Patent Application
20130128880
The present invention relates to telecommunication and a networked computer telephony system including the Internet and the Public Switched Telephone System, and more particularly to user-configurable allocation of telephony resources between a hosted facility and a subscriber's premise.
Two major telecommunication networks have evolved worldwide. The first is a network of telephone systems in the form of the Public Switched Telephone System (PSTN). This network was initially designed to carry voice communication, but later also adapted to transport data. The second is a network of computer systems in the form of the Internet. The Internet has been designed to carry data but also increasingly being used to transport voice and multimedia information. Computers implementing telephony applications have been integrated into both of these telecommunication networks to provide enhanced communication services. For example on the PSTN, computer telephony integration has provided more functions and control to the POTS (Plain Old Telephone Services). On the Internet, computers are themselves terminal equipment for voice communication as well as serving as intelligent routers and controllers for a host of terminal equipment.
The Internet 30 is a worldwide interconnection of IP (Internet Protocol) networks, with interconnecting computers communicating with each other using TCP/IP (Transmission Control Protocol/Internet Protocol). Some of the computers may also be interconnected by a private segment of the IP network with restricted access. On an IP network, data from a source node is cast into a number of packets that may individually be transported via multiple paths on the network to be reassembled at a destination node. The transmission on the IP network is packet-switched and asynchronous.
On an IP network, voice or multimedia information can also be digitized as data and transported over the network using the Internet Protocol (IP). In that case, it is generally referred to as VoIP or (Voice-over-IP). The VoIP protocol includes a number of standards. For example, one such standard is the H.323 standard promulgated by the ITU (International Telecommunication Union) aims to ensure VoIP interoperability. It provides a specification for communication of multimedia such as voice, data and video between terminal equipment over IP networks. The terminal equipment communicating on the Internet includes personal computers with telephony capabilities 40, VoIP phones 42 that can connect to the Internet directly, and other networked telephony appliances.
In recent years, the World Wide Web (WWW) has become a universal platform for information dissemination on the Internet. Web applications 44 in general and web pages in particular are written in HTML (HyperText Markup Language) and are hosted by web servers 46 on the Internet. Each web page can be called up by its URL (Uniform Resource Locator), which can translate to an IP address on the Internet. These web pages may be requested and processed by a web browser running on a computer connected to the Internet. The web browser retrieves the web page under HTTP (HyperText Transfer Protocol) and parses the HTML codes on the web page to execute it. Typically, the execution of HTML codes on a web page results in rendering it into a display page on the browser or client computer. In other instances, it may result in the execution of some backend functions on the client and/or server computers. One reason for the widespread acceptance of WWW is the relative ease web applications can be created and deployed, and the existence of standardized web browsers. HTML, with its tag-coding scheme, is now well known to everyone from the professional developer to the savvy end user. More recently, XML (Extensible Markup Language) has been introduced to extend HTML with enhanced features including customizable tags, which allow for more structural specification of data.
Telephony or Computer Telephony Integration (CTI) involves using a computer to control and manage a phone or a telephone system. When applied to a phone or a terminal equipment, CTI provides added features to an end user's phone. When applied to a telephone system whether as part of the PSTN or part of an IP telephony network system, CTI is usually implemented with a CT (Computer Telephony) server, such as CT server 50. Such a server executes telephony applications that can provide custom services such as interactive voice response, customer service or help desk for an organization. The CT server 50 can be configured to interface via a PSTN interface 52 with an exchange 12 to receive and process calls pertaining to a predefined set of telephone numbers on the PSTN. Similarly, it can also be configured to interface via an IP network interface 54 with the Internet to receive and process calls pertaining to a predefined set of telephone numbers or IP addresses. The CT server 50 is usually a computer operating under UNIX or Microsoft Windows NT and is running installed customized application software 56 for the various voice applications. The CT server provides a set of API 58 (Application Program Interface) which is a set of procedures, protocols and tools for building software applications. These APIs are generally proprietary and specific to the individual hardware manufacturers. Developing an application on an existing CT server would involve a highly specialized application developer undertaking a fairly complex task of coding the application in C++ or JAVA, programming language and employing and invoking the APIs specific to the hardware.
U.S. Pat. No. 6,011,844 discloses a distributed call center system in which a business call center running a custom interactive voice response application is essentially replicated in a number of local points of presence to reduce communication cost when connecting a local customer.
