摘要:
The specification relates to a broadband multiple access protocol for bi-directional hybrid fiber/coax (HFC) networks. The protocol supports downstream broadcast transmission from headend to cable modem, and also provides for allocation of bandwidth for cable modems to transmit back to the headend. Although the present invention is described in relation to an HFC network, it is also equally applicable to a wireless communications environment. The protocol supports different access modes such as synchronous transfer mode, asynchronous transfer mode, and variable length data. The protocol adapts to changing demands for a mix of circuit and packet mode applications and allocates upstream and downstream bandwidth in response to the a variety of bursty and isochronous traffic sources. In order to satisfy the quality of service requirements of varied applications, while maintaining high bandwidth efficiency, the protocol utilizes a frame structure with frame partitioning into regions; one region dedicated to STM payload and asynchronous, second region dedicated to ATM and VL payloads, messaging and control.
摘要:
The specification relates to a method and apparatus supporting multiple access, bi-directional data and multimedia transfer described over a hybrid fiber/coax (HFC) network, though applicable to transmissions over other media as well. The protocol associated with the present invention supports downstream broadcast transmission from headend to cable modem, and also provides for allocation of bandwidth for cable modems to transmit back to the headend. The protocol supports different access modes such as STM, ATM, and VL; within each subframe of a subframe/frame/masterframe structure. The protocol is utilized over a system which provides for forward error correction (FEC) and scrambling/descrambling, while eliminating the deleterious effects on the error correcting capability of the FEC due to the bit error spreading associated with a scrambling/descrambling function. The protocol adapts to changing demands for a mix of circuit and packet mode applications and allocates upstream and downstream bandwidth in response to the a variety of bursty and isochronous traffic sources.
摘要:
The specification relates to a broadband multiple access protocol for bi-directional hybrid fiber/coax (HFC) networks. The protocol supports downstream broadcast transmission from headend to cable modem, and also provides for allocation of bandwidth for cable modems to transmit back to the headend. Although the present invention is described in relation to an HFC network, it is also equally applicable to a cellular wireless communications environment. The protocol supports different access modes such as STM, ATM, and VL; within each subframe of a subframe/frame/masterframe structure. The protocol adapts to changing demands for a mix of circuit and packet mode applications and allocates upstream and downstream bandwidth in response to the a variety of bursty and isochronous traffic sources. In order to satisfy the quality of service requirements of varied applications, while maintaining high bandwidth efficiency, the protocol utilizes a subframe structure with subframe partitioning into regions; one region dedicated to STM payload transmission, a second region dedicated to ATM cell transmission, and a third region dedicated to VL payload transmissions.
摘要:
The specification relates to a system supporting the transmission of multiple protocols over a single bytestream. The multiple protocol types supported include asynchronous transfer mode (ATM) protocol data units (PDUs), synchronous transfer mode (STM) PDUs, and variable length (VL) PDUs, as well as subtypes included within the aforementioned multiple protocol types. PDUs from higher layers are processed at three intermediate protocol layers where application layer PDUs are prepared, segmented, and repacked as asynchronous block multiplexing (ABM) PDUs. ABM PDUs include a type identification field. Cyclical redundancy checks and other error detection/correction techniques are optionally supported. ABM PDUs are multiplexed within a multiple protocol bytestream. Multiple protocol bytestream support is provided between a transmitter and receiver over a plurality of mediums, including but not limited to coaxial cable, wireless, optical fiber, hybrid fiber/coax, satellite, and twisted pair. Despite the various modes, protocols, PDU lengths, and different quality of service (QOS) requirements, support is provided over a common bytestream with a common physical layer datalink in either a point-to-point or broadcast environment.
摘要:
In a packet voice system, discarding of a packet is performed as a function of previously discarded packets. In one embodiment, a packet voice system includes an ATM Adaptation Layer Type 2 (AAL2) and Service Specific Convergence Sublayer (SSCS) System. In this system, a transmission buffer stores AAL2 voice packets for transmission, each AAL2 voice packet comprising a sequence number, the values of which range from 0 to n−1, and a source identifier, k. When traffic congestion is detected, the transmitter portion of the SSCS System selectively discards one packet from a source k at the output of the transmission buffer if no packet from source k was dropped in either the last n−1 packets or over a predefined prior interval of time. Another embodiment of the invention discards packets at the input of the transmission buffer.
