Method and apparatus for transmission of voice data in a...

Multiplex communications – Pathfinding or routing – Switching a message which includes an address header

Reexamination Certificate

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Details

C370S395700

Reexamination Certificate

active

06639915

ABSTRACT:

TECHNICAL FIELD
The present invention relates generally to network structures, and more particularly to an apparatus for transmitting voice data from one network location to one or more other network locations.
BACKGROUND OF THE INVENTION
Network systems typically transmit data from one network location (a source location) to another network location (a destination location) by first placing the data into “packet” form. A packet can include information that allows a data packet to be directed from one network location to another by including certain information, such as a “destination address,” that identifies the destination location. Such packet identifying information is typically placed in an initial portion of the packet., often referred to as a “header.” The header will, include a number of fields, each of which includes a particular type of packet information. The data carried by the packet is often referred to as a “payload.” The process of generating packets from data is often referred to as “packetization.” In particularly large networks, packet data can be transferred from a source location to a destination location by way of one or more network nodes. A transmission from one node to the next can be referred to as a “hop.”
One particular type of data that is greatly desired but that can be problematic for many network transmission structures, is voice data. Voice data typically involves utilizing an analog-to-digital converter to convert an analog voice signal into digital form. The digital voice data can then be processed further. As just one example, digital voice data can be “compressed” into a more compact form. More compact voice data can be transmitted over a given bandwidth, but at the cost of some fidelity.
One particular way of transmitting voice data is “Voice over Internet Protocol” (often referred to as “VoIP”). Internet Protocol (IP) is one particular protocol for transmitting data over a network structure having devices that recognize the IP protocol. The wide reach of the Internet and wide use of IP in networks makes VoIP a compelling application. VoIP can enable a local network, such as that shared by a corporation, to allow voice communications between the various users in addition to conventional sharing of data and/or applications. Because most connections to the Internet are by way of a conventional telephone line other higher speed connections (IDSN, T-1, or xDSL, as just a few examples) the use of VoIP can provide long distance voice communications at considerable cost savings by avoiding the toll charges required by conventional long distance telephone calls. VoIP can prove extremely valuable to the telecommuter, as data and voice can both be transmitted to an office over the same wire, allowing efficient communication between the remote worker and other on-site workers.
Another advantage of VoIP is the ability to transmit the same data packets to multiple destinations (multicast). This enables teleconferencing to occur over very large distances without the unwanted expense of a conventional long distance telephone, service provider.
The transmission of voice data over the Internet is also referred to as “Internet Telephony.” It is expected that Internet Telephony will be a widely used application for the Internet.
The transmission of voice data over network structures can present different concerns from those present in the transmission of other data over network structures. One important concern is that of delay. For example, for many types of data, delay may be acceptable or not relevant. Whether or not text and/or images are not immediately viewable is of less concern than if the text and/or images are corrupted. Accordingly, most protocols are concerned with reassembling data in the proper order; and less concerned with the overall rate at which the data is transmitted. As just one example, a data file can be broken down into packets that are transmitted in a particular order. The packets may be received in a different order, however. Many protocols allow the packets to be subsequently reassembled in the correct order. This is in sharp contrast to voice data. Attempting to have a conversation over a data network can be extremely awkward when undue delay is introduced in the transmission of the voice data. The natural flow of conversation is often interrupted as a delay may be misinterpreted as the end of one person's conversation. Further, if data is lost, or multiple packets take too long to reassemble, speech can be broken up, with portions of the conversation being dropped.
Another concern in voice data over networks is the processing latency introduced by conventional approaches. In particular, the packetization of single channel data can result in inefficient use of network bandwidth, as an entire header must be created, even if the corresponding payload is relatively small.
One approach to voice over data networks is the personal computer (PC) Internet phone or Internet videophone. Such approaches typically involve utilizing the microphone and sound card of a PC in conjunction with associated software. The central processing unit (CPU) of the PC, in conjunction with an analog-to-digital converter (typically in the sound card), converts the voice signal into digital form, and then transmits the data in a conventional fashion as a series of packets.
Another more expensive approach, typically utilized for multiple voice channels, is that of the voice server. A voice data server system is set forth in
FIG. 1
, and designated by the general reference character
100
. A voice data server.system
100
is shown to include a number of different voice data sources. Set forth in
FIG. 1
are a number of devices, including conventional telephones
102
-
a
and
102
-
b
, as well as a fax machine
102
-
c
. The voice data server system
100
receives conventional voice data (and other data, such as fax data) from a number of different conventional sources and transmits the voice data over the Internet (or some other network structure) in packet form. Each particular conventional voice source (such as telephones
102
-
a
and
102
-
b
) can be considered a separate voice channel when a conversation is taking place.
As shown in
FIG. 1
, the devices (
102
-
a
to
102
-
c
) are coupled to a switching device
104
, such as a private branch exchange (“PBX”). The switching device
104
allows switching between devices (
102
-
a
to
102
-
c
), and in addition, allows the devices (
102
-
a
to
102
-
c
) to be switched to an internet server
106
. The connection to the internet server
106
can be a high-speed multiplexed line, such as a “T-1” carrier line. In such an arrangement, up to
24
voice channels can be carried by the T-1 line to the internet server
106
.
The internet server
106
includes a conventional multiplexer interface
108
that de-multiplexes interface
108
that de-multiplexes the conventional voice data. The conventional voice data is then processed by a central processing unit
110
, which packetizes data for each channel, and then transmits the packetized voice data to a network interface
112
. The packets of voice data can then be output to a network structure, such as the Internet, by way of the network interface
112
. The network interface
112
is coupled to a network media line
114
.
The voice data server system
100
can also receive voice data packets by way of the network media line
114
. Voice packets, each including voice data for a single channel, are stripped of their associated header (de-packetized), and the resulting voice data payload passed on to the multiplexer interface
108
. The multiplexer interface
108
then outputs the voice data on the appropriate multiplexed channel. The data is then provided to the desired phone (
102
-
a
and
102
-
b
) and/or other device (such as fax machine
102
-
c
) by the switching device
104
.
A drawback to many conventional voice-over-network approaches arises out of the intermittent or “asynchronous” nature of voice data from multiple sources (channels). In part

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