US20020141404A1 - Call routing using information in session initiation protocol messages - Google Patents

Call routing using information in session initiation protocol messages Download PDF

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Publication number
US20020141404A1
US20020141404A1 US10/020,783 US2078301A US2002141404A1 US 20020141404 A1 US20020141404 A1 US 20020141404A1 US 2078301 A US2078301 A US 2078301A US 2002141404 A1 US2002141404 A1 US 2002141404A1
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Prior art keywords
information
routing
server
caller
session
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US10/020,783
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English (en)
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Michael Wengrovitz
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Alcatel Lucent SAS
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Alcatel SA
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Assigned to ALCATEL reassignment ALCATEL ASSIGNMENT OF ASSIGNORS INTEREST (SEE DOCUMENT FOR DETAILS). Assignors: WENGROVITZ, MICHAEL
Priority to US10/020,783 priority Critical patent/US20020141404A1/en
Application filed by Alcatel SA filed Critical Alcatel SA
Priority to AT02400025T priority patent/ATE363799T1/de
Priority to JP2002099938A priority patent/JP4011950B2/ja
Priority to EP02400025A priority patent/EP1248440B1/de
Priority to DE60220329T priority patent/DE60220329T2/de
Priority to CNB021272689A priority patent/CN100428764C/zh
Publication of US20020141404A1 publication Critical patent/US20020141404A1/en
Priority to US11/427,513 priority patent/US8737272B2/en
Priority to US14/263,727 priority patent/US20140233560A1/en
Assigned to ALCATEL LUCENT reassignment ALCATEL LUCENT CHANGE OF NAME (SEE DOCUMENT FOR DETAILS). Assignors: ALCATEL
Abandoned legal-status Critical Current

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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066—Session management
    • H04L65/1101—Session protocols
    • H04L65/1104—Session initiation protocol [SIP]
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L45/00—Routing or path finding of packets in data switching networks
    • H04L45/42—Centralised routing
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066—Session management
    • H04L65/1069—Session establishment or de-establishment
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066—Session management
    • H04L65/1101—Session protocols
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M7/00—Arrangements for interconnection between switching centres
    • H04M7/006—Networks other than PSTN/ISDN providing telephone service, e.g. Voice over Internet Protocol (VoIP), including next generation networks with a packet-switched transport layer

