US 6614781 Voice over data telecommunications network architecture
ABSTRACT – The present invention describes a system and method for communicating voice and data over a packet-switched network that is adapted to coexist and communicate with a legacy PSTN. The system permits packet switching of voice calls and data calls through a data network from and to any of a LEC, a customer facility or a direct IP connection on the data network. The system includes soft switch sites, gateway sites, a data network, a provisioning component, a network event component and a network management component. The system interfaces with customer facilities (e.g., a PBX), carrier facilities (e.g., a LEC) and legacy signaling networks (e.g., SS7) to handle calls between any combination of on-network and off-network callers.
The present invention relates to telecommunications, and in particular to voice and data communication operating over a data network. The Public Switched Telephone Network (PSTN) is a collection of different telephone networks owned by different companies which have for many years provided telephone communication between users of the network. Different parts of the PSTN network use different transmission media and compression techniques.
Most long distance calls are digitally coded and transmitted along a transmission line such as a T1 line or fiber optic cable, using circuit switching technology to transmit the calls. Such calls are time division multiplexed (TDM) into separate channels, which allow many calls to pass over the lines without interacting. The channels are directed independently through multiple circuit switches from an originating switch to a destination switch. Using conventional circuit switched communications, a channel on each of the T1 lines along which a call is transmitted is dedicated for the duration of the call, whether or not any information is actually being transmitted over the channel. The set of channels being used by the call is referred to as a “circuit.”
Telecommunications networks were originally designed to connect one device, such as a telephone, to another device, such as a telephone, using switching services. As previously mentioned, circuit-switched networks provide a dedicated, fixed amount of capacity (a “circuit”) between the two devices for the entire duration of a transmission session. Originally, this was accomplished manually. A human operator would physically patch a wire between two sockets to form a direct connection from the calling party to the called party. More recently, a circuit is set up between an originating switch and a destination switch using a process known as signaling.
Signaling sets up, monitors, and releases connections in a circuit-switched system. Various signaling methods have been devised. Telephone systems formerly used in-band signaling to set up and tear down calls. Signals of an in-band signaling system are passed through the same channels as the information being transmitted. Early electromechanical switches used analog or multi-frequency (MF) in-band signaling. Thereafter, conventional residential telephones used in-band dual-tone multiple frequency (DTMF) signaling to connect to an end office switch. Here, the same wires (and frequencies on the wires) were used to dial a number (using pulses or tones), as are used to transmit voice information. However, in-band signaling permitted unscrupulous callers to use a device such as a whistle to mimic signaling sounds to commit fraud (e.g., to prematurely discontinue billing by an interexchange carrier (IXC), also known as a long distance telephone company).
More recently, to prevent such fraud, out-of-band signaling systems were introduced. Out-of-band signaling uses a signaling network that is separate from the circuit switched network used for carrying the actual call information. For example, integrated services digital network (ISDN) uses a separate channel, a data (D) channel, to pass signaling information out-of-band. Common Channel Interoffice Signaling (CCIS) is another network architecture for out-of-band signaling. A popular version of CCIS signaling is Signaling System 7 (SS7). SS7 is an internationally recognized system optimized for use in digital telecommunications networks.
SS7 out-of-band signaling provided additional benefits beyond fraud prevention. For example, out-of-band signaling eased quick adoption of advanced features (e.g., caller id) by permitting modifications to the separate signaling network. In addition, the SS7 network enabled long distance “Equal Access” (i.e., 1+ dialing for access to any long distance carrier) as required under the terms of the modified final judgment (MFJ) requiring divestiture of the Regional Bell Operating Companies (RBOCs) from their parent company, AT&T.
An SS7 network is a packet-switched signaling network formed from a variety of components, including Service Switching Points (SSPs), Signaling Transfer Points (STPs) and Service Control Points (SCPs). An SSP is a telephone switch which is directly connected to an SS7 network. All calls must originate in or be routed through an SSP. Calls are passed through connections between SSPs. An SCP is a special application computer which maintains information in a database required by users of the network. SCP databases may include, for example, a credit card database for verifying charge information or an “800” database for processing number translations for toll-free calls. STPs pass or route signals between SSPs, other STPs, and SCPs. An STP is a special application packet switch which operates to pass signaling information.
