KR20080031128A - System and method for originating a sip call via a circuit-switched network from a user equipment device - Google Patents

Abstract

A system and a method for originating an SIP call through a circuit-switched network from UE device are provided to enable an AS(Alternative Service) node to allocate a selected E.164 number dynamically to the URI of a called party received from the UE device, return the selected E.164 number to the UE device by an SIP 380 response message, and transmit the allocated E.164 number to the UE device by a call setting message, thereby routing the SIP call toward the called party. An SIP(Session Initiation Protocol) call routing method comprises the following steps of: transmitting an SIP invite message having call information associated with an SIP call to an IMS(Internet Protocol Multimedia Subsystem) network in response to the identification of a mobile originating call request for the SIP call, wherein the SIP invite message includes the URI(Uniform Resource Locator) of a called party(312); receiving an SIP 380 response message including E.164 number from the IMS network in response to the transmission of the SIP invite message(316); and transmitting a CS(Circuit-Switched) call setting message, which includes the E.164 number for the routing of the SIP call to the called party, to the IMS network for the SIP call after the reception of the SIP 380 response message(320).

Description

System and method for sending SIP call from user equipment device over circuit switched network {SYSTEM AND METHOD FOR ORIGINATING A SIP CALL VIA A CIRCUIT-SWITCHED NETWORK FROM A USER EQUIPMENT DEVICE}

This patent application is filed on April 25, 2007, filed with application number 11 / 740,102 entitled “SYSTEM AND METHOD FOR ORIGINATING A SIP CALL VIA A CIRCUIT-SWITCHED NETWORK FROM A USER EQUIPMENT DEVICE”. Filed March 3 and claims priority to Application No. 11 / 542,462, entitled “SYSTEM AND METHOD FOR MANAGING CALL CONTINUITY IN IMS NETWORK ENVIRONMENT USING SIP MESSAGING,” which is incorporated herein by reference.

This patent disclosure generally relates to call routing in a communication network. More specifically, but not limited to, the present disclosure discloses a system for managing call routing in a network environment including a circuit-switched (CS) network and an IP multimedia subsystem (IMS) network. And a method, wherein a CS-originated IP call (eg, based on session initiation protocol or SIP) is routed using the IMS network infrastructure.

Today's advanced communication devices can operate seamlessly in packet switched IP network domains (eg, using wireless LAN (WLAN) or Wi-MAX networks, etc.) as well as circuit switched cellular network domains. . To facilitate this capability, the current 3GPP (3 ^rd Generation Partnership project) standard, a communication device (or user equipment referred to as UE) is one using any one of a single domain IP- subscribers and conventional circuit-switched telephony subscribers It defines a new IP-based network architecture called IP multimedia subsystem (IMS) that can initiate calls to both. However, because no packet switched network is available or the network available in the packet switched domain does not support Voice-over-IP (VoIP) service, the wireless device, i. The situation may arise where a voice call can only be made on the called party by use. In this situation, when the called party is an IP-only subscriber and is identified by a Uniform Resource Indicator (URI), the originating UE may be able to make only E.164 number-based calls while operating in the circuit switched domain, thus making the originating UE an IP-based call. It may not be possible to form.

A method and apparatus for sending a Session Initiation Protocol (SIP) call from a user equipment (UE) device to a called party in a network environment including a circuit switched (CS) network and an IP multimedia subsystem (IMS) network. Is explained.

When a SIP call originates from a UE device in a CS network domain, a SIP Invite message containing a called party's SIP URI or Tel URI is sent from the UE device to an application server (AS) node in the IMS network. do. At the AS node, a pool of E.164 numbers is maintained as the IP multimedia routing number (IMRN) used to map to or associate with the called party URI. The AS node dynamically assigns an optional E.164 number to the called party's URI received from the UE device and returns it to the UE device in a SIP 380 (Alternative Service) Response message (SIP 380). do. The dynamically assigned E.164 number is then sent from the UE device in a call set up message for identification of the URI and other appropriate call information at the AS node. Thus, the dynamically assigned E.164 number can be used to route SIP calls towards the called party via queries of the SIP URI-IMRN mapping, which can then be released back to the IMRN's pool for later use. . To verify whether the call refnum associated with the call remains valid (eg has not timed out) or whether the dynamically allocated IMRN remains valid (eg has not timed out) Timers suitable for the device and AS node endpoints may be provided. Optionally, the distributed IMRN can be quarantined for a period of time.

According to the present invention, a SIP call can be sent from a user equipment (UE) device to a called party in a network environment including a circuit switched (CS) network and an IP multimedia subsystem (IMS) network.

A more complete understanding of the embodiments of the present invention may be made with reference to the following detailed description in conjunction with the accompanying drawings.

