WO2008040216A1 - Procédé de taxation d'appels et système et dispositif de taxation - Google Patents

Procédé de taxation d'appels et système et dispositif de taxation Download PDF

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Publication number
WO2008040216A1
WO2008040216A1 PCT/CN2007/070669 CN2007070669W WO2008040216A1 WO 2008040216 A1 WO2008040216 A1 WO 2008040216A1 CN 2007070669 W CN2007070669 W CN 2007070669W WO 2008040216 A1 WO2008040216 A1 WO 2008040216A1
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WIPO (PCT)
Prior art keywords
pulse
charging
call
sip
charging mode
Prior art date
Application number
PCT/CN2007/070669
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English (en)
French (fr)
Inventor
Xiaofeng Zheng
Ping Liu
Original Assignee
Huawei Technologies Co., Ltd.
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Huawei Technologies Co., Ltd. filed Critical Huawei Technologies Co., Ltd.
Priority to EP07801077A priority Critical patent/EP2088710A4/en
Publication of WO2008040216A1 publication Critical patent/WO2008040216A1/zh

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Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L12/00—Data switching networks
    • H04L12/02—Details
    • H04L12/14—Charging, metering or billing arrangements for data wireline or wireless communications
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066—Session management
    • H04L65/1101—Session protocols
    • H04L65/1104—Session initiation protocol [SIP]

Definitions

  • the present invention relates to the field of communications, and in particular, to a call charging method and a charging system and apparatus. Background technique
  • SIP Session Initiation Protocol
  • NGN Next Generation Net
  • IP Internet Protocol
  • SIP Internet Protocol
  • IETF Internet Engineering Task Force
  • SIP is built on the basis of Simple Mail Transfer Protocol (SMTP) and Hypertext Transfer Protocol (HTTP). Get up. SIP is used to establish, change, and terminate calls between IP-based users.
  • SMTP Simple Mail Transfer Protocol
  • HTTP Hypertext Transfer Protocol
  • SIP In order to provide telephone service, SIP also combines different standards and protocols to support the telephone service, such as Real Time Transport Protocol (RTP) for ensuring transmission, and resource reservation protocol for ensuring voice quality.
  • RTP Real Time Transport Protocol
  • RTP Resource reservation protocol
  • the Reservation Protocol (RSVP) which provides a Lightweight Direction Access Protocol (LDAP) for the directory, can authenticate the user's Remote Authentication Dial In User Service (RADIUS) and so on.
  • LDAP Lightweight Direction Access Protocol
  • RADIUS Remote Authentication Dial In User Service
  • the SIP network has two logical elements: a SIP user agent and a SIP network server.
  • the user agent is the terminal system element of the call, the user agent itself has the client element of the initial call and the server element that answers the call, and the SIP server is a network device that handles signaling associated with multiple calls for providing name resolution and user Positioning, that is, determining the specific server to resolve the address information according to the obtained email form address or the telephone number associated with the called party, currently has three server forms, SIP stateful proxy server, SIP stateless proxy server and SIP redirect server.
  • SIP proxy The server accepts the request, decides where to forward the requests, and passes them to the next server using the next hop routing principle; the stateful proxy server remembers the incoming request received, the echoed response, and the outgoing request it forwarded to Generate requests and try multiple possible user locations in parallel and send back the best response, which may be the local device closest to the user agent, used to control the user domain, is the primary platform for the application server; once the stateless proxy server forwards The request forgets all the information to quickly forward the request, and is the backbone in the SIP structure; the redirect server accepts the request, does not pass the request to the next server, but sends a response to the caller to indicate the called user the address of.
  • the communication system carrying the SIP protocol has user positioning, user capability, user availability, call setup, call processing, call forwarding, call number delivery, personal mobility, and terminal type negotiation. Services such as selection, terminal capability negotiation, calling and called authentication, uninformed and directed call forwarding, and multicast conference invitations.
  • the SIP protocol is increasingly favored by the communications industry and is becoming an important protocol in the NGN and 3G multimedia subsystem domains, and more and more SIP support is available on the market.
  • Protocol client software, intelligent multimedia terminals, and servers and softswitch devices implemented using the SIP protocol are implemented using the SIP protocol.
