EP1181805A1 - Verfahren und vorrichtung für integriertes sprach-gateway mit einer schnittstelle zum mobiltelefon, ip-telefon und un-pbx-systemen - Google Patents

Verfahren und vorrichtung für integriertes sprach-gateway mit einer schnittstelle zum mobiltelefon, ip-telefon und un-pbx-systemen

Info

Publication number
EP1181805A1
EP1181805A1 EP00930722A EP00930722A EP1181805A1 EP 1181805 A1 EP1181805 A1 EP 1181805A1 EP 00930722 A EP00930722 A EP 00930722A EP 00930722 A EP00930722 A EP 00930722A EP 1181805 A1 EP1181805 A1 EP 1181805A1
Authority
EP
European Patent Office
Prior art keywords
network
telephone
wireless
call
gateway
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Withdrawn
Application number
EP00930722A
Other languages
English (en)
French (fr)
Inventor
Judith Duffy
Stephen R. Raad
Gordon K. Chang
Richard B. Barry
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Starvox Inc
Original Assignee
Starvox Inc
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Starvox Inc filed Critical Starvox Inc
Publication of EP1181805A1 publication Critical patent/EP1181805A1/de
Withdrawn legal-status Critical Current

Links

Classifications

    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/10Architectures or entities
    • H04L65/102Gateways
    • H04L65/1043Gateway controllers, e.g. media gateway control protocol [MGCP] controllers
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066Session management
    • H04L65/1069Session establishment or de-establishment
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M7/00Arrangements for interconnection between switching centres
    • H04M7/0024Services and arrangements where telephone services are combined with data services
    • H04M7/0057Services where the data services network provides a telephone service in addition or as an alternative, e.g. for backup purposes, to the telephone service provided by the telephone services network
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M7/00Arrangements for interconnection between switching centres
    • H04M7/12Arrangements for interconnection between switching centres for working between exchanges having different types of switching equipment, e.g. power-driven and step by step or decimal and non-decimal
    • H04M7/1205Arrangements for interconnection between switching centres for working between exchanges having different types of switching equipment, e.g. power-driven and step by step or decimal and non-decimal where the types of switching equipement comprises PSTN/ISDN equipment and switching equipment of networks other than PSTN/ISDN, e.g. Internet Protocol networks
    • H04M7/1285Details of finding and selecting a gateway for a particular call
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M2207/00Type of exchange or network, i.e. telephonic medium, in which the telephonic communication takes place
    • H04M2207/20Type of exchange or network, i.e. telephonic medium, in which the telephonic communication takes place hybrid systems
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M2242/00Special services or facilities
    • H04M2242/22Automatic class or number identification arrangements
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M3/00Automatic or semi-automatic exchanges
    • H04M3/22Arrangements for supervision, monitoring or testing
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M3/00Automatic or semi-automatic exchanges
    • H04M3/22Arrangements for supervision, monitoring or testing
    • H04M3/2254Arrangements for supervision, monitoring or testing in networks
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M3/00Automatic or semi-automatic exchanges
    • H04M3/42Systems providing special services or facilities to subscribers
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M3/00Automatic or semi-automatic exchanges
    • H04M3/42Systems providing special services or facilities to subscribers
    • H04M3/42025Calling or Called party identification service
    • H04M3/42034Calling party identification service
    • H04M3/42042Notifying the called party of information on the calling party
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M3/00Automatic or semi-automatic exchanges
    • H04M3/42Systems providing special services or facilities to subscribers
    • H04M3/42025Calling or Called party identification service
    • H04M3/42085Called party identification service
    • H04M3/42102Making use of the called party identifier
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M3/00Automatic or semi-automatic exchanges
    • H04M3/42Systems providing special services or facilities to subscribers
    • H04M3/42314Systems providing special services or facilities to subscribers in private branch exchanges
    • H04M3/42323PBX's with CTI arrangements
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M3/00Automatic or semi-automatic exchanges
    • H04M3/42Systems providing special services or facilities to subscribers
    • H04M3/42314Systems providing special services or facilities to subscribers in private branch exchanges
    • H04M3/4234Remote access to features of PBX or home telephone systems-teleworking in a PBX
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M3/00Automatic or semi-automatic exchanges
    • H04M3/42Systems providing special services or facilities to subscribers
    • H04M3/428Arrangements for placing incoming calls on hold
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M3/00Automatic or semi-automatic exchanges
    • H04M3/42Systems providing special services or facilities to subscribers
    • H04M3/487Arrangements for providing information services, e.g. recorded voice services or time announcements
    • H04M3/493Interactive information services, e.g. directory enquiries ; Arrangements therefor, e.g. interactive voice response [IVR] systems or voice portals
    • H04M3/4931Directory assistance systems
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M3/00Automatic or semi-automatic exchanges
    • H04M3/42Systems providing special services or facilities to subscribers
    • H04M3/54Arrangements for diverting calls for one subscriber to another predetermined subscriber
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M3/00Automatic or semi-automatic exchanges
    • H04M3/42Systems providing special services or facilities to subscribers
    • H04M3/56Arrangements for connecting several subscribers to a common circuit, i.e. affording conference facilities
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04QSELECTING
    • H04Q3/00Selecting arrangements
    • H04Q3/72Finding out and indicating number of calling subscriber

Definitions

  • This application relates generally to telephony systems, and more particularly to an integrated voice gateway system with a wireless capability and a method of operating the same.
  • PBX private branch exchange
  • IP Internet protocol
  • company e.g., intranet, wide area network (WAN), Internet
  • PSTN public switched telephone network
  • PSTN public switched telephone network
  • current voice gateway systems are capable of only routing a nominal telephone call from a calling party at point A to a called party at point B.
  • current voice gateway systems do not provide important additional services to facilitate making a connection between the calling party and the called party at a later time or at another location or by an alternative method.
  • current voice gateway networks have no information regarding the calling party's name, telephone number, or status of the called party, e.g., busy or idle. It is this information about the calling and called parties which is not readily available, but which is necessary to provide important additional services.
  • the integrated voice gateway system should have the ability to route telephone calls between parties at two different locations over the IP network, the PSTN, a private-wireless-network and a public-wireless-network, and the capability to automatically select over which to route telephone calls.
  • the present invention overcomes the disadvantages of the prior art by providing a communication network having an integrated voice gateway system incorporating a private- wireless-network as a component of the system and method for operating the same that provides users of mobile phones and un-PBX systems access to features of the gateway network as described in U.S. Pat. App. Serial No. 09/061,802, which is incorporated herein by reference.
  • the present invention is directed to a communication system for providing communication between a plurality of sites within an enterprise.
  • the communication system comprises a public switched telephone network (PSTN), an internet protocol (IP) network, a public-wireless-network and a plurality of gateway networks coupled to the PSTN, IP network and the public-wireless-network to route a telephone call between a calling and a called party thereover.
  • PSTN public switched telephone network
  • IP internet protocol
  • Each of the plurality of gateway networks is configured to automatically select over which of the IP network, PSTN or the public-wireless-network to route the telephone call.
  • the plurality of gateway networks are configured to automatically reroute an in-progress telephone call routed over the IP network over the PSTN if a delay in transmission of data packets, losses in transmission of data packets, or jitter exceeds a specified maximum. More preferably, the gateway networks are configured so that the rerouting of the in-progress telephone call is substantially transparent to the calling party and to the called party.
  • At least one of the plurality of gateway networks comprises a private-wireless-network
  • the gateway network is configured to route a telephone call between the calling and the called party over the private- wireless-network and at least one of the IP network, the PSTN or the public-wireless-network.
  • one of the gateway networks comprises an UN-PBX system, and wherein the gateway network is configured to route a telephone call between the calling and the called party over the UN-PBX system.
  • the gateway networks when a called party's telephone is unavailable, or the called party does not answer, are configured to automatically set up a telephone call between the calling party and the called party as soon as the called party becomes available.
  • the gateway networks are configured to enable the calling party to forward the telephone call.
  • the gateway networks are configured to create a log of incoming telephone calls, call attempts and outgoing telephone calls.
  • the gateway networks may also be configured to enable a user of the system to forward telephone calls to a different telephone according to a time schedule predetermined by the user.
  • the gateway networks are configured to enable the calling party to send a computer message that will be immediately displayed on a computer screen co-located with the called party's telephone.
  • At least one of the plurality of gateway networks is coupled to a directory server comprising information on parties using the communication system, and wherein the gateway network is configured to (i) provide a single-point-of- entry for modifications to the information and (ii) to provide replication of these changes across all enterprise sites.
  • the gateway networks are further configured to provide information identifying the calling party. The identifying information can be displayed on a display of a called party's telephone or on a computer screen co-located with the called party's telephone.
  • the present invention provides a method of operating a communication system to connect a telephone call between a calling party on a first telephone and a called party on a second telephone.