Described herein are methods and apparatus for sending a Session Initiation Protocol (SIP) call from a user equipment (UE) device to a called party in a network environment including a circuit switched (CS) network and an IP multimedia subsystem (IMS) network. When a SIP call is originated from a UE device in the CS network domain, a SIP invitation message containing the called party's SIP URI or Tel URI is sent from the UE device to an application server (AS) node in the IMS network. At the AS node, a pool of E.164 numbers is maintained as the IP multimedia routing number (IMRN) used to map to or associate with the called party URI. The AS node dynamically assigns a selection E.164 number to the called party's URI received from the UE device and returns it to the UE device in a SIP 380 (Alternative Service) response message. The dynamically assigned E.164 number is then sent from the UE device in a call setup message for identification of the URI and other appropriate call information at the AS node. Thus, the dynamically assigned E.164 number is used to route the SIP call towards the called party via a query of the SIP URI-IMRN mapping, which can then be distributed back to the pool of IMRN for later use. To verify whether the call refnum associated with the call remains valid (eg has not timed out) or whether the dynamically allocated IMRN remains valid (eg has not timed out) Timers suitable for the device and AS node endpoints may be provided. Optionally, the deployed IMRN can be blocked for a period of time.

The system and method of the present invention are now described with reference to various examples of how embodiments can best be formed and used. Like reference numerals are used to designate similar or corresponding parts throughout the description and the various drawings, and various elements are not necessarily drawn to scale. Referring now to the drawings, and more specifically to FIG. 1, an exemplary network environment 100 in which embodiments of the present invention for routing SIP calls originated by UE devices in a circuit switched network or domain may be implemented. Shown. As shown, the network environment 100 includes an access space 104 that includes a number of access technologies available for the plurality of UE devices 102-1 through 102-N. For the purposes of the present disclosure, the UE device may be any tethered or untethered communication device, and any personal computer (eg, desktop, laptop, palmtop, or hand) with a suitable wireless modem. Handheld computing devices) or mobile communication devices (e.g., cellular phones or handheld devices capable of sending and receiving messages, web browsing, etc.), or email, video mail, Internet access, integrated data access, messaging, calendaring And any enhanced PDA device or integrated information appliance capable of utilizing scheduling, information management, and the like. Preferably, the UE device can operate in multiple modes capable of participating in both switched circuit (CS) as well as packet switched (PS) communications, and converts from one communication mode to another without loss of continuity can do.

Access space 104 may be comprised of both CS and PS networks that may involve wireless technology, wired technology, broadband access technology, and the like. For example, reference numeral 106 is a GSM (Global System for Mobile Communications) network, and CDMA (Code Division Multiple Access) refers to a wireless technology such as a network, however, its teachings may be any 3GPP (3 ^rd Generation Partnership Project) compatible It is contemplated that cellular networks (eg, 3GPP or 3GPP2) may also be extended. Reference numeral 108 denotes a fixed network, such as a DSL, cable broadband, etc., as well as a wireless local area network or a broadband access network including a WLAN, Wi-MAX network. Also illustrated as part of the access space 104 is a conventional wired PSTN infrastructure 110.

IP Multimedia Subsystem (IMS) core network 112 is connected to the various access networks described above, including any CS-based network. As is well known, the IMS standard defined by 3GPP is designed to allow service providers to manage the various services that can be delivered over IP over any network type, where IP is used for bearer traffic and It is used to transport both SIP-based signaling traffic. In a broad sense, IMS is a framework for managing applications (ie services) and networks (ie access) that can provide multimedia services. IMS is a network element that delivers services that subscribers use to “application servers,” for example, voice call continuity (VCC), push-to-talk (PTT), and IMS Centralized Services (ICS). It is defined as. IMS requires each application server (AS) to have, for example, subscriber profile, IMS mobility, network access, authentication, service authorization, billing and billing, inter-operator functions, and interoperation with legacy phone networks. Manage applications by defining common control components.

While IMS is defined primarily by the 3GPP standard body addressing GSM networks, it should be understood that another group, 3GPP2, is involved in defining a very similar architecture called the Multimedia Domain (MMD). Since MMD is essentially an IMS for a CDMA network, and MMD and IMS are generally equivalent, the term “IMS” in the present disclosure can be used to generically refer to both IMS and MMD where applicable.

With continued reference to FIG. 1, reference numerals 114-1 through 114 -N, as mentioned above, refer to a plurality of AS nodes operable to support various services, such as VCC, PTT, ICS, and the like. Also, to implement call control of SIP calls using CS as a voice bearer, one of the AS nodes, for example AS 114- (N-1), is referred to as an IMS Centralized Services Control Function (ICCF). Can be provided to implement functionality. ICCF resides in the home IMS network and can act as an IMS application server element that tracks all call sessions and associated mobile VoIP bearer traffic between the CS and IMS domains. The functionality of AS 114- (N-1) in the pending US patent application Ser. No. 11 / 328,875, filed January 10, 2006, entitled “SYSTEM AND METHOD FOR MANAGING CALL ROUTING IN A NETWORK ENVIRONMENT INCLUDING IMS”. Additional details regarding this can be found, see here.

In addition, another AS node, AS 114-N, is an IP that is sent by one of the UE devices in the CS domain while the connection in the PS domain is not available or the available PS network cannot support VoIP services. It is provided as part of the core IMS network 112 to facilitate routing of / SIP calls. As described in more detail below, a suitable database architecture is associated with the AS 114-N to configure and manage a pool of IP Multimedia Routing Numbers (IMRNs) to which a select IMRN can be dynamically assigned for SIP call routing [e.g. For example, DB 122, a timer mechanism (eg, timer 124), and appropriate logic 126 may be provided.

