Multiplex communications – Pathfinding or routing – Switching a message which includes an address header
Reexamination Certificate
1999-02-18
2003-02-11
Jung, Min (Department: 2731)
Multiplex communications
Pathfinding or routing
Switching a message which includes an address header
C370S395610, C370S468000
Reexamination Certificate
active
06519259
ABSTRACT:
FIELD OF THE INVENTION
The present invention relates generally to packet-based communication systems, such as asynchronous transfer mode (ATM) networks, and more particularly to techniques for transmitting voice and other information over such systems.
BACKGROUND OF THE INVENTION
Packet-based communication systems are coming into increasingly widespread use in compressed voice transport applications, such as applications involving transport of multiple encoded voice channels. For example, conventional ATM communication systems provide high-speed, low-delay switching of voice, data, video and other types of user information traffic. In an ATM system, the user information traffic is separated into fixed-length 53-byte cells. Each ATM cell typically includes a 5-byte header and a 48-byte payload. The header incorporates a virtual channel identifier (VCI) and a virtual path identifier (VPI) associated with the corresponding cell. The VCI and VPI together specify a virtual connection (VC) which is established when a user requests a network connection in the ATM system. Additional details regarding these and other aspects of ATM systems can be found, for example, in the ATM Forum, “ATM User-Network Interface Specification,” Version 3.1, September, 1994, and in Martin de Prycker, “Asynchronous Transfer Mode: Solution for Broadband ISDN,” Ellis Horwood, New York, 1993, both of which are incorporated by reference herein.
The ATM standard includes a number of ATM Adaptation Layers (AALs), each specifying different types of connections in an ATM system. One such AAL is known as AAL Type
2
, or simply AAL
2
. AAL
2
includes a Common Part Sublayer (CPS) which communicates with the ATM layer, and a Service Specific Convergence Sublayer (SSCS) that operates between the CPS and a communication service layer. The purpose of the SSCS is generally to convey narrowband voice and telephony service information, such as voice, voiceband data, or circuit mode data, associated with various voice and telephony communication services. The SSCS specifies packet formats and procedures to encode different call-related data streams for bandwidth-efficient transmit by AAL
2
. The SSCS accommodates known techniques of low rate audio encoding, silence compression, and facsimile modulation/demodulation. The CPS provides multiplexing functions which allow many calls to be sent over a single ATM connection. The single ATM connection therefore acts as a trunk group for transmission between two points of access. Additional details regarding AAL
2
and the SSCS can be found in, for example, “AAL Type
2
Service Specific Convergence Sublayer for Trunking,” ITU—Telecommunication Standardization Sector, Draft Recommendation 1.366.2, Temporary Document 24-E (PLEN), June 1998, which is incorporated by reference herein.
Existing techniques for supporting compressed voice transport in ATM networks and other types of packet-based communication systems include G.763 DSI and G.764, both defined by the CCITT-ITU standards body. The G.763 DSI technique deals with an overload condition, i.e., a situation in which an additional voice call is to be added to an existing set of active calls utilizing all of the available bandwidth of a given transport stream, by taking one bit from each sample in the given stream, and allocating the resulting freed-up bandwidth to the additional call. The G.764 technique signals the coding rate in each frame of a given transport stream, and organizes the voice samples so that blocks of bits, starting with the least significant bit (LSB), can be identified and dropped if an intermediate network node experiences an overload condition. Unfortunately, these and other known techniques for dealing with compressed voice transport fail to provide an efficient allocation of available bandwidth among multiple channels, and can thus lead to an excessive degradation in reconstructed signal quality in one or more of the channels.
SUMMARY OF THE INVENTION
The invention provides methods and apparatus for communicating information in a packet-based communication system, such that one or more of th e problems associated with the above-noted conventional techniques are overcome. An illustrative embodiment of the invention includes a transmitter which configures multiple channels of encoded voice information for transmission through the system to a receiver. In the illustrative embodiment, the transmitter determines a total number of active channels to be coded at each of a number of different available code rates, within a specified bandwidth constraint. The transmitter assigns the code rates to the active channels in accordance with a code rate assignment technique, such that the bandwidth constraint is satisfied for a defined time period, e.g., a voice channel sample period. The code rate assignment technique may be, e.g., a random rate assignment technique, a round robin rate assignment technique, or an oldest talking channel to lowest rate assignment technique. In the latter case, the code rates are assigned to the channels such that a channel which has been active for a longer period of time than another channel is assigned a lower code rate than the other channel.
In accordance with another aspect of the invention, the transmitter generates packets for the active channels, each encoded at a corresponding assigned code rate, such that a time period between consecutive packets generated for a given channel is substantially constant for the duration of a voice call associated with that channel. In other words, all packets may be transmitted at a substantially constant rate in the time domain. The packet period for a given call, i.e., the time period between consecutive voice packets for that call, thus remains substantially constant for the duration of the call regardless of the speech coder bit rate. This constant emission rate reduces the complexity of the transmitter and the receiver. It also allows the receiver to more readily detect lost, late-arriving or misinserted packets, and to apply corrective treatments in a timely and consistent manner.
Although the above-described illustrative embodiment is particularly well suited for use in the transmission of voice information over an ATM network connection, the invention can also be implemented in other types of packet-based communication systems including, for example, Frame Relay systems and Internet Protocol (IP) systems. In addition, the invention can be applied to other types of audio and non-audio information. These and other features and advantages of the present invention will become more apparent from the accompanying drawings and the following detailed description.
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patent: 5898696 (1999-04-01), Proctor et al.
patent: 5911128 (1999-06-01), DeJaco
patent: 5970412 (1999-10-01), Maxemchuk
patent: 6266322 (2001-07-01), Berger et al.
“AAL Type 2 Service Specific Convergence Sublayer for Trunking,” ITU—Telecommunication Standardization Sector, Draft Recommendation I.366.2, Temporary Document 24-E (PLEN), pp. 1-91, Jun. 1998.
“Voice over Frame Relay Implementation Agreement,” Frame Relay Forum Technical Committee Contribution, FRFTC 96-025R2, pp. 1-73, Sep. 1996.
“Voice Packetization—Packetized Voice Protocols,” ITU-CCITT Recommendation G.764, pp. 1-21, Geneva, 1990.
“Digital Circuit Multiplication Equipment Using ADPCM (Recommendation G.726) and Digital Speech Interpolation,” ITU-CCITT Recommendation G.763, pp. 1-50, Jan. 1994.
T. Chu et al., “A SSCS Design for the Landline Trunking Application,” ATM Forum, pp. 1-10, Apr.-May 1997.
T. Chu, “Proposed Procedure to Change the Encoding Rule in Mid-call,” ATM Forum, pp. 1-6, Apr.-May 1997.
Baker Albert D.
Spagnolo Anthony J.
Szajdecki Richard
Wierzbicki Craig P.
Avaya Technology Corp.
Jung Min
Ryan & Mason & Lewis, LLP
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