Definitions

  • This invention relates generally to Internet telephony systems, and more particularly, to intelligent call routing in Internet telephony systems that adhere to session initiation protocol.
  • Session Initiation Protocol is a signaling protocol for creating, modifying, and terminating multimedia sessions, including Internet telephone calls, with one or more SIP-endpoints. Details about the SIP signaling protocol is set forth in Internet Engineering Task Force Request for Comment 2543 entitled “SIP: Session Initiation Protocol,” March 1999 (hereinafter referred to as RFC 2543), which is incorporated herein by reference. SIP provides an alternative to PBX- or H.323-signaled telephony.
  • SIP servers including proxy and redirect servers, are typically engaged during the call set-up process to route calls.
  • Such call routing includes ascertaining called end-points in response to call establishment messages, referred to as INVITE messages, originated by calling end-points.
  • INVITE messages are either proxied to ascertained called end-points or the addresses of ascertained called end-points are returned to the calling end-points for allowing the calling end-points to directly place calls to the called end-points.
  • FIG. 1A is a functional block diagram for establishing a SIP call via a typical proxy SIP server 10 .
  • the proxy server 10 receives an invitation from a calling end-point 15 in the form of an INVITE request.
  • the INVITE request includes routing information in the “From:”, “To:”, “Contact:” and other standardized fields within the INVITE message header.
  • the “To:” field of the message header includes a generic SIP URL associated with a called end-point 20 .
  • the proxy server 10 accepts the INVITE request and in step 32 , preferably engages a location server 25 for routing the call based on the routing information in the SIP message header.
  • the location server 25 retrieves the SIP URL associated with the called end-point to resolve the URL to a more precise address. For example, a call directed to a generic SIP URL such as, for example, “sales@acme.com” may be resolved to a particular person, such as, for example, “bob@ny.acme.com.”
  • the retrieved address information is transmitted to the proxy server in step 34 .
  • the proxy server 10 issues a second INVITE request to the more precise address.
  • the called end-point 20 receives the second INVITE request and alerts the user of the request by, for example, causing the user's telephone to ring. If the call is answered, the called end-point 20 , in step 38 , returns a success indication to the proxy server 10 via an OK response.
  • the proxy server 10 forwards the OK response in step to the calling end-point 15 .
  • the receipt of the success result is confirmed by the calling end-point 15 by transmitting an ACK request to the proxy server 10 in step 42 , which then forwards it to the called end-point 20 in step 44 .
  • FIG. 1B is a functional block diagram of an alternative method for establishing a SIP call using a typical redirect SIP server 47 .
  • the redirect server 47 accepts the INVITE request and, as the proxy server 10 of FIG. 1A, contacts the location server 25 in step 33 for routing the call based on the routing information in the INVITE message header.
  • the redirect server 47 instead of directly contacting the newly found address received in step 35 , returns the address to the calling end-point in step 37 .
  • the calling end-point 15 confirms receipt of the address via an ACK request in step 39 .
  • the calling end-point 15 issues a new INVITE request to the address returned by the redirect server 30 in step 41 . If the call succeeds, the called end-point 20 transmits an OK response in step 43 , and the calling end-point 15 , in step 45 , completes the handshake with an ACK request.
  • IVR interactive voice response
  • DTMF dual tone multi-frequency
  • Conventional telephony systems may employ caller ID data that is automatically transmitted with an incoming call in lieu of DTMF key entries to route the call.
  • the data transmitted is limited to caller ID data, and does not include additional caller intent information that may be desirable to more intelligently route a call.
  • Newer systems may employ voice recognition techniques in response to IVR prompts to deduce the caller's intent.
  • voice recognition systems are also cumbersome and inconvenient for the caller, subject to error, and also yield slow call setup times.
  • the current invention is directed to a system and method for intelligently routing IP telephone sessions using information transmitted by the caller that extends beyond simple caller ID data.
  • the invention is directed to a method for establishing an IP telephony session between a first device and a second device where the method includes receiving a call establishment message from the first device, the call establishment message including a first routing information in a header portion of the message, and a second routing information in a body portion of the message.
  • the method further includes determining an address of the second device based on the first and second routing information and using the address for routing the session to the second device.
  • the invention is directed to a method for establishing an IP telephony session between a first device and a second device where the method includes retrieving caller intent information from a data store on the first device, creating a call establishment message including a header and a body, the body including the caller intent information, and transmitting the caller initiation request to a server for routing the session to the second device based on the caller intent information.
  • the invention is directed to an IP telephony system including a first device, a second device, and a server operative between the first device and the second device.
  • the first device creates a call establishment message including a header and a body, the body including caller intent information.
  • the server uses the caller intent information for determining an address of the second device for routing the session to the second device.
  • the invention is directed to a server in an IP telephony system operative between a first device and a second device.
  • the server is configured for receiving a call establishment message from the first device, the call establishment message including a first routing information in a header portion of the message and a second routing information in a body portion of the message, determining an address of the second device based on the first and second routing information, and using the address for routing the session to the second device.