In accordance with the teachings of the present disclosure, preferably, the AS 114-N maintains a pool of E.164 numbers operable as next steps, i.e., an IMRN terminated on an AS node, with the optional E.164 number limited thereto. Is not received but is received in a SIP invitation message that includes the recipient's SIP URI or Tel URI, P-Preferred Identity, Privacy Indication, and Network Access Info header. Which can be mapped to information; Dynamically assigns an optional E.164 number to the called party's URI (eg, SIP URI or Tel URI) and other received information, and sends the optional E.164 number via a SIP 380 (Alternative Service) response message. Providing to the originating UE device; Verifying that the selected E.164 number has not timed out when the selected E.164 number is returned to the AS 114-N (via conventional CS call setup) to conduct a SIP call session to the called party. ; And optionally, redistributing the selection E.164 number to a pool for later use to block the selection E.164 number for a predetermined time; call continuity control function (CCCF) 116, network Suitable logic / structure / software / firmware module (s) are provided, such as domain section (118), and gsm gsm service capability feature (SCF) 120.

Note that E.164 refers to the International Telecommunications Union (ITU) telephone numbering plan, which specifies how and by whom telephone numbers are assigned. The format and symbols of E.164 telephone numbers are specified, for example, in ITU standard E.123. To manage a pool of dynamically assignable IMRNs, an AS node (eg, AS 114-N) may be configured in a number of ways with respect to E.164 numbers. For example, a specific E.164 number may be provided as the "start address" number in the IMRN range. Another E.164 number can act as a range delimiter for the IMRN range. To allow flexibility, it may be desirable to provide a pool of various IMRNs to be constructed from various number ranges. In addition, in order to ensure that the assigned IMRN remains valid (e.g., has not timed out, i.e. used within a suitable time limit), or that a suitable blocking time is applied, the appropriate timer mechanism (s) is applied to the AS. It may be implemented at 114-N. As will be described in detail below, management of the timer associated with the IMRN in AS 114-N and the timer associated with the call reference number in the originating UE device can be used to conduct a SIP call by a UE device operating in the CS domain. Allows dynamic provisioning of an IMRN.

2 shows a flowchart of an exemplary embodiment of the overall method of the present invention for making a CS originating SIP call by a UE device to a called party identified by a URI (eg, a SIP URI or a Tel URI). . The SIP call is initiated either by the end user of the UE device, or by the calling party. Preferably, the originator enters the URI through a suitable interface (eg, MMI), or selects the URI from a list stored in the UE. As is well known, a typical SIP address may take the form sip : < username > @ < hostname > , for example the name SIP : Session Initiation Protocol RFC 3261 and its name Obtaining and Using Globally Routable User Agent ( UA ) URIs ( GRUU ) in the Session Initiation Additional systematic elements and parameters may be included, such as those described in the Internet draft, Protocol ( SIP ) (draft-ieth-sip-gruu-06) (expiration: April 23, 2006).

Note that “Tel URI” is currently defined in Request For Comments (RFC) 3966 (December 2004). Some examples of Tel URIs are as follows: (1) tel: + 1-201-555-0123: This URI points to a US telephone number. Hyphens are included to make it easier for people to read the number, which separates the country, area code, and subscriber number; (2) tel: 7042; phone-context = example.com: URI indicates a local telephone number valid within the context “example.com”; And (3) tel: 863-1234; phone-context = + 1-914-555: The URI indicates a local telephone number valid within a specific telephone local station number.

At block 202, various pieces of information about a SIP call (collectively referred to herein as “call information”) are provided from a UE device to a network node deployed in an IMS network. Call information includes the call reference number associated with the call, the SIP URI of the called party (or B-URI), the Opaque parameter (if applicable), the GRID parameter (if applicable), and additional URI related information (e.g. , A display name), an originating SIP URI (or A-URI), an OPAR parameter, a privacy indicator, a network access info header, and the like. If the calling party sends a B-URI containing the OPAR and GRID parameters as well as the Address of Record (AOR), they will be provided as part of the call information. In addition, if the calling party sends its own URI including the AOR, OPAR and GRID parameters, they will also be provided as call information.

As described above, a timer used to monitor at least a portion of the call information sent by the originating UE device may be started at the UE device. In particular, a timer may be implemented to monitor the elapsed time since a particular call reference number was generated and sent to the IMS network node. At the IMS network node, the IMRN selected from the pool of IMRNs is dynamically associated with the call reference number, where the IMRN is mapped to at least a portion of the call information, eg, the SIP URI of the called party (block 204). In some embodiments, the IMRN may be mapped to all received SIP call information. In addition, a timer may be started at the network node to monitor time-to-live variables associated with the dynamically allocated IMRN.