  • Analog user terminals such as coin-operated telephones, magnetic telephones, and IC card telephones, are still being used and will continue to be used for a certain period of time. Therefore, it is inevitable that the current operators The constructed SIP-based packet network communication system terminal must still support the access of the analog terminal, so that the packet network communication system must support the traditional Public Switched Telephone Network (PSTN) service.
  • PSTN Public Switched Telephone Network
  • the communication system includes a calling SIP user agent, a SIP call server, and a It is called a SIP user agent and a transport network.
  • the service control function entity in Figure 1 includes a SIP call server. The specific steps are as follows:
  • Step s101 the calling SIP user agent sends a SIP INVITE (invitation) message to the call server through the calling side transmission network to request to establish a call connection according to the call request of the simulated calling user; step s102, after the call server receives the SIP INVITE information, According to the routing related information in the SIP INVITE information, the control transmission network establishes a call connection, and the SIP INVITE information is sent to the called SIP user agent through the called side transmission network;
  • SIP INVITE invitation
  • Step sl03 the call server controls the entire communication system for session processing; Step s104, after the called SIP user agent detects the response response of the called user, sends a SIP 200 OK message indicating the response of the called user to the call server through the called side transmission network;
  • Step sl05 after receiving the SIP 200 OK message, the call server sends the SIP 200 OK message to the calling SIP user agent through the calling side transmission network, and establishes the call connection at the same time;
  • Step sl06 the calling SIP user agent sends a SIP ACK (acknowledgement) message to the call server for confirming that the calling SIP user agent receives the final response of the SIP INVITE message through the calling side transmission network;
  • SIP ACK acknowledgement
  • Step sl07 the call server receives the SIP ACK information, and sends the SIP ACK information to the called SIP user agent through the called side transmission network;
  • Step s108 the call server controls the entire system to perform the communication session of the call connection; in step s109, the calling SIP user agent sends a SIP BYE (end) information indicating that the calling user hangs up the call to the call server through the calling side transmission network;
  • SIP BYE end
  • Step s110 The call server receives the SIP BYE information, and sends the SIP BYE information to the called SIP user agent through the called side transmission network, and terminates the call connection.
  • the prior art SIP-based communication system implements the above-mentioned analog subscriber call service, but has the following drawbacks:
  • the user access device accesses the softswitch device by using the SIP protocol
  • the analog user call service in the SIP-based communication system cannot be performed. Pulse charging is performed, so that the operator cannot charge for the call service.
  • the technical problem to be solved by the present invention is to provide a call charging method and a charging system and apparatus.
  • the impulse charging of the analog service is implemented.
  • a call charging method including: a call charging method, including:
  • the call is pulsed according to the SIP control signal.
  • a billing system the billing system comprising:
  • the charging control device determines the pulse charging mode of the call according to the SIP request signal that is included in the simulated user side and includes the call attribute information, and generates a SIP control signal including the pulse charging mode;
  • the SIP control signal performs impulse charging on the call.
  • a charging control device comprising:
  • the pulse charging method configures a database unit to store a pulse charging method for performing pulse charging on the call
  • the pulse charging mode query unit queries the pulse charging mode configuration database unit according to the call attribute information in the SIP request signal, and obtains a query result including the call pulse charging mode; the charging control signal unit, according to the The result of the query is obtained by pulse charging mode, and a SIP control signal is generated.
  • a billing execution device includes:
  • the parsing unit parses the pulse charging mode of the call in the SIP control signal to obtain a pulse charging mode; the pulse sending unit sends a charging pulse according to the pulse charging mode;
  • the fee statistics unit performs impulse charging on the call according to the charging pulse.
  • the invention can realize the pulse charging of the analog user call service on the basis of simulating the user call service based on the SIP protocol-based communication system, and increase and improve the charging function of the SIP-based communication system.
  • FIG. 1 is a flow chart of a conventional SIP-based communication system simulating a user call service
  • FIG. 2 is a main flow chart of a call charging method according to an embodiment of the present invention
  • FIG. 3 is a flow chart of implementing call billing in a SIP-based communication system according to an embodiment of the present invention
  • FIG. 4 is a schematic diagram of a charging system according to an embodiment of the present invention.
  • FIG. 5 is a schematic structural diagram of a charging control apparatus according to an embodiment of the present invention.
  • FIG. 6 is a schematic structural diagram of a charge execution apparatus according to an embodiment of the present invention.
  • the basis of the embodiment of the present invention is to develop a simulated user call service in a communication system based on the SIP protocol.