  • aplurality of gateway networks are coupled to one another via a public switched telephone network (PSTN), an internet protocol (IP) network and a wireless-network.
  • PSTN public switched telephone network
  • IP internet protocol
  • the communication system automatically selects over which of the IP network, PSTN or the wireless-network to setup the telephone call, and routs the telephone call over at least one of the IP network, PSTN or the wireless-network.
  • At least one of the plurality of gateway networks comprises a private-wireless-network
  • the step of routing the telephone call between the calling and the called party comprises the step of routing the telephone call over the private- wireless-network and at least one of the IP network, the PSTN or the wireless-network.
  • at least one of the first and second telephones is a mobile phone coupled to the private-wireless-network
  • the step of automatically selecting over which of the IP network, PSTN or the wireless-network to setup the telephone call includes the step of determining whether the mobile phone is turned on.
  • the wireless-network is a public-wireless-network and at least one of the first and second telephones is a mobile phone coupled to the public- wireless-network, and the step of automatically selecting over which of the IP network, PSTN or the wireless-network to setup the telephone call includes the step of learning whether the mobile phone in public-wireless-network is turned on.
  • the method comprises the further steps of automatically rerouting an in-progress telephone call routed over a first call path over the public-wireless-network to a second call path over the private-wireless-network when the mobile phone needs to roam off the public-wireless-network to the private-wireless- network.
  • WAN wide area network
  • the integrated voice gateway system is able to do so in a manner that is transparent to both the calling and called parties, either or both of which can be using a mobile phone.
  • Telephone- users include users of PBX telephones and mobile phones, and modifications to the user- profile may include changes to the existing data for a user, addition of new users or deletion of users.
  • the identification of the calling party e.g., name, title, department, primary telephone number
  • the identification of the calling party e.g., name, title, department, primary telephone number
  • PC personal computer
  • IVR interactive voice response
  • FIG. 1 is a block diagram of a communication network having gateway networks according to an embodiment of the present invention
  • FIG. 2 is a block diagram of a gateway network according to an embodiment of the present invention.
  • FIG. 3 is a block diagram of a gateway server shown FIG. 2 configured to operate with a PBX according to an embodiment of the present invention
  • FIG. 4 is a block diagram of a gateway server shown FIG.2 configured to operate with an Un-PBX according to an embodiment of the present invention
  • FIG. 5 is a block diagram of a private-wireless-network shown FIG. 2 according to an embodiment of the present invention
  • FIG. 6 is a block diagram of an IP telephone network shown FIG. 2 according to an embodiment of the present invention
  • FIG. 7 illustrates operation of a gateway network according to an embodiment of the present invention to provide a connection between a mobile phone operating in the private-wireless-network and a PSTN telephone at a nearby location;
  • FIG. 8 illustrates an embodiment of a obile-to-PSTN call setup sequence for the gateway network of FIG. 7
  • FIG. 9 illustrates an embodiment of a PSTN- to-mobile call setup sequence for the gateway network of FIG. 7;
  • FIG. 10 illustrates an embodiment of a mobile-to-PBX call in progress for a telephone call routed within a single gateway network
  • FIG. 11 illustrates an embodiment of a mobile-to-PBX call setup sequence for the telephone call of FIG. 10;
  • FIG. 12 illustrates an embodiment of a mobile-to-mobile call in progress for a telephone call routed within a single gateway network
  • FIG. 13 illustrates an embodiment of a mobile-to-mobile call setup sequence for the telephone call of FIG. 12;
  • FIG. 14 illustrates operation of a gateway network according to an embodiment of the present invention to provide a mobile-to-PSTN telephone call, WAN VoIP and Hop-off to PSTN;
  • FIG. 15 illustrates operation of a gateway network according to an embodiment of the present invention to provide WAN VoIP for an in-progress mobile-to-PBX telephone call originating from a mobile phone;
  • FIG. 16 illustrates an embodiment of a mobile-to-PBX call setup sequence for the telephone call of FIG. 15;
  • FIG. 17 illustrates operation of a communication network according to an embodiment of the present invention to provide an in-progress mobile-to-mobile phone call for mobile phones coupled to different gateway networks;
  • FIG. 18 illustrates operation of a communication network according to an embodiment of the present invention to provide a telephone call for a mobile phone in private-wireless-network
  • FIG. 19 illustrates operation of a communication network according to an embodiment of the present invention to provide a telephone call for a mobile phone in public-wireless-network;
  • FIG. 20 illustrates operation of a communication network according to an embodiment of the present invention to provide a telephone call between a mobile phone in a public-wireless-network and a mobile phone homed at a remote site;
  • FIG. 21 illustrates operation of a communication network according to an embodiment of the present invention to provide an in-progress mobile-to-PBX telephone call for a mobile phone in a public-wireless-network calling to local PBX telephone;
  • FIG.22 illustrates an embodiment of a mobile-to-PBX call setup sequence for the telephone call of FIG. 21;
  • FIG. 23 illustrates operation of a communication network according to an embodiment of the present invention to provide an in-progress PBX-to- mobile phone call for a PBX telephone calling to a mobile phone in a public-wireless-network;
  • FIG.24 illustrates an embodiment of a PBX-to-mobile call setup sequence for the telephone call of FIG. 23, wherein the mobile phone is homed locally;
  • FIG.25 illustrates an embodiment of a PBX-to-mobile call setup sequence for the telephone call of FIG. 23, wherein the mobile phone is homed remotely;
  • FIG. 26 illustrates operation of a gateway network according to an embodiment of the present invention to provide a connection between a PBX telephone and a mobile phone operating in a public-wireless-network at a remote site;
  • FIG. 27 illustrates operation of a gateway network according to an embodiment of the present invention to provide a fallback-to-PSTN for an in-progress mobile-to-PBX telephone call originating from a mobile phone;
  • FIG. 28 illustrates operation of a gateway network according to an embodiment of the present invention to provide an in-progress PBX-to-mobile phone call for a PBX telephone calling to a mobile phone in a private-wireless-network of the same gateway server;
  • FIG. 29 illustrates a call path for the telephone call of FIG. 28 after mobile phone user's action to transfer the telephone call to another user within the ente ⁇ rise;
  • FIG. 30 illustrates an embodiment of a transfer sequence for the telephone call of FIG. 28
  • FIG. 31 illustrates operation of a gateway network according to an embodiment of the present invention to provide a three-way conference-call
  • FIG. 32 illustrates operation of a gateway network according to an embodiment of the present invention to provide dial call control via a PC browser
  • FIG. 33 illustrates operation of a gateway network according to an embodiment of the present invention to provide busy notification and alert pop-up.
  • An apparatus and method are provided for integrating a voice gateway with wireless telephone systems, internet protocol (IP) telephone systems, and un-PBX telephone systems to provide voice over internet protocol (VoIP) capability and feature transparency to users at geographically distinct sites.
  • IP internet protocol
  • VoIP voice over internet protocol
  • un-PBX system it is meant a dedicated or general pu ⁇ ose computer comprising a telephone interface and a software application that is capable of providing many features of a traditional private branch exchange (PBX).
  • PBX private branch exchange
  • FIG. 1 shows a sample configuration of a communication system or network 100 for multi-site company having four gateway networks 105 (individually 105 A, 105B, 105C and 105D) at a different company site.
  • Each gateway networks 105 comprises a gateway server 110, and one or more of an IP telephone subsystem 115, a private- wireless-network 120 and a private branch exchange (PBX) system 130 or an un-PBX telephone system 135 connected to each other by a local area network (LAN) 140.
  • PBX private branch exchange
  • LAN local area network
  • Each gateway network 105 can communicate to other gateway networks to send voice and data to each over an IP Network 145, a public-wireless-network 150 or the public switched telephone network (PSTN) 160.
  • PSTN public switched telephone network
  • the gateway server 110 operates in conjunction with the PBX telephone system 130 to coordinate both PBX telephone calls and calls over the PSTN 160.
  • the configuration of the gateway network 105 is described in greater detail in U.S. Pat. App. Serial No. 09/061,802, which is inco ⁇ orated herein by reference.
  • the private-wireless-network 120 supports mobile phone users using private mobile phones.
  • private mobile phones it is meant wireless phones that will operate with the private-wireless-network 120 and, preferably, also with the public-wireless- network 150.
  • the gateway server 110 coordinates the telephone-related activities for both PBX telephones (not shown in this figure) in the PBX telephone system 130 and mobile phones (not shown in this figure) in the private-wireless-network 120, as well as for calls incoming from and outgoing to the PSTN 160.
  • the private-wireless-network 120 can have a connection with the public-wireless-network 150, to allow for roaming between the private and public-wireless-networks.
  • the IP telephone subsystem 115 supports IP telephones that can be discrete, individual IP telephones (not shown) or can be inco ⁇ orated into computer work stations (not shown).