The dynamically assigned IMRN is then provided to the UE device via a SIP 380 (Alternative Service) response message. Upon receipt of the SIP 380 (Alternative Service) response message, the elapsed time associated with the call reference number is monitored to ensure that it is not invalidated (block 206). If the elapsed time meets the selection condition, for example within the active time value, the dynamically allocated IMRN is accepted by the UE device (block 208). In response, the appropriate call setup is then initiated by the UE device using the dynamic IMRN, whereby the accepted IMRN is terminated at the AS node and returned to the AS node. Upon receipt of the IMRN at the AS node, its active time variable is monitored to ensure that the IMRN has not timed out (block 210). The SIP session with the called party is then generated by generating and sending a SIP invite message (e.g., by inserting the A-side URI, privacy indicator, B-side URI, OPAR parameter, etc. in the SIP invite message and sending it to the called party). The recipient's SIP URI or Tel URI (and any other suitable SIP information originally received that is mapped to a dynamically assigned IMRN) is used by the AS node for implementation. In one implementation, the dynamic IMRN can optionally be returned back to the pool of IMRN and blocked for some time before it is reused or made available for later use (block 212).

Based on the foregoing, those skilled in the art will receive appropriate logic at the AS node when call information, i.e., the SIP URI or Tel URI, call reference number, etc. of the called party, is transmitted by the UE device to the serving AS node. It will be appreciated that a record may be generated that maps the received call information to an E.164 based IMRN to be sent back to the UE device. By correlating the IMRN with call reference information, the UE establishes a call using the IMRN terminated on the AS node. Then, to establish the SIP session with the called party (i.e., between the calling party (UE device) identified by the A side address and the called party identified by the B side address) to retrieve the called party's URI against the record. IMRN is queried.

In addition, message flow between a UE device and an AS node in a home IMS network can be mediated through a number of other suitable network infrastructure elements, and can be implemented in a number of ways depending on the device characteristics as well as the network characteristics and protocol to be used. It should be recognized by those skilled in the art. Typically, the message flow is a mobile switching center (MSC) and media gateway control function (MGCF) element disposed between the UE device and its home IMS AS node operable to facilitate CS outgoing SIP calls. It can be mediated through a network element such as

3A illustrates a message flow embodiment 300A for making CS outgoing SIP calls based on dynamic IMRN assignment when SIP messaging is implemented. The wireless UE device 302 with the functionality of the CS domain and IMS domain mode is directed towards the application server (AS) node 308 in response to detecting that a SIP call is originating from the UE device 302 in the CS domain. Is operable to generate a SIP invitation message 312. As described above, the SIP invitation message 312 may include a call reference number, a SIP URI of the called party, additional URI information, and other such as, for example, the A side AOR of the P_Priority identity, the privacy indicator, the Opark parameter, the GRID parameter, and the like. Contains applicable call information. As described above, the originating party initiates a call by entering a URI (or SIP address) or Tel URI via a suitable interface (e.g., MMI) or by selecting it from a list stored at the UE.

The SIP invitation message 312 may determine whether the message is for a circuit switched (CS) mobile-originated (MO) call (ie, if the UE device 302 intends to make this call through the CS domain). It may further include an indicator of the indicator field indicating whether or not. For example, a new network-access-info value such as “GERAN-CS” can be used. Alternatively, a new feature tag or new URI parameter may be provided in the SIP invitation message. However, note that the indication can only be assumed to include the SIP URI or Tel URI of the called party (“B side”) in the SIP invitation message. In one preferred embodiment, the TARGET address of the SIP invitation message is filled with either the SIP URI or Tel URI of the called party or B side. In this case, the SIP URI field of the SIP invitation message is filled with the public service identifier (PSI) of the AS node. The cause value of the SIP invitation message will be set appropriately to indicate that a radio bearer channel for the session will be established through the CS domain.

An appropriate timer mechanism 310 may be initiated at the UE device to monitor the activity time variable associated with the call reference number. It is to be appreciated that this operation may be provided in addition to the normal SIP timer since this operation is known to provide a SIP 380 response with specific information within a specific timeframe.

In response to the invitation message 312, which may be mediated via the I-CSCF and / or S-CSCF node, the AS node 308 deployed in the user's home IMS network as described above may be configured to use an appropriate SIP 380 (Alternative Service). ) Initiate the SIP-URI logic 313 to generate and populate a response message (eg, a SIP 380 response message). If the user is allowed to make a SIP call and verifies that the invitation message includes the appropriate CS MO indicator, the network node (in this example, the IMS AS node) may call the call information or parameter (eg, A of P_Priority identity). Dynamically assign a select IMRN mapped to the side AOR, privacy indicator, OPAR parameter, GRID parameter, etc.) and return it back to the UE device 302 via SIP 380 message 316. Again, the dynamically assigned IMRN may be referred to as an IMS Centralized Service Routing Number or “ICSRN”. Dialog information included in the invitation header or the body of the invitation may be used to correlate the call.

An appropriate timer mechanism may be started at the AS node 308 to monitor the activity time variable associated with the dynamically allocated IMRN (block 314). After verifying that the call reference has not timed out based on the timer mechanism of the UE device, in response to receiving the SIP 380 response message 316, the UE device 302 establishes a call including a dynamic IMRN (or ICSRN). Initiate message 320. In response, MSC 304 generates an Initial Address Message (IAM) 322 toward MGCF 306. A SIP invitation message 324 containing an IMRN is generated by the MGCF 306 towards the AS node 308, which then initiates the establishment of a SIP session or call to the called party (not shown). Use IMRN mapping for this. Although not specifically shown here, of course, various intermediate SIP messages and resource allocation / reservation negotiations may occur between the MGCF 306 and the called party following the SIP invite 324. In addition, additional ISUP messaging that may occur before the bearer path is established between the UE device 302 and the called party to be understood is not shown here.