The components in the SS7 network are connected together by links. Links between SSPs and STPs can be, for example, A, B, C, D, E or F links. Typically, redundant links are also used for connecting an SSP to its adjacent STPs. Customer premises equipment (CPE), such as a telephone, are connected to an SSP or an end office (EO) switch.
To initiate a call in an SS7 telecommunications network, a calling party using a telephone connected to an originating EO switch, dials a telephone number of a called party. The telephone number is passed from the telephone to the SSP at the originating EO (referred to as the “ingress EO”) of the calling party’s local exchange carrier (LEC). A LEC is commonly referred to as a local telephone company. First, the SSP will process triggers and internal route rules based on satisfaction of certain criteria. Second, the SSP will initiate further signaling messages to another EO or access tandem (AT), if necessary. The signaling information can be passed from the SSP to STPs, which route the signals between the ingress EO and the terminating end office, or egress EO. The egress EO has a port designated by the telephone number of the called party. The call is set up as a direct connection between the EOs through tandem switches if no direct trunking exists or if direct trunking is full. If the call is a long distance call, i.e., between a calling party and a called party located in different local access transport areas (LATAs), then the call is connected through an inter exchange carrier (IXC) switch of any of a number of long distance telephone companies. Such a long distance call is commonly referred to as an inter-LATA call. LECs and IXCs are collectively referred to as the previously mentioned public switched telephone network (PSTN).
Emergence of competitive LECs (CLECs) was facilitated by passage of the Telecommunications Act of 1996, which authorized competition in the local phone service market. Traditional LECs or RBOCs are now also known as incumbent LECs (ILECs). Thus, CLECs compete with ILECs in providing local exchange services. This competition, however, has still not provided the bandwidth necessary to handle the large volume of voice and data communications. This is due to the limitations of circuit switching technology which limits the bandwidth of the equipment being used by the LECs, and to the high costs of adding additional equipment.
Since circuit switching dedicates a channel to a call for the duration of the call, a large amount of switching bandwidth is required to handle the high volume of voice calls. This problem is exacerbated by the fact that the LECs must also handle data communications over the same equipment that handle voice communications.
If the PSTN were converted to a packet-switched network, many of the congestion and limited bandwidth problems would be solved. However, the LECs and IXCs have invested large amounts of capital in building, upgrading and maintaining their circuit switched networks (known as “legacy” networks) and are unable or unwilling to jettison their legacy networks in favor of the newer, more powerful technology of packet switching. Accordingly, a party wanting to build a packet-switched network to provide voice and data communications for customers must build a network that, not only provides the desired functionality, but also is fully compatible with the SS7 and other, e.g., ISDN and MF, switching networks of the legacy systems.
Currently, internets, intranets, and similar public or private data networks that interconnect computers generally use packet switching technology. Packet switching provides for more efficient use of a communication channel as compared to circuit switching. With packet switching, many different calls (e.g., voice, data, video, fax, Internet, etc.) can share a communication channel rather than the channel being dedicated to a single call. For example, during a voice call, digitized voice information might be transferred between the callers only 50% of the time, with the other 50% being silence. For a data call, information might be transferred between two computers 10% of the time. With a circuit switched connection, the voice call would tie-up a communications channel that may have 50% of its bandwidth being unused. Similarly, with the data call, 90% of the channel’s bandwidth may go unused. In contrast, a packet-switched connection would permit the voice call, the data call and possibly other call information to all be sent over the same channel.
Packet switching breaks a media stream into pieces known as, for example, packets, cells or frames. Each packet is then encoded with address information for delivery to the proper destination and is sent through the network. The packets are received at the destination and the media stream is reassembled into its original form for delivery to the recipient. This process is made possible using an important family of communications protocols, commonly called the Internet Protocol (IP).
In a packet-switched network, there is no single, unbroken physical connection between sender and receiver. The packets from many different calls share network bandwidth with other transmissions. The packets are sent over many different routes at the same time toward the destination, and then are reassembled at the receiving end. The result is much more efficient use of a telecommunications network than could be achieved with circuit-switching.
Recognizing the inherent efficiency of packet-switched data networks such as the Internet, attention has focused on the transmission of voice information over packet-switched networks. However, such systems are not compatible with the legacy PSTN and therefore are not convenient to use.
