Fast network SIP/SDP procedures for conference operations with optimization of network resources
Abstract
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Term
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Expired 1 April 2024, 2.5 years ago.
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20 claims: 12 independent, 8 dependent
- 13GPP無線会議呼動作のためのSIP/SDP(セッション開始プロトコル/セッション記述プロトコル)ベースの装置であって、 受信した要求に応じて、第1および第2の呼を少なくとも第3の呼を伴う会議呼に接続するSIP/SDPベースのネットワークコントローラを含み、 前記SIP/SDPベースのネットワークコントローラは、ネットワークリソースについて交渉し、前記ネットワークリソースを割り振り、そして前記ネットワークリソースを再構成するように構築され、前記SIP/SDPベースのネットワークコントローラはSIPバック-ツー-バックユーザエージェントを含み、前記SIP/SDPベースのネットワークコントローラは、第1ユーザから会議呼要求を受信するよう構成されており、前記会議呼要求はSIP/SDP信号指示により送信され、前記会議呼要求の受信に応じて、前記SIP/SDPベースのネットワークコントローラは、SIPユーザエージェントを含むリソースコントローラとSIP/SDP信号指示により通信し、前記リソースコントローラは、前記第1ユーザと第2ユーザと第3ユーザとの間の各呼区間に対する少なくともコーデック再ネゴシエーションおよびベアラ再構成を前記SIP/SDPベースのネットワークコントローラが独立して行うことを可能にする少なくとも1つのコーデックを前記ネットワークコントローラに提供し、前記リソースコントローラは、ベアラ経路リソースを、前記第1ユーザと前記第2ユーザと前記第3ユーザとの間の各呼区間に対する前記SIP/SDPベースのネットワークコントローラに通信し、前記ベアラ経路リソースは、インタネットプロトコルアドレス、使用可能なコーデック及び指向性のうちの少なくとも1つを含み、そして、前記SIP/SDPベースのネットワークコントローラは、前記第1ユーザと前記第2ユーザと前記第3ユーザとの間の各呼区間に対するベアラ経路リソースについて同時に交渉する、装置。 A SIP / SDP (Session Initiation Protocol / Session Description Protocol) based device for 3GPP wireless conference call operation. Depending on the request received、First and second callsLessSIP / SD to connect to a conference call with at least a third callP-beInternetLycium chinenseTrollaIncluding, Said SIP / SDP based networkLycium chinenseTrolla、NetworkChestnutSourceaboutNegotiateThe network resourceAllocate,And the network resourceBuilt to reconfigure, The SIP / SDP-based network controller includes a SIP back-to-back user agent.The SIP / SDP-based network controller is configured to receive a conference call request from a first user, and the conference call request is transmitted by a SIP / SDP signal instruction.In response to the reception of the conference call request, the SIP / SDP-based network controller communicates with the resource controller including the SIP user agent by SIP / SDP signal instruction, and the resource controller communicates with the first user and the second user. Provides the network controller with at least one codec that allows the SIP / SDP-based network controller to independently perform at least codec renegotiation and bearer reconfiguration for each call interval between and a third user. , The resource controller communicates the bearer route resource to the SIP / SDP-based network controller for each call interval between the first user, the second user, and the third user, and the bearer route resource Includes at least one of, Internet Protocol Address, Available Codecs and Directionality, andThe SIP / SDP-based network controller simultaneously negotiates bearer route resources for each call interval between the first user, the second user, and the third user.apparatus.
- 6A method of performing SIP / SDP (Session Initiation Protocol / Session Description Protocol) based 3GPP radio conference call operation. The step of putting a call between the first user and at least the second user on holdIncludingThe first user is placed on the action call and at least the second user is placed on the hold call.,further, With at least a third userUkaiDiscussionCallRequestBy SIP / SDP signal instruction from the first user to the network controllerStep to sendThe network controller includes a SIP back-to-back user agent, and furtherThe first user, the second user, and the third userFor each call interval betweenTo doAt least codec renegotiation and bearer reconstructionSaid networkIndependent controllerTo doBy SIP / SDP signal instructions to enable、Said networkLycium chinenseWith TrollaIncluding SIP user agentSteps to communicate with the resource controllerThe resource controller communicates the bearer route resource for each call interval between the first user, the second user, and the third user to the network controller, and the bearer route resource is an internet protocol address. Includes at least one of the available codecs and directivity, andThe resource controller simultaneously negotiates a bearer route resource for each call interval between the first user, the second user, and the third user.Method. SIP/SDP(セッション開始プロトコル/セッション記述プロトコル)ベースの3GPP無線会議呼動作を遂行する方法であって、 第1ユーザと少なくとも第2ユーザとの間の呼を保留状態に置くステップを含み、前記第1のユーザが動作呼上に置かれ、少なくとも前記第2ユーザが保留呼上に置かれ、さらに、 少なくとも第3ユーザを伴う会議呼要求を前記第1ユーザからネットワークコントローラにSIP/SDP信号指示により送信するステップを含み、前記ネットワークコントローラはSIPバック-ツー-バックユーザエージェントを含み、さらに、前記第1ユーザと前記第2ユーザと前記第3ユーザとの間の各呼区間に対する少なくともコーデック再ネゴシエーションおよびベアラ再構成を前記ネットワークコントローラが独立して行うことを可能にするSIP/SDP信号指示により、前記ネットワークコントローラとSIPユーザエージェントを含むリソース・コントローラとの間で通信するステップを含み、前記リソースコントローラは、前記第1ユーザと前記第2ユーザと前記第3ユーザとの間の各呼区間に対するベアラ経路リソースを前記ネットワークコントローラに通信し、前記ベアラ経路リソースは、インタネットプロトコルアドレス、使用可能なコーデック及び指向性のうちの少なくとも1つを含み、そして、前記リソースコントローラは、前記第1ユーザと前記第2ユーザと前記第3ユーザとの間の各呼区間に対するベアラ経路リソースについて同時に交渉する、方法。
Independent claims2
57 paragraphs, as filed
The present invention generally relates to communication, and more specifically to wireless communication.