  • the function of charging the analog user call is added, and the pulse charging mode is determined according to the SIP session request signal uploaded by the simulated calling party side, and the pulse delivery rule is obtained according to the parsing pulse charging mode, and the call is carried.
  • the subscriber line sends and receives the charging pulse for charging, which effectively solves the problem that the prior art cannot perform impulse charging for the simulated user calling service because there is no definition of the analog user call charging in the SIP protocol.
  • the embodiment of the present invention fully utilizes the SIP protocol-based communication system to perform the SIP signal in the process of simulating the user call, and the function of the pulse charging is loaded in the call connection process, and the following describes the embodiment of the present invention with reference to the accompanying drawings. Give specific instructions.
  • FIG. 2 the figure is a main flowchart of a call charging method according to an embodiment of the present invention, where the process mainly includes:
  • Step s201 Determine a pulse charging mode of the call according to a SIP session request signal that is sent by the calling user side and that includes the call attribute information, and generate a SIP control signal that includes the pulse charging mode.
  • Step s202 Perform pulse charging on the call according to the SIP control signal generated in step s201.
  • FIG. 3 is a flow chart of implementing call billing in a communication system based on the SIP protocol.
  • the communication system includes an analog user, a user access device, and a soft switch. Steps:
  • Step s301 The simulated user dials successfully, and initiates a dialing call to the user access device.
  • Step s302 The user access device receives the dialed call, and carries the SIP with the call attribute information.
  • the INVITE signal is sent to the softswitch, and the call attribute information may be the primary and called number;
  • Step s303 The softswitch establishes a call connection route between the primary and the called according to the call attribute information in the received SIP INVITE signal.
  • Step s304 The softswitch sends a SIP INVITE signal to the remote called user by using a call connection route.
  • Step s305 the soft switch controls the entire system for session processing
  • the softswitch determines, according to the call attribute information, a pulse charging mode for charging the call, the pulse charging mode may be a pulse sending interval information, a pulse number information in each pulse sending interval, and a pulse.
  • the determining process may be that the specific pulse charging mode is directly obtained from the pulse charging mode configuration database in the application server;
  • Step s307 The softswitch obtains a SIP 200 OK signal uploaded by the remote called party, and confirms the response of the called user.
  • Step s308, the softswitch transmits the SIP 200 OK signal carrying the pulse charging mode to the user access device that simulates the calling user terminal;
  • Step s309 The user access device parses the pulse charging mode in the received SIP 200 OK signal, and obtains a pulse charging mode to be captured.
  • Step s310 The user access device sends a charging pulse to and from the subscriber line carrying the call to perform charging for the call according to the determined pulse charging mode, and the number of pulses in the pulse sending interval and the pulse sending interval,
  • the frequency of the pulse issuance and the tariff value represented by each pulse can be determined by the rate of the current call, and the analog user terminal providing the public telephone service can identify the charging information by detecting the pulse to achieve the real-time charging purpose, wherein
  • the fee pulse is physically represented by a 16/12kHZ, a pulse signal having a width of not less than 50ms or a signal such as a reverse polarity pulse;
  • Step s311 the soft switch controls the entire system for subsequent session processing, and counts the current call cost.
  • the SIP 200 OK signal may be used to modify the SIP header field and/or the SIP message parameters to enable the SIP 200 OK signal to carry a pulse charging mode, wherein a new message media type may be defined in the SIP message to cover the pulse meter.
  • Fee mode information, description of the pulse billing method can refer to
  • Amet event package definitions in the H.248 protocol include the following:
  • EM Start Automatic Meter Type
  • MPB burst meter Meter Pulse Burst Type
  • PM-pulse-repetition-interval PM-pulse-repetition-interval
  • PM-Maximum pulse count per charge interval PM-MAX PCCI
  • Maximum number of billing intervals PM-repetition of Max PCCI
  • PM-REPX PM-REPX
  • PM-Minimum pulse per charge interval PM-MIN PCCI
  • minimum billing pulse The number of charging intervals (PM-repetition of Min PCCI, PM-PCN), the PM-charge interval (PM-CI), and the PM-phase duration (PM-PD).
  • the number of pulses in the interval of each pulse can be obtained by the number of pulses sent per interval, and the interval of the pulse is sent through the interval parameter of the event, so that the interval is sent through each pulse.
  • the number of pulses and the interval between the bursts of the bursts are used to implement billing. The process of how to implement billing according to which parameters are all prior art, and will not be described in detail herein.
