Multiplex communications – Data flow congestion prevention or control – Control of data admission to the network
Reexamination Certificate
1999-03-17
2003-09-23
Yao, Kwang Bin (Department: 2664)
Multiplex communications
Data flow congestion prevention or control
Control of data admission to the network
C370S468000
Reexamination Certificate
active
06625119
ABSTRACT:
BACKGROUND OF THE INVENTION
1. Field of the Invention
The present invention relates to telecommunications transmission systems and more particularly to a method and apparatus for facilitating increased call capacity in a transmission system when faced with a state of emergency.
The invention is particularly useful in the context of packet switched networks such as the Internet, and especially in the context of internet telephony (also known as voice-over-IP (VoIP) or IP telephony). For purposes of illustration, the invention will be described in the context of internet telephony, where “internet telephony” refers generally to the transmission of real-time media signals and/or data signals via a packet switched network (such as the Internet, for example). More generally, however, the invention can extend to use in connection with communication of any real-time media and/or data signals over any packet switched communications link, including, for instance, IP, ATM, frame relay, X.25 and SNA networks, whether local area, metropolitan area or wide area, and point-to-point or direct end-to-end connections. In addition, the invention can extend to use in connection with communications between a pair of terminals (e.g., users) or among multiple terminals (e.g., in a multicast environment).
2. Description of the Related Art
Those skilled in the art are familiar with the basic configuration of an internet telephony system. Architectural elements and functions suitable for use in one such system are described, for instance, by the H.323 standard for multimedia transmissions, as published by the International Telecommunications Union (ITU). The entirety of the H.323 standard is hereby incorporated herein by reference. The present invention, however, is not necessarily limited to use in the H.323 configuration but may extend to other configurations or other transmission protocols. For example, another protocol that can support internet telephony is Session Initiation Protocol, or SIP.
In general, an internet telephony system facilitates telephone communication between two or more users over a packet switched, such as an IP network for example. Each user may be positioned at a telephone device (hereafter “telephone”), which may be a personal computer, a digital telephone, an analog telephone (e.g., a “black box telephone”) or other suitable communications equipment. The telephone device may be any real-time media communication device, such as, for example, an audio telephone, a videophone, or a combination or subset of such devices and/or other devices. Each telephone device (and/or telephone number) is then typically served by a network access server, which provides connectivity to the packet switched network. In the context of internet telephony, the network access server may be referred to as an internet telephony gateway (“ITG” or “gateway”) and is typically owned and operated by an internet telephony service provider (ITSP).
The gateway serves as an interface between the packet switched network and the communications link, and in turn the telephone device. In this regard, for instance, the gateway typically performs translation between protocols, data formats and media types, to facilitate communication of information between two possibly different types of networks or links. For example, a gateway may be configured to receive a real-time media stream from the communications link and to encode (e.g., compress and packetize) the stream into a sequence of packets for transmission over the packet switched network to a remote destination. Similarly, a gateway may be configured to decode (e.g., de-packetize and decompress) data arriving from the packet switched network and to forward the resulting media stream via a communications link to a specified telephone device.
To place a two-party call to over a packet switched network, a user at an initiating telephone device establishes a connection with an initiating gateway via a suitable communications link such as the public switched telephone network (PSTN) and/or other circuit switched or packet switched network or direct link. Alternatively, the telephone device itself may be directly connected to the IP network. The communications link may be a permanent or semi-permanent connection (as in the case of a LAN connection between the telephone device and the gateway), which may facilitate direct dialing. Alternatively, to connect with the gateway, the user may need to place a call over the public switched telephone network call to the gateway, such as by dialing a telephone number designated by the user's ITSP. In any event, the user may specify the telephone number of the called party.
In a packet switched network, the location of each gateway and other element is identified by a network address. Therefore, given the telephone number of a called party, the initiating gateway must identify the network address of a terminating gateway that can serve the called number. To identify the network address, the gateway may query an address mapping database or may send a signaling message to another device or process that can provide the necessary address based on the dialed number. In some systems, this address translation is facilitated by a signaling server such as a gatekeeper or proxy.
Provided with the network address of the terminating gateway, the initiating gateway may then contact the terminating gateway via the packet switched network and notify the terminating gateway of the desire to establish a connection with the called party. The terminating gateway may then establish an appropriate connection (e.g., over a communications link such as the PSTN) with a telephone device at the called number and notify the initiating gateway that the call can proceed. With the end-to-end connection thus established between the calling and called parties, the parties may then communicate with each other over the packet switched network, sending and receiving various communications signals, such as voice, video, audio and/or data.
The present invention relates to the capacity of telephony systems to receive and handle call traffic and specifically to the number of calls that the system can handle at any given time. One of the factors that is known to affect call capacity in a telephony network is the statistical analysis and engineering design that is involved in aggregating traffic in the network. In general, this analysis focuses on so-called “aggregation points,” where traffic from a number of locations arrives to be processed and/or funneled through to a next point in the network.
An aggregation point typically has an input capacity (e.g., physical input ports, time slots, or channels), which may define a maximum amount of data that the aggregation point can receive and/or process at once (e.g., multiplexed, in parallel, etc.). In turn, the aggregation point typically has a processing capacity and/or output capacity, which defines a maximum amount of data that the aggregation point can process and/or output at once for transmission to the next (downstream) element in the network.
In many cases, it would be too expensive to build an aggregation point to be able to process and output all of its potential input in real time. Therefore, aggregation points are instead usually designed to have higher input capacity than processing capacity and/or output capacity. This design is based on statistical modeling and desired probabilities of call blocking. The statistical modeling assumes that, at any given time, something less than all of the input capacity will be filled, and that the input at any given time can therefore be statistically multiplexed among the available processing capacity and output capacity. If more than the statistically assumed input capacity arrives at once, some of the input may be blocked.
As an example of an aggregation point, consider the central office (CO) switch in the PSTN. A CO switch has a potential input capacity defined by a set number of input ports, each of which is permanently hard-wired to a telephone subscriber. The CO
Grabelsky David A.
Kostas Thomas J.
Schuster Guido M.
Sidhu Ikhlaq S.
3Com Corporation
McDonnell & Boehnen Hulbert & Berghoff
Yao Kwang Bin
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