Prior computer telephony systems have infrastructures that do no allow easy development and deployment of telephony applications. The system illustrated in
Even with the deployment of telephony systems in hosted facilities, customers and subscribers have a need to install their own telephony resource on premise for economic, control and security reasons. Issues with capacity and equipment failure can not be handled in an expedient manner with existing infrastructures.
It is therefore a general object of the invention to provide a computer telephony system that allows easy allocation of resources for execution of telephony applications and call processing between premise and hosted facilities.
It is another general object of the invention to provide an infrastructure in which users from premise or hosted facilities can quickly effect user-specified provisioning of telephony resources for controlling and managing telephone calls on the PSTN and the Internet.
It is another object of the invention to provide an infrastructure in which calls are routed to be handled by one of a plurality of resource centers based on user-specifiable routing rules.
It is another object of the invention to provide an infrastructure in which calls are routed to be handled by one of a plurality of resource centers based on criteria including the availability or capacity of the individual resource centers.
It is another object of the invention to allow a customer or subscriber of a telephony service to configure allocation of resources between premise and hosted facilities so that the hosted facilities act as an extension and backup of the telephony service provided at the premise.
It is another object of the invention to allow the resource centers to share call traffic not just between premise and hosted facilities, but also between multiple premise installations.
According to a general aspect of the invention, a network system and method allows sharing of resources between voice application gateway centers (“voice centers”) operated by a hosted facility and by a subscriber or customer on premise. In this way, a customer is able to configure the network system to have some calls processed on premise and some calls processed by the host facility. Similarly, a hosted facility is able to configure the network system to have some calls originally to be processed by the hosted facility to be processed by an available premise voice center. The invention is accomplished by a semi-real-time manipulation of a voice-center directory by which a destination voice center is selected to process a call.
According to one aspect of the invention, there is provided at least a provisioning management server with a user interface for a user to easily update lookup information in the voice-center directory. The lookup information enables a voice center to be selected as a function of the dialed number as well as routing rules. The user, either an operator from the hosted facility or a customer with premise equipment, is able to update the routing rules in the voice-center directory.
According to another aspect of the invention, the voice-center selection function of the voice-center directory also depends on the degree of availability or current capacity of individual voice centers. This is accomplished by having the individual voice centers updating the voice-center directory regarding their current capacity at predetermined time intervals.
According to another aspect of the invention, the network of telephony system includes a voice center receiving a routed call and retrieving a voice application associated with the dialed number and appropriate for that voice center. As the call can potentially be routed to any one of a number of voice centers, preferably there is a version of the voice application appropriate for each of the number of voice centers. In the preferred embodiment a local network address of the appropriate version of the voice application is cached in each voice center so that the voice center can use it to retrieve the voice application without having to perform a look up in an external directory.
By using a unified voice service providing platform for both premise and hosted deployments and the provision of near real-time user-configurable routing, premise customers can commit to install on premise a predefined amount of resources without over budgeting and yet be able to overflow and failover to hosted resource for unplanned call volume growth, peak call seasons, and disaster recovery plans.
According to another aspect of the invention, the hybrid resource allocation system described also offers the following business advantages. It allows a service provider to sell premise-based telephony software and hosted telephony services as a combined product/service, such that the customer's premise telephony software ports integrate seamlessly with the service provider's hosted telephony software ports. The customer will be allowed to use an online management console to configure and manage his premise-based software, making it possible for him to set parameters for how many calls he wants the premise system (as opposed to the hosted system) to take, under what circumstances to route calls through the hosted or premise system.
Additional objects, features and advantages of the present invention will be understood from the following description of its preferred embodiments, which description should be taken in conjunction with the accompanying drawings.
As mentioned in an earlier section, the Internet is a worldwide network of IP networks communicating under TCP/IP. Specifically, voice and other multimedia information are transported on the Internet under the VoIP (Voice-over-IP) protocol, and under the H.323 standard that has been put forward for interoperability. Another implementation of VOIP protocol that is being coming into currency is SIP (“Session Initiation Protocol”.)
On the other hand, the PSTN 10 is a network of exchanges. Each exchange is provisioned with a plurality of telephone lines or nodes having designated call numbers. Two PSTN nodes are connectable by switching the intervening exchanges to form a circuit.