This application includes content related to the content of the following applications assigned to the same assignee. Thereby, the applications listed below are incorporated herein by reference in their entirety. "INTERNET PROTOCOL MULTIMEDIA SUBSYSTEM COMPONENT PROVIDING OF PACKET-SWITCHED SWITCHING FUNCTIONS TO SERVING MOBILE SWITCHING CENTER FEATURE SERVER", Cyr, et al., Application No. 10/295774, filed on November 14, 2002 "COMMUNICATION BETWEEN USER AGENTS THROUGH EMPLOYMENT OF CODEC FORMAT UNSUPPORTED BY ONE OF THE USER AGENTS", Ejzak, et al., Application No. 10/295775, filed on November 14, 2002.
Current wireless communication systems provide the ability for a user to communicate with a wireless user or a mobile user. In general, there are two wireless communication system systems, namely, circuit-switched (CS) and packet-switched (PS).
In a typical circuit-switched wireless communication system, the mobile switching center (MSC) connects the public switched telephone network (PSTN) system of the terrestrial communication line to the wireless communication system. Normally, a mobile exchange center is divided into a mobile exchange center server and a media gateway (MGW), and bearer independent call control (MGW) is used to transfer calls between mobile exchange centers. It incorporates the call control protocol of BICC) or integrated services digital network user part (ISUP).
Internet Protocol ("IP") multimedia services for Universal Mobile Telecommunications Systems ("UMTS") and Code Division Multiple Access ("CDMA") third generation ("3G") systems The current approach to implementing is to define the latest Internet Protocol Multimedia Subsystem (IMS), which is within the Internet Protocol Multimedia Subsystem that uses packet-switched services. Internet Protocol Consists of a set of connected network entities. These network entities use session initiation protocol (SIP) as the primary means of call control to provide Internet Protocol multimedia features and services.
As those network entities become more centralized and use different codes for communication, it becomes more difficult to allow end users to communicate with each other. It is desirable to optimally configure conference calls by allowing codecs to negotiate between remote network resources and end users.
The operator of the wireless service 1) packets the line voice over the backbone core network, 2) negotiates with the codecs and modifies them with Transcoder Free Operations (TrFO) over the packet network. We are looking for an efficient solution to facilitate both remote transcoder operations (RTO). TrFOs and RTOs have the effect of improving sound quality, saving transport equipment and reducing network resources.
In the existing wireless network, the conference management function is a value-added service. While deploying TDM-based transport to packet-based networks, it quickly configures conferencing bridging capabilities that model existing line conferencing operations, minimizes network resource usage, and provides good sound quality. It is highly desirable to find a solution that maintains. This solution should also support SIP-enabled packet mobile phones with end-to-end VoIP applications.
In this way, there is a need for a rapid network SIP / SDP (Session Initiation Protocol / Session Description Protocol) procedure for running a conference in response to a request from an end user by optimizing network resources for the prior art. Has been done.<patcit num="1"><text>"INTERNET PROTOCOL MULTIMEDIA SUBSYSTEM COMPONENT PROVIDING OF PACKET-SWITCHED SWITCHING FUNCTIONS TO SERVING MOBILE SWITCHING CENTER FEATURE SERVER", Cyr, et al., Application No. 10/295774, filed on November 14, 2002</text></patcit><patcit num="2"><text>"COMMUNICATION BETWEEN USER AGENTS THROUGH EMPLOYMENT OF CODEC FORMAT UNSUPPORTED BY ONE OF THE USER AGENTS", Ejzak, et al., Application No. 10/295775, filed on November 14, 2002</text></patcit>
<p> In one embodiment, the present invention includes one device. The device comprises a SIP / SDP (Session Initiation Protocol / Session Description Protocol) -based network controller that connects the first and second calls to conference calls with at least a third call in response to received requests. .. The SIP / SDP-based network controller is constructed to negotiate, allocate, and reconfigure network resources.</p><p> In another embodiment the invention includes one device. This device is a conference involving the first user and the second user with at least a third user in response to a request from a first user who is on an active call and has at least one hold call to the second user. It has a network controller to connect to the call. SIP / SDP (Session Description Protocol / Session Description Protocol) that allows the media resource function controller to perform at least codec renegotiation and bearer reconstruction independently for each call interval between users. It is operably connected to the network controller via an open interface by signal instruction.</p><p> In another embodiment the invention includes one method. This method is a step of putting a call between the first user and at least the second user on hold, with the first user on the action call and at least the second user on the hold call. A step to be placed, a step to send a request for a conference call from the first user to the network controller by SIP / SDP signal instruction, accompanied by at least a third user, and at least a codec recurrence for each call interval between users. With the step of communicating between the network controller and the media resource function controller via an open interface by SIP / SDP signal instruction to allow the resource controller to perform negotiation and bearer reconstruction independently. including.</p><p> The features of the invention that are considered novel are detailed in the appended claims. The present invention will be best understood by reference to the following description in conjunction with the accompanying drawings. In some of the drawings, similar reference numerals indicate similar elements.</p>
As multiple service providers deploy their current TDM-based networks into packet-based networks, they need to maintain service continuity for their multiple customers. Various embodiments of the devices and methods according to the invention provide a conferencing solution using an IP-based SIP network architecture and SIP / SDP signaling for network controlled conferencing functions over packet networks. The network control conferencing feature is most desirable for radiotelephone users due to the scarcity of resources on the air interface. The solution is also applicable to end users enabled by SIP.
Typically, a SIP / SDP-based solution enables the network controller to be used at the request of the end user and performs the following conferencing function operations: two current calls, one on hold and one. (One is the operating state) to connect to the conference call, and to clear the conference call and resume the two-way conversation.
This allows the network controller (ie, SIP back-to-back User Agent, B2BUA) to be controlled by a SIP User Agent (for example, a network resource). It will be possible to negotiate, allocate and reconfigure the conference bridge and voice transcoder). This method optimizes parallel processing operations between all participating parties and enables quick call setup. It conforms to standard SIP / SDP procedures and can be easily extended to support other normal conference operations such as adding another party, excluding one party from a conference call, and so on.