Upon receipt of the dynamically allocated IMRN via the SIP invitation 324 at the AS node 308, the timer mechanism may be stopped to verify whether the IMRN has timed out (block 326). If timed out, the SIP invitation message may be discarded and the call routing process may end. If the IMRN has not timed out, the AS node 308 can establish a SIP session based on the IMRN correlation. After using the IMRN for correlation, it may be returned to the IMRN pool, and a block timer may be started to prevent further use of the IMRN until the block timer is stopped after a certain time (block 330).

As noted above, a timer mechanism at the device side (eg, using the timer mechanism 318) may also be used to ensure that the call reference number has not timed out, and the reference number is received from the network node. Used by the UE device to correlate with information (eg, dynamic IMRN). If the timer expires before the same reference number is received again from the network node, the UE device may retry the call process a predetermined number of times (eg, five attempts), after which the call if no response is received The procedure may be considered failed. In other words, if the UE device receives a reference number that is no longer valid, it may be discarded and the call procedure may end.

3B shows a message flow diagram 300B for a mobile originating SIP call by employing a SIP invite message with a SIP-URI in the request URI, where some intermediate node is illustrated in the home network 350. Like the flow chart embodiment 300A described above, the UE device 302 is operable to generate a SIP invitation message 356 towards the I-CSCF 352, where the SIP invitation message is included in the TARGET field. It includes the SIP-URI of the called party. This invitation message is delivered to the AS node 308 either directly as the SIP invitation 361 or via the S-CSCF 354 by the SIP invitation messages 358 and 360. As described above, the SIP 380 (Alternative Service) response message 364 with ICSRN is generated by the AS node 308 towards the S-CSCF 354 and then via the SIP response 366 the UE device ( 302). Call setup message 368 with ICSRN is provided to MSC 304 and initiates a CS origination procedure 370. IAM messaging 372 from MSC 304 to MGCF 306 is operable to generate SIP invite 374 towards I-CSCF 352, which is an AS node as invitation message 380 with ICSRN. It may be delivered directly to 308. Alternatively, I-CSCF 352 first provides SIP invite 376 to S-CSCF 354, and then S-CSCF 354 forwards SIP invite 378 to AS node 308. . Somehow, once the ICSRN is received at the AS node 308, a suitable call correlation is established to establish a SIP call between the UE and the called party.

In one variation of the technique described in connection with FIGS. 2, 3A and 3B, it is noted that the E.164 number is not only dynamically assigned, but merely identified, calculated or otherwise selected according to any suitable algorithm. Pay attention.

In more detail the technique of the present invention, when the UE device detects that it should call a CS call origination, it can generate and send a SIP invitation message to the R-URI known to terminate at the IMS central service node. In this case, the target parameter of the SIP invite message is filled with the B side address (SIP URI or Tel URI), and the Cause value of the SIP invite message is set to indicate that the call should be set through the CS. . Alternatively, the R-URI may be populated with the B-side address, and the indicator field indicates whether the message is for a circuit switched (CS) mobile originating (MO) call (ie, the UE device 302 makes this call through the CS domain). It may further include an indicator indicating whether or not to form]. For example, a new network-access-info value such as “GERAN-CS” can be used. Alternatively, a new feature tag or new URI parameter can be provided in the SIP invitation message. In the first case, the SIP invitation message includes the GRUU of the UE / Public ID combination. The P-Preferred-ID is set to a calling line identity (CLI) associated with the user or subscriber of the UE device for identification in the CS network. The B side public user address (Tel URI, SIP URI) is set in the SIP URI target parameter, and the cause value indicates CS bearer request = YYY.

Note that when the target parameter is used to convey the B side address, the SIP R-URI may be one of many that have been provisioned to the UE device to indicate the ICCF. In such a case, the UE device may select one of them at random and the URI may have some indicia identifying the priority.

An example is provided below:

If the R-URI is set to B side, the invitation includes the GRUU of the UE / public ID combination. The P-Priority-ID is set to the originating line identity that the user wants to be known in the CS network. The network-access-info header is set to a value indicating that the call is SIP controlled but the radio bearer is going through the CS domain, in this example the setting is 3GPP-GERAN-CS and in another example 3GPP-UTRAN-CS Can be.

An example is provided below:

Upon receiving a 380 (Alternative Service) response to the SIP Invite message, the UE establishes a CS call using the ICSRN provided in the 380 (Alternative Service) response as an E.164 number. This E.164 number may be in the contact header of 380 (alternative service) or in fact in the XML body.

Upon receipt of the R-URI, the S-SCSF recognizes that the invitation is for an IMS center service node that has been assigned to the UE and sends it to this AS node. The IMS Central Service Node configuration information may comprise (a) an ICSRN E.164 start address number; And (b) the number of the ICSRN to be assigned or the latest E.164 start address number. To allow flexibility in the numbering plan, more occurrences of (a) and (b) can occur so that a pool of ICSRNs can be allocated from various number ranges. In addition to (a) and (b), other configuration information may include (c) an active time for the ICSRN to act; And (d) the blocking time of the ICSRN (how long cannot be used after the ICSRN is reassigned to the ICSRN pool).

