Lucent Technologies Inc.

Alcatel-lucent USA Inc.

Avaya Technology LLC

Behrokh Samadi

Wassim A Matragi

Qinqing Zhang

Bharat Tarachand Doshi

Yung-Terng Wang

Eric Jonathan Bauer

Joseph Fried

An architecture for quality-of-service (QoS) management creates a logical circuit-switched network within a packet network to support QoS-sensitive demands levied on the network. This QoS-managed network can serve to interwork, e.g., a PSTN with VoIP networks. The architecture can include a connection resource manager (CRM), which oversees bandwidth availability and demand admission/rejection on dynamically provisioned virtual trunk groups (VTGs) within the packet network, and a transport bandwidth controller (TBC). The VTGs serve to transport QoS-sensitive demands across the packet network. The TBC serves the CRM by providing an interface to routers and/or OAM systems of the packet network to size VTGs to meet QoS requirements. Media switches located at the packet network borders serve to mux/demux the demands into/from VTGs. CRMs and TBCs can be implemented as centralized, distributed, or hierarchical, and flat and aggregated variants of the architecture are supported. VTGs can be implemented using MPLS LSPs, VPNs, or source-based routing.

Network quality of service management

The present invention is a methodology for providing packet aggregation based on transmission window size to achieve improved transport efficiency. One embodiment of the invention includes the steps of receiving an indication of a transmission window size for packets to be transmitted from a first location to a second location where the transmission window size provided by a flow control message. The transmission window size is utilized to determine a bundling factor for transmission of packets from said first location to said second location, the bundling factor being a lesser of a maximum bundling factor and the transmission window size. A transmission window size threshold represents a boundary for a size of a transmission window when transmission of existing frames may be delayed. The frames are bundled according to the bundling factor if the transmission window size is greater than the transmission window size threshold.

Methods and apparatus for flow control based packet aggregation in a communication network

The present invention is a method and apparatus to aggregate or bundle packets to optimize the resource utilization and improve the transport efficiency of packets which are transferred between various protocol layers in a communications network. One embodiment of the invention sets forth a method of aggregating packets of a first size to packets of a second size for transmission in a wireless communications network between, for example, a base station controller and base transmitting station. The method includes the steps of calculating a maximum bundling factor represented by (an IP packet MTU size at a first location minus protocol overhead) divided by an application layer packet size and determining a most efficient bundling factor in terms of packet fragments, where the most efficient bundling factor is determined by a largest packet per fragment ratio. If the number of packets to be bundled is less than or equal to the maximum bundling factor then all packets are bundled together for a next step in the transmission. If the number of packets to be bundled is greater than the maximum bundling factor, then packets are bundled according to the most efficient bundling factor for the next step in the transmission. If there are multiple bundling factors having a same bundling efficiency, a larger of the multiple bundling factors having the same bundling efficiency is selected as the most efficient bundling factor.

Methods and apparatus for optimum packet aggregation in a communication network

A method and apparatus are disclosed for congestion management in a multi-branch Internet Protocol-based private branch exchange switch. The multi-branch Internet Protocol-based private branch exchange switch is interconnected through (i) a packet network referred to as the primary network, such as a wide area network, and (ii) an alternate network, such as the public switched telephone network. Packet phone adapters associated with each packet telephone unit monitor packet telephone calls and report delay information to communication servers. The communication server can reroute the packet telephony calls through the secondary network upon detection of congestion in the underlying primary network, thereby preserving voice quality. The packet phone adapter will discard records collected from calls whose duration is below a minimum value, to ensure reliable congestion information. Each communication server records reported voice quality of service information in a congestion control database. An overload control process processes each call set up request and determines if the requested path is congested. If a requested path is congested, then the overload control process may forward the call using the secondary network.

Method and apparatus for overload control in multi-branch packet networks

A method and apparatus are disclosed for alleviating congestion and overload in a distributed call-processing system interconnected through a packet based network. The illustrative Internet Protocol network includes a plurality of end terminals and distributed call processors. According to an aspect of the invention, the call processor will determine whether to process a call request or to forward the request to another call processor. Generally, the call processor will declare an overload condition if sufficient resources (including processing or memory resources) are not available to process a given call. If a call processor determines that it is too congested to process a call, the call processor enters an overload condition, selects an alternate call processor and forwards the request to the alternate call processor. Each call processor maintains an ordered list of call processors that indicates whether or not each call processor is overloaded.

Method and apparatus for message-based overload control in a distributed call-processor communication system

A method and apparatus are disclosed for dynamically adapting the play-out delay for voice packets transmitted over a local area network using the Ethernet standard as an access mechanism. It has been observed that the distribution of voice packet delays in a LAN (shared by voice and data traffic) follows a log-normal distribution. An adaptive algorithm is disclosed to estimate the parameters of the log-normal distribution and to apply a dynamic play-out delay to improve the quality of packetized voice conversations, quantified by minimum delay and packet loss. For example, the size, B, of the play-out buffer can be established to ensure that the packet loss does not exceed one percent (1%). The distribution parameters are continuously updated and the size of the play-out buffer, B, is modified at the beginning of every calibration interval according to the illustrative 99% percentile of the delay distribution.

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Saratoga, CA, United States of America

Behrokh Samadi