The PSTN and the Internet are interconnected by means of access servers such as an access server 14. This enables communication between a PSTN node and an Internet node. A telephonic call transported between two network nodes comprises a signaling portion for setting up and tearing down the call and a media portion for carrying the voice or multimedia data. The access server 14 essentially converts both of these portions to an appropriate format across the interface between the two types of networks. On the PSTN side the digital interface is PRI and on the Internet side the interface is VoIP. A wireless or mobile telephone network (not shown) may similarly be considered as an extension of the PSTN. It is typically connected to the PSTN via a suitable interface implemented by a gateway.
The set of designated call numbers handled by the vAGC 100 are registered in a directory, such as DIR0. When a call to one of the designated call numbers is made from the PSTN, it is switched to the access server 12 and a lookup of the directory DIR0 allows the call to be routed to vAGC 100 for processing. Similarly, if the call originates from one of the terminal equipment on the Internet, a directory lookup of DIR0 provides the pointer for routing the call to the vAGC 100.
The plurality of telephony applications vAPP 110, . . . , 112, each associated with at least one designated call number, is accessible by the vAGC from the Internet. Each application is coded in vXML and is being hosted as a webpage on a web server on the Internet. A directory DIR1 provides the network address of the various applications. When the vAGC 100 received a call, it uses the call number (or dialed number DN) to look up DIR1 for the location/address (whether a URL or an IP address or some other location method) of the vAPP associated with the DN. The vAGC 100 retrieves the vXML webpage and executes the call according to the vXML scripts.
A similar networked computer telephony system is disclosed in U.S. Pat. No. 6,922,411, the entire disclosure is incorporated herein by reference.
Step 130: For a given call number DN, create an associated telephony application, vAPP in vXML, and deploy it on the Internet with a specific IP address or URL.
Step 132: Provide any media, files and web applications that are requested or act on by vAPP.
Step 134: Update the directory DIR1 so that the address of vAPP can be obtained by querying with its associated call number DN.
Call processing by vAGC 100 is described in steps 140, 142, 144 and 146.
Step 140: vAGC receives a call with DN routed thereto.
Step 142: vAGC uses DN to look up DIR1 for the address of the webpage for vAPP.
Step 144: vAGC requests and retrieves the webpage containing vXML scripts for vAPP.
Step 146: vAGC processes the call according to the retrieved vXML scripts for vAPP.
Thus, the present system allows very power yet simple telephony applications to be built and deployed on the Internet. The following are some examples of the vAPP telephony applications contemplated. A “Follow me, find me” application sequentially calls a series of telephone numbers as specified by a user until one of the numbers answers and then connects the call. Otherwise, it does something else such as takes a message or sends e-mail or sends the call to a voice center, etc. In another example, a Telephonic Polling application looks up from a database the telephone numbers of a population to be polled. It then calls the numbers in parallel, limited only by the maximum number of concurrent sessions supported, and plays a series of interactive voice prompts/messages in response to the called party's responses and records the result in a database, etc. In another example, a Help Desk application plays a series of interactive voice prompts/messages in response to the called party's responses and possibly connects the call to a live agent as one option, etc. In yet another example, a Stock or Bank Transactions application plays a series of interactive voice prompts/messages in response to the called party's responses and conducts appropriate transactions with a backend database or web application, etc.
The application gateway server 200 exchanges data with the Internet indirectly through the cache server 310 and possibly the media conversion proxy server 320. As will be described in more detail later, upon receiving a call, the AGS 200 retrieves the associated vAPP from a website and proceeds to execute the vXML scripts of the vAPP. During the course of executing the vXML scripts, associated media and/or files may also be retrieved from various sites as part of the vAPP suite.
In the preferred embodiment, in order to increase performance, the vXML scripts, media and files that are retrieved into the vAGC are cached by the cache server 310. They are requested by the AGS through the cache server 310. If a cached copy of the requested data exists in the cache server, it is delivered directly to the AGS. If not, the cache server retrieves the data, caches it and delivers the data to the AGS to fulfill the request.
In the preferred embodiment, in order to simplify the design of the AGS and to improve the performance and scalability of it, the AGS is designed to handle only one native media format. For example, one suitable format for audio is G.711 or GSM. Media that come in different format are handed over to the media conversion proxy server 320, which coverts the media to the native format of the AGS 200.