  • the softswitch and the user access device may be defined by other agreed pulse charging methods according to the needs of the operator, and are not limited thereto.
  • the charging system of the embodiment of the present invention is described below with reference to FIG. 4.
  • the charging system includes: a charging control device 41, which determines a pulse meter of the call according to a SIP session request signal that includes call attribute information uploaded by the simulated user side. a fee method, generating a SIP control signal including the pulse charging mode;
  • the charging executing means 42 performs impulse charging on the call based on the SIP control signal.
  • the charging control apparatus includes:
  • the pulse charging mode configuration database unit 51 stores a pulse charging mode for performing pulse charging on the call
  • the pulse charging mode query unit 52 queries the pulse charging mode configuration database unit according to the call attribute information in the SIP session request signal, and obtains a query result including the call pulse charging mode;
  • the charging control signal unit 53 obtains a pulse charging mode according to the query result, and generates the SIP control signal.
  • the charging executing apparatus includes:
  • the parsing unit 61 parses the pulse charging mode of the call in the SIP control signal, and obtains a pulse charging mode to be captured;
  • the pulse sending unit 62 sends a charging pulse to and from the subscriber line carrying the call according to the pulse charging mode
  • the fee statistics unit 63 charges the call according to the charging pulse.
  • the charging control device is a softswitch
  • the charging executing device may be an integrated access device in the user access device, in order to implement pulse charging for the calling service of the analog user, in the above integrated access device and the analog user terminal.
  • the Z interface can be used for connection.
  • the SIP protocol interface can be used between the integrated access device and the softswitch.
  • the integrated access device is an analog user terminal connected to the next generation network (Next Generation Net,
  • the physical entity of the NGN supports the subscriber line signaling and the SIP protocol for processing the mutual conversion function of the circuit voice and the NGN voice between the analog user terminal and the NGN domain, and generates the call user attribute when receiving the analog user dialing call.
  • the SIP INVITE signal of the information and sends the SIP INVITE signal to the softswitch;
  • the softswitch provides the main functions of call control, routing connection, resource allocation, protocol processing, authentication, and accounting for users accessing the network in the NGN.
  • the call connection route between the master and the called party is established according to the primary and called numbers in the received SIP INVITE signal, and the SIP INVITE signal is sent to the remote called user, and at the same time, according to the call attribute information, the pair is determined.
  • the pulse charging mode of charging for the call after receiving the SIP 200 OK call answering signal uploaded by the remote called party, transmitting the SIP 200 OK signal carrying the pulse charging mode information to the analog calling party
  • the integrated access device of the user end controls the integrated access device to send a charging pulse to the subscriber line for statistical charging.
  • the charging executing device may be an access gateway in the user access device, and perform pulse charging for the calling service of the analog user, between the access gateway and the analog user terminal. It can be connected by Z interface, and the SIP protocol interface can be used between the access gateway and the server.
  • Z interface Z interface
  • SIP protocol interface can be used between the access gateway and the server.
  • the access gateway is a physical entity that simulates the user terminal accessing the NGN, and supports the subscriber line signaling and the SIP protocol, and is used for processing the mutual conversion function of the circuit voice and the NGN voice between the analog user terminal and the NGN domain, when receiving the analog user dialing When calling, generating a SIP INVITE signal containing the information of the calling user attribute, and sending the SIP INVITE signal to the charging server;
  • the accounting server provides a charging function for the user accessing the network in the NGN. After receiving the SIP INVITE signal uploaded from the access gateway of the analog calling user side, according to the call attribute information in the received SIP INVITE signal, Determining the pulse charging mode for charging the call; after receiving the SIP 200 OK call response signal uploaded by the remote called party, it will carry the pulse charging mode.
  • the SIP 200 OK signal of the information is transmitted to the access gateway of the analog calling client, and the access gateway is controlled to send a charging pulse to the subscriber line for statistical charging.
  • the SIP 200 OK signal may cause the SIP 200 OK signal to carry a pulse charging mode by modifying the SIP header field and/or the SIP message parameter, wherein a new message media type may be defined in the SIP message to cover the pulse meter.
  • the fee mode information, the description of the pulse charging mode can refer to the type defined by the Amet event package in the H.248 protocol as described above.