One approach that implements voice communications over an IP network requires that a person dial a special access number to access an IP network. Once the IP network is accessed, the destination or called number can be dialed. This type of call is known as a gateway-type access call.
Another approach involves a user having a telephone that is dedicated to an IP network. This approach is inflexible since calls can only be made over the UP network without direct access to the PSTN.
What is needed is a system and method for implementing packet-switched communications for both voice calls and data calls that do not require special access numbers or dedicated phones and permit full integration with the legacy PSTN.
SUMMARY OF THE INVENTION
The present invention is a system and method for communicating both voice and data over a packet-switched network that is adapted to coexist and communicate with a PSTN. The system permits efficient packet switching of voice calls and data calls from a PSTN carrier such as, for example, a LEC, IXC, a customer facility or a direct IP connection on the data network to any other LEC, IXC, customer facility or direct IP connection. For calls from a PSTN carrier, e.g., LEC or IXC, the invention receives signaling from the legacy SS7 signaling network or the ISDN D-channel or from inband signaling trunks. For calls from a customer facility, data channel signaling or inband signaling is received. For calls from a direct IP connection on the data network, signaling messages can travel over the data network. On the call destination side, similar signaling schemes are used depending on whether the called party is on a PSTN carrier, a customer facility or a direct IP connection to the data network.
The system includes soft switch sites, gateway sites, a data network, a provisioning component a network event component and a network management component. The system of the invention interfaces with customer facilities (e.g., a PBX), carrier facilities (e.g., a PSTN carrier, a LEC (e.g., ILECs and CLECs), an independent telephone company (ITC), an IXC, an intelligent peripheral or an enhanced service provider (ESP)) and legacy signaling networks (e.g., SS7) to handle calls between any combination of on-network and off-network callers.
The soft switch sites provide the core call processing for the voice network architecture. Each soft switch site can process multiple types of calls including calls originating from or terminating at off-network customer facilities as well as calls originating from or terminating at on-network customer facilities. Each soft switch site receives signaling messages from and sends signaling messages to the signaling network. The signaling messages can include, for example, SS7, integrated services digital network (ISDN) primary rate interface (PRI) and in-band signaling messages. Each soft switch site processes these signaling messages for the purpose of establishing new calls through the data network and tearing down existing calls and in-progress call control functions. Signaling messages can be transmitted between any combination of on-network and off-network callers.
Signaling messages for a call which either originates off-network or terminates off-network can be carried over the out-of-band signaling network of the PSTN via the soft switch sites. Signaling messages for a call which both originates on-network and terminates on-network can be carried over the data network rather than through the signaling network.
The gateway sites originate and terminate calls between calling parties and called parties through the data network. The soft switch sites control or manage the gateway sites. In a preferred embodiment, the soft switch sites use a protocol such as, for example, the Internet Protocol Device Control (IPDC) protocol, to manage network access devices in the gateway sites to request the set-up and tear-down of calls. However, other protocols could be used, including, for example, network access server messaging interface (NMI) and the ITU media gateway control protocol (MGCP).
The gateway sites can also include network access devices to provide access to network resources (i.e., the communication channels or circuits that provide the bandwidth of the data network). The network access devices can be referred to generally as access servers or media gateways. Exemplary access servers or media gateways are trunking gateways (TGs), access gateways (AGs) and network access servers (NASs). The gateway sites provide for transmission of both voice and data traffic through the data network. The gateway sites also provide connectivity to other telecommunications carriers via trunk interfaces to carrier facilities for the handling of voice calls. The trunk interfaces can also be used for the termination of dial-up modem data calls. The gateway sites can also provide connectivity via private lines and dedicated access lines (DALs), such as T1 or ISDN PRI facilities, to customer facilities.
The data network connects one or more of the soft switch sites to one or more of the gateway sites. The data network routes data packets through routing devices (e.g., routers) to destination sites (e.g., gateway sites and soft switch sites) on the data network. For example, the data network routes internet protocol (IP) packets for transmission of voice and data traffic from a first gateway site to a second gateway site. The data network represents any art-recognized data network including the global Internet, a private intranet or internet, a frame relay network, and an asynchronous transfer mode (ATM) network.