  • the present invention allows a more intelligent determination of a callee address than when only using standard routing information typically included in the header portion of a call establishment message or traditional caller ID information. Additional information about the caller included in the body portion of the call establishment message allows an efficient determination of the caller's intent for routing the call to a most appropriate callee. Because the caller intent information is already made available at the placement of the call, back and forth interactions with the caller for obtaining such information is no longer required.
  • FIG. 1A is a functional block diagram of establishing a SIP call using a typical proxy SIP server
  • FIG. 1B is a functional block diagram of an alternative method for establishing a SIP call using a typical redirect SIP server
  • FIG. 2 is a schematic block diagram of a communication system adhering to an IP telephony protocol according to one embodiment of the invention
  • FIG. 3 is a functional block diagram for establishing a SIP call based on caller information according to one embodiment of the invention where a proxy SIP server is used;
  • FIG. 4 is a functional block diagram for establishing a SIP call based on caller information according to one embodiment of the invention where a redirect SIP server is used;
  • FIG. 5 is a functional block diagram for determining an address of a called end-point using caller information according to one embodiment of the invention.
  • FIG. 6 is a flow diagram of a process for an IP telephony call setup according to one embodiment of the invention.
  • FIG. 2 is a schematic block diagram of a communication system adhering to an IP telephony protocol such as, for example, a session initiation protocol (SIP), according to one embodiment of the invention.
  • IP IP telephony protocol
  • the system includes a calling end-point 50 initiating a call that is directed to a called end-point 52 over a wide area network, such as, for example, a public internet 54 .
  • the calling end-point 50 , called end-point 52 , and internet 54 preferably adhere to the SIP signaling protocol set forth in RFC 2543 .
  • RFC 2543 a public internet 54 .
  • a person skilled in the art should recognize, however, that the any other IP signaling protocol conventional in the art may be used instead of SIP.
  • the calling and called end-points 50 , 52 are preferably SIP-enabled telephones, hand phones, personal computers, switches, routers, and/or the like.
  • each calling and called end-point is associated with an input device 59 , 65 receiving input data, such as, for example, an input cable, a keyboard, a keypad, or the like.
  • Each calling and called endpoint is further associated with an output device 61 , 63 presenting output data, such as, for example, an output cable, a display, or the like.
  • the calling and called end-points 50 are preferably also associated with handsets 60 , 62 for allowing a voice conversation between a caller and callee.
  • the communication system of FIG. 2 further includes a server preferably routing calls between the calling and called endpoints 50 , 52 .
  • the server 56 is preferably a SIP proxy or redirect server similar to the proxy and redirect servers 10 , 47 of FIGS. 1 A- 1 B.
  • the server 56 is enhanced with the capability of using information transmitted by the calling endpoint along with standard routing information, for more intelligently routing the call.
  • the server 56 is preferably coupled to a database 57 and location server 58 .
  • the location server contains location information used for routing calls between the calling end-point and the called end-point.
  • the location server may be similar to the location server 25 of FIGS. 1 A- 1 B.
  • the database 57 is a data store, such as a hard disk drive or drive array, storing caller information and associated routing information for use in addition to the location information to route the calls.
  • the server 56 , database 57 , and/or location server 58 reside in a single machine. In another embodiment, the servers reside in two separate machines coupled to each other over a local area network, private wide area network, or the public internet 54 .
  • the calling end-point 50 initiates a call by transmitting a call initiation message to the server 56 .
  • the call initiation message may be, for example, a SIP INVITE message.
  • the call initiation message includes caller information in addition to standard routing information that may be used by the server 56 to route the call.
  • Such caller information preferably relates to the caller's intent, and may include, by way of example, a name, an address, a phone number, an email address, an account number, a transaction history, a billing history, a department name or selection, an agent preference or selection, a language preference or selection, a product preference or selection, or any other information that may be used to route a call, such as, for example, information typically gathered via an IVR system.
  • the caller information may also include user profile information, user habit information, or information contained in traditional browser cookies associated with a web site domain.
  • the caller information is preferably generated and stored in the calling end-point 50 according to various mechanisms.
  • the caller information data may be generated by completing an electronic form that may have a click-to-call button for initiating a call.
  • the caller information may also be written by the server 56 or calling end-point 50 during or after a call.
  • the caller information may, for instance, update transaction history information, call history information, called agent information, and/or the like.
  • the server 56 receives the call initiation message from the calling end-point 50 and determines an address to which to route the call using information retrieved from the location server 58 and database 57 . The call is then routed to the called end-point based on the address, or the address returned to the calling end-point 50 for allowing it to establish a direct communication with the called end-point.
  • FIG. 3 is a functional block diagram for establishing a SIP call based on caller information according to one embodiment of the invention where the server 56 is a proxy server 56 a .
  • the proxy server 56 a receives an INVITE message from the calling end-point 50 .
  • the INVITE request preferably includes standard routing information in the “From:”, “To:”, “Contact:” and other fields within the INVITE message header, referred to as the header routing information (HRI).
  • the INVITE request further includes caller information in the body of the INVITE message, referred to as the body routing information (BRI).
  • the proxy server 56 a ascertains an address of the called end-point 52 based on the header and body routing information.
  • the header routing information may contain an address of a general appliance company to which the call is directed.
