WO1996031993A1 - Systeme de transmission vocale a haute capacite - Google Patents

Systeme de transmission vocale a haute capacite Download PDF

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Publication number
WO1996031993A1
WO1996031993A1 PCT/CA1996/000206 CA9600206W WO9631993A1 WO 1996031993 A1 WO1996031993 A1 WO 1996031993A1 CA 9600206 W CA9600206 W CA 9600206W WO 9631993 A1 WO9631993 A1 WO 9631993A1
Authority
WO
WIPO (PCT)
Prior art keywords
signals
link
identification pattern
voice
undecompressed
Prior art date
Application number
PCT/CA1996/000206
Other languages
English (en)
Inventor
Eric Verreault
Original Assignee
Newbridge Networks Corporation
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 by Newbridge Networks Corporation filed Critical Newbridge Networks Corporation
Priority to AU52637/96A priority Critical patent/AU5263796A/en
Publication of WO1996031993A1 publication Critical patent/WO1996031993A1/fr

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Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04Q—SELECTING
    • H04Q11/00—Selecting arrangements for multiplex systems
    • H04Q11/04—Selecting arrangements for multiplex systems for time-division multiplexing

Definitions

  • This invention relates to a method of transporting a call through a communications network.
  • HCV High Capacity Voice
  • a voice signal is coded with HCV, decoded to recover the voice signal, and then sent through another coder/decoder pair.
  • HCV head-to-headset
  • a regional division places a call to another regional division.
  • the call must be decompressed to pass through the PBX interface and then recompressed if it gets routed back into the corporate network.
  • An object of the invention is to alleviate the signal degradation that occurs in such situations.
  • a method of establishing a call through a communications network with voice compression where the call is routed through a digital exchange, characterized in that a digital identification pattern is transmitted over a digital link through the exchange between transmitting and receiving compression modules; the receiving module monitors incoming signals from the link to detect said digital identification pattern; and when said identification pattern is detected at the receiving module, undecompressed voice signals are exchanged directly over said digital link.
  • the system recognizes the existence of a tandeming situation and is able to avoid signal degradation due to multiple compression and decompression operations.
  • the identification signal is preferably a framing pattern in a bit of small weight of a PCM coded voice signal.
  • the bit rate of the signalling pattern is low enough to ensure that the user will not perceive it as noise.
  • This signal is a unique identification tag sent by an stHCV (Super tandem HCV) device to indicate to another stHCV device at the end of the link that it is talking to another stHCV.
  • stHCV Super tandem HCV
  • the stHCV receiver When the stHCV receiver detects the identification pattern, it indicates to its transmitter section to start sending the compressed voice digitally over the voice link
  • one stHCV module is compressing the signal at a certain rate (e.g. 8K) and it is talking to another device compressing to another rate (e.g. 16K) .
  • the digital information should not be propagated in this case because the compressed information is not the same between the devices; no data scaling can be performed to down rate or up rate the signal.
  • This issue is resolved by transmitting different identification patterns depending on the compression rate of the HCV device. If an 8K HCV is talking to a 16K HCV, they will not recognize each other because the signalling pattern sent by one is different from the pattern expected by the other.
  • the HCV receiver has to become synchronized with the incoming 8K or 16K data stream on the voice side. Only when synchronization is achieved on the voice side is the data passed transparently. Synchronization has to be achieved since otherwise there could be a loss of synchronization from the far-end HCV because the incoming data on the voice channel is not necessarily synchronized with the out-going data.
  • an accumulation buffer can be included to re-synchronize the received digital signal with the transmitted one.
  • the transmission of the identification pattern should be synchronized with the incoming digital data.
  • the incoming data alignment will already be known and it will be properly aligned before being re-transmitted once the slow rate identification pattern is detected.
  • the voice side data can be further analyzed to detect the presence of valid compressed voice data. If the compressed voice contains a synchronization pattern, then this pattern can be searched for, and only when detected can the transfer between the voice side to the digital side be performed. This eliminates any dead time after detection of the low-rate framing pattern.
  • the invention also a communications network comprising a pair of voice compression modules communicating over a digital link, characterized in that each module includes means for inserting a digital identification pattern into outgoing voice signals transmitted over said link, means for detecting the identification pattern in incoming voice signals from said link, and means for exchanging undecompressed signals directly over said link on detection of said identification signal.
  • FIG. 1 shows a communications system employing super tandem HCV
  • Figure 2 is a simplified diagram of two stHCV units communicating over a digital link
  • FIG. 3 is a detailed block diagram of an stHCV system.
  • a telephone set 1 is connected to a Newbridge Networks DTE 3606/12/00/45 unit 2 including an HCV module 3, which compresses outgoing voice signals and passes them through network 4 to Newbridge MainstreetTM 3600 multiplexer 5 containing a pair of back- to-back super tandem HCV units 6, 7.
  • HCV Units 6, 7 are linked through PBX 8 connected to telephone extensions 9.
  • StHCV unit 7 passes compressed signals through network 10 to HCV unit 12 in DTE unit 11.
  • the super tandem HCV unit 6 when put in the super tandem mode, sends a digital identification signal to the super tandem HCV unit 7 through the PBX, and if unit 7 detects this identification signal, thus indicating the presence of a digital link between units 6, 7, the signals are forwarded compressed (i.e. undecompressed) through the PBX 8 so as to eliminate the intervening compression and decompression operation entirely.
  • stHCV transmitter 6 (the voice decompressor in this case) continuously sends an identification pattern in a bit of small weight of the PCM voice signal.
  • the bit rate in this identification channel is low enough to ensure that the user will not perceive it as noise.
  • This identification information is transmitted whenever the super-tandem feature is enabled.
  • This signal is a unique identification tag sent by an stHCV to indicate to another stHCV device at the end of the voice link that it is talking to another stHCV.
  • this identification pattern is detected, it means for the stHCV that it can talk digitally with the other end because the communication path is digital and the compression rate of the two devices is the same.
  • the stHCV receiver When the stHCV receiver detects the identification pattern, it indicates to its transmitter section to start sending the compressed voice digitally over the voice link
  • the HCV receiver has to become synchronized with the incoming 8K or 16K data stream on the voice side. Only when synchronization is achieved on the voice side is the data passed transparently. Synchronization has to be achieved otherwise there could be a lost of synchronization from the far-end HCV 12 because the incoming data on the voice channel is not necessarily synchronized with the out-going data.