The network event component collects call events recorded at the soft switch sites. Call event records can be used, for example, for fraud detection and prevention, and billing.
The provisioning event component receives provisioning requests from upstream operational support services (OSS) systems such as, for example, for order-entry, customer service and customer profile changes. The provisioning component distributes provisioning data to appropriate network elements and maintains data synchronization, consistency, and integrity across multiple soft switch sites.
The network management component includes a network operations center (NOC) for centralized network management. Each network element(NE) (e.g., soft switch sites, gateway sites, provisioning, and network event components, etc.) generates simple network management protocol (SNMP) events or alerts. The NOC uses the events generated by each network element to determine the health of the network and to perform other network management functions.
In a preferred embodiment, the invention operates as follows to process, for example, a long distance call (also known as a 1+ call). First, a soft switch site receives an incoming call signaling message from the signaling network. The soft switch site determines the type of call by performing initial digit analysis on the dialed number. Based upon the information in the signaling message, the soft switch site analyzes the initial digit of the dialed number of the call and determines that it is a 1+ call. The soft switch site then queries a customer profile database to retrieve the originating trigger plan associated with the calling customer. The query can be made using, for example, the calling party number provided in the signaling message from the signaling network. This look-up in the customer profile database returns subscription information. For example, the customer profile may indicate that the calling party has subscribed to an account code verification feature that requires entry of an account code before completion of the call. In this case, the soft switch site will instruct the gateway site to collect the account code digits entered by the calling party. Assuming that the gateway site collects the correct number of digits, the soft switch site can use the customer profile to determine how to process the received digits. For account code verification, the soft switch site verifies the validity of the received digits.
Verification can result in the need to enforce a restriction, such as a class of service (COS) restriction (COSR). In this example, the soft switch site can verify that the account code is valid, but that it requires that an intrastate COSR should be enforced. This means that the call is required to be an intrastate call to be valid. The class of service restriction logic can be performed within the soft switch site using, for example, pre-loaded local access and transport areas (LATAs) and state tables. The soft switch would then allow the call to proceed if the class of service requested matches the authorized class of service. For example, if the LATA and state tables show that the LATA of the originating party and the LATA of the terminating party are in the same state, then the call can be allowed to proceed. The soft switch site then completes customer service processing and prepares to terminate the call. At this point, the soft switch site has finished executing all customer service logic and has a 10-digit dialed number that must be terminated. To accomplish the termination, the soft switch site determines the terminating gateway. The dialed number (i.e., the number of the called party dialed by the calling party) is used to select a termination on the data network. This termination may be selected based on various performance, availability or cost criteria. The soft switch site then communicates with a second soft switch site associated with the called party to request that the second soft switch site allocate a terminating circuit or trunk group in a gateway site associated with the called party. One of the two soft switch sites can then indicate to the other the connections that the second soft switch site must make to connect the call. The two soft switch sites then instruct the two gateway sites to make the appropriate connections to set up the call. The soft switch sites send messages to the gateway sites through the data network using, for example, IPDC protocol commands. Alternately, a single soft switch can set up both the origination and termination.
The present invention provides a number of important features and advantages. First, the invention uses application logic to identify and direct incoming data calls straight to a terminating device. This permits data calls to completely bypass the egress end office switch of a LEC. This results in significant cost savings for an entity such as an internet service provider (ISP), ILEC, or CLEC. This decrease in cost results partially from bypass of the egress ILEC end office switch for data traffic.
A further advantage for ISPs is that they are provided data in the digital form used by data networks (e.g., IP data packets), rather than the digital signals conventionally used by switched voice networks (e.g., PPP signals). Consequently, the ISPs need not perform costly modem conversion processes that would otherwise be necessary. The elimination of many telecommunications processes frees up the functions that ISPs, themselves, would have to perform to provide Internet access.
Another advantage of the present invention is that voice traffic can be transmitted transparently over a packet-switched data network to a destination on the PSTN.
Yet another advantage of the invention is that a very large number of modem calls can be passed over a single channel of the data network, including calls carrying media such as voice, bursty data, fax, audio, video, or any other data formats.
Further features and advantages of the invention, as well as the structure and operation of various embodiments of the invention, are described in detail below with reference to the accompanying figures.