Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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 a methodology for aggregating packets of a first size to packets of a second size for transmission in a communications network. The method 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.

Abstract: 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.

Abstract: 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.

Abstract: A method and apparatus is disclosed for providing continuation of a communication call as a user moves from the coverage area of one communication network to the coverage area of another communication network having different access media and different applications. A voice, data, video or multimedia call involving multiple end-points which may be located on different networks may be placed on hold pursuant to a pause request made by a user of one of the end-points. The networks include processors programmed to interrupt and recognize specified signals, such as a pause request signal, and to respond by providing reconnection identification information. The user making a pause request may generate a reconnection request in a different network to continue the paused call on the new network by using a calling device that is compatible with the access media and applications provided by the new network.