  • an accumulation buffer is included to re-synchronize the received digital signal with the transmitted one.
  • the only impact of this will arise when switching between the compressor and the data coming from the voice link.
  • a discontinuity in the frame packets will occur.
  • the audible result should be a short glitch preceding an improvement of the voice quality.
  • All digital voice links are standardized to rate of 64kbps with data words of 8 bits.
  • the bit numbering of the data byte is defined as depicted in the table below:
  • El links are guaranteed to be clear channels, which means that the bits propagated, not considering any bit errors, are unaffected by any outside interference.
  • Tl links can use robbed bit signaling that will corrupt bO at every 6 data byte. This bit position cannot be used to propagate the identification channel of stHCV because of the possible deterioration of the communication channel.
  • the next smallest weight bit is bl .
  • This bit can also be deteriorated on a Tl link by the forced 1 insertion that occurs when too many consecutive 0 bits have been sent on the Tl link. This condition should not occur too often because voice signal is supposedly sent and therefore the identification channel will use bl .
  • HCV can be modified to make sure it never output all O's on the PCM side and Jam Bit 7 will never be seen.
  • the compressed digital information is sent in the same channel (position bl in the incoming byte) as the identification pattern with this identification pattern still present.
  • a 16K bandwidth stHCV transmits the compressed signal over the voice link, it uses b2 and bl and also continues to generate the identification pattern in bl.
  • the digital data is transmitted over the voice link, the higher order bits of the PCM representation of the decompressed analog voice are also continuously transmitted.
  • the voice signal is always transmitted, and the compression operation of stHCV is never disabled at any time. This is required because there are no handshake exchanges between the two stHCV devices 6, 7; therefore the transmitter cannot know if the far-end receiver has detected the identification signal and that is extracting the digital information from the voice link. This also provides a safeguard in case the tandeming operation fails.
  • the identification pattern has to co-exist with the 8 or 16K HCV digital data. Because the HCV compressed signal is quite sensitive to bit errors, the identification information should not corrupt the HCV digital data.
  • HCV processes the voice by chunks of 20 ms ) there are bits dedicated to hold a sync, pattern.
  • the identification pattern can overwrite one bit of the sync pattern, on the condition that when the data is extracted from the voice link that the sync pattern bit that was destroyed is replaced by its original value. That dictates that the identification information is sent at a rate of n*20 ms, in other word, every 160 byte.
  • the longest identification pattern possible should be used.
  • a pseudo-random bit generator can be used instead of pre-defining the identification pattern over 20 or 40 bits.
  • the resulting identification pattern is of length of 2 N -1 where N is any integer.
  • the generation of such patterns is reduced to an N bit past data shift and a few exclusive ORing operations.
  • the synchronization of the incoming identification pattern can be performed by the same sequence of operation where the past generated data is replaced by the past received data and a comparison is performed to know the validity of the received pattern.
  • pseudo ⁇ random bit generation as the identification pattern, it guarantees the uniqueness of the two different patterns when the two N are different.
  • bit block processing can be performed for the pseudo-random bit comparison (instead of performing the operation on a bit per bit basis, the operation of shifting and exclusive ORing can be performed on words built with bit extracted from successive time slices.
  • PBX 8 is connected to super tandem HCV units 6, 7, each connected over incoming and outgoing lines 20, 21 to respective digital networks.
  • Each unit 6, 7 has an input side 15, and an output side 16.
  • Compressed digital voice signals are sent and received over the lines 20, 21 to and from the digital networks.
  • Incoming digital voice on line 20 of unit 6 is passed on a per channel basis to decompressor 22, which continuously decompresses the signal and outputs PCM voice over line 23 to a first input of a digital merging unit 24.
  • the digitized voice is also passed directly to a second input of the unit 24 over line 25.
  • Unit 24 merges the digital voice signals and decompressed PCM voice signals when an st framing pattern is detected in framing detector
  • the incoming digital voice signals are continually decompressed in voice decompressor 22.
  • super tandem frame generator 26 inserts an st framing sequence, which is a framing pattern at a very low bit rate, in second merge unit 27 connected to the output of merge unit 24 that merges the compressed and digital signals when indicated by the framing detector 33.
  • the combined signals (digital signals, decompressed signals, and identification pattern) are passed to PBX 8, and from there out through unit 7.
  • the signals on line 30 coming from the PBX 8 are continually recompressed in voice compressor 31 and passed to switch 32.
  • Framing detector 33 continually monitors line 30 for the super tandem identification pattern. When this is detected, the framing detector 33 instructs data extractor
  • the framing detector 34 to extract the compressed digital signals from line 30 and pass them through delay line 35, included for network alignment purposes, to switch unit 32. Then, when the framing detector recognizes the signature of the digital voice on the signal line 30 coming from the PBX, it controls the switch 32 to replace the re-compressed voice signals with the undecompressed (i.e. compressed) signals coming directly from the data extractor 30.
  • Framing unit 33 also instructs the switch 34 to merge the incoming undecompressed digital voice signals on line 20 into the PCM decompressed voice signals in the input side 15 so that they can be passed to the PBX 8.
  • the input side of the st HCV unit is instructed to pass the undecompressed digital signals going in the opposite direction through the PBX, along with the PCM signals, so that the output side of the other st HCV unit can extract the undecompressed output signals and forward them on to the network.
  • Unit 24 has been described as a merge unit for merging the decompressed and uncompressed signals for transmission through the PBX when the identification is detected.
  • the advantage of sending both decompressed and compressed signals is that the system can be very quickly switched back to the normal mode with minimal loss of data if synchronization is lost.
  • unit 24 could be replaced by a simple switch so that only undecompressed signals are sent in the super tandem mode and only decompressed signals are sent in the normal mode. This embodiment risks a loss of data in switching between modes.
  • the invention thus results in improved voice transmission in tandem situations due to the elimination of unnecessary multiple compressions and decompressions. No handshaking takes place and the compressed signals are transmitted back directly on receipt of the identification pattern. This arrangement prevents signal degradation that occurs when a signal must undergo multiple compressions and decompressions.