With reference to FIG. 1, a communication network 200 that provides a conference function is shown by an exemplary configuration. In one embodiment, the communication network 200 is part of the Internet Protocol multimedia subsystem (IMS). The communication network 200 in one embodiment includes one or more network controllers 202, one or more resource controllers 204, one or more media servers 206, 208 and 210, and one or more. SIP servers 212, 214 and 216 (for one or more end users UE-A 218, UE-B 220, UE-C 222, respectively) and one or more network resources 224. ing. Network controller 202 is a SIP back-to-back user agent (SIP) within the 3GPP IP Multimedia Subsystem (IMS). B2BUA "), for example an application server ("AS "). The resource controller 204 in one embodiment includes a SIP user agent, such as the media resource function controller 112 (FIG. 1). The resource controller 204 may be physically located anywhere in the network to support a plurality of geographically distributed network controllers 202. Network controller 202 optimizes the allocation of network resource 224 using the specified or found information about the functionality of resource controller 204. Media servers 206, 208 and 210 in one embodiment include media gateway 118 (Figure 1). The SIP servers 212, 214 and 216 in one embodiment are serving mobile switching center servers 3GPP IMS CSCF, 3GPP IMS. It has an MGCF or SIP proxy server. One or more network resources 224 in one embodiment include a media resource function processor 116 (FIG. 1).
End users 218, 220 and 222 use communication link 1 to carry call control information to SIP servers 212, 214 and 216, respectively. Communication link 1 in one embodiment is a call control protocol for exchanging call control information between the end user 218 and the SIP server 212, such as a session start protocol, bearer independent call control (BICC). ), Comprehensive service digital communication network user section (ISUP), or radio interface line call control protocol for CDMA or UMTS. UE-A 218 uses the conference control interface 2 to control the conference function. UE-A 218 may use any suitable protocol for conference function control interface 2, including the call control protocol used for communication link 1. End users 218, 220 and 222 use communication link 3 to send or receive media for conference calls.
SIP servers 212, 214 and 216 control media servers 206, 208 and 210 via communication link 4, respectively. In one embodiment, SIP servers 212, 214 and 216 use the H.248 protocol to control media servers 206, 208 and 210, respectively. Communication links 5 and 10 provide a route for call control information between SIP servers 212, 214 and 216 and network controller 202. Communication link 5 in one embodiment uses a session initiation protocol to exchange call control information. In another embodiment, the SIP server 212 translates call control information between each protocol used in the corresponding communication link 1 and communication link 5. Similarly, SIP servers 214 and 216 translate call control information between each protocol used by the corresponding communication link 1 and communication link 10. The SIP server 212 also translates between the protocols used in the conference function control interface 2 and the conference function control interface 6 as needed. Call control interface 6 in one embodiment is SIP Use the INFO method to convey conference function control information. The communication link 8 in one embodiment provides a route for resource control information between the network controller 202 and the resource controller 204. In one embodiment, network controller 202 uses a session initiation protocol within SIP to exchange resource control information over communication link 8 in a third-party call control procedure. The resource controller 204 of one embodiment uses H.248 interface 9 to communicate with network resource 224. The media of the conference call is transmitted via the bearer pathway formed by the communication pathways 3, 7 and 11. Media servers 206, 208 and 210 and network resource 224 perform media conversions between their respective users, end user 218, end user 220, and end user 222.
The general explanation of the conference function allocation procedure is given below. An end user, such as UE-A 218, has two calls established using the call control protocol of communication link 1, one in the working state and the other in the pending state. The end user signals the network using interface 2 and connects the above two calls to the conference call. Requests and corresponding parameters are sent to network controller 202. In one embodiment, the request and corresponding parameters are sent in a SIP INFO message. Upon receiving the request, network controller 202 may check the authorization database for service authorization and proceed with the configuration of network resource 224. Network controller 202 maintains previous call information such as endpoint codec functionality and call identification. Network controller 202 SIP without SDP attachment Send INVITE to resource controller 204 to request conference media resource 224, allowing individual session capability negotiation with each participating party. When network conferencing resource 224 is allocated, network controller 202 simultaneously sends each end user a SIP UPDATE containing send / receive IP addresses, port numbers, and codec information for the corresponding conferencing port on network resource 224. Submit and initiate SDP offer / response negotiations to optimally reconfigure bearer resources. After completing the SDP offer / answer procedure in the SIP UPDATE transaction to optimally allocate network resources for each section of the conference bridge, network controller 202 transfers remote media endpoint information in SDP to resource controller 204. As a result, the configuration of network resource 224 can be completed. A conference call is established here.
The network controller 202, resource controller 204, and end users 218, 220, and 222 in one embodiment are geographically separated. By using the session initiation protocol on communication link 8, network controller 202 communicates with one or more remote resource controllers 204 to optimize one or more network resources 224 required for conference calls. Can be allocated to.
Referring to FIG. 2, the apparatus 100 of one embodiment includes a plurality of components such as computer software and / or hardware components. Some of the above components may be combined or separated within the apparatus 100.
In one embodiment, device 100 uses one or more computer-readable signal indicating media. An example of a computer-readable signal instruction medium for device 100 is a recordable data storage medium 101 such as one or several of magnetic, electrical, optical, biological, and atomic data storage media. This is an example. In another embodiment, the computer readable signal indicator medium for device 100 is one of networks including or coupled to device 100, such as a telephone network, a local area network (LAN), the Internet and a wireless network. Or it contains a modulated carrier signal transmitted via some. As will be appreciated by those skilled in the art, the exemplary components of device 100 use and / or comprise a set and / or column of computer instructions written or implemented in any of a number of programming languages.
In one embodiment, device 100 comprises a mobile exchange center (MSC) 102. The mobile exchange center 102 comprises one or more mobile exchange center service components and one or more Internet Protocol multimedia subsystem (IMS) components. This one or more Internet Protocol multimedia subsystem components provide packet-switched exchange capabilities for the one or more mobile exchange center service components described above. The one or more Internet Protocol multimedia subsystem components described above are Session Initiation Protocol (SIP)-based call signaling, calling routing, and signaling interactions. interworking), and provide media processing services to one or more of the above mobile exchange center service components. The one or more mobile exchange center service components and the one or more Internet Protocol multimedia subsystem components include a network for communication between endpoints. The radio part of the network is a global system for mobile communications ("GSM") or a universal mobile telecommunications system ("GSM") via a cellular system protocol, such as code division multiple access ("CDMA"). Communicate with UMTS "). The one or more mobile exchange center service components and the one or more Internet Protocol multimedia subsystem components may be in the same mobile exchange center 102, or the mobile exchange center. It may be distributed among multiple instances of 102.