In the preferred embodiment, the AGS 200 is a set software modules running on a Windows NT or UNIX server. For example, the AGS is implemented as a Windows NT machine on a card, and multiple cards are installed on a caged backplane to form a high scalable system.
The AGS 200 comprises four main software modules, a session manager 210, an I/O abstraction layer 220, a computer telephony (CT) abstraction layer 230, and a telephony scripting language parser 240. The telephony scripting language parser 240 further comprises a telephony XML or vXML parser 242 and a generic XML parser 244. In addition, a streaming interface 250 provides a direct streaming path for media data between the I/O abstraction layer 220 and the CT abstraction layer. Each of these modules is designed to be a separate DLL (Dynamically Linked Library) and perform a specific task. In the preferred embodiment, the AGS is a console only application with no user interface for any of these modules. Several of these modules incorporate commercial, third party software components in performing their tasks. These components will be discussed along with the appropriate modules.
The session manager 210 is the centerpiece of the AGS 200. It is responsible for creating new sessions, deleting terminated sessions, routing all actions and events to the appropriate modules and maintaining modularity between each session. It responds to I/O and vXML goto requests, and other additional events. In one embodiment, it employs commercially available software libraries containing thread and string classes from PWLib, a product of Equivalence Pty Ltd, Erina, New South Wales, Australia.
The session manager interfaces to the external of the AGS via the I/O abstraction layer 220 and the CT abstraction layer 230. It accesses the I/O and CT layers as a set of classes and member functions that are individual DLLs. The Session Manager 210 runs as a single-threaded processor of actions and event.
A session begins with the reception of an asynchronous event from the CT abstraction module 230 signaling an incoming call. The Session Manager then creates a session for this call by accessing a database (e.g. DIR1 of
Each session is assigned a unique session identification, SID (session ID). For example, in the Microsoft Win32 platform, the SID is conveniently implemented by the creation of 128 bit globally unique Ids (GUIDs).
In the preferred embodiment, the session manager 210 is accessed or invoked via a number of interface points of its DLL as described in TABLE 1.
The I/O abstraction layer 220 performs all input and output operations for the AGS 200. Essentially, it renders transparent to the internal of the AGS the variety of I/O formats and protocols that might be encounter externally. To the session manager 210, most HTTP, FTP, File, and memory-mapped I/O requests are reduced to four commands: open, close, read, and write. This allows access to a stream from any of these sources with the same procedure calls once the stream is open. In one embodiment, it incorporates available commercial software libraries, such as Winlnet from Microsoft Corporation, Seattle, Wash., U.S.A and PWLib from Equivalence Pty Ltd. Winlnet is a windows-specific DLL that allows the I/O abstraction layer to communicate to outside sources using HTTP and FTP. PWLib also used by the session manager 210 contains strings and threads classes.
In the preferred embodiment, the I/O abstraction layer 220 is accessed or invoked via a number of interface points of its DLL as described in TABLE 2. A single thread per active stream is created by instantiating a VXEIOStream when accessed by the session manager 210. If the stream is FTP or HTTP-based, then the user will need to provide the appropriate login data, submission method, and CGI variables. Next, the user calls the Open method and then uses the Read and Write methods to operate upon the stream until closing it with the Close method. At this point, this instance of the VXEIOStream is available for use on another stream source or it can be deleted.
The computer telephony (CT) abstraction layer 230 is a thin abstraction layer that makes it possible for the AGS 200 to communicate with several computer telephony devices and/or protocols. In one direction, the CT abstraction layer receives requests for computer telephony actions from the session manager 210 and translates those requests to a CT module. In the other direction the CT abstraction layer receives user events directed to that CT module and relates them back to the session manager. In the preferred embodiment, the CT modules include a H.232 stack for handling VoIP signals, a SIP (Session Interface Protocol), a MGCP (Media Gateway Control Protocol) as well as other CT modules such as Dialogic CT modules. Since several CT modules can be placed below the CT abstraction layer and the CT abstraction will talk to all of the CT modules, the modular design allows the AGS to communicate with a new computer telephony device or protocol simply with the addition of a new CT module.
The CT abstraction layer 230 will preferably make use of PWLib's platform-independent thread class. The CT Abstraction layer is instantiated by the Session Manager 210. It then seeks out a vXML configuration file that contains information on the number and type of telephony boards in its system. The member functions represent generic functionality that should be supportable across a wide variety of telephony hardware. The motivation for this abstraction layer is to make the AGS 200 both platform and protocol independent.
