Multiplex communications – Communication techniques for information carried in plural... – Combining or distributing information via time channels
Reexamination Certificate
1998-09-09
2003-11-18
Nguyen, Steven (Department: 2665)
Multiplex communications
Communication techniques for information carried in plural...
Combining or distributing information via time channels
C370S352000
Reexamination Certificate
active
06650662
ABSTRACT:
TECHNICAL FIELD
The present invention relates to voice transmission in high speed packet switching networks and more particularly to a method and system for ensuring the integrity of a Dual-Tone MultiFrequency (DTMF) signal in a compressed voice connection.
BACKGROUND ART
The evolution of digital networks in the last past years caused a fundamental shift in the customer traffic profile. Now, high speed packet switching networks allow the customer to integrate data, voice and video information which is digitally encoded, chopped into small packets and transmitted through the network. The efficient transport of mixed traffic streams on very high speed lines imposes a set of requirements for new network architectures in term of performance and resource consumption. The requirements are summarized below:
One key requirement is that the nodes must provide total connectivity; that is, support for attachment of a user's devices, regardless of vendor or protocol, and support for an ability to communicate with any other device. Networks have to support any type of traffic including data, voice, video, fax, graphic or image. Nodes must be able to take advantage of all common carrier facilities and to be adaptable to a plurality of protocols. All needed conversions must be automatic and transparent to the end user.
Another key requirement of high speed packet switching networks is reduction of end-to-end delay in order to satisfy real time delivery constraints and to achieve the necessary high throughput for the transport of voice and video. Increases in link speeds have not been matched by proportionate increases in the processing speeds of communication nodes and a fundamental challenge for high speed networks is to minimize the packet processing time within each node. In order to minimize the processing time and to take full advantage of the high speed/low error rate technologies, most of the transport and control functions provided by the new high bandwidth network architectures are performed on an end-to-end basis.
Communication networks have at their disposal limited resources to ensure an efficient packets transmission. An important requirement is an efficient bandwidth management scheme to take full advantage of a high speed network. While transmission costs per byte continue to drop year after year, transmission costs are likely to continue to represent the major expense of operating future telecommunication networks as the demand for bandwidth increases driven by new applications and new technologies.
A major goal of almost every customer is to reduce transmission costs in its networks by minimizing required bandwidth. One solution is to use bandwidth management algorithms to adjust the bandwidth according to the quality of service requested. For voice transmission, bandwidth can be saved by using voice compression algorithms capable of reducing significantly the data rate in voice circuits without measurable loss of quality. There are various possible ways to reduce the data rate required in a voice circuit from the 64 kbps standard data rate. Many voice compression algorithms rely on the fact that a voice signal has considerable redundancy, so the characteristics of the next few samples can be predicted from the last few samples. One of the most common voice compression algorithms based on the prediction method is the GSM technique. The GSM voice compression algorithm has been defined as a standard for the European digital cellular telecommunications system by the European Telecommunications Standards Institute (recommendation I-ETS 300 036).
According to international standards, when voice is converted to digital form, the analog signal is sampled at the rate of 8000 times per second (one sample every 125 microseconds) and each sample is represented or coded in 8 bits. This gives a constant bit rate of 64 000 bits per second. The coding system is called “Pulse Code Modulation” (PCM). The basic concept of PCM is that each eight bit sample is simply a coded measure of the amplitude of signal at the moment of sampling. This process is improved upon by a system called “Companding” (Compression/Expansion) where the lower amplitude parts of the scale are coded with more precision that the peaks. In practice, PCM is always encoded generally this way but the details of the standard differ in specific countries. One system is called “Mu-law” and the other is called “A-law”. The “Companding” process, defined in CCITT recommendation G.711, performs a conversion between the 8-bit A-law or Mu-law companded format and the 13-bit uniform format used in the GSM algorithm. The GSM algorithm takes a block or window of 160 samples in this 13-bit uniform PCM format and encodes it in a compressed data stream of 260 bits. Therefore, the average bit rate of this compressed data stream is 13 kbps compared to the initial bit rate of 64 kbps. In the receive node, the voice decoder performs inverse operations.
This coding scheme is well suited for pure voice traffic in high speed digital networks and efficiently reduces the bandwidth actually occupied. However, for voice connections, some control signals such as Dual-Tone MultiFrequency (DTMF) signals, which have different characteristics than voice traffic, may be transmitted over the network.
Dual-Tone MultiFrequency DTMF signals are used either during the call establishment, from the customer telephone set, for pushbutton signaling, or once the call established, for signal recognition in particular applications.
In both cases, DTMF signals are transmitted through the packet switching network using the voice transmission path. The DTMF code provides 16 distinct signals. the signalling is based on the simultaneous transmission of two frequencies each one belonging to a group of 4 frequencies. The signal frequencies are geometrically spaced and are not harmonically related. The CCITT recommendation Q.23 defines the characteristics of the DTMF signalling in term of frequencies (to generate a tone), as well as in term of tolerance.
When the voice traffic in a digital network is compressed using the GSM techniques briefly described above, the DTMF signals may be corrupted by the voice compression algorithm. After compression and decompression, the DTMF signals may not be recognizable by a DTMF Detector at the destination node of the network. Such a situation is often unacceptable to an end user. This problem of DTMF corruption does not occur with low compression rate speech algorithms because such algorithms are able to transmit DTMF signals without any deterioration. For example, Adaptive Differential PCM (ADPCM) reduces the data rate required in a voice circuit from the 64 kbps standard rate to 32 kbps without measurable loss of quality. In concept, the ADPCM algorithm encodes each sample as the difference between it and the last sample, rather that as an absolute value. Voice is real time traffic. Voice packets must be delivered to the receiver at a steady, uniform state and not in burst. No transit delay is permitted and a short response time is required to satisfy the CCITT recommendations. That means it is not possible to wait for receiving a complete DTMF signal before resending it towards the destination node. Otherwise: the receiver could detect the same DTMF signal more than once; and the voice signal received on the other side by the destination node during the DTMF regeneration process, could be lost.
For these reasons, early detection of candidate DTMF signal is essential for triggering the DTMF process as soon as possible. Early detection consists of analyzing a window of 160 samples to find a candidate for a DTMF signal at the beginning of a DTMF cycle. An obvious alternative would be to stop the compression process as soon as the candidate DTMF signal is detected, and to transmit it at 64 kbps standard rate (clear channel). This raises a problem when the connection has reserved bandwidth for compressed voice but not enough reserved capacity for transmitting data in clear channel.
As illustrated in
FIGS. 7
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, Europea
Arnaud Charles
Cesaro Claude
Richter Gerard
International Business Machines - Corporation
Munoz-Bustamante Carlos
Nguyen Steven
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