Electrical computers and digital processing systems: multicomput – Computer-to-computer session/connection establishing
Reexamination Certificate
2000-07-18
2004-12-14
Vu, Thong (Department: 2142)
Electrical computers and digital processing systems: multicomput
Computer-to-computer session/connection establishing
C709S223000, C709S229000
Reexamination Certificate
active
06832254
ABSTRACT:
BACKGROUND
1. Field of the Invention
This invention is related to multimedia packet networks. Specifically, this invention relates to a mechanism to allow such a packet network to more effectively carry telephony messages, and to more efficiently interface with the public switched telephone network (PSTN).
2. Description of the Problem Solved
Evolution of the PSTN has accelerated in recent years; however, most of the PSTN still operates on circuit switched, time division multiplexed (TDM) connections. Integrated services digital network (ISDN) bearer channels often provide transport. In parallel with the PSTN, a packet based data network has evolved. This data network has largely been used for Internet traffic and data networking. Although these networks have been mostly separate until recently, the two networks are starting to merge. The merger of these networks requires that voice traffic be carried over packet networks, and further that such packet networks be able to seamlessly integrate with traditional circuit switched networks, as the two types of networks may carry different call legs of the same call.
FIG. 1
illustrates a typical TDM, PSTN call. Caller
101
places a call to callee
105
. The call goes through end office A,
102
, over some type of trunk bearer channel to toll office
103
, then to end office B,
104
, and finally to the callee. Such calls may pass through multiple toll offices, or may be connected directly from one end office to another. In any case, a path of circuits for the call is maintained throughout the call. Signaling between offices is typically provided by an ISUP (ISDN user part) connection. ISUP signaling is well understood and is standard in the telecommunications industry. For more information on ISUP signaling, see the various International Telecommunications Union (ITU) Recommendations pertaining to telephone signaling, including Q.761, Q.764 and Q.931, the most recent versions of which at the time of filing this application am incorporated herein by reference.
FIG. 2
illustrates a call which is similar to the TDM call of
FIG. 1
; however, in this case, the call is transported from one end office to another (called switch offices,
202
and
204
, in this case) via a packet switched network
203
. This fact is, in theory, transparent to caller
201
and callee
205
. ISUP+ or SIP+ provides signaling in this case. ISUP+ is essentially the same as ISUP except that ISUP+ signals contain extra fields for packet or cell based network information. An International Telecommunications Union (ITU) recommendation has been proposed for ISUP+ as of the filing date of this application as ITU Q.BICC. SIP stands for “session initiation protocol” and is a well-known standard. SIP and SIP+ are described in document RFC 2543, published by the Internet Engineering Task Force (IETF), March, 1999 which is incorporated herein by reference. SIP and SIP+ provide the same type of signaling for control of calls, but are more oriented towards packet based networks.
The network of
FIG. 2
has been conceptualized for some time, and standards groups and conference groups have written extensively about how to make such a network work in everyday use. In order for the call leg which is handled by the packet network to seamlessly connect with the call legs handled by TDM switching offices, media provided by one type of network must be converted into media provided by the other. This conversion is referred to as circuit emulation services (CES) in an ATM network. The device that provides this conversion is called a media gateway (MG). In the network of
FIG. 2
, a media gateway handles each end of the bearer connection through packet network
203
. The media gateway terminates bearer media streams from both the switched circuit TDM network, and the packet network. The media gateway and the network it serves may be capable of processing audio and video (hence the term “multimedia packet network”). The media gateway is capable of full duplex media translations. It may also provide other features such as conferencing.
Each media gateway is associated with a media gateway controller (MGC). The media gateway is “dumb” in that it does not have call processing capabilities. The call processing capabilities for the network reside in the MGC. An MGC provides the signaling for call control and controls the call state of a media gateway. The protocol used by the MGC to control the MG is called the media gateway control protocol (or the “Megaco” protocol). Megaco is an application layer protocol which is also described in ITU Recommendation H.248, which shares a common text with the IETF Internet Draft “Megaco Protocol,” and which is incorporated herein by reference. The “Megaco Protocol” Internet Draft first became an IETF working document in March, 1999. Within the Megaco protocol, session description protocol (SDP) can be used to describe bearer channel terminations, which are being controlled by the MGC's. SDP is described in document RFC 2327, published by the IETF, April 1998, which is incorporated herein by reference. Throughout the rest of this disclosure we will refer to Megaco as “Megaco/H.248.”
Despite the fact that the theoretical workings of a network like that shown in
FIG. 2
have been widely explored, such networks have seen relatively little everyday use. The reason is that there are still problems to be overcome before such networks exhibit the same very high quality of service for voice traffic that users of the PSTN have come to expect. One such problem stems from the fact that there is no dedicated physical path for a call through a packet network, and therefore no way to identify a particular media stream to be associated with a particular call.
A packet switched network, used for transport of audio and video media streams, is typically based on asynchronous transfer mode (ATM), frame relay (FR), and Internet protocol (IP) technologies. Public ATM networks generally operate according to the user network interface (UNI). The UNI is described in the book, “ATM User Network Interface (UNI) Specification Version 3.1” by the ATM Forum, published by Prentice Hall PTR, June, 1995, which is incorporated herein by reference. An update to the UNI version 3.1, “ATM User-Network Interface (UNI) Signaling Specification 4.0” was published by the ATM Forum in July, 1996, and is incorporated herein by reference. For private ATM networks, the private network to network interface (PNNI) describes the ATM interface. PNNI is covered in the ATM forum document “PNNI addendum for the network call correlation identifier” published by the ATM forum in July 1999, which is incorporated herein by reference. In ATM, fixed length cells carry packetized data. Each cell that is sent through the network has a virtual channel identifier, and other addressing information; however, each node in the network handles many cells that are associated with different media streams. Therefore, each call leg on the ATM network may actually go through many different network nodes and many different virtual circuits to complete the network path. It is impossible for an MGC and a media gateway to correlate the call leg throughout its path with a particular call. Since the nodes in the network are unaware of which call's cells are being sent when, it is difficult to maintain control of the call throughout the network in real time to maintain an appropriate level of quality of service. What is needed is a way within the Megaco/H.248 protocol to absolutely identify a media stream in the network as being associated with a particular call.
SUMMARY
The present invention solves the above-described problem by providing an end-to-end call identifier (EECID) for use in an ATM or other type of packet switched network which serves as a transport network for real-time audio and video media streams. The EECID is used to identify a call leg uniquely across the packet network, regardless of the number of nodes used in completing the network path. The EECID allows a call t
Brown Charles Michael
Jarzemsky David John
Joyner Stanley Wayne
Schellenberger Kathleen Kelley
Scoggins Shwu-Yan Chang
Nortel Networks Limited
Vu Thong
Withrow & Terranova , PLLC
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