In the preferred embodiment, the Session Manager 210, XML Parser 240, and CT Abstraction layer 230 cooperate via the following protocol. First, the telephony scripting language parser 240 locates a vXML element which requires a telephony task. Next, the telephony scripting language parser sends this task to the Session Manager in a microXML action string. The Session Manager then parses the microXML action string and determines the appropriate call to the CT abstraction layer along with its associated parameters. The Session Manager now calls the CT abstraction layer asynchronously and the CT abstraction layer returns an event signaling the completion of the CT task and the Session Manager resumes parsing.
In the preferred embodiment, the CT abstraction layer 230 is accessed or invoked via a number of interface points of its DLL as described in TABLE 3.
The streaming interface 222 provides a direct streaming transfer between the I/O abstraction layer 220 and the CT abstraction layer 230 when media data, such as audio or other multimedia is involved. For example, the streaming interface facilitates the AGS to play audio from URL's and to record audio to URL's in a streaming manner. In the preferred embodiment, the interface is generic and passes the burden of buffer management to the CT module in use. This allows specific CT modules to buffer information as appropriate for the corresponding telephony hardware or protocol. The streaming interface is implemented through the readAsynchronous and writeAsynchronous interface points in the I/O abstraction layer.
The telephony scripting language parser 240 is responsible for parsing the vXML scripts handed to it by the session manger 210. It in turn informs the session manager of the described actions coded in the vXML scripts. The telephony scripting language parser is modular and can accommodate additional parsers such as that for voiceXML and parsers for other telephony scripting language that may arise. In the present preferred embodiment, it comprises the vXML parser 242 and the generic XML parser 244.
The generic XML parser 244 parses the vXML scripts, which are essentially XML scripts with embedded custom telephony tags, and puts them in a format that the vXML parser 242 can expediently act on. In the preferred embodiment, the generic XML parser 244 conveniently employs CueXML components available from CueSoft, Inc, Brighton, Colo., U.S.A. These components enable parsing of vXML documents into an object model, DOM (Document Object Model) listing the parsed objects in a hierarchical tree structure. This allows the vXML parser 242, which in the preferred embodiment is a DLL written in Delphi 5.0, to “walk” through the tree of objects and interpret them into microXML codes that can be understood by the session manager 210.
The vXML parser 242 behaves as follows: when called it will examine the incoming microXML and determine if there is a buffer of new vXML to parse, if such a buffer exists then the parser uses the generic XML parser 244 to construct a new object model for this buffer, the session object model is set to that model and the session state is cleared. The vXML parser 242 begins parsing from the session state in the session object model (an empty state implies the beginning of a document). As the parse traverses the document model the state is updated and events are generated. If these events are internal to the processor they are handled (i.e. assigns update the session variables, blocks may cause looping to occur), if the events are not internal then they are buffered for return to the session manager. When an event needs to be reported to the session manager the event buffer is processed so that variables are replaced with their values, wildcards are properly expanded, etc. This negates the need for any other module to maintain information about session variables.
The vXML parser 242 is required to maintain state per session so that each invocation of the vXML parser will continue where the previous invocation within the same session ended. The maintenance of state includes preserving the DOM for the current instance of vXML, the node in the DOM that the parser is currently examining, and any variables that are associated with the session.
In the preferred embodiment, the vXML parser 242 is accessed or invoked via a number of interface points of its DLL as described in TABLE 4.
As mentioned earlier, microXML is a subset of simple vXML used for communication between the session manager 210 and the telephony scripting language parser 240. MicroXML is the native codes of the virtual machine of the session manager 210. In one direction, the vXML parser 242 communicates with the session manger 210 in a synchronous manner using microXML. In another other direction, user events may also be reported to the vXML parser via microXML. If a user event is reported the parser will find the appropriate event handler by first looking locally for a valid handler. If a handler is not found there then the parent node in the document model is examined for a valid handler. The search continues in this manner until either a handler is found or there is no parent to examine. If a handler is found then the parser sets the state to the handler and begins parsing as described above. If a handler is not found then an error is returned via microXML.
In the preferred embodiment, MicroXML is composed of a limited number of tags, these tags do not have any attributes, and CDATA sections are not supported. Table 5 shows examples of microXML tags:
vXML is XML with additional custom tags for telephony applications. TABLE 6A-6D lists example tags useful for creating telephony applications. A user or developer need only code his or her telephony application in these vXML tags and deploy the resulting scripts as a webpage on the Internet for the vAGS 200 to access.