 Describe the behavior at the IMS-centric service node. When the IMS Central Service Node receives an invitation that includes an R-URI, it checks the R-URI to determine whether the R-URI is associated with a request to initiate a MO call via CS. In an alternative implementation, the IMS central service node will examine the P-Access-Network-Info header. If it is set to GERAN-CS or some other value to indicate call-set-up via CS, then the IMS central service node assumes that the received invitation ends here.

The IMS Central Service Node will assign the ICSRN to the received GRUU. The following shows possible mappings to other information elements of ICSRN.

The IMS Central Service Node will respond with an 380 (Alternative Service) response to the INVITE request. An example of coding of this response, including ICSRN, radio access type to be handed over, is shown below, where the radio access type is "IEEE-802.11" / "IEEE-802.11a" / "IEEE-802.11b" / " IEEE-802.11g "/" 3GPP-GERAN "/" 3GPP-UTRAN-FDD "/" 3GPP-UTRAN-TDD "/" ADSL "/" ADSL2 "/" ADSL2 + "/" RADSL "/" SDSL "/" HDSL "/" HDSL2 "/" G.SHDSL "/" VDSL "/" IDSL "/" 3GPP2-1X "/" 3GPP2-1x-HRPD "/" DOCSIS "/ token, 3GPP -GERAN CS, 3GPP -GERAN PS, 3GPP-UTRAN CS, 3GPP-UTRAN PS, 802.11b, 802.11a, 802.11g, EVDO, CDMA1X, WiMAX, and the like. ICSRN is also included in the contact header. For the assignment of ICSRN, it will start a timer that has ICSRN as R-URI and that the origination is released upon receipt of the invitation from MGCF. If the timer expires, the ICSRN is put in the blocking pool.

If the IMS Central Service Node receives a subsequent request from the same UE identified by the GRUU at the invitation, then the IMS Central Service Node will: (a) retransmit the same ICSRN and reset the timer; (b) assigning a new ICSRN, starting a timer associated with the ICSRN and putting the old one into the blocking pool; And (c) reject the request completely and put the old into the blocking pool.

The following is example code that codes the 380 alternative service response.

4 is a block diagram of an embodiment of a mobile communication device operable as a wireless UE device, eg, a UE 302 for the purposes of the disclosure. Embodiments of the UE 302 may include an arrangement similar to that shown in FIG. 4, but it will be appreciated by those skilled in the art that there may be many variations and modifications in hardware, software or firmware for the various modules shown. Thus, the arrangement of FIG. 4 should be considered as illustrative rather than limiting to the embodiment of the present invention. The microprocessor 402, which provides overall control of the embodiment of the UE 302, is operatively connected to a communication subsystem 404 capable of multi-mode communication (eg, CS domain, IP domain such as IMS, etc.). . The communication subsystem 404 generally includes one or more receivers 408 and one or more transmitters 414, as well as associated components such as one or more local oscillator (LO) modules 410, and digital signal processors (DSPs). a processing module such as a digital signal processor (412). As will be apparent to those skilled in the telecommunications art, the particular design of the communication module 404 may depend on the communication network (eg, CDMA network, GSM network, WLAN, etc.) for which the mobile device is intended to operate. However, regardless of the particular design, the signal received by the antenna 406 via a suitable access infrastructure 405 (eg, cellular base station tower, WLAN hot spot, etc.) is provided to the receiver 408, and the receiver 408 may perform these common receiver functions such as signal amplification, frequency down conversion, filtering, channel selection, analog-to-digital (A / D) conversion, and the like. Similarly, signals to be transmitted, including modulation and encoding, for example, are processed by the DSP 412, and are subjected to digital-to-analog (D / A) conversion, frequency up conversion, filtering, amplification, and antenna 416 Provided to the transmitter 414 for transmission over the air interface.

The microprocessor 402 may also include auxiliary input / output (418), serial port 420, display 422, keyboard / keypad 424, speakers 426, microphone 428, RAM ( random access memory) 430, short-range communication subsystem 432, and any other device subsystem, collectively referred to as 433, such as an additional device subsystem, such as a timer mechanism. In this example, display 422, keyboard / keypad 424, speaker 426, microphone 428 are part of the user interface of the mobile communication device through which calls can be initiated and maintained by the end user. Can be. To control access, a SIM / RUIM 434 may also be provided to communicate with the microprocessor 402. In one implementation, the SIM / RUIM interface 434 can operate as a SIM / RUIM card with multiple key configurations 444 and other information 446 such as identification as well as identification and subscriber related data. Do. Note that in the absence of a SIM / RUIM, the UE device is referred to as mobile equipment (ME), but the technique of the present invention is applicable to any device.