  • the body routing information may contain keywords associated with the caller, such as, for example, “washing machine,” “service,” and “model ABC.” Based on this information, the proxy server 56 a may deduce the caller intent and route the call to an agent skilled in answering questions about the servicing of washing machine model ABC using information retrieved from the database 57 and location server 58 .
  • the proxy server 56 a issues a second INVITE message to the ascertained address. All or portions of the received body routing information may also be provided to the called end-point in the body of the second INVITE message. This may be desirable, for example, to facilitate discussions between a customer representative at the called end-point 52 and a caller at the calling end-point 50 .
  • the called end-point 52 receives the second INVITE message and alerts the user of the request by, for example, causing the user's telephone to ring. If the call is answered, the called end-point 52 , in step 82 , returns a success indication to the proxy server 56 a via an OK response. The proxy server 56 a in turn forwards the OK response in step 84 to the calling end-point 50 . The receipt of the success result is confirmed by the calling end-point 50 by transmitting an ACK request to the proxy server 56 a in step 86 , which then forwards it to the called endpoint in step 88 . Voice conversation between users of the calling and called end-points 50 , 52 ensues in step 90 via their respective handsets 60 , 62 .
  • FIG. 4 is a functional block diagram for establishing a SIP call based on caller information according to another embodiment of the invention where the SIP server 56 is a redirect server 56 b .
  • the redirect server 56 b accepts an INVITE message including header and body routing information, and, as the proxy server 56 a for FIG. 3, examines the header and body information for ascertaining an address to the called end-point using information retrieved from the database 57 and location server 58 .
  • step 112 the redirect server 56 b returns the ascertained address to the calling end-point 50 which confirms receipt of the address via an ACK request.
  • step 114 the calling end-point issues a new INVITE request to the address returned by the redirect server 56 b . If the call succeeds, the called end-point transmits an OK response, and the calling end-point 50 completes the handshake with an ACK request. Voice conversation between the users of the calling and called end-points 50 , 52 ensues in step 116 via their respective handsets 60 , 62 . All or portions of the received body routing information may also be provided to the called end-point in the body of the second INVITE message. This may be desirable, for example, to facilitate discussions between a customer representative at the called endpoint and a caller at the calling end-point 50 .
  • FIG. 5 is a functional block diagram for determining an address of a called end-point according to one embodiment of the invention.
  • a location service 100 receives an INVITE message 102 from the calling end-point 50 .
  • the message preferably contains header routing information 102 a preferably including “From:,” “To:,” “Contact:,” and other standardized routing fields set forth in RFC 2543 .
  • the message preferably also contains body routing information 102 b .
  • the body routing information 102 b includes information associated with the caller that may be used to deduce the caller's intent.
  • the location server 100 produces an address 106 as a function of the header routing information and body routing information.
  • the location server 100 may deduce the address using information in the location server 58 and database 57 and/or based on rules and processing logic for ascertaining a most appropriate address.
  • the address may be, for instance, a specific IP address.
  • the location service is preferably a software module residing in the server 56 .
  • the location service may be a software module residing in the location server 58 and/or database 57 .
  • the location service may be implemented in firmware, hardware, or in any combination of software, firmware, and/or hardware.
  • FIG. 6 is a flow diagram of a process for an IP telephony call setup according to one embodiment of the invention.
  • the process starts, and in step 210 , the calling end-point 50 determines if a request was received to initiate a call.
  • a caller may request that a call be initiated by, for example, selecting a click-to-call button from a web page provided by a browser resident in the calling end-point.
  • the calling end-point searches a data store, in step 212 , for the caller information.
  • the data store may be a PAM, disk drive, or any other storage device conventional in the art.
  • the calling end-point searches for a particular file-name, file extension, or the like, where caller information is known to be stored.
  • the calling end-point retrieves caller information that is relevant to the destination of the call and the associated server 56 .
  • the retrieved caller information is used to fill entries in the web page for user verification.
  • the web page may provide entries for a last called department, a last called agent, a product name, an account number, and the like, which the calling endpoint automatically fills out based on the retrieved caller information.
  • the user may modify the automatically filled information after review.
  • the calling end-point creates a call initiation request message, such as for example, a SIP INVITE message, with standard header routing information for transmitting to the server 56 .
  • the calling end-point in step 216 , includes the retrieved caller information in the body of the call initiation request message as body routing information.
  • the calling end-point transmits the call initiation request message including the header and body routing information to the server 56 .
  • the server 56 receives the call initiation request message and retrieves from it the header routing information and the body routing information.
  • the server 56 determines an address of the called end-point as a function of the header routing information and the body routing information. For example, while the header routing information may indicate a general customer service number in the “To:” field, the body routing information may indicate a particular department or agent to which to route the call.
  • the SIP server routes the call to an end-point based on the determined address.
  • step 224 a determination is made as to whether new caller information is to be written to the calling end-point 50 .
  • Such information may be provided by the server 56 , called endpoint 52 , or the calling end-point itself. If the answer is YES, the calling end-point stores the information in an appropriate storage location. The information is then used for routing future body routing information.