  • the gateway server 110 coordinates the activities of all internal, company telephone systems including the PBX telephone system 130, the private-wireless-network 120 and the IP telephone subsystem 115. PSTN calls and the activities of roaming mobile phones are also coordinated by the gateway server 110.
  • the gateway network 105 can also include an un-PBX telephone system 135.
  • un-PBX telephone system is a cheaper alternative to a traditional PBX system.
  • the un- PBX system consists of a software application plus a set of telephony cards which can be plugged into a standard PC (not shown), such as a Windows NT workstation. Because of its' relatively low cost, un-PBX telephone systems 135 have become a popular alternative for small offices.
  • gateway network configurations that can be supported by the invention are not limited to those shown in FIG. 1.
  • the table below shows some combinations of network telephone system elements that can be supported by the invention.
  • more than one of that kind of system can be equally well supported by the invention.
  • a configuration consisting of two PBX telephone systems 130 and one private- wireless-network 120 can all be controlled by a single gateway server 110.
  • the architecture of the gateway network 105C as shown in FIG. 2 corresponds to the third configuration from the table.
  • the gateway server 110, IP telephone systems 115, and private-wireless-networks 120 are all connected to the LAN 140.
  • the gateway server 110 is coupled with one or more PBX telephone systems 130. Each PBX telephone system 130 is coupled to the PSTN 160.
  • the gateway server 110 is coupled to the IP Network 145 via a router 165. Most routers 165 have configurations which can handle separating the VoIP data versus other data when routing packets incoming from the IP Network 145.
  • the communication system can further include a directory server 170.
  • the directory server 170 maybe on the same server as the gateway server 110 or it may be physically located on a different computer (not shown). Directory servers are described in greater detail in U.S. Pat. App. Serial No. 09/061,802.
  • the gateway server 110 comprises gateway server software 175 some key components of which are shown, including a database 180, a web server 190, a directory services module 195 and a computer telephony integration (CTI) services module 200.
  • the database 180 is where configuration and user data is stored.
  • the web server 190 supports web browser and graphical user interface (GUI) applications, described infra.
  • the web server 190 can be any suitable, commercially available web server.
  • the directory services module 195 refers to an interface software being used to access the user information.
  • the CTI module 200 provides an interface to a CTI driver 210, which is typically supplied by the PBX vendor and is specific to the PBX telephone system 130.
  • the gateway server 110 further includes a VoIP driver 220 which consists of hardware and software that handles the conversion between the voice data in the format required for a trunk or station telephony interface and the IP packet format.
  • a trunk driver 225 includes hardware and software to interface the trunk with the PBX.
  • a station driver 240 consists of hardware and software that manages the analog station interface to the PBX telephone system 130.
  • the PBX telephone system 130 consists of a PBX 250 and one or more PBX telephones 255.
  • PBX telephone systems 130, PBXs 250 and PBX telephones 255 that are widely known have been omitted from the figures and the description.
  • PBX telephone systems 130 and the PBX telephones 255 and how they work are discussed in greater detail in U.S. Pat. App. Serial No. 09/061,802.
  • FIG. 4 shows a sample architecture for the gateway server 110 integrated with an un-PBX telephone system 135.
  • Conventional un-PBX systems use un-PBX software on a separate or dedicated server platform to control an office's telephone system to provide both person-to-person calls in the office and outside calls.
  • One shortcoming of such a configuration is that integration with other company sites is not possible.
  • the components of the un-PBX telephone system 135 are part of the gateway server 110, enabling integration with other company sites and resulting in a substantial cost savings by eliminating the need for a dedicated server platform for the un- PBX telephone system.
  • the un-PBX telephone system 135 of the present invention includes hardware (not shown) and software 260 to interface with and/or control the trunk and the station.
  • the un-PBX telephone system 135 can include a basic VoIP hardware/software package to convert the digital IP packets to an analog format compatible with the telephones of the UN-PBX telephone system.
  • the un-PBX software 260 typically contains PBX switching software 280 that has an open application programming interface (API) (not shown).
  • API application programming interface
  • the gateway server software 175 uses this API to control the call setup of the office telephones in a manner similar to how it controls PBX telephones 255 in a PBX configuration.
  • the CTI services module 200 of the gateway server software 175 also communicates with the un-PBX software 260 to coordinate these features.
  • the gateway server 110 of the present invention provides seamless and consistent usage of the communication system for users at all sites and is an attractive alternative to a full PBX configuration for a small satellite office.
  • FIG.5 shows the major components of the private-wireless-network 120 including a mobile phone 285 and a wireless base station 290 (also known as a wireless base transceiver station).
  • a wireless base station 290 also known as a wireless base transceiver station.
  • mobile phone 285 it is meant any a portable communications device such as a cellular telephone or a personal communication service (PCS) telephone. Users can place and receive telephone calls from the mobile phone 285. Users are typically employees at a private company site or campus that carry a mobile phone 285 with them when they leave their desk to perform company business at other locations or offices within the company campus, or when they visit other company locations. In some implementations, this mobile phone 285 can roam into the public-wireless-network 150.
  • the wireless base station 290 comprises hardware and software that enable it to receive and transmit voice and data to or from the mobile phones 285. The wireless base station 290 converts the voice data received via the mobile phone 285 to H.323 IP packets and routes them to the gateway server 1110 via the LAN 140.
  • H.323 IP packets for voice communication destined for the mobile phone 285 are routed from the gateway server 110 out to the wireless base station 290 via the LAN 140, where the packets are converted back to a form compatible with the mobile phone 285 and transmitted to the mobile phone.
  • multiple wireless base transceiver stations 290 will be placed on a given campus, each with a defined wireless range. As a mobile phone user travels across campus, the current call is handed off from one wireless base station 290 to another.
  • a wireless gatekeeper server 295 coordinates control of the wireless base stations
  • the wireless gatekeeper server 295 keeps track of which mobile phones 285 are currently operating (i.e., recently registered or handling an in-progress call) and which wireless base stations 290 each is communicating with.
  • the wireless gatekeeper server 295 communicates to the wireless base stations 290 using H.323 signaling.
  • the wireless gatekeeper server 295 may also retain a list of IDs identifying mobile phones that are permitted to access the wireless service, or it may request this information as needed from the gateway server 110.
  • the wireless gatekeeper server 295 is the primary point within the private-wireless-network 120 for call control interaction with the gateway server 110.
  • the wireless gatekeeper server 295 may control both local wireless base stations 290 and remote wireless base stations 300.
  • the private- wireless-network 120 may be so configured to handle wireless services for employees at a small satellite sales office.
  • the wireless gatekeeper server 295 comprises software (not shown) which can reside on its own single-pu ⁇ ose server, on the gateway server 110 or on one of the wireless base stations 290.
  • the wireless gatekeeper server 295 is scalable so that a single one can serve a site, with the wireless gatekeeper server interacting with its associated gateway server 110, the wireless base stations 290, 300.
  • the associated gateway server 110 and the wireless base stations 290, 300, whose tasks are coordinated by wireless gatekeeper server 295 make up the private-wireless-network 120.
  • one of the wireless gatekeeper servers 295 may be configured as a master, and the other, slave wireless gatekeeper servers would communicate only with that wireless gatekeeper server.
  • the gateway server 110 would simply communicate to each wireless gatekeeper server 295 separately, and the wireless gatekeeper servers would need to communicate with each other for pu ⁇ oses of handoff only.
  • a handoff is when a mobile phone user moves from the area covered by one wireless base station 295 to the area covered by an adjacent wireless base station 300. "Handoff is the term used in North America; "handover" is the European term.
  • the public-wireless-network 150 shown in FIG. 5 is connected with the PSTN 160; the PSTN can route calls into the public-wireless-network, and vice versa. Furthermore, some private-wireless-networks 120 support configurations whereby the private-wireless-network and the public-wireless-network 150 are connected, as shown in FIG. 5, via a signaling interface 310; thus the public-wireless-network 150 can keep the private-wireless-networks 120 apprized regarding which mobile phones 290 are operating in the public-wireless-network. Thus, the communication system 100 of the present invention enables a mobile phone user (using either a private or public mobile phone) to roam back and forth between the two wireless-networks 120, 150. Given the complex security, authentication and billing issues involved, such configurations are rare in conventional communication systems.
  • the public-wireless-network 150 may include conventional cellular communications standards, including GSM, IS-136, AMPS, TAC, CDMA, and so on.
  • the public-wireless-network 150 may also include other services, such as PCS, for implementing wireless-networks for voice and/or data communication.
  • FIG. 6 shows the IP telephone subsystem 115 which is managed by a centralized server called the IP call manager 315.
  • the call manager 315 may reside on its own server platform (as shown), or it may reside on the same platform as the gateway server 110.