One or more of the above mobile exchange center service components are the service mobile switching center feature server ("SMSC-FS") 104, the gateway mobile exchange center function server ("GMSC-FS"). It has 106 and a media coordinator (MC) 108. The one or more Internet Protocol multimedia subsystem components described above include the Media Gateway Control Function (MGCF) 110, the Media Resource Function Controller (MRFC) 112, and the Breakout Gateway Control Function (MRFC). It has a "BGCF") 114, a media resource function processor ("MRFP") 116, and a media gateway ("MGW") 118.
The service mobile exchange center function server 104 provides the mobile exchange center 102 with the functions of the service mobile exchange center, but by one or more of the Internet Protocol multimedia subsystem components and media coordinator 108 described above. The features provided are excluded. Service Mobile Exchange Center Function Server 104 generates an intelligent network (IN) trigger, mobility management, subscriber function control, call-related replenishment services, numerical analysis, emergency services, billing, and media. Supports coordinator interface.
In one embodiment, the service mobile exchange center function server 104 comprises a session initiation protocol user agent (UA). Service Mobile Exchange Center Function Server 104 supports session initiation protocol call control procedures. The session initiation protocol call control procedure supports session initiation protocol-based call signal instruction, call routing, signal instruction interaction, and media processing. An additional instance of the Service Mobile Exchange Center Function Server 104 may support other call control protocols such as Bearer Independent Call Control (BICC) or Integrated Services Digital Communications Network User Unit (ISUP). The mobile exchange center 102 may support multiple call control protocols using additional instances of the service / mobile exchange center function server 104.
The service mobile exchange center function server 104 provides the media coordinator 108 with internal outgoing / outgoing call functionality and interaction between the service and the external session initiation protocol interface. The external session initiation protocol interface supports the transmission of mobile outgoing and mobile terminated calls between the Service Mobile Exchange Center function server 104 and the rest of the network. The external session initiation protocol interface also supports the transmission of media requests for tones, announcements, or conferencing between the Service Mobile Exchange Center feature server 104 and the media coordinator 108. Propagation of media requests for a conference between Service Mobile Exchange Center Function Server 104 and Media Coordinator 108 requires the definition of an additional session initiation protocol header in some cases. Unlike the session initiation protocol interface between other entities within the mobile exchange center 102, the external session initiation protocol interface between the service mobile exchange center function server 104 and the media coordinator 108 is typically for a single vendor. It remains private and allows the use of private session initiation protocol extensions by the X-header mechanism defined by the Internet Engineering Task Force (IETF).
The gateway / mobile exchange center function server 106 provides the gateway / mobile exchange center service to the mobile exchange center 102 via the session start protocol call control procedure. In one embodiment, the gateway mobile exchange center function server 106 includes a session initiation protocol back-to-back user agent (B2BUA). The gateway mobile exchange center function server 106 supports the session start protocol call control procedure. An additional instance of the gateway mobile exchange center function server 106 may support other call control protocols such as bearer independent call control or integrated service digital communication network user unit. The mobile exchange center 102 may support multiple call control protocols using additional instances of the gateway mobile exchange center function server 106.
The gateway mobile exchange center function server 106 supports termination services, basic system call transmission, termination intelligent network triggers, secondary processing, and billing. In one embodiment, the gateway / mobile exchange center function server 106 transmits a call progress or service control instruction as out-of-band call progress information to the caller by means of a session start protocol. The origin in the network (ie, service-mobile exchange center function server 104, media coordinator 108, or media gateway control function 110) converts out-of-band call progress information into in-band call progress information. Call progress and service control instructions are typically carried via existing session initiation protocol messages and headers, but in some cases additional session initiation protocol headers or attachments may be required.
In one embodiment, the media coordinator 108 comprises a session initiation protocol back-to-back user agent between the service mobile exchange center function server 104 and the network. For calls made by mobile phones, media 108 assists in the propagation of basic call status information between the service / mobile exchange center function server 104 and the network. The media coordinator 108 uses the resource allocation of the media resource function controller 112 and the media resource function processor 116 to provide in-band call progress information to the mobile phone for out-of-band call progress or call-release information from the network. Support conversion to. The media coordinator 108 assists in media negotiation between endpoints through resource allocation of the media resource function controller 112 and the media resource function processor 116 as needed for media conversion. Media coordinator 108 forwards the media when the call is answered within the network. Supports cut-through) control. The media coordinator 108 supports the session initiation protocol / third party call control procedure for performing media functions under the direction of the service / mobile exchange center function server 104. Media features include resource allocation capabilities for the media resource function controller 112 and the media resource function processor 116, as needed to control conference functions, tones, announcements, or inter-system handoffs.
For calls terminated on a mobile phone, Media Coordinator 108 supports the propagation of basic call status information between the Service Mobile Exchange Center Function Server 104 and the network. The media coordinator 108 supports media negotiation between endpoints in the network through resource allocation of the media resource function controller 112 and the media resource function processor 116 as needed for media conversion. The media coordinator 108 supports the session initiation protocol / third party call control procedure for performing media functions under the direction of the service / mobile exchange center function server 104. Media features include resource allocation capabilities for the media resource function controller 112 and the media resource function processor 116, as needed to control conference functions, tones, announcements, or inter-system handoffs.
The media gateway control function 110 acts as a call control interface and converter between the mobile exchange center 102 and the public switched telephone network (PSTN) 120 or other networks. In one embodiment, the media gateway control function 110 comprises a session initiation protocol user agent (UA) for the mobile exchange center 102. For example, the media gateway control function 110 has a call control message of the session start protocol of the mobile exchange center 102 and a call control message of the bearer independent call control or the integrated service digital communication network user section of the public switched telephone network 120. Perform a conversion between. The media gateway control function 110 communicates with the public switched telephone network 120 via a communication link, for example, bearer independent call control or integrated service digital communication network user interface 122.
