Multiplex communications – Communication techniques for information carried in plural... – Combining or distributing information via frequency channels
Reexamination Certificate
1999-11-12
2004-03-16
Sam, Phirin (Department: 2661)
Multiplex communications
Communication techniques for information carried in plural...
Combining or distributing information via frequency channels
C370S494000, C370S495000, C370S529000
Reexamination Certificate
active
06707827
ABSTRACT:
FIELD OF THE INVENTION
The present invention relates to computer network systems in general, and to telephony-over-LAN systems in particular.
BACKGROUND OF THE INVENTION
As the speed of computer systems increases, the line between traditional telephony and computer systems is becoming less defined. For example, many companies are now using their local area computer networks (LANs) to transmit real time telephone signals between users on the network. These systems are generally referred to as telephony-over-LAN (ToL) or Voice over IP (VOIP) systems. The computer networks may be local area networks such as those contained within a single building, campus area networks, which may extend between several buildings, or may be a global or wide area network such as the Internet. One benefit of using a ToL system to transmit telephone signals is that the cost associated with installing and maintaining a separate telephone system, including separate wiring and dedicated telephone equipment, such as a private branch exchange (PBX), can be eliminated.
Despite the benefits of ToL systems, the sound quality of such systems is often not as good as a dedicated telephone system. One reason for the degradation of the sound quality is due to differences in the audio components of the call originating and destination computers. These audio components include microphones, sound cards, speakers and as well as the codecs that encode the voice signals into packets for transmission on the LAN. While there exists software for setting user-defined parameters of these components such as volume, treble or bass or a particular encoding method, these settings are static and are not varied with changes in network conditions or the capabilities of different users. Therefore, the audio settings used are not always optimal to produce the best possible sound quality for each call placed over the LAN. Given these shortcomings, there is a need for a system that can dynamically adjust the audio characteristics of telephonic signals transmitted over a LAN to ensure the best possible audio quality.
SUMMARY OF THE INVENTION
To improve the quality of audio signals transmitted over a packetized computer network, the present invention operates to exchange audio component data between a call source computer and a call destination computer on the network. Once the source and destination computers know the audio component data of the corresponding computer system with which they are to communicate, the audio components are dynamically programmed to optimize the transmitted sound quality for that particular call.
In one embodiment of the invention, a source computer and a destination computer determine characteristics of their audio components by looking in a local database. These characteristics can include the gain of signals transmitted from a sound card, a encoding method used by a codec and the frequency response of a microphone. The information received from the local database is transmitted to the corresponding computer such that the computers can collaborate in optimizing the component parameters to produce a quality audio signal.
In another embodiment of the invention, each of the source and destination computers determines the characteristics of its audio components by searching a database which is connected to the computer network. Once the audio characteristics are known, they are transmitted to the corresponding computer to optimize the audio components.
In yet another embodiment of the invention, the characteristics of the audio components in the source and destination computers can be determined empirically by transmitting a test signal from the source or destination and monitoring the amplitude and frequency characteristics of the audio signals produced in response to the test signal.
Once the characteristics of the audio components in the source and destination computers are known, the source computer can adjust the parameters of both the destination computer and its own audio components to produce an optimal audio signal. Alternatively, each of the source and destination computers can adjust their own audio components to produce an optimum audio signal.
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Adil Benyassine, Eyal Shlomot and Huan-Yu Su, Rockwell International, Dominique Massaloux, Claude Lamblin, and Jean-Pierre Petit, France Telecom-CNET, ITU-T Recommendation G.729 Annex B: A Silence Compression Scheme for Use with G.729 Optimized for V.70 Digital Simultaneous Voice and Data Applications, IEEE Communications Magazine, Sep. 1997, XP-000704425, 10 pgs.
Beyda William J.
Carter George E.
Shaffer Shmuel
Sam Phirin
Siemens Information & Communication Networks, Inc.
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