  • the individuals that use the IP telephone services are equipped with one of (i) an IP phone 320, (ii) an IP telephone and a work station 325, or (iii) a work station that has a built-in IP telephone 330.
  • the IP call manager 315 is the server that first receives a call setup request from an IP telephone user who initiates a call on the IP phone 320.
  • the call manager 315 connects the two IP phones so a call can take place.
  • the call manager 315 exchanges information with the gateway server 110 including caller ID information (normally stored at the gateway server).
  • the gateway server 110 can log the start of the call to a call log (not shown). If a call is initiated to an extension or to a location ID plus an extension that is not known to the call manager 315, then the call manager passes the call setup request to the gateway server 110.
  • the gateway server 110 determines how to route the call, sets up the other leg of the call, and returns to the IP call manager 315 the IP address to which the IP telephone needs to route the IP packets.
  • FIG. 7 MOBILE-TO-PSTN ROUTING SAME GATEWAY NETWORK
  • a call in progress is shown between a mobile phone 285 operating in the private-wireless-network 120 and a PSTN telephone 340 at a location nearby the gateway network's location.
  • the voice data travels to the wireless base station 290 where is it converted into IP packets which are dropped onto the local LAN 140.
  • the packets are then routed to the VoIP driver 220 of the gateway server 110.
  • the VoIP driver 220 the data is removed from the IP packets and decoded from whatever encoding and compression format that was applied within the wireless private network 120.
  • the voice data is then routed via the trunk driver 225 to the PBX telephone system 130 and from there to the PSTN 160, which in turn routes it in an analog format to the PSTN telephone 340.
  • the voice which travels from the called party to the caller follows the same path, but goes through the process in reverse; at the VoIP driver 220, the voice data is encoded and inserted into IP packets. The data from the IP packets are extracted and decoded within the private-wireless-network 120.
  • a caller and called telephone have been identified. However, it should be noted that the same routing path would result if the caller where using the PSTN telephone 340 and the called party on the mobile phone 285.
  • FIG. 8 MOBILE TO PSTN CALL SETUP SEQUENCE
  • Mobile phone 285 communicates with wireless base station 290 to exchange initial setup information.
  • Wireless base station 290 communicates to wireless gatekeeper server 295 that a call setup is requested.
  • Wireless base station 290 verifies the mobile ID for the user is OK to access the private-wireless-network 120. 5. Wireless base station 290 communicates to gateway server 110 that call setup is requested.
  • Gateway server 110 authenticates, via the ente ⁇ rise directory 205, the mobile ID and user privilege to call to PSTN 160, and verifies that PBX telephone system 130 resources are available to connect to the PSTN. 7. Gateway server 110 communicates call setup request to PBX telephone system
  • PBX telephone system 130 communicates call setup request to PSTN 160 via the station driver 240.
  • PSTN 160 rings called party PSTN telephone 340. 10. Called party answers. 11. PSTN 160 informs PBX telephone system 130 that called party has answered. PBX telephone system switches the routing through the Station/Trunk Driver from the station over to the trunk.
  • PBX telephone system 130 informs gateway server 110 that called party has answered.
  • Gateway server 110 informs the wireless gatekeeper server 295 that called party has answered.
  • the wireless gatekeeper server 295 informs the wireless base station 290 that the called party has answered. 15. The wireless base station 290 then connects with the VoIP Driver 220.
  • Wireless base station 290 coordinates with mobile phone 285 to start routing IP voice packets to the VoIP driver 220.
  • FIG. 9 PSTN TO MOBILE CALL SETUP SEQUENCE 1.
  • PSTN user dials PSTN telephone 340.
  • PBX telephone system 130 receives call setup request via PSTN 160, and routes request to gateway server software 175.
  • Gateway server 110 validates the direct inward dialing (DID) or extension number and finds that this user currently has a mobile phone 285 that is ON. It passes on the call setup request, along with mobile phone ID and other user identification information (such as user's name, for caller ID), to the wireless gatekeeper server 295.
  • DID direct inward dialing
  • Wireless gatekeeper server 295 looks up which wireless base station 290 is currently associated with the mobile phone 285 being called and requests that wireless base station to ring through to the mobile phone. 5. Called mobile user answers. This information is relayed from the wireless base station 290 back to the wireless gatekeeper server 295.
  • the wireless gatekeeper server 295 responds to the gateway server 110 that the call is connecting OK.
  • the gateway server 110 completes the call circuit setup, configuring the PBX telephone system 130 to begin routing voice packets via the trunk driver 240 to the VoIP driver 220.
  • Wireless base station 290 connects with the VoIP Driver 220.
  • Wireless base station 290 coordinates with mobile phone 285 to start routing IP voice packets to the VoIP driver 220.
  • FIG. 10 MOBILE-TO-PBX ROUTING WITH SAME GATEWAY NETWORK
  • a mobile phone user uses the mobile phone 285 to call another employee on the campus (or in same building).
  • the called user is on the PBX telephone system 130.
  • the caller's voice is transmitted to the wireless base station 290 where is it digitized, compressed, packetized into H.323 data packets which are then deposited on the LAN 140 and routed to the gateway server's VoIP driver 220.
  • the VoIP driver 220 and its' associated digital signal processor (DSP) decompress the IP packets' voice data and push the data across a time division multiplexed (TDM) bus 355 where is it sent to the PBX telephone system 130 via an analog or Tl/El digital trunk 360 coupled to the PBX 250. From the PBX telephone system 130 the analog voice goes to the PBX telephone 300.
  • TDM time division multiplexed
  • a handoff may be required if the user leaves the range of one wireless base station 290 and enters the range of another. This will be handled via H.323 messaging between the wireless base stations 290 and the wireless gatekeeper server 295; the wireless gatekeeper server must inform the gateway server 110 of the IP address of the new wireless base station so voice data can be properly routed.
  • QoS Quality of Service
  • VoIP Voice over IP
  • an alert is provided that could be forwarded to a network management system 350, where a suitable action might be taken to correct the problem.
  • an occasional QoS problem resulting from network congestion is not likely to be an issue, given the expectations of wireless users.
  • the caller's mobile phone 285 In order to set up the call, the caller's mobile phone 285 first performs a handshake between itself and the wireless base station 290. The exact format and data exchanged in this handshake varies according to the wireless protocol being used, but generally involves the mobile phone 285 transmitting its mobile ID (usually a ten digit telephone number) and other identifying information.
  • the wireless base station 290 then converts the identification information to H.323 signaling format and sends it to the wireless gatekeeper server 295, which validates that the mobile phone 285 is allowed access to the private-wireless-network 120.
  • the wireless gatekeeper server 295 routes the access request to the gateway server software 175, along with the IP address of the wireless base station 290.
  • the gateway server 110 may also fetch additional information about the handling of the call. For example, it can update its internal tables and call log, and has the PBX telephone system 130 set up the call to the requested extension.
  • FIG. 12 MOBILE-TO-MOBILE ROUTING WITH SAME GATEWAY NETWORK
  • FIG. 12 shows a first mobile phone 285A being used to call another employee on the same campus (or in same building) who is also using a second mobile phone 285B.
  • the caller's voice is transmitted to the wireless base station 290 where is it digitized, compressed, packetizedinto H.323 data packets which are then deposited on the LAN 140 and routed directly to another wireless base station 300.
  • the called party's wireless base station 300 then de-packetizes the IP voice packets, decompresses the data, and converts it back to the format compatible with the called mobile phone 285B, and transmits it out to the called mobile phone.
  • the same flow occurs in the reverse direction as well.
  • FIG. 13 MOBILE-TO-MOBILE SETUP WITH SAME GATEWAY NETWORK
  • the wireless base station 290B currently associated with the called mobile phone 285B is notified of the call setup request.
  • This request is forwarded to the wireless gatekeeper server 295, which may authenticate the mobile identification of the caller mobile phone 285A. Assuming the authentication passes, the wireless gatekeeper sever 295 forwards the call setup request with the caller mobile identification to the gateway server software 175.
  • the gateway server software 175 checks its database 180 to obtain the user information for the caller, and it also uses the dialed office extension to look up the current location of the called party and checks whether the called party has any special call forwarding enabled.
  • the gateway server 110 determines that the user's mobile phone 285B is currently operating, obtains the corresponding mobile identification of the called party, and routes the call setup request with mobile identification of the called party back to the wireless gatekeeper server 295.
  • the wireless gatekeeper server 295 determines which wireless base station 290B the mobile phone 285B is currently associated with, and routes the call setup request to that wireless base station.
  • the called wireless base station 290B dials the called party's mobile phone 285B.
  • the called wireless base station 290B now begins routing IP packets to the caller wireless base station 290A (which was known to it via the call setup request).
  • the success of the call setup operation is communicated from the called wireless base station 290B back to the wireless gatekeeper 295, which now informs the caller wireless base station 290 A of the address of the called wireless base station, and the caller wireless base station can now begin routing its voice IP packets.
