摘要:
A protocol for handling multiple access on broadband communication networks, e.g., fiber/coax networks and wireless networks, supports both continuous bit rate (CBR) and variable bit rate (VBR) traffic representing voice, video telephony, interactive television, and data. The invention is carried out both in customer premise equipment (CPE) at stations, and in a common controller with which all stations communicate. A medium access control (MAC) processor provided in each of the stations and in the common controller divides the time domain for a given RF channel into a series of successive frames, each having a plurality of time slots. Because of the architecture of the communication network, individual stations do not communicate directly with each other, but can receive broadcast messages indicating the status of each time slot, which messages are generated in the common controller and transmitted in a downstream channel. When a station desires to transmit information in the upstream direction, it inserts the information into an available time slot, with availability being determined in accordance with time slot status. Depending upon the type of traffic being originated, a station can indicate to the common controller a need for continued use of the “same” time slot in successive frames. This permits a station, such as a station requiring a CBR connection, to avoid having to contend repeatedly for continued access to the transmission network. In the case of a wireless communication network, the invention is carried out both in mobile stations, and in a base station which acts as a common controller and with which all mobile stations communicate.
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 an automatically switched optical network, the wavelengths are assigned to optical path based on their intrinsic physical performance and on the current network operating parameters. The wavelength performance information is organized in binning tables, based primarily on the wavelength reach capabilities. A network topology database provides the distance between the nodes of the network, which is used to determine the length of the optical path. Other network operating parameters needed for wavelength selection are also available in this database. Once a bin corresponding to the path length is identified in the binning table, the wavelength for that path is selected based on length only, or based on the length and one or more additional parameters. The optical path performance is estimated for the selected wavelength, and the search continues if the estimated path performance is not satisfactory. Several available wavelengths are searched and of those, the wavelength that is most used along the optical path in consideration or alternatively network-wide is selected and assigned. This method helps minimize wavelength fragmentation. The binning tables may have various granularities, and may be organized by reach, or by reach, wavelength spacing, the load on the respective optical path, the fiber type, etc.
摘要:
An AAL2/SSCS packet voice system multiplexes various forms of voice-band traffic including voice packets, fax packets, and data packets into a virtual circuit (VC). This AAL2/SSCS packet voice system executes a dynamic call admission algorithm that takes into account call type in deciding whether to admit a new call to the VC. In particular, this approach takes into account different bandwidth needs for different call types. The AAL2/SSCS packet voice system also performs bit or block dropping on voice packets to mitigate the effects of traffic congestion. The bit or block dropping is done based on the packet queue fill value exceeding at least one queue threshold. Further, the AAL2/SSCS packet voice system also dynamically varies a queue threshold as a function of capacity.
摘要:
An AAL2/SSCS packet voice system multiplexes various forms of voice-band traffic including voice packets, fax packets, and data packets into a virtual circuit (VC). This AAL2/SSCS packet voice system executes a dynamic call admission algorithm that takes into account call type in deciding whether to admit a new call to the VC. In particular, this approach takes into account different bandwidth needs for different call types. The AAL2/SSCS packet voice system also performs bit or block dropping on voice packets to mitigate the effects of traffic congestion. The bit or block dropping is done based on the packet queue fill value exceeding at least one queue threshold. Further, the AAL2/SSCS packet voice system also dynamically varies a queue threshold as a function of capacity.
摘要:
A packet voice system includes an ATM Adaptation Layer Type 2 (AAL-2) and Service Specific Convergence Sublayer (SSCS) System comprising a transmitter and a receiver. A portion of the 32 codepoints of the five bit RES (or UUI) field of the AAL-2 header are predefined to indicate an extended header. This a prior definition is stored in tables in both the transmitter and the receiver. The extended header itself comprises an additional octet appended to the AAL-2 header. A sequence number is normally carried in the RES field for the packet stream of voice calls that require sequence numbers. When the extended header is used, the sequence number is part of the extended header and thus messages are conveyed from the transmitter to the receiver without causing an interruption to sequence number assignment messages The transmitter dynamically uses the extended header and discontinues use of the extended header after a predefined duration of time or upon receiving an acknowledgment from the receiver.
摘要:
A communication node comprises a receiver for receiving packets and a routing device for routing the received packets to one of two output ports. For each output port, the communication node further comprises a classifier for classifying a received packet based on its traffic characteristic and storing that packet in a corresponding queue for that traffic characteristic, and a concatenated packets preparer for concatenating n received packets from each queue to form a concatenated packet to be transmitted by the associated output port, where the maximum of n is determined based on the traffic characteristic of the associated queue.
摘要:
A wide variety of call traffic is effectively integrated in a single broadband communications network. Calls having widely differing bandwidth requirements and sensitivities to delay are handled by the network with efficient, effective, and fair bandwidth allocation and transmission scheduling. This is accomplished by classifying each call in accordance with certain signal characteristics, such as required bandwidth and sensitivity to delay. Each call class is directed to a separate queuing circuit. Some calls in certain classes, such as those associated with high-bandwidth constant bit rate services, are each directed to their own individual queuing circuits. Other calls within a class are statistically multiplexed into a single queuing circuit for that class. A multiplexing circuit operates in accordance with a dynamic time slice scheme which involves defining a service cycle time period during which the multiplexer withdraws a predetermined number of information packets from each of a plurality of queuing circuits containing information packets and places those predetermined numbers of packets onto an output link. The multiplexer breaks up the cycle time period into a plurality of time slices, each of which determines how many information packets are transmitted from each queuing circuit during the cycle time period. Efficient resource usage and congestion avoidance are further achieved by using one of a number of alternative scheduling techniques for delay insensitive traffic.
摘要:
An integrated voice and data network includes a multiplexer arranged with a voice queue for storing voice packets and a data queue for storing data packets. Voice packets are transmitted for a predetermined interval T1. Data packets are transmitted for a predetermined interval T2. The predetermined intervals T1 and T2 may be of different durations. A separate signaling queue can be provided for storing received signaling messages. If a signaling message is moved into the separate signaling queue during either interval T1 and T2, that interval is suspended and the transmission of voice or data packets is interrupted until the entire signaling message is transmitted. Then the interrupted voice or data transmission is resumed for the remainder of the suspended interval T1 or T2. As an alternative, signaling messages can be transmitted during predetermined intervals between the intervals T1 and T2. Block dropping of low order voice bits also is described for reducing congestion at the node. The multiplexer guarantees a certain minimum bandwidth for voice traffic and data traffic. Concurrently, the multiplexer allows each type of traffic to utilize any spare bandwidth momentarily available because it is not being utilized by the other type of traffic. Signaling messages are serviced with very low delay and zero packet loss.