In one embodiment, the media gateway control function 110 controls the media gateway 118. The media gateway control function 110 includes a signal indicator layer controller, and the media gateway 118 includes a media layer controller. The media gateway control function 110 enables connection control for media channels within the media layer controlled by the media gateway 118. The media gateway control function 110 controls the media gateway 118 via a communication link, for example, the H.248 interface 124, via the Internet Protocol network 128.
In another example, the media gateway control function 110 controls a plurality of media gateways 118. The media gateway control function 110 controls the plurality of media gateways 118 via one or more communication links, for example, one or more instances of the H.248 interface 124 for the Internet Protocol network 128. The plurality of media gateways 118 are registered in the media gateway control function 110. After registering with the media gateway control function 110, the plurality of media gateways 118 can start bearer processing. The media gateway control function 110 controls the establishment of bearer resources for communications requiring interaction between the mobile exchange center 102 and the public switched telephone network 120. The media gateway control function 110 requests port allocation for communications requiring the services of the media gateway 118.
The media gateway control function 110 signals the media gateway 118 to perform one or more media operations using the H.248 interface 124 for the Internet Protocol network 128. The one or more media operations described above include registration of media gateway 118, control of bearer establishment between mobile exchange center 102 and public switched telephone network 120, media conversion resources (ie, compression, echo cancellation and statistic). Includes vocoding request, control of events detected at media gateway 118, tone and announcement applications, and collection of statistics.
The media gateway control function 110 uses the session initiation protocol network 130 to receive commands from other signaling entities in the network. The media gateway control function 110 performs functions related to call control. The media gateway control function 110 enables negotiation of media attributes with other endpoints in the network. For calls issued by the public switched telephone network 120 and entering the mobile exchange center 102, the media gateway control function 110 enables conversion of out-of-band call progress information into in-band call progress information. The out-of-band call progress information includes a signal instruction message that is not heard by the communicating user. The in-band call progress information includes a signal instruction message heard by the user in communication. For example, the media gateway control function 110 converts an out-of-band ringing instruction into an in-band ringing tone. In another embodiment, media gateway control function 110 translates out-of-band network error instructions (ie, session start protocol server internal error response messages) into in-band network error signals (ie, fast busy instructions). Allows you to.
The Media Resource Capability Controller 112 provides packet-based media services (ie, latest announcement generation and detection), conferences, tones and announcements, more up-to-date media services (ie, video mix), transcoding, and interactive voice. Allows control of the response. The media resource function controller 112 controls the media resource function processor 116 via a communication link, such as the H.248 interface 126, over the Internet Protocol network 128. By controlling the media resource function processor 116, the media resource function controller 112 controls the use of vocoders, transcoders, compression entities, bearer stream mixers, and echo cancellers. Vocoders have different media coding formats (eg, selectable mode vocoder "SMV", enhanced variable rate). Codec) Required for transcoding between "EVRC" and G.711). Media Resource Capability Controller 112 supports Real Time Protocol, User Datagram Protocol, and Internet Protocol (RTP / UDP / IP) as a transport protocol stack for packet media.
The breakout gateway control function 114 includes a signal indicating entity for call / session control. In one embodiment, the breakout gateway control function 114 comprises a session initiation protocol proxy server for the mobile exchange center 102. In another embodiment, the breakout gateway control function 114 comprises a session initiation protocol redirect server or a session initiation protocol back-to-back user agent. The breakout gateway control function 114 selects the media gateway control function 110 to connect the mobile exchange center 102 to the public switched telephone network 120. Calls from radiotelephones to telephones within the public switched telephone network 120 include signaling messages. The signaling message contains connection information for a telephone address (ie, an E.164 address) within the public switched telephone network 120. The breakout gateway control function 114 uses the E.164 address to locate the Internet Protocol network destination address of the call. In one embodiment, the breakout gateway control function 114 looks up the address translation table to find the Internet Protocol network destination address that corresponds to the E.164 address. The address translation table contains other information (including port numbers, forwarding protocols, and security parameters) needed to establish communication between the breakout gateway control function 114 and the next hop destination for signaled messages. May include. The breakout gateway control function 114 sends a signal instruction message to the destination address.
The breakout gateway control function 114 may use information from multiple sources to determine the destination address. These sources include call origination points in the network, E.164 address locations, local policies, and business contracts between visited and home networks, and public switched telephones. We seek to minimize the path length within the network 120 and seek the lowest cost path.
The breakout gateway control function 114 selects the media gateway control function 110 based on the destination address, hides network information from other networks, and regulates security by approving peer network elements. If the first breakout gateway control function 114 is in the first network and the second breakout gateway control function 114 is in the second network and those networks are combined, then the first and The second breakout gateway control function 114 may hide local network information from other networks.
Media gateway 118 performs the conversion between media flow (eg, acoustics) on a given Internet Protocol network and bearer data on the public switched telephone network 120. Media gateway 118 terminates circuit-switched (CS) bearer traffic on the public switched telephone network 120 and uses Internet Protocol media flow as a packet stream from other endpoints within the mobile exchange center 102. To terminate. In one embodiment, the other endpoint comprises a media endpoint associated with a media gateway control function 110, or a service mobile exchange center function server 104. Media gateway 118 is bocoding and provides tones and announcements. Media gateway 118 includes resources to modify (ie, encode, compress, echo offset, packetize, transcode, packet timing synchronize, and packet loss handle) bearer streams.
Media gateway 118 supports one or more voice-coded formats (ie, codec formats). One or more of the above audio coding formats include G.711, Enhanced Variable Rate Codec (EVRC), and Adaptive Multi-Rate (AMR). Media gateway 118 can use the G.711 codec format to encode and decode audio on trunks connected to the public switched telephone network 120.
Media gateway 118 registers multiple signal directives, such as media gateway control function 110, event detection (eg, dual tone multi-frequency (DTMF) detection), tone and announcement bearer streams. Supports application to, graceful teardown and random restart, notifications, statistic generation, and support for the H.248 package. Media gateway 118 organizes bearer connections with H.248 context-containing terminations. Media gateway 118 may contain a large number of simultaneous contexts.
