収用
バツフアメモリ15に蓄積させた後該順次蓄積さ
れた音声パケツトをFirst―In―First―Out方式
で音声パケツト出力端子21に出力して個々の音
声パケツトの伝送遅延時間の変動を吸収できる。
That is, the number of received voice packets X NM (≒delay time D NM / voice packet transmission time interval) corresponding to the delay time D NM in the first embodiment is stored in the storage table shown in FIG. 5 as the average value Es and is stored in correspondence with the number K of relay packet switching devices,
Similarly to the first embodiment, in the n-th talk spurt in FIG. 6, the delay time storage unit 1 uses the average value Es and the number K of relay packet switching devices as parameters.
The received number X is read from 8. Next, at the n-th pause, the buffer memory 15 for absorbing delay time fluctuations
After confirming that no voice packets have been accumulated in , the number of received packets is updated to the number of received packets
For the +1st talk spurt, fluctuations in transmission delay time are absorbed by the newly updated reception number X. In the (n+1)th talk spurt, the first received voice packet is stored in the buffer memory 15 for absorbing delay time fluctuations until the voice packets following the first voice packet by the number of received voice packets are received. The sequentially accumulated voice packets are outputted to the voice packet output terminal 21 in a first-in-first-out manner, thereby absorbing variations in the transmission delay time of individual voice packets.

本発明は伝送遅延時間の変動をパケツト交換網
での伝送遅延の分布に従つて補正している為、伝
送遅延時間の変動の大小にかかわらず通話の品質
を保持することが可能であり、又伝送遅延時間の
変動が小さい時は遅延量を小さくできるので相手
方の音声が遅れて聞こえ会話がやりづらくなると
いう欠点を除くことができ、音声パケツト交換シ
ステムにおいて非常に有効な伝送遅延制御方式を
提供するものである。
Since the present invention corrects variations in transmission delay time according to the distribution of transmission delays in the packet switching network, it is possible to maintain call quality regardless of the magnitude of variation in transmission delay time, and When the fluctuations in transmission delay time are small, the amount of delay can be reduced, which eliminates the disadvantage that the other party's voice is delayed and makes it difficult to have a conversation, providing a very effective transmission delay control method in voice packet switching systems. It is something to do.

【図面の簡単な説明】[Brief explanation of drawings]

第1図は音声パケツト交換システムの概略図、
第2図は音声パケツト交換システムにおける伝送
遅延時間の分布図、第3図は従来方式において用
いられていた伝送遅延時間の近似分布図、第4図
は本発明の実施例であつて第1図の遅延時間変動
吸収装置10に相当するブロツク図、第5図は第
4図の遅延時間記憶部18の記憶内容を示す説明
図、第6図は音声信号のタイムチヤートである。 10…遅延時間変動吸収装置、11…中継パケ
ツト交換装置数入力端子、12…中継パケツト交
換装置数受信回路、13…処理装置、14…音声
パケツト入力端子、15…遅延時間変動吸収用バ
ツフアメモリ、16…音声パケツト受信時刻測定
回路、17,20…タイマ回路、18…遅延時間
記憶部、19…制御回路、21…音声パケツト出
力端子。
Figure 1 is a schematic diagram of a voice packet switching system.
Figure 2 is a distribution diagram of transmission delay time in a voice packet switching system, Figure 3 is an approximate distribution diagram of transmission delay time used in the conventional system, and Figure 4 is an embodiment of the present invention. 5 is an explanatory diagram showing the storage contents of the delay time storage section 18 of FIG. 4, and FIG. 6 is a time chart of the audio signal. 10...Delay time fluctuation absorbing device, 11...Relay packet switching device number input terminal, 12...Relay packet switching device number receiving circuit, 13...Processing device, 14...Audio packet input terminal, 15...Buffer memory for absorbing delay time variation, 16 ...Voice packet reception time measuring circuit, 17, 20...Timer circuit, 18...Delay time storage section, 19...Control circuit, 21...Voice packet output terminal.

Claims (1)