  • FIG. 14 MOBILE-TO-PSTN ROUTING DIFFERENT GATEWAY NETWORKS
  • FIG. 14 shows a scenario for routing a telephone call over the LP Network 145 which will typically allow the users to bypass toll charges that would be incurred using the PSTN or a public-wireless-network.
  • a first user or employee uses a mobile phone 285 to call another employee located at another, remote company site, and wherein the called employee is on a PSTN telephone 340 that is located near the remote company site.
  • the caller's voice is transmitted to the wireless base station 290 where is it encoded and packetized into H.323 IP data packets which are then deposited on the LAN 140 and routed to the IP network 145.
  • the called gateway server's VoIP driver 220 de- packetizes and decodes the IP packets' voice data and sends the data across the TDM bus to the trunk driver 225 where is it sent to the PBX 250 via an analog or digital Tl/El trunk 360.
  • COl telephone company central office
  • C02 telephone company central office
  • a mobile phone user in the private-wireless-network 120 can also use this scenario to directly dial a PSTN telephone number, where the gateway server 110 near the caller identifies (via the dialed telephone number) which other gateway network 105 is closest in location to the targeted PSTN telephone.
  • the routing allows the bypass of toll charges when the called party is located in the PSTN near a remote gateway network 105; this type of call routing is termed a "hop-off to PSTN".
  • FIG. 15 shows a second scenario for routing a call in which the routing of the call over the IP Network 145 will typically allow the users to bypass toll charges.
  • a mobile phone user uses the mobile phone 285 to call another employee located at another company site, and the called employee is on a PBX telephone 345.
  • the caller's voice is transmitted to the wireless base station 290 where is it encoded and packetized into H.323 IP data packets which are then deposited on the LAN 140 and routed to the IP network 145.
  • the gateway server's VoIP driver 220 de-packetizes and decompresses the IP packets' voice data and sends it to the PBX 250 via the TDM bus 355, trunk driver 240 and an analog or digital Tl/El trunk 360. From the PBX 250 an analog voice signal is transmitted to called user's PBX telephone 345. The reverse of this process flow occurs from the PBX telephone back to the mobile as well.
  • the same codec algorithm must be used in both the gateway server's VoIP driver and the wireless base station.
  • FIG. 16 MOBILE-TO-PBX SETUP DIFFERENT GATEWAY NETWORKS
  • FIG. 16 shows the call setup steps that might result in the call routing shown in
  • FIG. 15 In this case the mobile caller is homed in the private-wireless-network 120 from which the caller is originating the call.
  • the call is originated at the caller's mobile phone 285 which coordinates with the wireless base station 290.
  • the wireless base station 290 forwards the call setup request to the wireless gatekeeper server 295, which may verify that the mobile phone 285 is allowed access to the private-wireless-network 120.
  • the wireless gatekeeper server 295 in turn routes the call setup request to the gateway server software 175 A.
  • the caller gateway server software 175 A inspects the dialed digits and determines that the called party is homed at another gateway network 105B.
  • the caller gateway server software 175 A then queries the gateway server 110B in the called gateway network 105B for the routing information of how to get to the called telephone. In this case, since the called party is at a PBX telephone 345 of the called gateway server HOB, the called gateway server responds with the IP routing address of the VoIP driver 220B of the caller gateway server. The caller gateway server software 175 A then relays this information back to the wireless gatekeeper server 295, which then coordinates with the wireless base station 290 to set the routing of voice IP packets directly to the called gateway server's VoIP driver 220B. The called VoIP driver 220B routes the call setup request to the PBX 250 via the station and trunk drivers 240, 225, and finally on to the called party's PBX telephone 345.
  • FIG. 17 shows a third scenario for routing a call in which the routing of the call over the IP Network 145 will typically allow the users to bypass toll charges.
  • The-voice IP packets are generated at the caller wireless base station 290A, and routed directly to the called wireless base station 290B via the IP network 145.
  • the called wireless base 145B station de-packetizes the voice IP packets and routes the data to the called mobile phone 285B.
  • FIG. 18 MOBILE PHONE BEING TURNED ON IN Private-wireless-network
  • the mobile phone 285 When the mobile phone 285 is turned on when the telephone's user is within the coverage area of the private-wireless-network 120 the mobile performs a registration process with the nearest wireless base station 290.
  • the wireless base station 290 if it hasn't seen this user in at least some time period (a period similar to the registration interval of the mobile phones, for example 10 minutes), will send a message to the wireless gatekeeper server 295 to inform it of the mobile phone's presence.
  • the wireless gatekeeper server 295 authenticates the ID of the mobile phone 285, it sends on a notification to the gateway server 110 that this particular mobile is now available for calls.
  • the gateway server 110 checks its database to get the user and location information associated with the mobile phone 285.
  • the gateway server 110 will update its database with this information, so that any subsequent calls to the employee's office extension can be automatically forwarded by the gateway server to the mobile phone. If the mobile phone 285 is an employee's primary (only) telephone, the status of the user is updated so that, for example, the user can now be called at the mobile phone instead of routed off to voicemail. If the gateway server 110 determines that the mobile phone 285 is associated with a visiting user from a different company site, the gateway server notifies the remote gateway server at the visitor's home location regarding the new location of its mobile phone; the remote gateway server stores this information for the pu ⁇ ose of routing subsequent calls for the user.
  • this scenario of a mobile phone 285 being turned on is the same scenario as occurs if a user comes back in range of the gateway network 105 after having been out of range, and with the assumption that there is no roaming between the private and public-wireless-networks 120, 150.
  • Such a scenario might occur, for example, if an employee goes out to lunch and forgets to turn off the mobile phone which he carries with him.
  • the sequence of notification events for a mobile phone 285 being turned off is virtually identical to the sequence of events for a mobile phone being turned on, which is illustrated in FIG. 18.
  • the wireless base station 290 to which a mobile phone 285 has been registering determines that the mobile phone has been turned off, or has not registered or made calls for some time period (generally a period greater than the registration interval of the mobile phone), then this information is forwarded to the wireless gatekeeper server 295.
  • the wireless gatekeeper server 295 may verify that the mobile phone 285 has not appeared at another wireless base station 290, and forwards the notification that the mobile phone is no longer present to the gateway server 110.
  • the local database 180 is updated to note that the mobile phone is no longer available for calls. If the mobile phone 285 that went off the air belongs to a visitor from another site, then the notification is also passed to the remote gateway server at the visitor's home site.
  • this scenario of a mobile phone 285 being turned off is the same scenario as occurs if a user exits from the range of the gateway network 105, and with the assumption that there is no roaming between the private and public-wireless- networks 120, 150. This might occur, for example, if an employee goes out to lunch and forgets to turn off the mobile phone 285 which he carries with him.
  • FIG. 19 MOBILE PHONE TURNED ON IN Public-wireless-network.
  • the gateway server 110 it is important to the proper operation of the gateway server 110 in conjunction with the private-wireless-network 120 that the gateway server maintain the knowledge of (1) where (i.e. at which site or associated gateway network 105) all the operating mobile phones 285 are homed, and (2) which mobile phones (either homed locally or roaming from another gateway network) are currently operating in the gateway network' s wireless private network 120, and (3) which mobile phones (either homed locally or roaming from another gateway network) are currently operating in the gateway network's nearby wireless public network 150.
  • FIG. 19 The scenario for a mobile phone 285 being turned on in the wireless public network 150, where the mobile phone's home site is in the locale of the gateway network 105 is shown in FIG. 19.
  • the mobile phone 285 is turned on and registered in the public-wireless-network 150.
  • the public-wireless-network 150 determines this is a roamer mobile phone 285 and routes the information of the presence of the mobile phone to the wireless gatekeeper server 295 in the private-wireless-network 120 via a signaling link 380 connecting the two wireless-networks.
  • the wireless gatekeeper server 295 then routes the notification to the gateway server software 175, so it can update its database 180 and properly route subsequent calls to the mobile phone 285.
  • FIG. 20 MOBILE PHONE TURNED ON IN Public-wireless-network.
  • the scenario for a mobile phone 285 roaming in the locale of another gateway network 105 is slightly different.
  • the mobile phone registration is received in the public- wireless-network 150.
  • the public-wireless-network 150 knows to route the registration notification to the wireless gatekeeper sever 295 in the mobile phone's home private-wireless-network 120.
  • the wireless gatekeeper sever 295 notifies the gateway server software 175 in the associated gateway server 110 of the mobile phone's registration.
  • the gateway server software 175 updates its database 180 with the new location information of the mobile phone, and, because it is further determined that this mobile phone is operating from a remote location, any other gateway network 105 associated with that location is also notified that this mobile phone is operating in its area.