The mobile exchange center 102 includes one or more interfaces with communication support entities located on the network outside the mobile exchange center 102. The mobile exchange center 102 includes a signaling system seven (SS7) interface 132 to the home location register (HLR) 134. The home location register 134 includes a database that stores registration information about users on the network. The mobile exchange center 102 comprises an intelligent network application protocol (INAP) or wireless intelligent network (WIN) interface 136 to an intelligent network (IN) 138. Intelligent Network 138 provides one or more telephone services from the Mobile Exchange Center 102, such as number translation, local number portability (LNP), call transfer, call screening, and wireless integration. Allow access to integration). The mobile exchange center 102 has an interface 140 to the radio access network (RAN) 142. Radio access network 142 comprises an interface between the mobile phone and the network (ie, mobile exchange center 102). The radio access network 142 is a code split multi-connection radio access network, universal mobile telecommunications system terrestrial radio access network (UTRAN), or global. -Mobile communication / enhanced data for the global evolution radio access network ("GERAN") May include a global system for rate). The radio access network 142 is coupled to the mobile phone via a radio interface, such as a 3rd generation (3G) radio interface. Radio access network 142 may use an instance of media gateway 118 to convert radio interface media flows into packet streams.
Referring to FIG. 3, the exemplary configuration of the communication network in FIG. 2 shows a preliminary conference function configuration. User UE-A 218 has established two separate media sessions (calls) with UE-B 220 and UE-C 222, respectively. Here, user UE-A 218 may make a conference (multiparty) call. Media gateway 206 is operably connected to media servers 208 and 210. SIP servers 212, 214 and 216 are operably connected to network controller 202. Users UE-A 218, UE-B 220, and UE-C 222 are connected to SIP servers 212, 214, and 216, respectively, and to media gateways 206, 208, 210, as shown in Figure 3. ing. Users UE-A 218, UE-B 220, and UE-C 222 may be mobile phone users, PSTN users, or native SIP endpoints.
With reference to FIG. 4, the communication network exemplary configuration of FIG. 3 shows the conference function configuration. End users UE-A 218, UE-B 220, and UE-C 222 exchange call control information with network controller 202 via SIP server 212, SIP server 214, and SIP server 216, respectively. User UE-A 218 is the originator of the conference (multiparty) call. Media server 206 is operably connected to media servers 208 and 210 via network resource 224. The network controller 202 is operably connected to the resource controller 204, and the resource controller 204 is operably connected to the network resource 224. Network controller 202 performs SIP third-party call control procedures to allocate suitable network conferencing resources 224 through resource controller 204 and optimal bearer resources between endpoints 206, 108, 210 and 224. Allocate in the form of
Referring here to FIGS. 5-7, the exemplary configuration illustrates a conference call scenario in a communication network that provides conference functionality. In FIG. 5, user MS-A 502 has one call configured for user PSTN-B 504 in conference call preconditions. The access network 508 in one embodiment includes a media server 206 (Fig. 2). The MSC / CSCF506 in one embodiment includes a SIP server 212 (Fig. 2). The SIP B2BUA 510 is equipped with a network controller 202 (Figure 2). The MGCF 512 in one embodiment is equipped with SIP server 214 (Fig. 2). The MGW 514 in one embodiment is equipped with a media server 208 (Fig. 2).
In FIG. 6, the call between user MS-A 502 and user PSTN-B 504 is put on hold, and the call between user MS-A 502 and user UA-C 602 is established. The user UA-C 602 in one embodiment is equipped with UE-C 222 (Fig. 2). The SIP proxy 604 in one embodiment includes a media server 210 (Figure 2). User MS-A 502 requests the construction of a conference call and connects the existing call to the pending call.
In Figure 7, signal indications and bearer paths are shown for established conference calls. User MS-A 502 signals network controller 510 to initiate conference call setup by call identification for users PSTN-B 504 and UA-C 602. In one embodiment, end users 504 and 602 include PSTN users or SIP line users. At this time, the network controller 510 gives a signal instruction to the resource controller 702 to seize the conference bridge 704 on the three ports. The resource controller 702 in one embodiment includes a resource controller 204 (FIG. 2). The conference bridge 704 in one embodiment includes network resource 224 (Figure 2). The network controller 510 updates the network to reconfigure the bearer route as shown in FIG.
Referencing Figures 8 and 9 shows an exemplary method of invoking a conference call. UE-A 502 represents the end user who is first connected to the other two users PSTN-B 504 and UA-C 602 via their respective call intervals. The network controller 510 acts as a manager for establishing conference calls between call intervals. Resource controller 702 uses multiple codecs and conference ports to act as a conference bridge for conference calls. The SIP server 506 acts as a converter for call control messages between the UE-A 502 and the network controller 510. SIP server 506 first provides a communication path between UE-A 502 and PSTN-B 504 via MGCF / MGW 512 and between UE-A 502 and UA-C 602 via SIP proxy 604. ..
End-user UE-A 502 is first connected to PSTN-B 504 and UA-C 602 with separate calls, which are identified by their corresponding call IDs. Calls to PSTN-B 504 are initially put on hold. UE-A 502 sends a request for a conference call to the mobile exchange center 506. Mobile exchange center 506 is PSTN-B 504 and UA-C Signals network controller 510 to build a conference call for each call ID in 602 (Message 1). Upon receiving the request, the network controller 510 sends an acknowledgment of the request to the mobile exchange center 506 (message 2). Network controller 510 then sends a request (message 9) to resource controller 702 to allocate one or more network resource 704s, such as one or more transcoders and network ports, for conference calls. .. In one embodiment, message 9 has three separate messages that allocate one or more of the network resources 704 above for a conference call. The resource controller 702 allocates one or more of the above network resources 704 and uses control information corresponding to the allocated network resource 704 for each section of the conference call, such as the Internet Protocol (IP) address. Returns a possible codec and directionarity (message 11). In one embodiment, message 11 includes an offer message for available network resource 704 for a conference call.