摘要:
The specification relates to a device and method utilized for packaging voice data (and other delay critical ‘connection’ or ‘flow’ type application data) for point-to-point transport from one Packet Circuit Gateway (PCG) to a second PCG over Label Switching Routers (LSRs) within an Internet Protocol (IP) network; the beneficial aspects of the packaging format being: (i) a reduced overhead requirement when compared to conventional IP telephony due to inclusion of a switching label in lieu of an appended IP header, thereby increasing network bandwidth efficiency, and (ii) the increased transport speed associated with layer two label switching when compared to layer three forwarding.
摘要:
A quality of service guarantee for voice and other delay sensitive transmissions within an Internet Protocol (IP) network is provided by identifying the IP network path utilized for IP packet transmission between source and destination edge devices and virtually provisioning IP network path bandwidth for priority voice traffic. Priority for voice packets and admission control of new voice calls (and other delay sensitive traffic) based on the remaining available capacity over the IP network path guarantees that high priority voice (and other delay sensitive traffic) meet stringent delay requirements. A Virtual Provisioning Server is utilized to maintain bandwidth capacity data for each path segment within the IP network and to forward the bandwidth capacity data to a Signaling Gateway. The Signaling Gateway determines whether to accept or reject an additional delay sensitive traffic component based upon available bandwidth capacity for an IP network path. The Signaling Gateway then signals the originating source edge device as to its determination to accept or reject. Quality of Service guarantees concerning acceptable delay and jitter characteristics for real-time transmission over an IP network are therefore provided without the need to directly signal the individual IP routers over which an IP network path is established.
摘要:
STM traffic, e.g. voice and video telephony (VT), as well as packet mode (e.g. ATM) traffic, e.g. broadcast digital video, interactive television, and data, are transmitted via a multiple access broadband fiber/coaxial cable network. Customer premises equipment (CPE) at stations, and a bandwidth controller, which may be at a head end or central office, with which all stations communicate, work together to adapt to the changing demands of the traffic mix, and efficiently allocate bandwidth to a variety of bursty and isochronous traffic sources. The bandwidth allocation defines two types of time slots, STM and ATM, and divides each frame into two corresponding STM and ATM regions. The boundary between the regions can be changed dynamically. A contention access signaling channel is provided in the STM region, for call control and set-up requests. Within the STM region, the time slots can be of variable length and be allocated on a per call basis; the length of the time slots is proportional to the bandwidth requirement of STM calls. Within the ATM region, the time slots are of fixed length, each capable of accommodating one ATM cell. Further, the fixed length ATM time slots may be reserved for a particular user for the duration of a call, or may be shared through a contention process. At least one contention ATM time slot is always made available for signaling messages related to ATM call control and set-up requests. The downstream time frame is structured in a similar manner, but includes an additional MAP field to transmit to the stations ATM time slot allocation and status information for time slots in the upstream channel.
摘要:
A head-end dynamically allocates bandwidth of a communications channel as a function of the type of communications traffic. In particular, the head-end communicates to subscriber stations via a broadband cable network using an access protocol, which is modified to provide a variable number of mini-slots and a variable number of data slots in each frame. Each mini-slot is used to request assignment of a data slot(s) to subscriber stations for the communication of information and, also, as a vehicle to resolve contention between subscriber stations. The head-end dynamically adjusts the number of mini-slots over a period time as a function of the type of communications traffic, e.g., bursty and isochronous traffic sources. Any variation in the number of mini-slots concomitantly effects the number of data slots available to communicate information. For example, less mini-slots provides more data slots. As a result, the dynamic adjustment of the number of mini-slots allows the head-end to more efficiently allocate bandwidth on the communications channel.
摘要:
In a packet voice system, a dynamic build-out delay approach in a receiver during the duration of a call. In particular, the build-out delay is applied at least twice during the duration of the call. In one embodiment, a packet voice system includes an ATM Adaptation Layer Type 2 (AAL-2) and Service Specific Convergence Sublayer (SSCS) System. The receiver portion of the SSCS System recovers AAL-2 packets and plays back the compressed audio to a voice decoding element. In providing playback, the receiver applies the build-out delay at the start of each talk-spurt. The voice decoding element provides an uncompressed audio stream. In another embodiment, the receiver portion of the SSCS System applies the build-out delay at the start of the each talk-spurt as a function of the length of the previous silence interval.