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  • Engineering & Computer Science (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Time-Division Multiplex Systems (AREA)

Abstract

L'invention concerne un procédé pour établir un appel par un réseau de communication avec compression de la voix, où l'appel est acheminé par une centrale. Ce procédé consiste è envoyer un motif d'identification numérique sur une liaison en passant par le central entre des modules de compression émetteurs et récepteurs. Quand le motif d'identification est détecté au module récepteur, les signaux de voix comprimés sont échangés sur la liaison numérique sans l'intervention d'une décompression et d'une recompression. Aucun message d'établissement de la liaison n'a lieu et les signaux sont retransmis directement à la réception du motif d'identification. Cet agencement empêche une dégradation des signaux qui se produit quand un signal doit subir des compressions et des décompressions multiples.
PCT/CA1996/000206 1995-04-05 1996-04-04 Systeme de transmission vocale a haute capacite WO1996031993A1 (fr)

Priority Applications (1)

Application Number Priority Date Filing Date Title
AU52637/96A AU5263796A (en) 1995-04-05 1996-04-04 High capacity voice transmission system

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
GBGB9507058.7A GB9507058D0 (en) 1995-04-05 1995-04-05 Super tandem high capacity voice transmission system
GB9507058.7 1995-04-05

Publications (1)

Publication Number Publication Date
WO1996031993A1 true WO1996031993A1 (fr) 1996-10-10

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Country Status (3)

