摘要:
Overhead is reduced and packet transport efficiency is increased for a flow of switched packets from a router by identifying a plurality of packets having a common destination node within a network and transmitting at least one control message to establish the flow of switched packets; the at least one control message including: (i) a label mapping message corresponding to the flow of switched packets and (ii) a header removal field. Packet headers corresponding to packets of a switched packet flow are not parsed, therefore either the entire header, or a portion of the header, may be removed from each packet assigned a label. A header removal field is shared among routers while signaling to establish a labeled flow. The header removal field is used to provide header structure information to those routers which will be utilized for transport of the subsequent labeled flow. Packet transport densities are monitored at individual routers. A dual-value threshold density methodology is applied at routers to determine whether packet transport density is great enough to warrant a dedicated label, a shared label applied to merged switched packet flows, or no label at all.
摘要:
The present invention provides an improved RACH access burst arrangement and frame structure. That is, the invention provides methods and apparatus for supporting more than one access burst length in the UMTS access channel structure. Preferably, two access burst lengths are supported, e.g., 5 ms and 10 ms. Such an arrangement is advantageous in applications where it is beneficial to have fast access latency such as, for example, voice or other forms of real-time traffic. Also, the invention provides methods and apparatus for supporting multiple frame sizes. It is to be appreciated that further enhancement to access latency can be obtained by having the UMTS physical layer support multiple frame sizes. The access burst signal transmitted by a remote terminal over the RACH may be an access request or data packets in the case where the RACH is being utilized for UMTS short message services.
摘要:
Overhead is reduced and packet transport efficiency is increased for a flow of switched packets from a router by identifying a plurality of packets having a common destination node within a network and transmitting at least one control message to establish the flow of switched packets; the at least one control message including: (i) a label mapping message corresponding to the flow of switched packets and (ii) a header removal field. Packet headers corresponding to packets of a switched packet flow are not parsed, therefore either the entire header, or a portion of the header, may be removed from each packet assigned a label. A header removal field is shared among routers while signaling to establish a labeled flow. The header removal field is used to provide header structure information to those routers which will be utilized for transport of the subsequent labeled flow. Thus, since layer two switching utilizes the appended label instead of the packet header for conveying a labeled packet, the packet header may be partially or completely removed. The header removal field is used to inform routers utilized for a labeled packet flow which portions of the subsequently conveyed packet headers will be removed and which portions will be present.
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:
A Universal Mobile Telecommunications System (“UMTS”) is modified to provide location-determining services similar to Wireless-Assisted-GPS (“WAG”) services to mobiles without requiring the deployment of a GPS receiver inside the UMTS network. The resulting network is known as a UMTS Network Assisted GPS (“UNAG”) network. A UNAG network is created by deploying a UNAG server inside a UMTS network. The UNAG server provides WAG-like services to mobiles. Only one UNAG server is needed in an entire system, allowing the cost of the UNAG server to be amortized over a number of Mobile Switching Centers (“MSCs”) in the UMTS network. The UNAG server provides excellent location-prediction accuracy and reduces the amount of time required to determine the location of a mobile in a UMTS network.
摘要:
The present invention relates to a time compandor (40) including a cascaded pair of chirp transformers (41,42) of chirp rates .beta. and .sigma., respectively, capable of companding a continuous time signal x(t) to form a companded and delayed output signal x(a[t-.tau.]). Each chirp transformer includes a pi-network arrangement of linear dispersive filters (44,45,46,47,48,49), wherein the first chirp transformer (41) performs a Fourier transform on a continuous time input signal x(t) to produce X(.beta.t) and the second chirp transformer (42) performs an inverse Fourier transform on the product of the output of the first chirp transformer and a signal associated with the desired time delay .tau., to form the desired companded output signal x(.sigma./.beta.[t-.tau.]), where .sigma./.beta.=.alpha. is the desired companding rate, and .tau. is the desired time delay.
摘要:
A system is disclosed for overload control in a hybrid switching system that separately transports each monomedia stream, such as video, voice and data, of a composite multimedia signal. The overload control is based on the time delay for completion of call establishment of each multimedia session. When the average time delay for completion of call establishment of each multimedia session is below a predefined threshold, the system has sufficient capacity and overload corrective measures are not required. When the average time delay for completion of call establishment of each multimedia session, or a connection for any individual monomedia component within the multimedia session, exceeds a predefined threshold, the switching system is approaching an overload condition, and overload control must be implemented to reduce the number of calls accepted in the next time interval. A decision is made at the beginning of each time interval based on the average call completion time delay over some predefined period of time, to accept a determined number of calls during each time interval.
摘要:
Call admission methods for admitting connections into ATM/IP networks having a plurality of communication channels are disclosed. An overbooking technique is utilized which distinguishes among the different service classes. Each service class is assigned an overbooking factor. The call admission is determined based on the overbooking factor assigned to the class and the effective bandwidth for that service class. In addition, methods are disclosed for performing appropriate bookkeeping, i.e., updating and maintaining information concerning the state of the system.
摘要:
Methods and systems are provided for improving frame selection in wireless communications networks. During decoding of a frame, a base station generates an error burst representation associated with error bursts and stores the representation within the frame, thus forming an enhanced frame. The base station then transfers the enhanced frame to a network controller. A frame selection unit (“FSU”) within the network controller thereafter applies frame selection to the enhanced frame. The error burst representation can be analyzed to determine the quality of the enhanced frame. A “combined” frame, generated by combining an “acceptable” portion of the enhanced frame and an acceptable portion of a copy of the enhanced frame, can then be generated to substantially eliminate errors. The present frame selection methods and systems enable superior quality frames to be passed on to higher layers in a network's communications protocol (hereafter collectively referred to as “network”).