  • the pulse charging mode is realized by a combination of pulse sending interval information, pulse number information in each pulse sending interval, pulse sending frequency information, and a tariff value represented by each pulse, and both It can be determined by the rate of the current call.
  • the pulse sending interval information can be configured according to local data, and the entire pulse sending billing time is divided into several intervals, so that different charging intervals can be used at different charging intervals.
  • the charging policy; the analog user terminal providing the public telephone service can identify the charging information by detecting the pulse to achieve the real-time charging purpose, wherein the charging pulse is physically represented as a 16/12 kHZ pulse signal with a width of not less than 50 ms. Or a signal such as a reverse polarity pulse.

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Description

呼叫计费的方法及计费系统和装置 本申请要求于 2006 年 9 月 29 日提交中国专利局、 申请号为 200610122520.5、 发明名称为"呼叫计费的方法及计费系统 "的中国专利申请的 优先权， 其全部内容通过引用结合在本申请中。
技术领域
本发明涉及通信领域， 尤其涉及一种呼叫计费的方法及计费系统和装置。 背景技术
会话发起协议 ( Session Initiation Protocol , SIP ) 是下一代网络 ( Next Generation Net, NGN )中的重要协议， 越来越得到通信业界的重视。 开发 SIP 的目的是为了解决因特网协议（Internet Protocol, IP ) 网中的信令控制， 并同 软交换进行通信， 提供跨越因特网 （Internet ) 的高级电话业务。 而 IP电话正 在向一种正式的商业电话模式演进， SIP就是用来确保这种演进实现的 NGN 一系列协议中重要的一员。 SIP 作为互联网工程任务组 （ The Internet Engineering Task Force, IETF )标准进程的一部分， 是在诸如简单邮件传送协 议（ Simple Mail Transfer Protocol, SMTP )和超文本传送协议（ Hypertext Transfer Protocol, HTTP )基础上建立起来的。 SIP用来建立、 改变和终止基于 IP网 络的用户间的呼叫。 为了提供电话业务， SIP还结合了不同的标准和协议对电 话业务进行有力支持， 如用于确保传输的实时传输协议（Real Time Transport Protocol , RTP ) , 能确保语音质量的资源预留协议 ( Resource Reservation Protocol , RSVP ), 能够提供目录的轻量级目录服务（ Lightweight Direction Access Protocol, LDAP ), 能够鉴权用户的远程认证拨号用户服务（ Remote Authentication Dial In User Service, RADIUS )等等。