Operating system software and applicable service logic software may be implemented in a persistent storage module (ie, non-volatile storage) such as flash memory 435. In one implementation, the flash memory 435 is not only a storage area 436 for various areas, for example computer programs (eg, service processing logic), but also device states 437, address books 439, and the like. Data storage area, such as personal information manager (PIM) data 441, and other data storage areas, collectively referred to as 443. FIG. A delivery stack 445 may be provided to implement one or more suitable wireless packet delivery protocols. In addition, as described above, a call control logic module 448 for proper call messaging processing in accordance with the techniques of the present invention may be used to effect SIP-URI and call reference ID generation, validity, validation, correlation with IMRN, and the like. Is provided.

Thus, a method and apparatus for sending a Session Initiation Protocol (SIP) call from a user equipment (UE) device to a called party in a network environment including a circuit switched (CS) network and an IP multimedia subsystem (IMS) network have been described. When a SIP call is originated from a UE device in a CS network domain, a SIP invitation message containing a called party's SIP URI or Tele URI is sent from the UE device to an application server (AS) node in the IMS network. At the AS node, a pool of E.164 numbers is maintained as the IP Multimedia Routing Number (IMRN) used to map to or associate with the called party URI. Thus, the AS node dynamically assigns a selection E.164 number to the called party's URI received from the UE device and returns it to the UE device in a SIP 380 (Alternative Service) response message. The dynamically assigned E.164 number is then sent from the UE device in a call setup message for identification of the URI at the AS node. Thus, the dynamically assigned E.164 number is used to route the SIP call towards the called party via a query of the URI-IMRN mapping, which can then be distributed back to the pool of IMRN for later use. To verify whether the call refnum associated with the call remains valid (eg has not timed out) or whether the dynamically allocated IMRN remains valid (eg has not timed out) Timers suitable for the device and AS node endpoints may be provided. Optionally, the deployed IMRN can be blocked for a period of time.

In an AS node, the techniques of the present invention include maintaining access to a pool of E.164 numbers as an IP Multimedia Routing Number (IMRN); Receiving a SIP invite message for a SIP call originating from a user equipment (UE) device via a circuit switched network domain, the SIP invite message including a called SIP SIP or Tel URI; Selecting one of the E.164 numbers and storing a mapping between the selected E.164 number and call information; Sending a SIP 380 (Alternative Service) response message to the UE device that includes the selected E.164 number in response to receiving a SIP invitation message; And after sending a SIP 380 (Alternative Service) response message, receiving a call setup message comprising the selected E.164 number from a UE device for a SIP call; Identifying a URI identified from a call establishment message using the selected E.164 number via a stored mapping; And establishing a SIP session with the called party using the URI identified through the stored mapping.

The operation and configuration of the embodiments of the present invention will be apparent from the above detailed description. While illustrative embodiments have been shown and described which may be characterized as being preferred, it should be understood that various modifications and changes can be made without departing from the scope of the invention as set forth in the following claims.

1 illustrates a network environment including a circuit switched network infrastructure and an IM multimedia subsystem (IMS) infrastructure in which embodiments of the present invention may be practiced.

2 depicts a flow diagram associated with one or more exemplary embodiments of the present invention.

3A and 3B illustrate an example of originating a SIP call by employing a SIP invite message having a called party's SIP-URI in a request URI for mapping with an dynamically allocated IP multimedia routing number (IMRN) at an application server (AS) node. Shows a typical message flow diagram.

4 shows a block diagram of an embodiment of a communication device operable for the purposes of the present invention.

Claims (20)