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AU (1) AU5263796A (fr)
GB (1) GB9507058D0 (fr)
WO (1) WO1996031993A1 (fr)

Cited By (6)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0705052A3 (fr) * 1994-09-22 1997-01-15 Eci Telecom Ltd Système de communication pour signaux de parole numériques
WO1998028936A1 (fr) * 1996-12-05 1998-07-02 Nokia Telecommunications Oy Detection du retrobouclage de voie de conversation
EP0895381A1 (fr) * 1997-07-30 1999-02-03 AT&T Corp. Réseau de télécommunication avec suppression de multiples compressions/décompressions de paquets de voix
WO2001056207A1 (fr) * 1999-10-18 2001-08-02 Nuera Communications, Inc. Procede de detection tandem et tunnelisation tandem
US6512790B1 (en) 1998-12-23 2003-01-28 Eci Telecom Ltd. Method, system and apparatus for transmitting coded telecommunication signals
US9172726B2 (en) 2008-09-30 2015-10-27 Alcatel Lucent Impairment reduction for tandem VoIP calls

Citations (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JPS63172555A (ja) * 1987-01-12 1988-07-16 Nec Corp 音声符号化構内交換方式
EP0332345A2 (fr) * 1988-03-11 1989-09-13 AT&T Corp. Codeurs-décodeurs avec suppression de codage multiples à travers une connexion
JPH02298132A (ja) * 1989-05-11 1990-12-10 Mitsubishi Electric Corp 時分割多重化装置
JPH03181263A (ja) * 1989-12-08 1991-08-07 Matsushita Electric Ind Co Ltd 中継交換装置
EP0495128A1 (fr) * 1990-08-06 1992-07-22 Fujitsu Limited Materiel de telecommunications avec fonction de commutation a repetition

Patent Citations (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JPS63172555A (ja) * 1987-01-12 1988-07-16 Nec Corp 音声符号化構内交換方式
EP0332345A2 (fr) * 1988-03-11 1989-09-13 AT&T Corp. Codeurs-décodeurs avec suppression de codage multiples à travers une connexion
JPH02298132A (ja) * 1989-05-11 1990-12-10 Mitsubishi Electric Corp 時分割多重化装置
JPH03181263A (ja) * 1989-12-08 1991-08-07 Matsushita Electric Ind Co Ltd 中継交換装置
EP0495128A1 (fr) * 1990-08-06 1992-07-22 Fujitsu Limited Materiel de telecommunications avec fonction de commutation a repetition

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Title
PATENT ABSTRACTS OF JAPAN vol. 012, no. 440 (E - 684) 18 November 1988 (1988-11-18) *
PATENT ABSTRACTS OF JAPAN vol. 015, no. 082 (E - 1038) 26 February 1991 (1991-02-26) *
PATENT ABSTRACTS OF JAPAN vol. 015, no. 434 (E - 1129) 6 November 1991 (1991-11-06) *
R.V. COX ET AL: "Application and implementation of an embedded subband coder", IEEE INTERNATIONAL CONFERENCE ON COMMUNICATIONS - PAPER 3.5, vol. 1, 12 June 1988 (1988-06-12) - 15 June 1988 (1988-06-15), PHILADELPHIA (US), pages 90 - 95, XP002007369 *

Cited By (9)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0705052A3 (fr) * 1994-09-22 1997-01-15 Eci Telecom Ltd Système de communication pour signaux de parole numériques
WO1998028936A1 (fr) * 1996-12-05 1998-07-02 Nokia Telecommunications Oy Detection du retrobouclage de voie de conversation
US6230120B1 (en) 1996-12-05 2001-05-08 Nokia Communications Oy Detection of speech channel back-looping
EP0895381A1 (fr) * 1997-07-30 1999-02-03 AT&T Corp. Réseau de télécommunication avec suppression de multiples compressions/décompressions de paquets de voix
US6021136A (en) * 1997-07-30 2000-02-01 At&T Corp. Telecommunication network that reduces tandeming of compressed voice packets
US6512790B1 (en) 1998-12-23 2003-01-28 Eci Telecom Ltd. Method, system and apparatus for transmitting coded telecommunication signals
WO2001056207A1 (fr) * 1999-10-18 2001-08-02 Nuera Communications, Inc. Procede de detection tandem et tunnelisation tandem
US6498796B2 (en) * 1999-10-18 2002-12-24 Nuera Communications, Inc. Method for tandem detection and tandem tunneling
US9172726B2 (en) 2008-09-30 2015-10-27 Alcatel Lucent Impairment reduction for tandem VoIP calls

Also Published As

Publication number Publication date
GB9507058D0 (en) 1995-05-31
AU5263796A (en) 1996-10-23

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