SIP网络有 SIP用户代理和 SIP网络服务器两个逻辑元素。 用户代理是呼 叫的终端系统元素，用户代理本身具有初始呼叫的客户机元素和应答呼叫的服 务器元素， 而 SIP服务器是处理与多个呼叫相关联信令的网络设备， 用于提供 名字解析和用户定位，即根据获得的电子邮件形式的地址或与被叫方关联的电 话号码确定特定服务器来解析地址信息， 目前具有三种服务器形式， SIP有状 态代理服务器， SIP无状态代理服务器和 SIP重定向服务器。 其中， SIP代理 服务器接受请求， 决定将这些请求传送到何处， 并且使用下一跳路由原理将它 们传送到下一服务器； 有状态代理服务器记住接收的入请求、 回送的响应和它 转送的出请求，以生成请求并且并行地尝试多个可能的用户位置并送回最好的 响应， 它可能是离用户代理最近的本地设备， 用于控制用户域， 是应用服务器 的主要平台； 无状态代理服务器一旦转送请求就忘记所有的信息， 以快速地转 送请求， 并且是 SIP结构中的骨干； 重定向服务器接受请求， 并不将这些请求 传递给下一服务器， 而是向呼叫者发送响应以指示被呼叫用户的地址。
由于 SIP协议提供的逻辑元素及功能机制，使得承载 SIP协议的通信系统 具有用户定位、 用户能力、 用户可用性、 呼叫建立、 呼叫处理、 呼叫前转、 呼 叫号码传递、 个人移动性、 终端类型的协商和选择、 终端能力协商、 主叫和被 叫鉴权、 不知情和指导式的呼叫转移及多播会议邀请等业务。
SIP协议凭借其简单、 易于扩展、 便于实现等诸多优点， 越来越得到通信 业界的青睐， 正逐步成为 NGN和 3G多媒体子系统域中的重要协议， 并且市 场上出现越来越多的支持 SIP协议的客户端软件、 智能多媒体终端， 以及用 SIP协议实现的服务器和软交换设备。 但现阶段通信技术发展处于过渡时期， 模拟用户终端， 如投币电话、 磁话机、 IC 卡话机， 仍被使用并将在一定时间 内被继续使用， 因此， 不可避免的是， 现阶段运营商建设的基于 SIP协议的分 组网通信系统终端仍必须支持模拟终端的接入，这样分组网通信系统必须要支 持传统的公共电话交换网 （Public Switched Telephone Network, PSTN )业务。
下面参照图 1 , 对现有技术的基于 SIP协议的通信系统， 实现上述模拟用 户终端呼叫连接及呼叫终止业务的流程进行说明， 在该通信系统中包括主叫 SIP用户代理、 SIP呼叫服务器、 被叫 SIP用户代理、 传输网络， 其中， 图 1 中的业务控制功能实体包括 SIP呼叫服务器， 具体步骤如下：
步骤 slOl ,主叫 SIP用户代理根据模拟主叫用户的呼叫请求，通过主叫侧 传输网络向呼叫服务器发送 SIP INVITE (邀请）信息， 请求建立呼叫连接； 步骤 sl02, 呼叫服务器接收 SIP INVITE信息后， 根据 SIP INVITE信息 中的路由相关信息， 控制传输网络建立呼叫连接， 并将 SIP INVITE信息通过 被叫侧传输网络向被叫 SIP用户代理下发；
步骤 sl03 , 呼叫服务器控制整个通信系统进行会话处理； 步骤 sl04,被叫 SIP用户代理检测到被叫用户的应答响应后，通过被叫侧 传输网络向呼叫服务器发送指示被叫用户应答的 SIP 200 OK信息；
步骤 sl05, 呼叫服务器接收 SIP 200 OK信息后， 将 SIP 200 OK信息通过 主叫侧传输网络向主叫 SIP用户代理下发， 同时建立此次呼叫连接；
步骤 sl06,主叫 SIP用户代理通过主叫侧传输网络向呼叫服务器发送用于 确认主叫 SIP用户代理收到 SIP INVITE信息最终响应的 SIP ACK (确认）信 息；
步骤 sl07, 呼叫服务器接收所述 SIP ACK信息， 并将该 SIP ACK信息通 过被叫侧传输网络发送至被叫 SIP用户代理；
步骤 sl08, 呼叫服务器控制整个系统进行此次呼叫连接的通信会话； 步骤 sl09,主叫 SIP用户代理通过主叫侧传输网络向呼叫服务器发送指示 主叫用户挂断呼叫的 SIP BYE (结束）信息；
步骤 sllO, 呼叫服务器接收到所述 SIP BYE信息， 将该 SIP BYE信息通 过被叫侧传输网络下发至被叫 SIP用户代理， 同时终止此次呼叫连接。
综上，现有技术的基于 SIP协议的通信系统虽然实现了上述模拟用户呼叫 业务， 但存在如下缺陷：
当用户接入设备釆用 SIP协议接入软交换设备时， 由于 SIP协议中目前没 有对模拟用户呼叫业务进行脉冲计费的定义过程， 因此， 无法对基于 SIP协议 的通信系统中模拟用户呼叫业务进行脉冲计费，使得运营商无法实现对该项呼 叫业务的收费。
发明内容