  • FIG. 21 MOBILE IN Public-wireless-network CALLING TO LOCAL PBX TELEPHONE
  • FIG. 21 shows the routing that results from a mobile phone 285 in the public- wireless-network 150 calling to a PBX telephone 345, where both telephones are associated with the same gateway network 105.
  • the call path starts at the mobile phone 285, is routed through the public-wireless-network 150 and the PSTN 160 to the PBX 250, where it is routed directly to the PBX telephone 345.
  • the public-wireless-network 150 receives the call setup request, forwards it into the PSTN 160, which in turn dials into the PBX 250.
  • the PBX 250 sends the call setup request on to the gateway server software 175, which looks up the incoming dialed number in its database 180 to determine where the user for this number is currently located, or whether and where the user has requested the call be forwarded to.
  • the software 175 determines that the called party should be dialed at a local PBX telephone 345, it responds back to the PBX 250, which completes the call routing through to the PBX telephone.
  • the call routing for this scenario is similar to the call routing described for the that shown in FIG. 21 except that two channels on the PBX 250 are used. This is due to the fact that once a telephone call is routed to the gateway server 110, the PBX 250 does not support re-routing of that leg of the call that has already been set up to the gateway server.
  • FIG. 24 CALL SETUP FROM PBX TELEPHONE TO MOBILE PHONE IN PUBLIC TELEPHONE NETWORK.
  • the caller at the PBX telephone 345 initiates the call to another employee homed at the same office, typically by dialing the office extension of the called party.
  • the call setup request is routed from the PBX telephone 345 to the gateway server software 175 via the PBX 250.
  • the gateway server software 175 looks up the called user in its database 180 and finds that the user with this extension currently has an active mobile phone 285, and that this telephone is currently operating in the public-wireless-network 150. Because the public-wireless-network 150 may be re-assigning the mobile phone temporary mobile identification numbers at regular intervals, the gateway server now queries the wireless gatekeeper server 295 for the current temporary mobile identification of the mobile phone.
  • the wireless gatekeeper server 295 maintains this information via its signaling link 280 to the public-wireless-network 150.
  • the wireless gatekeeper server 295 then returns the requested mobile identification information to the gateway server software 175 which has the PBX 250 dial out to the PSTN 160 using the mobile identification obtained from the wireless gatekeeper server.
  • MOBILE HOMED REMOTELY the caller at the PBX telephone 345 initiates the call to another employee homed at a different office by dialing the location ID and office extension of the called party.
  • the call setup request is routed from the PBX telephone 345 to the gateway server software 175 via the PBX 250.
  • the gateway server software 175 looks up the called user in its database 180 and finds that the user with this extension is homed at a remote site and currently has an active mobile phone 285, and that this mobile phon is currently operating in the public-wireless-network 150.
  • the gateway server 110 Because the public-wireless- network 150 may be reassigning the mobile phone 285 temporary mobile identification numbers at regular intervals, the gateway server 110 now must obtain this temporary mobile identification. It does so by querying the mobile phone's home gateway server 110 via the IP network 145. The remote gateway server 110 in turn queries its wireless gatekeeper server 295 for the current temporary mobile identification of the mobile phone 285. The wireless gatekeeper server 295 maintains this information via its signaling link 380 to the public-wireless-network 150. The wireless gatekeeper server 295 then returns the requested mobile identification information to the remote gateway server software 175, which is then returned to the original requesting gateway server software 175, which in turn has the PBX 250 dial out to the PSTN 160 using the mobile identification obtained from the wireless gatekeeper server.
  • FIG. 26 - PBX TELEPHONE CALLING TO PRIVATE MOBILE IN Public-wireless- network AT A REMOTE SITE
  • the caller at a PBX telephone 345 at one site is calling an employee at another site who is using his mobile phone 285 the public- wireless-network 150.
  • the call is routed through the PBX 250 to the gateway server 110 via the trunk driver 225.
  • the trunk driver 225 routes the call data across the TDM bus 355 to the VoIP driver 220.
  • the VoIP driver 220 converts the data into IP packets and routes it across the IP network 145 to the remote gateway server's VoIP driver 220.
  • the remote VoIP driver 220 reverses the process of the other VoIP driver, extracting the data from the IP packets and routing the voice to the trunk driver 225, which in turn routes it to the PBX 250 and then out to the PSTN 160.
  • the PSTN 160 is connected with the public-wireless-network 150 which in turn routes the call through to the called mobile phone 145.
  • the communication system 100 of the present invention is adapted to provide a plurality of features among the several gateway networks 105 at different sites over the IP network 145 regardless of the configuration of the gateway networks or the telephones used.
  • These features can be broadly classified as automated call control features and as usability features.
  • the automated call control features include quality of service monitoring, fallback to the PSTN 160, monitoring of mobile phones 285 and support of multiple different private-wireless-networks 120.
  • the usability features include support for multiple different numbering plans, caller ID, call control features available at mobile phones 285, call control at browser applications, support for handling a busy destination at browser application, support for a phone directory at a browser application, a call log, follow me control and integrated operations, administration, and maintenance, all of which are described in greater detail below.
  • Quality of Service Monitoring As described in U.S. Pat. App. Serial No. 09/061,802, which is inco ⁇ orated herein by reference, there are several quality of service indicators which can be placed in the IP voice packets at the time when the packets are generated in preparation for routing onto the LAN 140. These quality of service indicators include indicators to test for delay in transmission of data packets, losses in transmission of data packets, or jitter. At the later point in the call path routing, when the data in the IP packets are extracted and decoded, the quality of service indicators are assessed to determine if a reasonable quality for data or voice conversation is being maintained. In the VoIP call routing paths discussed in the above referenced application, the voice IP packets are always generated/encoded by the VoIP driver in one gateway server and extracted/decoded by a VoIP driver in another gateway server.
  • the quality of service (QoS) determination may involve two separate legs of a call routing path over the IP network 145.
  • the voice IP packets that are generated at the wireless base station 290 are de-packetized and decoded at the VoIP driver 220.
  • the multiple vendors agree on which QoS indicators to use. Assuming that both the VoIP driver vendor and the base station vendor agree on how to use QoS, then we need to assess where in the call path the quality degradation is taking place. In many cases, such as that shown in FIG.
  • the degradation may occur on the IP Network 145, or it may occur on the LAN 140, along the routing leg between the wireless base station 290 and the LAN. Since there is a method to improve degraded performance of the IP network, it is most useful to detect a IP Network problem.
  • the VoIP driver 220 of the invention continues to monitor QoS for the entire path segment over which voice LP packets travel. Once the QoS for this entire segment degrades to a given threshold, a secondary simple test of the QoS between the two gateway servers 110 (the gateway server of the caller and the gateway server of the called party) will be executed. This secondary test may involve sending bursts of IP packets over the IP network 145 and having the VoIP drivers 220 on either end assess QoS. Alternatively, the test may be more simplistic, such as issuing a series of "ping" commands across the IP network. If it is found that the secondary QoS is also degraded past a threshold, then the fallback to PSTN process can be initiated.
  • the gateway server 110 can do to alleviate the problem, other than alarm to an external system, such as a network management system (not shown).
  • a network management system not shown.
  • Such an alarm in more sophisticated network configurations, could activate load balancing algorithms which might eliminate high-load network activities or users.
  • FIG. 15 illustrates an original VoIP call involving a mobile phone
  • FIG. 27 illustrates the modified call path once the fallback to PSTN operation has completed.
  • a fallback to PSTN 160 is accomplished as follows. The sequence of events starts when the VoIP driver 220 in called gateway server 11 OB detects QoS degradation. It performs a secondary test by exchanging packets between its VoIP driver 220 and the VoIP driver in the caller gateway server 110 A. If this test also indicates degradation, the called gateway server HOB initiates a fallback to PSTN 160. First, the called gateway server HOB announces its intention via a notification sent to the caller gateway server 110 A. The caller gateway server 110A then coordinates with the called gateway server 110B to set up a PSTN call between the two gateway servers via the PBXs 250A, 250B. As soon as the PSTN call is set up between the two gateway servers 110A, 110B, the caller gateway server must command the caller wireless base station 290 to start routing its packets to the local VoIP driver 220A instead of the remote one.
  • the gateway server 110 must retain the knowledge of which mobile phones 285 are operating (e.g., turned on) in the private- wireless-network 120.
  • the wireless gatekeeper server 295 must keep the gateway server 110 apprized of the status of the mobile phones 285 in several categories including: (1) mobile phones in the home private-wireless-network 120 (2); mobile phones in the local public-wireless-network 150; (3) roamer mobile phones in the home private-wireless- network; and (4) roamer mobile phones in the local public-wireless-network.