【特許請求の範囲】 1 1つあるいは複数のパケツト交換装置を中継
し、呼ごとに同一の経路で音声パケツトを伝送す
る音声パケツト交換システムにおいて、受信側に
中継パケツト交換装置数を受信する手段と、音声
パケツトの受信時刻を測定する手段と、音声信号
の各トークスパート(有音部)ごとに複数の音声
パケツトの各々連続する2つの音声パケツトの受
信時刻の差の平均値を求める手段と、前記中継パ
ケツト交換装置数と前記平均値に対応して音声パ
ケツトの伝送遅延時間の変動の分布に従つて求め
られた遅延時間を記憶する手段と、受信する音声
パケツトを順次蓄積するバツフアメモリと、該バ
ツフアメモリに入力される音声パケツトを指定さ
れた時間蓄積後一定間隔で取り出す制御手段とを
有し、前記受信した中継パケツト交換装置数と前
記平均値に対応する遅延時間を前記記憶手段から
読み出し、該読み出した遅延時間だけ音声信号の
各トークスパート(有音部)における最初の音声
パケツトを前記バツフアメモリに蓄積後、前記バ
ツフアメモリに蓄積された音声パケツトを順次と
り出すことにより伝送遅延時間の変動を吸収する
ことを特徴とする音声パケツト伝送遅延制御方
式。 2 1つあるいは複数のパケツト交換装置を中継
し、呼ごとに同一の経路で音声パケツトを伝送す
る音声パケツト交換システムにおいて、受信側に
中継パケツト交換装置数を受信する手段と、音声
パケツトの受信時刻を測定する手段と、音声信号
の各トークスパート(有音部)ごとに複数の音声
パケツトの各々連続する2つの音声パケツトの受
信時刻の差の平均値を求める手段と、前記中継パ
ケツト交換装置数と前記平均値に対応して音声パ
ケツトの伝送遅延時間の変動の分布に従つて求め
られた遅延時間に相当する音声パケツトの受信個
数を記憶する手段と、受信する音声パケツトを順
次蓄積するバツフアメモリと、該バツフアメモリ
に入力される音声パケツトを指定された時間蓄積
後一定間隔で取り出す制御手段とを有し、前記受
信した中継パケツト交換装置数と前記平均値に対
応する音声パケツト受信個数を前記記憶手段から
読み出し、音声信号の各トークスパート(有音
部)における最初の音声パケツトを該最初の音声
パケツト受信から続いて前記読み出した受信個数
だけ音声パケツトを受信する迄前記バツフアメモ
リに蓄積後、前記バツフアメモリに蓄積された音
声パケツトを順次とり出すことにより伝送遅延時
間の変動を吸収することを特徴とする音声パケツ
ト伝送遅延制御方式。
[Scope of Claims] 1. In a voice packet switching system that relays voice packets through one or more packet switching devices and transmits voice packets through the same route for each call, there is provided means for receiving the number of relay packet switching devices on a receiving side. , means for measuring the reception time of a voice packet, and means for determining the average value of the difference in reception time of two consecutive voice packets of the plurality of voice packets for each talk spurt (sound part) of the voice signal; means for storing a delay time determined according to a distribution of variations in transmission delay time of voice packets corresponding to the number of relay packet switching devices and the average value; a buffer memory for sequentially accumulating received voice packets; control means for extracting voice packets input to the buffer memory at regular intervals after accumulation for a specified time, reading out the delay time corresponding to the number of received relay packet switching devices and the average value from the storage means; After storing the first audio packet in each talk spurt (sound part) of the audio signal in the buffer memory for the read delay time, the fluctuations in the transmission delay time are absorbed by sequentially extracting the audio packets stored in the buffer memory. A voice packet transmission delay control method characterized by: 2. In a voice packet switching system in which voice packets are relayed through one or more packet switching devices and voice packets are transmitted through the same route for each call, there is a means for receiving the number of relay packet switching devices on the receiving side and a method for indicating the reception time of voice packets. means for determining the average value of the difference in reception time of two consecutive audio packets of a plurality of audio packets for each talk spurt (sound part) of the audio signal; and means for determining the number of relay packet switching devices. means for storing the number of received audio packets corresponding to a delay time determined according to a distribution of variations in transmission delay time of audio packets corresponding to the average value; and a buffer memory for sequentially storing received audio packets. , a control means for extracting voice packets inputted to the buffer memory at regular intervals after accumulation for a designated time, and a control means for storing the number of received voice packets corresponding to the number of received relay packet switching devices and the average value. , and store the first audio packet in each talk spurt (sound part) of the audio signal in the buffer memory until the number of audio packets read out is received continuously from the reception of the first audio packet, and then store the first audio packet in the buffer memory. A voice packet transmission delay control method characterized by absorbing variations in transmission delay time by sequentially extracting accumulated voice packets.
JP57015472A 1982-02-04 1982-02-04 Delay control system for voice packet transmission Granted JPS58134557A (en)

Priority Applications (1)

Application Number Priority Date Filing Date Title
JP57015472A JPS58134557A (en) 1982-02-04 1982-02-04 Delay control system for voice packet transmission

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
JP57015472A JPS58134557A (en) 1982-02-04 1982-02-04 Delay control system for voice packet transmission

Publications (2)

Publication Number Publication Date
JPS58134557A JPS58134557A (en) 1983-08-10
JPS6365181B2 true JPS6365181B2 (en) 1988-12-14

Family

ID=11889737

Family Applications (1)

Application Number Title Priority Date Filing Date
JP57015472A Granted JPS58134557A (en) 1982-02-04 1982-02-04 Delay control system for voice packet transmission

Country Status (1)

Country Link
JP (1) JPS58134557A (en)

Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JPH0454069U (en) * 1990-09-04 1992-05-08

Families Citing this family (5)

* Cited by examiner, † Cited by third party
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EP0215526B1 (en) * 1985-09-19 1991-05-08 BELL TELEPHONE MANUFACTURING COMPANY Naamloze Vennootschap Data communication system
US4748620A (en) * 1986-02-28 1988-05-31 American Telephone And Telegraph Company, At&T Bell Laboratories Time stamp and packet virtual sequence numbering for reconstructing information signals from packets
JP2569493B2 (en) * 1986-08-08 1997-01-08 日本電信電話株式会社 Transfer fluctuation absorption processing method in packet communication
JPH04314236A (en) * 1991-04-12 1992-11-05 Ando Electric Co Ltd Delay time addition circuit for atm cell
JP4454338B2 (en) * 2004-02-17 2010-04-21 富士通株式会社 Packet shaping device and packet shaping method

Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JPH0454069U (en) * 1990-09-04 1992-05-08

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