Upon receiving control information from resource controller 702, network controller 510 updates the network to reconfigure the bearer route. In one embodiment, network controller 510 performs end-to-end negotiation with UE-A 502, PSTN-B 504, and UA-C 602. In another embodiment, the network controller 510 uses the SDP offer / response model to perform end-to-end negotiations with the UE-A 502, PSTN-B 504, and UA-C 602. For example, network controller 510 sends the corresponding conference port, the codecs available on the conference port, and the orientation of the conference port to the PSTN-B 504 and UA-C 602 (messages 16 and 19). PSTN-B 504 and UA-C 602 responds to end-to-end negotiation by accepting one of the codecs offered by network controller 510 (messages 18 and 21). Message groupings 802, 804 and 902 in one embodiment include the exchange of offer / response negotiations. An example of offer / response negotiation is shown in the patent application named "COMMUNICATION BETWEEN USER AGENTS THROUGH EMPLOYMENT OF CODEC FORMAT UNSUPPORTED BY ONE OF THE USER AGENTS", application number 10/295775, filing date November 14, 2002. This patent application has already been incorporated into this application for reference. In one embodiment, message groupings 802, 804 and 902 are negotiated simultaneously. The PSTN-B 504 and UA-C 602 are connected to their respective conference ports using the accepted codec. Network controller 510 is PSTN-B Signals resource controller 702 to configure each network resource 704 for the call interval according to the accepted codecs received from 504 and UA-C 602 (message 23). The network controller 510 then configures the UE-A 502 call interval (messages 27 and 29) and confirms that the conference call setup was successful (messages 31, 32).
Next, referring to FIGS. 10 and 11, an exemplary configuration diagram shows a request to cancel the scenario in the communication network shown in FIGS. 11 and 12. The UA-A 502 sends a request to the MSC / CSCF 506 to release the UA-C 602 (message 0). The MSC / CSCF 506 signals the network controller 510 to terminate the call identified by the call ID corresponding to the UA-C 602 and clear the conference bridge (Message 1). The network controller 510 signals the UA-C 602 via the SIP proxy 604 to disconnect from the conference call (Message 2). The UA-C 602 sends an acknowledgment of the signal instruction to disconnect from the conference call to the network controller 510 (message 4). The network controller 510 is then on the MSC / CSCF 506, the UA-C. Notifies that 602 has successfully disconnected from the conference call (Message 5). The network controller 510 notifies the resource controller 702 that it will release the network resource 704 allocated for the conference call (message 6). Resource controller 702 responds by deleting the allocated network resource 704 (message 7) and signals network controller 510 to confirm the closing of the conference bridge (message 8). The network controller 510 signals the MGCF / MGW 512 to disconnect from the conference bridge and connect to the MSC / CSCF 506 (message 17). The MGCF / MGW 512 disconnects from the conference bridge and signals the network controller 510 (messages 18 and 19). The network controller 510 then signals the MSC / CSCF 506 to connect to the MGCF / MGW 512 (message 20). MSC / CSCF 506 disconnects from conference call and MRFC / MGW Connect to 512 (message 21). The MSC / CSCF 506 then responds to signal instructions connecting to the MGCF / MGW 512 (Message 22). Message groupings 1002 and 1102 in one embodiment include the exchange of offer / response negotiations. An example of offer / response negotiation is shown in the patent application named "COMMUNICATION BETWEEN USER AGENTS THROUGH EMPLOYMENT OF CODEC FORMAT UNSUPPORTED BY ONE OF THE USER AGENTS", application number 10/295775, filing date November 14, 2002. This patent application has already been incorporated into this application for reference. In one embodiment, message groupings 1002 and 1102 are negotiated at the same time. The network controller 510 then sends a confirmation of the successful clearing of the conference bridge to the MSC / CSCF 506 (message 31). MSC / CSCF 506 returns an acknowledgment that the conference bridge was successfully cleared for network controller 510 (message 32).
Next, referring to FIG. 12, an exemplary configuration diagram shows a call release scenario by user UE-A 502 in a communication network that provides conferencing functionality. UE-A 502 is initially in a conference call with user PSTN-B 504 and user UA-C 602. UE-A 502 sends a signal to MSC / CSCF 506 to cancel the conference call (message 0). The MSC / CSCF 506 signals the network controller 510 to release the calls associated with the PSTN-B 504 and UA-C 602 (messages 1 and 11). The network controller 510 signals the UA-C 602 through the SIP proxy 604 to disconnect from the conference bridge (Message 2). Upon disconnecting from the conference bridge, the UA-C 602 signals the network controller 510 to confirm the disconnection (Message 4). The network controller 510 forwards a signal confirming the disconnection of the UA-C 602 to the MSC / CSCF 506 (message 5). Network controller 510 is PSTN-B Signal MGCF / MGW 512 to disconnect the call to 504 (Message 12). MGCF / MGW 512 disconnects the call to PSTN-B 504, removes the allocated port on media server 514 (message 13), and sends a disconnect acknowledgment to network controller 510 (message 14). ). The network controller 510 forwards the disconnect acknowledgment to the MSC / CSCF 506. The MSC / CSCF 506 sends an acknowledgment of disconnecting the PSTN-B 504 and UA-C 602 to the network controller 510 (message 15). The network controller 510 then signals the MRFC / MRFP 702 to release the allocated conference port (message 21). The MRFC / MRFP 702 releases the allocated conference port and sends an acknowledgment of the port release (messages 22 and 23).
Next, referring to FIGS. 13 and 14, an exemplary configuration diagram shows the cancellation of the call interval / scenario by the user PSTN-B 504 in the communication network that provides the conference function. The PSTN-B 504 cancels the call (message 0), which causes the MGCF / MGW 512 to signal the network controller 510 about the release (message 1). The network controller then signals the MSC / CSCF 506 to release the PSTN-B 504 (Message 3). The MSC / CSCF 506 sends an acknowledgment of the release of the PSTN-B 504 to the network controller 510 and signals the MRFC / MRFP 702 to release the allocated conference port (Message 4). The network controller 510 then forwards the acknowledgment acknowledgment to MGCF / MGW 512 (message 5). The MRFC / MRFP 702 releases the allocated conference port and sends an acknowledgment of the port release (messages 7 and 8). Network controller 510 is UA-C 602 and MSC / CSCF Update 506 to connect UE-A 502 and UA-C 602 (messages 17, 19, 20-22). In one embodiment, message grouping 1402 involves exchanging offer / response negotiations. An example of offer / response negotiation is shown in the patent application named "COMMUNICATION BETWEEN USER AGENTS THROUGH EMPLOYMENT OF CODEC FORMAT UNSUPPORTED BY ONE OF THE USER AGENTS", application number 10/295775, filing date November 14, 2002. This patent application has already been incorporated into this application for reference. The network controller 510 sends an acknowledgment that the conference bridge was successfully cleared (message 31). The MSC / CSCF 506 returns an acknowledgment of the clear to the network controller 510 (message 32).
