Abstract:
Inband Controlling of a packet-based Communications NetworkA method of handling control information (CMR latest, CMR latest new) sent inband in a packet-based communications network with traffic data, wherein the network has a buffer for temporarily storing received packets, is described. The method comprises receiving one or more packets and determining (312) first control information (CMR latest new) comprised therein, deciding (314) if the first control information (CMR latest new) is newer than second control information (CMR latest) received earlier, and, if the first control information (CMR latest new) is newer than the second control information (CMR latest), handling (318) the first control information (CMR latest new) independently from a buffering of the one or more received packets which comprise the first control information (CMR latest new). The invention further relates to a network component like a packet terminal or a converting gateway for performing the method and to communications systems comprising such a network component.
Abstract:
A system for optimally mapping circuits into packets based on round trip delay (RTD), and a system for measuring RTD for use in packet communications systems such as circuit emulation (CEM) systems is disclosed. The measured RTD value can be used in a system that adjusts packet size to reduce capture delay to partially offset an increase in RTD. As the use of smaller packets increases the overhead burden on the packet communication system, the packet size can be increased to reduce the overhead burden when the size of the current RTD becomes appropriately short. The disclosure also teaches the placement of data from two or more circuits destined for the same emulation endpoint into the same transmission packet in order to improve system performance. The abstract is a tool for finding relevant disclosures and not a limitation on the scope of the claims.
Abstract:
A data transmission method and system is disclosed in which a transmission rate formula is used to calculate the network bandwidth available per data stream to be transmitted. Multiple data streams are then transmitted at respective bit-rates the sum of which is not greater than the product of the number of data streams and the calculated bandwidth available per stream, but where individual data stream's rates can be increased at the expense of other of the transmitted streams. The invention is of particular use in streaming multimedia, and provides for controlled streaming rates of each type of media at the proper ratios. A data receiving method and system adapted to receive the data streams is also disclosed.
Abstract:
A packet data communication system (100) that includes multiple mobile stations (MS's) (102-104), each having a jitter buffer (324), and a wireless infrastructure (130) having corresponding base sites (116, 122, 126) serving the MS's, controls a size or depth of each jitter buffer. The jitter buffer size or depth is controlled based on a number of retransmissions of erroneously received data employed by the system, a radio frequency load of a base site, and a round trip time period in the system for acknowledgments and corresponding retransmissions. The jitter buffer size or depth may be further controlled by use of a supplemental channel in at least one of multiple forward links (140, 144, 148) and multiple reverse links (142, 146, 150) to expedite a transmission of data and a corresponding filling up of the jitter buffer, and by reduction of a waiting period for retransmission of the acknowledgments.
Abstract:
A method of controlling a queue buffer (2), said queue buffer (2) being connected to a link (1) and being arranged to queue data units (30) that are to be sent over said link (1) in a queue (20), comprising: determining (S1) a value (QL; QL
Abstract:
Methods and apparatuses for obtaining delay, jitter, and loss statistics of a path between server and an end user coupled via an internetwork are described. The serve may comprise a web server in communication with the end user via the Internet. Statistics are obtained by analyzing the details of a TCP connection underlying an HTML transaction. Robust measurements of jitter, delay, and loss are ensured by maximizing traffic between the web server and the surfer in order to generate a robust sample of TCP connections. Content may be updated with one or more html link(s). This existing content may reside on a highly trafficked portal, such as a web portal, and may be encoded in a markup language, such as Hyper Text Markup Language (HTML). The Uniform Resource Locators (URLs) corresponding to the one or more links resolve to the server from which the statistics are to be measured. The actual content supplied by the server may be minimized, in order to preserve bandwidth.
Abstract:
Improved data transport and management within a network communication system may be achieved by utilizing a transmit timer incorporated within the sender device and exploiting host-level statistics for a plurality of connections between a sender and receiver. The period of the transmit timer may be periodically adjusted based on a ratio of the smoothed round-trip time and the smoothed congestion window, thereby reducing or eliminating bursty data transmission commonly associated with conventional TCP architectures. For applications having a plurality of connections between a sender and a receiver that share a common channel, such as web applications, the congestion window and smoothed round trip time estimates for all active connections may be used to initialize new connections and allocate bandwidth among existing connections. This aspect of the present invention may reduce the destructive interference that may occur as different connections compete with one another to maximize the bandwidth of each connection without regard to other connections serving the same application. Error recovery may also be improved by incorporating a short timer and a long timer that are configured to reduce the size of the congestion window and the corresponding transmission rate in response to a second packet loss with a predefined time period in order to increase resilience to random packet loss.
Abstract:
A technique for providing high speed data service over standard wireless connections via an unique integration of protocols and existing cellular signaling is disclosed. Channel resources are allocated according to a buffer monitoring scheme between a base station (104) and multiple subscriber units (101-1,...,101-n). Each buffer (440-1,...,440-N) is monitored over time for threshold levels of data and a probability is calculated that takes into account the arrival of data into the buffer (440-1,...,440-N).
Abstract:
A system and method for guaranteeing a delay jitter bound when scheduling bandwidth grants for voice calls via a communication medium is provided. The method includes the steps of: determining the delay jitter bound, based on the determined delay jitter bound, dividing a packetization frame period into phases; assigning a voice call to one of the phases; and scheduling a bandwidth grant to the voice call during the assigned phase, thereby guaranteeing the delay jitter bound. The system includes a scheduler, where the scheduler determines the delay jitter bound, divides a packetization frame period into phases based on the determined delay jitter bound, assigns a voice call to one of the phases, and schedules a bandwidth grant to the voice call during the assigned phase, thereby guaranteeing the delay jitter bound. A dejitter buffer implements a way to provide zero jitter service, even though the packet transmission on the cable network has jitter, by delaying the packet and thus converting jitter into delay.
Abstract:
A system for the assessment of network performance criteria, and applying this criteria to the classification of network addresse s into appropriate ranges, using these ranges to consolidate performance measurements for the associated addresses, and applying these metrics toward the optimization of the network towards performance or policy objectives.