The following are examples of microXML communication between the session manager 210 and the vXML parser 242.
The following is an example of a vXML file:
The example vXML file results in the following corresponding microXML being generated by the vXML parser and sent to the session manager:
Each vAGC site is provided with a traffic monitor 400 that periodically pings the pluarality of vAPP sites and detects the return signals. The response time of each vAPP site to any given vAGC is collected by a network monitoring server 410. Since each vAPP is associated with a dialed number (DN), the network monitoring server computes a listing of DNs versus vAGCs sorted in order of fastest response time. This information is used to update the DIR0 directory (see
A plurality of voice application gateway centers (“vAGC”) 100 (also referred to as “voice centers”) is deployed on the Internet. As described in connection with
As described in connection with
In the preferred embodiment, the access server 14 routes the call to a vAGC routing proxy 580 which then takes over the job of the access server and routes the call to a destination vAGC after a DIR 0 lookup. As different LECs may set up access servers with varying amount of features and capabilities, it preferable for voice centers to rely on a set of vACG routing proxy 580 with guaranteed specification and capabilities.
According to a general aspect of the invention, a network system and method allows sharing of resources between voice application gateway centers (“voice centers”) operated by a hosted facility and by a subscriber or customer on premise. In this way, a customer is able to configure the network system to have some calls processed on premise and some calls processed by the host facility. Similarly, a hosted facility is able to configure the network system to have some calls originally to be processed by the hosted facility to be processed by an available premise voice center. The invention is accomplished by a semi-real-time manipulation of a voice-center directory by which a destination voice center is selected to process a call.
According to one aspect of the invention, there is provided at least a provisioning management server with a user interface for a user to easily update lookup information in the voice-center directory. The lookup information enables a voice center to be selected as a function of the dialed number as well as routing rules. The user, either an operator from the hosted facility or a customer with premise equipment, is able to update the routing rules in the voice-center directory.
In operation, the manager 610 manages user access and responds to the event monitor 620 to initiate various operations. Through the user interface a user can add modify or delete data in the replica 642 and 646. In particular, the data pertaining to provisioning and routing rules may be selected from the predefined set 644. The directory replicator replicates relevant portion of the data between the replica 642 and DIR 0550 as well as between the replica 646 and DIR 1570 so that changes made to the replica in the provisioning management server are reflected in DIR 0550 and DIR 1570.
Through the provisioning management server 600, a user including a subscriber can quickly make changes to DIR 0550 to implement close to real-time re-allocation of resources and services between premise and hosted facilities.
The voice-center lookup directory DIR 0550 essentially functions as a directory for looking up call-processing resources analogous to a domain name server (“DNS”) for locating the IP address of an Internet resource given an URL address. In the preferred embodiment, DIR 0 is an ENUM directory. ENUM is the name of a protocol that resolves fully qualified telephone numbers to fully qualified domain name addresses using a DNS-based architecture. It is the name of a chartered working group of the Internet Engineering Task Force (IETF) chartered to develop protocols that map telephone numbers to resources found on the Internet using the Domain Name System.
According to another aspect of the invention, the voice-center selection function of the voice-center directory also depends on the degree of availability or current capacity of individual voice centers. This is accomplished by having the individual voice centers updating the voice-center directory regarding their current capacity at predetermined time intervals.
Referring to
According to another aspect of the invention, the network of telephony system includes a voice center receiving a routed call and retrieving a voice application associated with the dialed number and appropriate for that voice center. As the call can potentially be routed to any one of a number of voice centers, preferably there is a version of the voice application appropriate for each of the number of voice centers. In the preferred embodiment a local network address of the appropriate version of the voice application is cache in each voice center so that the voice center can use it to retrieve the voice application without having to perform a look up in an external directory.
In a preferred embodiment, the premise vAGC P1 maintains a local replica of the premise version 574″ of DIR 1 data by a replication agent 552. When the DIR 1 data or even subsets of Dir1 such as customer-specific subsets is maintained on all premise nodes, then a premise-based vAGC can operate even when the link to the hosted DIR 1 is severed.