Next, referring to FIGS. 15 and 16, the exemplary configuration diagram shows the cancellation of the call interval / scenario by the user UA-C in the communication network that provides the conference function. The UA-C 602 releases the call and sends a release signal to the network controller 510 via SIP proxy 604 (messages 0, 1). The network controller 510 then signals the MSC / CSCF 506 to release the UA-C 604 (Message 3). The MSC / CSCF 506 sends an acknowledgment of the release of the UA-C 604 to the network controller 510 and signals the MRFC / MRFP 702 to release the allocated conference port (messages 4 and 6). The network controller 510 then forwards the acknowledgment acknowledgment to the UA-C 602 via the SIP proxy 604 (message 5). MRFC / MRFP 702 releases the allocated conference port and sends an acknowledgment of port release (messages 7 and 8). Network controller 510 updates MGCF / MGW 512 and MSC / CSCF 506 to UE-A Connect the 502 to the PSTN-B 504 (messages 17-19, 20-22). Message grouping 1602 in one embodiment involves exchanging offer / response negotiations. An example of offer / response negotiation is shown in the patent application named "COMMUNICATION BETWEEN USER AGENTS THROUGH EMPLOYMENT OF CODEC FORMAT UNSUPPORTED BY ONE OF THE USER AGENTS", application number 10/295775, filing date November 14, 2002. This patent application has already been incorporated into this application for reference. Network controller 510 sends an acknowledgment that the conference bridge was successfully cleared (message 31). The MSC / CSCF 506 returns an acknowledgment of the clear to the network controller 510 (message 32).
Each step or operation described herein is merely exemplary. These steps or operations can be modified in various ways without departing from the spirit of the invention. For example, each step may be performed in a different order, or steps can be added, omitted, or modified.
Although exemplary embodiments of the invention have been described in detail herein, various modifications, additions, substitutions, etc. can be made without departing from the spirit of the invention, as will be appreciated by those skilled in the art. Yes, they are therefore considered to be within the scope of the invention as defined in the claims.
<figref num="1">It is a figure which shows an example configuration of the apparatus which carries out a call conference function.</figref><figref num="2">FIG. 5 illustrates an example of a device with one or more mobile exchange center service components and one or more Internet Protocol multimedia subsystem components.</figref><figref num="3">It is a figure which shows an example configuration of the apparatus which carries out the preliminary meeting function.</figref><figref num="4">It is a figure which shows an example configuration of the apparatus which carries out a call conference function.</figref><figref num="5">It is a figure which shows the other exemplary configuration of the apparatus which carries out an example of a conference call scenario.</figref><figref num="6">It is a figure which shows the other exemplary configuration of the apparatus which carries out an example of a conference call scenario.</figref><figref num="7">It is a figure which shows the other exemplary configuration of the apparatus which carries out an example of a conference call scenario.</figref><figref num="8">It is another figure which shows an exemplary conference call setting operation.</figref><figref num="9">It is another figure which shows an exemplary conference call setting operation.</figref><figref num="10">It is another figure which shows an exemplary request for a release operation.</figref><figref num="11">It is another figure which shows an exemplary request for a release operation.</figref><figref num="12">It is another figure which shows an exemplary release call operation.</figref><figref num="13">It is a figure which shows an exemplary optional release operation for a call interval.</figref><figref num="14">It is a figure which shows an exemplary optional release operation for a call interval.</figref><figref num="15">It is another figure which shows an exemplary optional release operation for a call interval.</figref><figref num="16">It is another figure which shows an exemplary optional release operation for a call interval.</figref>
Every citation, both ways
| Document | Relation | Office |
|---|---|---|
| JP2001298493A | Cites | Japan |
| EP01246442A1 | Cites | European Patent Office (EPO) |
| JP2001111699A | Cites | Japan |
13 members in 7 offices
Priority claims5
| Document | Office | Kind | Date |
|---|---|---|---|
| 10405764 | United States of America | – | |
| 40576403 | United States of America | A | |
| 40576403 | United States of America | A | |
| 2003405764 | – | – | – |
| US20030405764 | – | – | – |
Members13
| Document | Office | Kind | |
|---|---|---|---|
| CN1534972A | China | A | |
| EP1465386A1 | European Patent Office (EPO) | A1 | |
| US2004196867A1 | United States of America | A1 | |
| KR20040086589A | Republic of Korea | A | |
| JP2004312738A | Japan | A | |
| EP1465386B1 | European Patent Office (EPO) | B1 | |
| AT307454T | Austria | T | |
| DE602004000139D1 | Germany | D1 | |
| DE602004000139T2 | Germany | T2 | |
| US7586857B2 | United States of America | B2 | |
| CN1534972B | China | B | |
| JP4567359B2This record | Japan | B2 | |
| KR101129264B1 | Republic of Korea | B1 |
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Numbers
- Publication
- 4567359
- Publication, DOCDB
- 4567359
- Publication, EPODOC
- JP4567359B
- Application
- 108742
- Application, DOCDB
- 2004108742
- Application, EPODOC
- JP20040108742
Titles2
- Japanese
- ネットワーク・リソースの最適化による、エンド・ユーザの要求に応じた会議運営のための迅速なネットワークSIP/SDP手順
- English
- Rapid network SIP / SDP procedures for end-user-demanded conference operations by optimizing network resources
Classification
- CPC, 9
- H04M3/56
- H04L65/4038
- H04M7/006
- H04M2203/5018
- H04L65/1016
- H04L65/1043
- H04L65/1104
- H04L9/40
- H04M2201/14
- IPC, 8
- H04W28 18
- H04W80 10
- H04W88 18
- H04W4 16
- H04L12 24
- H04L29 06
- H04M3 56
- H04M7 00