Telecommunications – Transmitter and receiver at same station – Radiotelephone equipment detail
Reexamination Certificate
1998-07-17
2001-11-27
{haeck over (S)}mits, T{overscore (a)}livaldis Ivars (Department: 2641)
Telecommunications
Transmitter and receiver at same station
Radiotelephone equipment detail
C455S436000
Reexamination Certificate
active
06324409
ABSTRACT:
FIELD OF THE INVENTION
The present invention relates to telecommunication call transmissions. In particular, the present invention relates to a system and method for selecting voice coding at various junctions in a telecommunication call.
BACKGROUND OF THE INVENTION
With the proliferation of voice over data network technology has come the deployment of numerous voice coding and compression algorithms. Examples of these voice coding and compression algorithms include G.723, Global System for Mobil Communication (GSM), Pulse Code Modulation (PCM), G.711, and Adaptive Differential Pulse Code Modulation (ADPCM). G.723, PCM, G.711, and ADPCM are protocols which are defined by the International Telecommunications Union (ITU), and are well known to those reasonably skilled in the art. GSM is a protocol defined by European Telecommunication Standards Institute and is also well known to those reasonably skilled in the art. These protocols seek to compress and/or code a voice signal into an optimal number of bits for transmission over a data network, attempting to balance quality of service with affordability.
Protocols currently exist which allow two users on the same local access network, for example, to set up an optimal connection. A telephony feature server (TFS) may determine the coding abilities of each of the clients and find the best choice for the connection. However, when users are on a wide area network (WAN), separate local area networks (LANs), or there are intermediate public network connections, the ability to perform optimal connection negotiation on a system wide basis is typically lost. When users are part of such systems, it can result in multiple coding and compression algorithms used throughout a connection, resulting in gross distortion and loss of voice quality. The resulting signal distortion and loss of quality typically make the circuit unsuitable for fax or computer modem connection. Further, a call is often connected to a voice mail system, using an entirely different coding algorithm from a live call, which can further add to the resulting distortion.
For example, assume a telephony over LAN (TOL) system in which a caller making a call has several compression options. In this example, assume that a call from the caller is set up using a compression algorithm defined by G.723. Assume also that the caller is connected to an Ethernet LAN, which has a TOL gateway X. The call may be received by gateway X and converted to pulse code modulation (PCM) coding. From gateway X to a second gateway, gateway Y, the call is transmitted in PCM coding. Assume also that the call receiver is connected to gateway Y and that the receiver only has GSM capabilities currently available. Accordingly, the call is then converted from PCM coding to GSM coding.
Typically, there may be multiple compression coding options that may be selected for use between the caller and gateway X. However, the coding selection is typically only an educated guess since there is typically no prior knowledge of the capabilities of the receiver. In this example, the voice compression of G.723 introduces a first distortion. When the coded G.723 voice arrives at gateway X, it is then converted to PCM voice. At this point, a second distortion is introduced due to the G.723 decompression and PCM coding. When the PCM voice arrives at gateway Y, gateway Y negotiates with the call receiver and determines that GSM is the only option. Accordingly, a third distortion is introduced when the connection is made to the call receiver using GSM. It is not necessarily the severity of the individual compressions, but the transcoding back and forth and back again to different compression methods that typically causes unnecessary signal distortion.
The resulting signal in this example is substantially more distorted than a signal produced by a single family of coding algorithms. The various conversions produced a suboptimal series of events on a system wide basis, even though each point-to-point decision may have been optimal. It would be desirable to produce a series of connections which optimizes overall quality of the transmission from the sender to the receiver. The present invention addresses such a need.
SUMMARY OF THE INVENTION
An embodiment of the present invention provides a system and method for selecting voice compression and coding based on capabilities of all intermediary networks and links in addition to the capabilities of the end points. The present invention optimizes voice quality by determining a minimum number of new conversions, herein referred to as transcodings, necessary in voice connections, on a system wide basis. According to an embodiment of the present invention, a sender sends a signaling message to a receiver. The signaling message is sent to the receiver prior to sending an actual message. Each entity or device capable of converting voice coding between the sender and the receiver identifies their capabilities to the signaling message. An entity or device, including end devices as well as intermediary devices (such as gateways), capable of converting voice coding between a sender and a receiver is herein referred to as a “station”. The receiver also identifies its capabilities to the signaling message. The signaling message may then be sent back to the sender for the sender to determine an optimized series of voice coding to be used for the call. Alternatively, the call receiver, or an intermediate station located along the call route, may determine the optimized series of voice coding to be used for the call and communicate to the caller how the call should be coded.
According to an embodiment of the present invention, the sender may maintain a prioritized list of preferred coding methods. Each coding method included in the prioritized list of preferred coding methods are compared with the capabilities of all the stations to determine if all the stations have the capability to perform each of the preferred coding. It is also determined whether the transmission from the sender to the receiver can be sent with only uncompressed codes and a single preferred coding method. These determinations are made for each preferred coding method of the sender. The results are stored and analyzed. If there is a result which allows transmission from the sender to the receiver with no transcoding, then that result is selected. Otherwise, the result with the minimum number of transcodings is selected. If there is a tie in the results, then the result with the most number of hops with compressed coding is selected. A hop is a telecommunication signal coded section between two stations. If there is a tie with the result with the most number of hops with compressed coding, then the result which is higher on the sender's preference list is selected. Each network or link is then instructed to use a coding method which has been predetermined by the selected result. Note that although a specific order is described herein for exemplary purposes, the various resolutions of the ties may be performed in any order.
A method according to an embodiment of the present invention for optimizing telecommunication signal quality is presented. The method comprising steps of sending a signaling message; collecting at least one capability of at least one station that interacts with the signaling message; and determining a coding scheme based on the at least one collected capability.
In another aspect of the present invention, a system according to an embodiment of the present invention for optimizing telecommunication signal quality is presented. The system comprising a telecommunication device configured to send a signaling message to a receiving location. The signaling message collects at least one capability of at least one station that interacts with the signaling message. The system also includes a processor coupled with the telecommunication device configured to determine a coding scheme based on the at least one collected capability.
REFERENCES:
patent: 5408419 (1995-04-01), Wong
patent: 5546395 (1996-08-01), Sharma e
Beyda William J.
Shaffer Shmuel
Siemens Information and Communication Systems, Inc.
{haeck over (S)}mits T{overscore (a)}livaldis Ivars
LandOfFree
System and method for optimizing telecommunication signal... does not yet have a rating. At this time, there are no reviews or comments for this patent.
If you have personal experience with System and method for optimizing telecommunication signal..., we encourage you to share that experience with our LandOfFree.com community. Your opinion is very important and System and method for optimizing telecommunication signal... will most certainly appreciate the feedback.
Profile ID: LFUS-PAI-O-2578594