STEP 710: In a networked computer telephony system operating under the Internet Protocol, providing a first set of application gateway centers in a hosted facility and a predetermined second set of application gateway centers at a subscriber's premise for receiving and processing telephone calls.
STEP 720: Providing a directory for routing a telephone call identified by a telephone number to an application gateway center from either the first set or the second set, the directory designating an application gateway center as a function of the telephone number, predefined routing rules and operating capacities of individual application gateway centers.
STEP 730: Updating the directory at predefined time intervals on the operating capacities of the individual application gateway centers.
STEP 740: Providing a user interface for a user to update user-specified routing rules in the directory.
STEP 742-759, any one of them is optional (see
STEP 760: Routing the telephone call according to the directory.
STEP 742: Wherein the user is a subscriber, and routing rules configurable by the subscriber include telephone calls to be preferentially processed by an application gateway center on premise from the second set and to be processed by a hosted application gateway center from the first set in the event of the operating capacity of the second set fall below a predetermined threshold.
STEP 744: Wherein the user is a host operator, and routing rules configurable by the host operator include telephone calls to be processed by an application gateway center from the second set in the event of the operating capacity of the first set falls below a predetermined threshold.
STEP 746: Wherein the user is a subscriber, and routing rules configurable by the subscriber include telephone calls each having one of a predefined dialed numbers must be processed by an application gateway center on premise from the second set for security reasons.
STEP 748: Wherein the user is a subscriber, and routing rules configurable by the subscriber include telephone calls to be preferentially processed by a hosted application gateway center from the first set when the telephone calls are associated with a first predefined set of telephone numbers.
STEP 750: Wherein routing rules configurable by the user include routing a predetermined percentage of call traffic to application gateway centers on premise from the second set relative those from the first set.
STEP 752: Wherein routing rules configurable by the user include basing on geographical location of the call to be processed.
STEP 754: Wherein routing rules configurable by the user include basing on time schedule.
STEP 756: Wherein routing rules configurable by the user include basing on the carrier the call originates from.
STEP 758: Wherein routing rules configurable by the user include avoidance of predefined network routes.
STEP 759: Wherein the user is a subscriber, and routing rules configurable by the subscriber include routing from one premise to another.
According to another aspect of the invention, the hybrid resource allocation system described also offers the following business advantages. It allows a service provider to sell premise-based telephony software and hosted telephony services as a combined product/service, such that the customer's premise telephony software ports integrate seamlessly with the service provider's hosted telephony software ports. The customer will be allowed to use an online management console to configure and manage his premise-based software, making it possible for him to set parameters for how many calls he wants the premise system (as opposed to the hosted system) to take, under what circumstances to route calls through the hosted or premise system.
Conventional systems requires a customer buying telephony software to be forced to predict how many ports (concurrent calls) he will need and then to hope that he guesses correctly. If he guesses too many, than he has built out too much infrastructure too fast and is thus wasting money on premature capital expenditures. If he guesses too few, then his installation is too small for his traffic and his users get busy signals or catastrophic failures.
With the hybrid scheme of providing telephony software, the customer does not have to guess what his traffic will be. He can buy 100 ports of software, and simply use the hosted network to take all calls after the first 100 concurrent calls. Or he can buy 20 ports and take only the calls about a certain subject on premise, with the rest going through the hosted service. Or he can take calls on-premise during business hours, and run them through hosted at night. Whatever the customer wants to do, he now has the flexibility of the premise and hybrid models.
Very few businesses sell both premise telephony software and a hosted telephony service. And none of them offers the kind of integrated service that is contemplated here. At most, the prior art entails simple fail-over to a hosted solution. The current hybrid scheme entails seamlessly integrating the premise installation so that there is virtually no logical distinction between the premise and hosted installations.
While the embodiments of this invention that have been described are the preferred implementations, those skilled in the art will understand that variations thereof may also be possible. Therefore, the invention is entitled to protection within the full scope of the appended claims.
This application is a continuation of U.S. application Ser. No. 12/195,298, filed on Aug. 20, 2008, now U.S. Pat. No. 8,355,394, which claims the benefit of U.S. provisional patent application No. 60/957,151, filed Aug. 21, 2007.
| Number | Date | Country | |
|---|---|---|---|
| 60957151 | Aug 2007 | US |
| Number | Date | Country | |
|---|---|---|---|
| Parent | 12195298 | Aug 2008 | US |
| Child | 13739983 | US |
