Multiplex communications – Data flow congestion prevention or control – Control of data admission to the network
Reexamination Certificate
1999-04-30
2003-02-04
Nguyen, Chau (Department: 2663)
Multiplex communications
Data flow congestion prevention or control
Control of data admission to the network
C370S252000, C370S352000
Reexamination Certificate
active
06515964
ABSTRACT:
FIELD OF THE INVENTION
The invention relates to network call admission. More particularly, the invention relates to a method and apparatus for dynamically controlling the admission of calls to a network.
BACKGROUND OF THE INVENTION
The present invention relates to a technique for dynamically controlling the admission of traffic to a network based in part on the state of the network.
One known type of network is an Internet Protocol (“IP”) network. An IP network implements the protocol specified in RFC 791, Internet Protocol <www.cis.ohio-state.edu/htbin/rfc/rfc791.html, visited November 30, 1998>. One type of traffic carried by known IP networks is voice traffic, called Voice over IP (“VoIP”) traffic.
FIG. 1
is an example of a prior art VoIP system. It is known to initiate a voice call from a phone set
1
over a conventional circuit-switched network
21
(such as the public switched telephone network (PSTN)) and route the calling party's voice signals to a first gateway
11
connected to the IP network
23
. The first gateway
11
packetizes the voice signals using the Internet Protocol and transmits the packets as VoIP traffic over the IP network
23
to a second gateway
12
closer to the called party than the first gateway
11
. The packets are converted back into voice signals at the second gateway
12
, and those voice signals are routed via the conventional circuit-switched network
22
, to the called party's phone set
2
.
One of the problems with VoIP services is latency. Latency is the delay between the time a signal is sent and the time it is received. Latency adversely affects the quality of service of real-time communications (e.g., voice communications) and is dependent upon the state of the network over which the communications are carried. For example, a heavily burdened network is likely to have more latency than an underutilized network.
A similar problem arises in the context of users making other types of calls over a packet-switched network, such as the Internet. At present, a user can be connected to the Internet by an Internet Service Provider (ISP) and can make a number of calls over the Internet via HTTP (Hypertext Transfer Protocol) commands (using a Web browser such as Microsoft Internet Explorer or Netscape Navigator), FTP (File Transfer Protocol) commands, TELNET connections, and the like. The user may encounter significant delays in accessing, for example, Web sites. Those delays can be caused by a number of factors, including a Web site's inability to respond to all of the users that concurrently seek information from that Web site. A user also may experience significant delays in accessing a particular Web site, not due to that Web site's inability to meet the demand for that site, but due to poor performance characteristics of one or more networks which couple the user to the Web site, or the internetwork routers.
In known VoIP systems, a gateway will pass traffic into a network whenever the gateway has an incoming port that is available to do so. Thus, certain networks must disadvantageously be over-engineered to be able to carry a peak load equal to the traffic that flows when all of the ports of all of the gateways connected to the network are in use. If the traffic sent through the network approaches or exceeds the network's capacity, then the network disadvantageously drops packets (i.e., experiences packet loss) and/or introduces unacceptable delays into communications. In known networks, it is difficult or impossible to guarantee a high quality of service when the network is operating near or at its capacity.
The International Telecommunications Union (“ITU”) has established the H.323 standard, which encompasses audio, video and data communications across packet-switched networks, such as the Internet. The H.323 standard was principally developed and established to allow multimedia products and applications from multiple vendors to interoperate. H.323 systems may include a gatekeeper, which can provide bandwidth management. For example, the gatekeeper can reject calls from a terminal if it determines that sufficient bandwidth is not available. H.323 bandwidth management also operates during an active call if a terminal requests additional bandwidth, and the gatekeeper may grant or deny the request for additional bandwidth. Likewise, there are other Internet protocols that provide for establishing or rejecting calls based on bandwidth requirements (e.g., RFC 2211, Specification of the Controlled-Load Network Element Service, <www.cis.ohio-state.edu/htbin/rfc/rfc2211.html, visited Jan. 11, 1999>; RFC 2210, The Use of RSVP with IETF Integrated Services, <www.cis.ohio-state.edu/htbin/rfc/rfc2211.html, visited Jan. 11, 1998>). These bandwidth management protocols do not provide for admitting or rejecting calls based on delay characteristics of the network.
SUMMARY OF THE INVENTION
The present invention provides a system for regulating the call traffic into a packet-switched network based in part upon delay characteristics of the network. In an embodiment of the present invention, a call delay characteristic requirement for a call is determined, a delay characteristic parameter of the packet-switched networks is determined, and a call action based at least partly upon the determined delay characteristic requirement and the determined delay characteristic parameter is performed.
In one embodiment of the invention, the network is an Internet Protocol (IP) network carrying Voice over IP (VOIP) traffic. A voice call made in connection with a VoIP service is not admitted to the IP network and is held if one or more current delay characteristic parameters of the IP network do not satisfy one or more prescribed delay characteristic requirements. Delay characteristic parameters can be periodically updated, and when the current value of one or more delay characteristic parameters satisfy one or more prescribed delay requirements, the VoIP call is admitted to the IP network.
Another embodiment of the present invention dynamically controls the admission of other traffic to an IP network, including multimedia communications, HTTP commands, FTP commands, TELNET connections, and the like. This embodiment allows such data calls to be admitted to the IP network when the IP network satisfies the delay requirements.
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Gordon, Shykeh, H.323: The Multimedia Communications Standard Moves From Consensus to Compliance, CTI Developer, 2(2):108-113.
Blank Michelle, H.323 Gatekeepers: Essential Software for IP Telephony and Multimedia Conferencing, CTI Developer, pp. 94-98, Feb. 1998.
Shenker, et al., RFC 2212 Specification of Guaranteed Quality of Service, pp. 1-20, Sep. 1997.
International Telecommunication Union, H.323, Series H: Audiovisual and Multimedia Systems: Infrastructure of audiovisual services—Systems and terminal equipment for audiovisual services, Visual telephone systems and equipment for local area networks which provide a non-guaranteed quality of service, 79 pp., Nov. 1996.
Cheung Hay Yeung
Hosseini-Nasab Louise E.
Yaniro, Jr. Daniel J.
AT&T Corp.
George Keith M.
Nguyen Chau
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