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  • Engineering & Computer Science (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Signal Processing (AREA)
  • Business, Economics & Management (AREA)
  • General Business, Economics & Management (AREA)
  • Multimedia (AREA)
  • Telephonic Communication Services (AREA)
  • Data Exchanges In Wide-Area Networks (AREA)
  • Mobile Radio Communication Systems (AREA)
US10/020,783 2001-04-03 2001-12-12 Call routing using information in session initiation protocol messages Abandoned US20020141404A1 (en)

Priority Applications (8)

Application Number Priority Date Filing Date Title
US10/020,783 US20020141404A1 (en) 2001-04-03 2001-12-12 Call routing using information in session initiation protocol messages
AT02400025T ATE363799T1 (de) 2001-04-03 2002-04-02 Anrufweglenkung mittels informationen aus sip- nachrichten
JP2002099938A JP4011950B2 (ja) 2001-04-03 2002-04-02 セッション開始プロトコルメッセージの情報を使用する通話のルーティング
EP02400025A EP1248440B1 (de) 2001-04-03 2002-04-02 Anrufweglenkung mittels Informationen aus sip-Nachrichten
DE60220329T DE60220329T2 (de) 2001-04-03 2002-04-02 Anrufweglenkung mittels Informationen aus sip-Nachrichten
CNB021272689A CN100428764C (zh) 2001-04-03 2002-04-03 使用会话启动协议消息中的信息的呼叫路由选择
US11/427,513 US8737272B2 (en) 2001-04-03 2006-06-29 Call routing using information in session initiation protocol messages
US14/263,727 US20140233560A1 (en) 2001-04-03 2014-04-28 Call routing using information in session initiation protocol messages

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US28114301P 2001-04-03 2001-04-03
US10/020,783 US20020141404A1 (en) 2001-04-03 2001-12-12 Call routing using information in session initiation protocol messages

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US11/427,513 Expired - Fee Related US8737272B2 (en) 2001-04-03 2006-06-29 Call routing using information in session initiation protocol messages
US14/263,727 Abandoned US20140233560A1 (en) 2001-04-03 2014-04-28 Call routing using information in session initiation protocol messages

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US14/263,727 Abandoned US20140233560A1 (en) 2001-04-03 2014-04-28 Call routing using information in session initiation protocol messages

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EP (1) EP1248440B1 (de)
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AT (1) ATE363799T1 (de)
DE (1) DE60220329T2 (de)

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US20140233560A1 (en) 2014-08-21
CN100428764C (zh) 2008-10-22
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CN1398103A (zh) 2003-02-19
JP4011950B2 (ja) 2007-11-21
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DE60220329D1 (de) 2007-07-12
JP2002335267A (ja) 2002-11-22
ATE363799T1 (de) 2007-06-15
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US20060250989A1 (en) 2006-11-09

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