Multiplex communications – Pathfinding or routing – Switching a message which includes an address header
Reexamination Certificate
1997-08-19
2001-02-27
Pham, Chi H. (Department: 2731)
Multiplex communications
Pathfinding or routing
Switching a message which includes an address header
C370S466000
Reexamination Certificate
active
06195358
ABSTRACT:
BACKGROUND OF THE INVENTION
The present invention relates to Internet Telephony Signal Conversion.
Although originally intended for the transmission of computer data, more recently the Internet has been exploited to provide real time telephony communications. The primary attraction of the Internet for telephony communications is the low charge compared with conventional telephony. Many Internet users have a dial-up connection to an access provider over a local telephone line, and therefore such users pay only local telephone charges when logged on. Some access providers charge a monthly description, whilst others charge on the basis of connection time (some may do both). However, there is generally no charge associated with actual data transfer over the network. As a result, the effective cost of an international call over the Internet may be no more than that of a local call of the same duration to the access provider. In addition, the fully digital nature of the Internet may potentially offer a richer functionality (eg in terms of conference calling) than conventional telephone networks. Internet phones are surveyed in the article “Dial 1-800-Internet” in Byte Magazine, February 1996, pages 83-88 and in the article “Nattering On”, in New Scientist, Mar. 2, 1996, pages 38-40.
The transmission of voice signals over a packet network is described for example in “Using Local Area Networks for Carrying Online Voice” by D. Cohen, pages 13-21, in “Voice Transmission over an Ethernet Backbone” by P. Ravasio, R. Marcogliese, and R. Novarese, pages 39-65, both in “Local Computer Networks” (edited by P. Ravasio, G. Hopkins, and N. Naffah; North Holland, 1982) and also in GB 2283252. The basic principles of such a scheme are that a first computer digitally samples a voice input signal at a regular rate (eg 8 kHz). A number of samples are then assembled into a data packet for transmission over the network to a second terminal, which then feeds the samples to a loudspeaker or equivalent device for playout, again at a constant 8 kHz rate. Voice transmission over the Internet is substantially similar to transmission over a LAN (which may indeed provide part of the Internet transmission path), but there tends to be less spare bandwidth available on the Internet. As a result, Internet phones normally compress the voice signal at the transmitting end, and then decompress it at the receiving end.
One of the major draw backs of the Internet telephony is that few of the Internet phones use the same standards. At present unless you have the same software as the person you are calling then it is impossible to connect with them. There are a variety of sound compression methods used in processing the voice signal and a large number use the Global Standard for mobile Communications (GSM) although some use proprietary compression algorithms. Even when vendors use the same compression algorithms there may be slight variations which can lead to compatibility problems. Another compatibility problem is that products tend to have different connection protocols and there seems little movement towards a common standard.
Another drawback of Internet telephony is that the quality of the voice signal is diminished by compressing and decompressing. In some circumstances a diminished quality signal can be so distorted as to make the message difficult to understand. Difficulty in understanding the message is further compounded if there is background noise, electrical interference, unclear speech or an unfamiliar accent or language.
A further drawback is that long delays can be introduced between messages from the participants which especially in the case of half-duplex transmission increases the difficulty in understanding the message. The factors which contribute to the delays include the time taken to compress and decompress the voice signal and transmission delays on the Internet when the network is busy and when the user has a slow modem connection.
SUMMARY OF THE INVENTION
According to one aspect of the invention there is provided a method of processing an internet telephony message comprising voice data in a first internet telephony format said method comprising the steps of:
receiving said message in a first internet telephony format from a first computer connected to the internet;
performing a first conversion of the message from the first internet telephony format to an intermediate format;
performing a second conversion of the message from the intermediate format to a second format;
sending the message in the second format to a second computer connected to the internet.
According to a second aspect of the present invention there is provided a system of processing an internet telephony message, said message comprising voice data in a first internet telephony format, said system comprising
means for receiving said message in a first internet telephony format from a first computer connected to the internet;
means for performing a first conversion of the message from the first internet telephony format to an intermediate format; and
means for performing a second conversion of the message from the intermediate format to a second internet telephone format;
means for sending the message to a second computer connected to the internet.
One advantage of the above is that the resources required to support a large number of protocols and compression schemes by using intermediate conversion is less than that when direct conversion is used since only one pair of algorithms is needed per protocol and per compression scheme. Furthermore the intermediate formatted message may be processed for tone detection, voice recognition and the recording of the message. The resources needed to perform such functions on a plurality of formats would be very large compared to those needed for one intermediate format.
In the preferred embodiment the intermediate format is a standard companding format used for voice signals in a voice processing system. Such a voice processing system requires little modification for converting Internet telephony signals and is ideally suited to such an application.
Preferably the intermediate message is an uncompressed digital message. Alternatively the intermediate message is a G.711 digital message.
The converting may be performed by at least one Digital Signal Processor (DSP) chip in a voice processing system. Normally a DSP chip has the capacity to handle multiple voice signals simultaneously. In the embodiment a DSP chip can process 6 voice data streams all in real time allocating fixed time slots to each data stream. One DSP chip converts the first formatted first message into an intermediate format in a first time slot and in an adjacent time slot converts the intermediate formatted first message to the second format. The ability to handle multiple voice channels in real time on a single DSP chip makes a voice processing system an ideal candidate for an Internet telephony conversion application.
A voice processing system comprises a DSP board which includes several DSP chips—the embodiment has a total of six DSP chips on one board. Another method of performing the voice conversion using the intermediate format is to perform the conversion on two DSP chips. A first chip converts the message from one format to the intermediate format, the message is then passed to a second DSP chip and the conversion performed from the intermediate format to the second format. A voice processing system further comprises several DSP boards. A further method of performing the voice conversion using the intermediate format is to perform the first part of the conversion on a DSP chip on one board, pass the intermediately formatted signal to another chip on another board for the second part of the conversion.
In the cases where the two part processing is not performed on a single chip the intermediate message is passed from one DSP chip to another DSP chip via a data bus. In such a case it is a particular advantage that the DSP chips communicate with each other over a dedicated bus such as a Time Division Multiplexed bus
Bowater Ronald John
Davies Simon Robert
Porter Lawrence Leon
Thomas Mark George Stewart Reid
Herndon Jerry W.
International Business Machines Corp.
Pham Chi H.
Yao Kwang B.
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