A method for routing Session Initiation Protocol (SIP) calls in a network environment including a circuit switched (CS) network and an Internet Protocol (IP) multimedia subsystem (IMS) network, wherein the SIP call is a SIP Uniform Resource Indicator (SUR). Or originating from the CS network by the mobile communication device towards the called party identified by the Tel URI. In response to identifying a mobile outgoing call request for a SIP call, sending a SIP Invite message with call information associated with the SIP call to an IMS network, wherein the SIP invitation message includes a called party's URI; That, step; Receiving a SIP 380 (Alternative Service) Response message containing an E.164 number from an IMS network in response to sending the SIP invitation message; And After receiving the SIP 380 (Alternative Service) response message, sending a CS call setup message to the IMS network for the SIP call, the CS call setup message comprising an E.164 number for routing of the SIP call to the called party; SIP call routing method comprising a. The method according to claim 1, And filling the TARGET field of the SIP invitation message with a URI of the called party. The method according to claim 1, Filling the URI field of the SIP invitation message with a public service identifier (PSI) of an application server (AS) of an IMS network. The method according to claim 1, Populating the SIP invite message with an indicator field indicating whether the SIP invite message is for a circuit switched mobile outgoing call from the mobile communication device. The method according to claim 1, Identification of the mobile outgoing call request comprising the called party's URI comprises receiving a selection of the URI in a user interface of the mobile communication device. The method according to claim 1, The E.164 number is an E.164 number dynamically selected from a pool of E.164 numbers in an IMS network, and mapped to a URI in an IMS network. The method according to claim 1, A SIP program embodied in a computer program product comprising a computer readable medium and computer instructions stored on the computer readable medium, the computer instructions executable by one or more processors of a wireless device for performing the method. Routing method. A mobile communication device operable to originate a SIP call in a network environment including a circuit switched (CS) network and an Internet Protocol (IP) multimedia subsystem (IMS) network, the call being a SIP Uniform Resource Indicator (URI) or Tele URI. A mobile communication device which is directed towards a called party identified by One or more processors; And A wireless transceiver coupled to the one or more processors; Including, The wireless transceiver is adapted for communication via a CS network; The one or more processors, In response to the identification of the mobile outgoing call request for the SIP call, a SIP invite message having a call information associated with the SIP call and including a called party's URI to the IMS network via the wireless transceiver and the CS network. Send; Receive via the wireless transceiver and a CS network a SIP 380 (Alternative Service) Response message comprising an E.164 number from an IMS network in response to sending the SIP invitation message; After receiving the SIP 380 (Alternative Service) response message, a CS call setup message comprising an E.164 number for routing of the SIP call to a called party is sent to the IMS network for the SIP call via the wireless transceiver and a CS network. To send to; The mobile communication device being adapted. The method according to claim 8, And the one or more processors are further operable to populate the TARGET field of the SIP invitation message with the called party's URI. The method according to claim 8, And the one or more processors are further operable to populate the URI field of the SIP invitation message with a public service identifier (PSI) of an application server (AS) of an IMS network. The method according to claim 8, And the one or more processors are further operable to populate the SIP invitation message with an indicator field indicating whether the SIP invitation message is for a circuit switched mobile outgoing call from the mobile communications device. The method according to claim 8, And the one or more processors are operable to send the SIP invitation message in response to the detection of a user initiated call request at a wireless device that includes the called party's URI. A method operable with an application server (AS) node for making a Session Initiation Protocol (SIP) call in a network environment including a circuit switched (CS) network and an Internet Protocol (IP) multimedia subsystem (IMS) network, the SIP In the SIP call execution method is that the call is directed to the called party identified by the SIP Uniform Resource Indicator (URI) or Tel URI, Maintaining access to the pool of E.164 numbers; Receiving a SIP invite message for a SIP call originating from a mobile communication device over a CS network, wherein the SIP invite message has call information including a called party's URI; Selecting one of the E.164 numbers from the pool and storing a mapping between the selected E.164 number and call information for the SIP call; Sending a SIP 380 (Alternative Service) Response message to the mobile communication device that includes the selected E.164 number in response to receiving the SIP invitation message; After sending the SIP 380 (Alternative Service) response message, receiving a CS call setup message from the mobile communication device for a SIP call comprising the selected E.164 number; Identifying the URI identified from the CS call setup message using the selected E.164 number through the stored mapping; And Establishing a SIP session with a called party using the URI identified through the stored mapping; SIP call implementation method comprising a. The method according to claim 13, And identifying the called party's URI in the TARGET field of the SIP invitation message. The method according to claim 13, And the URI field of the SIP invitation message is filled with a public service identifier (PSI) of an AS node of an IMS network. The method according to claim 13, Reading an indicator of an indicator field in the SIP invitation message; Identifying whether the indicator is set to indicate a circuit switched mobile outgoing call from the mobile communication device; If the indicator is set to indicate the circuit switched mobile outgoing call, selecting, as a subsequent step, an E.164 number, storing a mapping, and sending a SIP 380 (Alternative Service) response message to the mobile communication device; Performing a step of causing a; And Otherwise, refraining from performing the subsequent step if the indicator is not set to indicate the circuit switched mobile outgoing call; SIP call implementation method further comprising. Application server (AS) used in Internet Protocol (IP) Multimedia Subsystem (IMS) networks to make Session Initiation Protocol (SIP) calls originated by mobile communication devices in a network environment including circuit switched (CS) networks. A node, wherein the call is directed toward a called party identified by a SIP Uniform Resource Indicator (URI) or Tele URI, Maintain access to the pool of E.164 numbers; Receive a SIP Invite message for a SIP call originating from the mobile communication device over the CS network, the call invitation having call information including the URI of the called party; Select one of the E.164 numbers from the pool to store a mapping between the selected E.164 number and call information for the SIP call; Send a SIP 380 (Alternative Service) Response message to the mobile communication device that includes the selected E.164 number in response to receiving the SIP invitation message; After sending the SIP 380 (Alternative Service) response message, receive a CS call setup message from the mobile communication device for a SIP call comprising the selected E.164 number; Identify the URI identified from the call establishment message using the selected E.164 number through the stored mapping; Establish a SIP session with a called party using the URI identified through the stored mapping; An application server node comprising one or more processors adapted. The method according to claim 17, Wherein the one or more processors are further adapted to identify the called party's URI in the TARGET field of the SIP invitation message. The method according to claim 17, Wherein the one or more processors are further adapted to select an E.164 number by dynamically assigning an E.164 number from the pool. The method according to claim 17, The one or more processors, Read an indicator of an indicator field in the SIP invitation message; Identify whether the indicator is set to indicate a circuit switched mobile outgoing call from the mobile communication device; If the indicator is set to indicate the circuit switched mobile outgoing call, selecting, as a subsequent step, an E.164 number, storing a mapping, and sending a SIP 380 (Alternative Service) response message to the mobile communication device; Performing a step of causing the; Otherwise, refrain from performing the subsequent step if the indicator is not set to indicate the circuit switched mobile outgoing call; Application server nodes that are more adapted.

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