  • the wireless gatekeeper server 295 When updating the gateway server 110 regarding mobile phones 285 on the public network, the wireless gatekeeper server 295 must provide not the temporary mobile LD, but rather the mobile ID by which the mobile is known when operating within the private-wireless-network 150. This allows the gateway server 110 to determine which mobile phones 285 are active, for subsequent routing pu ⁇ oses. In other wireless-network arrangements, the mobile LD of the mobile phones 285 may be unique within both public and private-wireless-networks 150, 120. Multiple Private-wireless-network Vendors Supported
  • the communication system 100 of the invention is capable of supporting multiple private-wireless-networks 120 manufactured by different vendors.
  • the wireless-network equipment associated with one gateway server 110 may differ from the wireless equipment vendor associated with another gateway server.
  • both private-wireless-networks must support and be using the same kind of encoding and compression algorithms (such as enhanced full rate GSM codec), and they further must support the same voice over IP protocol, such as H.323.
  • the mobile identification number is a 10-digit string which uniquely identifies the mobile phone 285 within the private-wireless-network 120 and within all roaming partners of the wireless-network.
  • the mobile phone 285 is used as an employee resource within a company, it is most convenient for users not to have to remember a separate telephone number for the mobile phone, which a caller might dial after dialing to the office extension fails because the employee didn't answer.
  • the employees and the users of the mobile phones 285 are not required to remember mobile phone numbers. Instead the gateway server 110 automatically maps a dialed number to the mobile phone number, and the mobile phone number need only be known within the system 100.
  • the gateway server software 175 receives the call setup request and looks up which user has the given Location + Extension. If the called user does not have an actual PBX telephone 345 but does have an operating mobile phone 285, then the 10-digit mobile identification for the user's mobile phone is automatically passed to the relevant wireless gatekeeper server 295 along with a request to set up the call. If instead the mobile phone 285 (of the called user who doesn't have a PBX telephone 345) is not currently operating, then the gateway server 110 can route the call directly to the called party's voicemail.
  • ETN number i.e., 8+Location+Extension
  • the gateway server 110 configuration (or the individual user's configuration) may specify that if the mobile phone is operating, a call attempt to the mobile phone be initiated; and if the mobile phone is not operating, the call be attempted to the PBX telephone. Since the mapping of the dialed number to the ten-digit mobile identification number is handled entirely within the gateway server 110, the dialing plan for the mobile phone 285 can be configured for any of the plans discussed in the original patent. Furthermore, the convenience of having to recall only a single telephone number for each user is a unique productivity enhancement offered by the invention as it encompasses a wireless capability.
  • the dialing plan can be coordinated such that both kinds of telephone users have the same location LD, with possibly different extension ranges. Users at other sites make calls to both kinds of telephones, dialing 8 + location B) + extension, and the callers don't need to know which kind of telephone a particular user has.
  • Caller ID As described in U.S. Pat. App. Serial No.09/061, 802, the integration of a gateway server 110 with an ente ⁇ rise directory 205 provides a central repository for modifying user information.
  • User information includes many different attributes describing the particular user, including name, and office extension number for the user, etcetera.
  • additional attributes are added to support the additional user information required to be maintained regarding a user's mobile phone 285 and a user's IP telephone 320.
  • Table II shows a set of additional attributes that are used to support the wireless private network's inclusion in the gateway network 105. TABLE II
  • caller ID information is always added to the call setup request, which flows from the gateway server 110 to the wireless gatekeeper server 295 to a wireless base station 290, where it is finally made available to the mobile phone.
  • the display of caller LD information is also dependent on what a handset of the mobile phone 285 can support, and whether the wireless protocol being used by the vendor of the wireless-network equipment supports the exchange of caller ID information; for example, GSM Phase 2 supports caller ID, but GSM Phase 1 does not.
  • Outgoing Calls from Mobile Outgoing calls from the mobile phone 285 can support the caller ID, as the mobile ID is guaranteed to be routed to the gateway server 110, which can then look up the caller information in its database 180 and forward it to the called party, and the information can thus be displayed as described in U.S. Pat. App. Serial No. 09/061,802.
  • IP telephone display (not shown).
  • IP Telephone On Other Telephones. Outgoing Calls from IP Telephone
  • IP telephone 320 can support the caller ID, as the call is guaranteed to be routed to the gateway server 110, which can then look up the caller ID information in its database 180 and forward it to the called party, and the information can thus be displayed as described in the original invention.
  • the gateway server 110 has the knowledge as to which users currently are logged into the gateway server via a browser GUI interface 385 on the users' PCs. If the browser GUI interface 385 is up for the given user, the gateway server 110 will cause the browser GUI interface to display the caller ID information for the call as part of a pop-up dialog window (not shown).
  • the communication system of the present invention is adapted to make the same call control features generally available to the user on PBX telephone are also be available to the user on the mobile phone. These features include, call hold/retrieve, transfer, forwarding and conferencing, all of which are described in more detail below.
  • a second call can be routed to the user on a mobile phone 285 while a call is already in progress.
  • the second call may be signaled to the mobile phone user by a special tone played over the incoming voice, and the caller ID for the new caller may be shown on the display of the mobile phone 285.
  • the mobile user wishes to, he may verbally ask the original caller to hang on, and then dial a sequence, such as "*5' to put the first caller on hold and answer the new caller. Subsequently, the mobile phone user may dial another sequence, such as "*6" to retrieve the original and resume that conversation.
  • FIG. 28 shows a called in progress from a caller on a PBX telephone 345 to a called party using a mobile phone 285 in the private- wireless-network 120 of the same gateway server 110.
  • FIG. 29 shows the end result of the mobile phone user's action to transfer the call to another user within the ente ⁇ rise. In this case the second called user is located at another site, and thus the resulting routing is accomplished via the IP network 145.
  • FIG.30 Exemplary steps describing how the transfer of the call is accomplished by the communication system 100 are shown in FIG.30 and described in the following sequence of steps:
  • caller 1 In conversation with the caller, called party 1 (in FIG. 28) determines that the caller should be transferred to speak with the called party 2 (in FIG. 29). So the caller keys a special sequence, such as "*3" followed by the location ID and extension of the called party 2. 2. The key sequence is inte ⁇ reted by the wireless base station 290 which forwards the transfer request to the wireless gatekeeper server 295. 3. The wireless gatekeeper server 295 in turn forwards the transfer request to the gateway server software 175.
  • the gateway server software 175 forwards the setup request to the remote gateway server software with the given location LD defined in the dialed digits and the extension. 5.
  • the remote gateway server software 175 checks its database 180 and determines that the called party is in its gateway network 105.
  • the remote gateway server software 175 sets up the routing for the new call via its PBX 250, and dials through to the PBX telephone 345.
  • the remote gateway server 110 commands its VoIP driver 220 to route packets for the call to the other gateway server's VoIP driver.
  • the remote gateway server software 175 sends a response back to the first gateway server's software with its VoIP address and the information that the other end of the call has been routed. 8.
  • the gateway server software 175 commands its VoIP driver 220 to start routing the voice packets to the VoIP driver in the remote gateway server via the LP network 145.
  • the gateway server software 175 messages to the wireless gatekeeper server 295 that it can take down its leg of the call.
  • the wireless gatekeeper server 295 relays this information to the wireless base station 290.
  • the wireless base station 290 terminates its call with the mobile phone 285.
  • a method for forwarding a call using the gateway server 110 is described in U.S. Pat. App. Serial No. 09/061 ,802.
  • the method can be extended to handle communication systems 100 having mobile phones 285.
  • a user who has set a "follow-me" function to route calls first to his mobile phone, and secondly to a telephone at a secretary's desk.
  • An incoming call is routed by the gateway server 110 to the wireless gatekeeper server 295, and in turn to the wireless base station 290 with which the mobile phone 285 is currently registering. If after several rings the mobile user doesn't answer, the wireless base station 290 routes a negative acknowledgment back to the wireless gatekeeper server 295, which in turn routes it back to the gateway server 110.
  • the gateway server 110 receives the acknowledgment and then routes the call via the PBX 250 to the secretary's desk telephone.
  • FIG. 31 shows an example of a 3-way conference call in progress.
  • the first caller at the PBX telephone 345 has called a co-worker at a mobile phone 285A (the second party).
  • a third party calling from another mobile phone 285B at a remote office, joins the conversation.
  • the second party has two call paths set up to it.
  • the base station is combining the two incoming voice streams as needed, and funneling the combined stream to the mobile phone 295.
  • the CTI services module 200 may be working with the PBX 250 to implement the conferencing, for example, by selecting the higher volume of the two incoming voices and patching that voice to the first party.
  • the gateway server 110 of the present invention the usage protocol for conferencing can be offered in the same way for the different kinds of telephone users.
  • Call control features can also be offered to the mobile phone user who is logged into the LAN 140 and running a browser application that provides a browser GUI interface 385 to the gateway server 110.
  • the mobile phone 285 may be the user's primary telephone, and for convenience, the user may want to control the telephone's features through the browser GUI on the user's personal computer.
  • FIG. 32 illustrates an example wherein a user clicking on the "dial" button in the PC browser GUI interface 385, after having selected another user to dial from the ente ⁇ rise directory 205.
  • the PC browser messages to the gateway server software 175 that the user wants to initiate a call (step 1 in the figure), and to whom.
  • the message also contains the identification of the calling user, and so the gateway server 110 checks its database 180 and finds that the primary telephone for the user is the user's mobile phone 285.
  • the gateway server software 175 then messages to the wireless gatekeeper server 295 (step 2), which in turn messages to the relevant wireless base station 290 regarding the call initiation (step 3).
  • the wireless base station 290 coordinates with the mobile phone 285 to have it initiate the call automatically (step 4).
  • the telephone call is treated like any other telephone call.
  • the call setup would be exactly as described in FIG. 8 above.
  • the dotted line represents the call path after it has been set up.
  • the gateway server 110 must update the caller's browser GUI interface 385 to indicate the call has now started (step 5 in FIG. 32).
  • the call control functions handled in this manner can include dial, answer, hold/retrieve, hang-up, transfer, and conferencing. It is important to note that the private wireless system 120 and the mobile phones 285 must support this functionality in order to provide such call control from the PC.
  • the newer standards, such as the GSM phase 2 are supporting such sophisticated functions, where the older ones did not.
  • the do not disturb and forwarding functions are handled simply by the browser GUI interface 385 messaging the request to the gateway server 110. Since both these actions are simply state changes that will affect future calls, the gateway server 110 simply notes this information in its database 180 and will use it to make routing decisions when incoming calls arrive for the user.
  • Destination Busy handling for mobile phones 285 must be discussed in two different contexts, when the busy destination is a mobile phone, and when the caller initiating the call is on a mobile phone. Both scenarios are supported.
  • the caller is provided with several options by the browser GUI interface including, as described in U.S. Pat. App. Serial No. 09/061,802. Namely, the caller is provided with the options of send alert, request callback, ring through and cancel call, which is equivalent to hanging up.
  • a call flow sequence is shown wherein a first user on a mobile phone 285 with the browser GUI interface 385 running at one location calls a second user at another location, where the second user is on a PBX telephone 345 and also has the browser GUI interface running. The second user is already engaged in a telephone conversation on his PBX telephone 345.
  • the steps are as follows. 1. Mobile phone user dials using location ID plus extension and the wireless base station receives 290 the call setup request. 2. Wireless base station receives 290 routes setup request to the wireless gatekeeper server 295.
  • Wireless gatekeeper server 295 routes the setup request to gateway server software 175.
  • Gateway server software 175 looks up which gateway server 110 serves the dialed location ID and routes the setup request to that gateway server.
  • the remote gateway server software 175 receives the request and already knows that the called party is engaged in a call, so it immediately updates the called party's browser GUI interface with a pop-up message (not shown) regarding who is calling.
  • the remote gateway server software 175 notifies the original gateway server software that the called party is busy.
  • the gateway server software 175 routes this information to the browser GUI interface 385 of the caller, and gives the caller several destination busy options, including send alert, request callback, ring through, and cancel call. 8.
  • the caller now enters a one line alert message, such as urgent - client wants to back out of the deal.
  • Gateway server software 175 relays the alert message to the remote gateway server 110.
  • the remote gateway server software 175 displays the alert on the called party's browser GUI interface 385 pop-up window. 11. The called party sees the alert message and responds by quickly putting his other party on hold and answering the mobile phone caller's call by clicking on the answer button in the window.
  • the remote gateway server software 175 requests the call be answered via the CTI driver 210 and the other caller be put on hold.
  • the CTI driver 210 and PBX 250 cooperate to put the other part on hold, and switch the called party's routing in the PBX.
  • the remote gateway server software 175 gives the VoIP driver 220 the new routing information, including the address of the wireless base station 290 where the IP packets are going to be routed.
  • the remote gateway server software 175 acknowledges that all this has been accomplished back to the gateway server software associated with the calling gateway network 105.
  • the gateway server software 175 provides the wireless gatekeeper server 295 with the new address where the LP packets are to be routed (i.e., the remote VoIP driver 220).
  • the wireless gatekeeper server 295 requests the wireless base station 290 to reroute the IP packets to the remote VoIP driver 220.
  • the gateway server software 175 updates the caller's po-up window with the information that the call is now connected. Both parties can hear and speak. In the event that the called party is speaking on a mobile phone 285 and a new call arrives, the mobile phone user may have the alert displayed on his PC if the browser GUI interface 385 is up and the mobile phone has been marked in the system as the user's primary telephone.
  • a telephone directory (not shown) associated with the browser can be used for convenience when dialing to other users, or when transferring a call, as described in U.S. Pat. App. Serial No. 09/061,802. Call Log
  • Methods of logging of calls to and from mobile phones 285 and LP telephones 320 in a communication system 100 to create a call log according to the present invention are similar. However, it is critical that the gateway server 110 know when a call is being set up, and when a call is terminated, and also when a call changes state (such as when a call is transferred, for example). By considering the call setup flows for setup of mobile phone calls described above, it is dear that the gateway server 110 has the knowledge of when calls are initially set up, and thus can write the needed log information for the call initiation event. On the other hand, it is not so obvious that the gateway server 110 would be aware of call termination events. For example, consider the case of a mobile-to-mobile call as shown in FIG. 12.
  • a call could be ended entirely within the private-wireless- network 120.
  • the communication system component that first discovers that the termination has occurred e.g., the wireless base station 290 in the case of a mobile phone 285 and the IP telephone subsystem 115 software in the case of an LP telephone 320, must send out a notification to its controller, e.g., the wireless gatekeeper server 295 or IP call manager 315, which must in turn route the notification to the gateway server 110.
  • PBX telephones 345 are also relevant to mobile phones 285.
  • part of the data stored for each user will include an identification of which telephone is the primary telephone for the user.
  • the primary telephone might be a PBX telephone 345 in the office (the default), a mobile phone 285 or an LP telephone 320.
  • users may be given the ability to set up filters (not shown) on how to route their calls at certain times of the day. For example, the user may want to route telephone calls directly to voicemail during nights and weekend times. This can be accomplished by building call filters. A user can build up a set of favorite call filters and apply these at different times of the day. For example, a software engineer user may want to take only particular calls during the 7 to 9 am time-frame to guarantee uninterrupted work time.
  • a filter such as the following could be constructed by the user, using simple point-and-click methods on the browser-based GUI interface 385:
  • Filters could then be applied to the hours of 7 to 9 am in the user's calendar. Filters can also be stacked one upon the other. For example, a user might, in addition to the above filter, have a certain filter applied all the time, such as the following: Filter ALWAYS AVOID If caller is "ex-husband" then
  • the browser GUI interface 385 for filter setup is integrated with an ente ⁇ rise calendar management system (not shown).
  • calendar management tools include Novell GroupWise and ON Technology's Meeting Maker.
  • users can request to have their schedule for a given time period imported from the scheduling tool. Then, using a simple point and click method, users can select a scheduled item (such as a staff meeting) and apply an already set-up filter for the time covered by the meeting.
  • Integrated OA&M (Operations. Administration, and Maintenance)
  • OA&M operations, administration and maintenance
  • the invention provides a common browser based interface, by which typical OA&M functions can be managed from a single point.
  • An advantage of this is the consistency of service that can be provided across the multiple different kinds of telephone systems.
  • User management is one of the most common administrative functions which is typically carried out on a daily basis at an ente ⁇ rise.
  • the use of the ente ⁇ rise directory 205 which is the central point for managing all information about a user or employee, and is used to enter all kinds of information regarding usage of different kinds of telephone systems.
  • an administrator could, from the browser GUI interface 385 to add a new employee/user, including name, department info, e-mail address, and the information that an IP telephone 320 is the user's primary telephone and secondary telephone is a mobile phone 285, the mobile phone's ID, and what location and extension is associated with the user.
  • Previous communication systems 100 have not been integrated in this way.
  • an administrator might have to log into several different systems, including a call manager server (not shown) for specifying the IP address, another OA&M system (not shown) for setting up the mobile phone 285, and so on.
  • the invention supports open standards for alarms, including SNMP.
  • Alarms from the gateway network 105 can be sent to a commercially available network management system (not shown) , or alternatively all alarms can be monitored on a browser-based OA&M monitor (not shown).
EP00930722A 1999-05-12 2000-05-12 Verfahren und vorrichtung für integriertes sprach-gateway mit einer schnittstelle zum mobiltelefon, ip-telefon und un-pbx-systemen Withdrawn EP1181805A1 (de)

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US13378999P 1999-05-12 1999-05-12
US133789P 1999-05-12
PCT/US2000/013247 WO2000069156A1 (en) 1999-05-12 2000-05-12 Method and apparatus for integrated voice gateway with interface to mobile telephone, ip telephone and un-pbx systems

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