本发明所要解决的技术问题在于，提供一种呼叫计费的方法及计费系统和 装置。 在基于 SIP协议的通信系统模拟用户呼叫业务的基础上， 实现对模拟 用户的呼叫业务进行脉冲计费。
为了解决上述技术问题， 本发明提出了一种呼叫计费的方法， 包括： 一种呼叫计费的方法， 包括：
根据模拟用户侧上传的包含呼叫属性信息的 SIP请求信号，确定所述呼叫 的脉冲计费方式， 生成包含该脉冲计费方式的 SIP控制信号；
根据所述 SIP控制信号， 对所述呼叫进行脉冲计费。 一种计费系统， 该计费系统包括：
计费控制装置， 根据模拟用户侧上传的包含呼叫属性信息的 SIP请求信 号，确定所述呼叫的脉冲计费方式，生成包含该脉冲计费方式的 SIP控制信号； 计费执行装置， 根据所述 SIP控制信号， 对所述呼叫的进行脉冲计费。 一种计费控制装置， 包括：
脉冲计费方式配置数据库单元，存储用于对呼叫进行脉冲计费的脉冲计费 方式；
脉冲计费方式查询单元，根据 SIP请求信号中的呼叫属性信息， 查询所述 脉冲计费方式配置数据库单元， 获得包含所述呼叫脉冲计费方式的查询结果； 计费控制信号单元， 根据所述查询结果获得脉冲计费方式， 生成 SIP控制 信号。
一种计费执行装置， 包括：
解析单元，解析 SIP控制信号中呼叫的脉冲计费方式，得到脉冲计费方式； 脉冲下发单元， 根据所述脉冲计费方式， 下发计费脉冲；
费用统计单元， 根据所述计费脉冲， 对所述呼叫进行脉冲计费。
实施本发明， 具有如下有益效果：
釆用本发明可在基于 SIP协议的通信系统模拟用户呼叫业务的基础上， 实现对模拟用户呼叫业务进行脉冲计费，增加和完善了基于 SIP协议的通信系 统的计费功能。
附图说明
图 1是现有的基于 SIP协议的通信系统模拟用户呼叫业务的流程图； 图 2是本发明实施例的一种呼叫计费方法的主要流程图；
图 3是本发明实施例的基于 SIP协议的通信系统中实现呼叫计费的流程 图；
图 4是本发明实施例的计费系统示意图；
图 5是本发明实施例的计费控制装置的结构示意图；
图 6是本发明实施例的计费执行装置的结构示意图。
具体实施方式
本发明实施例在基于 SIP 协议的通信系统开展模拟用户呼叫业务的基础 上， 增加了对模拟用户呼叫进行计费的功能， 根据模拟主叫用户侧上传的 SIP 会话请求信号确定脉冲计费方式， 根据解析脉冲计费方式得到脉冲下发规则， 并在承载所述呼叫的用户线路上下发计费脉冲进行计费，有效解决了现有技术 的由于 SIP协议中无对模拟用户呼叫计费定义而无法对模拟用户呼叫业务进 行脉冲计费的问题。
需要说明的是，本发明实施例充分利用了基于 SIP协议的通信系统进行模 拟用户呼叫过程中的 SIP信号， 在该呼叫连接过程中加载了脉冲计费的功能， 下面结合附图对本发明实施例进行具体说明。
参照图 2, 该图是本发明实施例提供的一种呼叫计费方法的主要流程图， 该流程主要包括：
步骤 s201 ,根据模拟主叫用户侧上传的包含呼叫属性信息的 SIP会话请求 信号， 确定所述呼叫的脉冲计费方式， 生成包含该脉冲计费方式的 SIP控制信 号；
步骤 s202, 根据步骤 s201中生成的 SIP控制信号， 对所述呼叫进行脉冲 计费。
在上述呼叫计费主要流程基础上参考图 3 , 该图为基于 SIP协议的通信系 统中实现呼叫计费的流程图， 该通信系统包括模拟用户、用户接入设备和软交 换， 流程具体包括如下步骤：
步骤 s301 , 模拟用户拨号成功， 向用户接入设备发起拨号呼叫； 步骤 s302, 用户接入设备接收拨号呼叫， 并将携带有呼叫属性信息的 SIP
INVITE信号发至软交换， 该呼叫属性信息可以是主、 被叫号码；
步骤 s303 , 软交换根据接收到的 SIP INVITE信号中的呼叫属性信息建立 主、 被叫间的呼叫连接路由；
步骤 s304, 软交换通过呼叫连接路由， 将 SIP INVITE信号发送至远端被 叫用户；
步骤 s305 , 软交换控制整个系统进行会话处理；
步骤 s306 , 软交换根据呼叫属性信息， 确定对该次呼叫进行计费的脉冲 计费方式， 该脉冲计费方式可以是通过脉冲下发间隔信息、各脉冲下发间隔内 的脉冲数目信息、脉冲下发频率信息、每脉冲代表的资费值中的一个或多个组 合来实现，确定过程可以是从应用服务器中的脉冲计费方式配置数据库中直接 获取上述具体的脉冲计费方式；
步骤 s307 , 软交换获取远端被叫用户侧上传的 SIP 200 OK信号， 确认被 叫用户的应答；
步骤 s308, 软交换将携带有脉冲计费方式的 SIP 200 OK信号下传至模拟 主叫用户端的用户接入设备；
步骤 s309, 用户接入设备对接收到的 SIP 200 OK信号中的脉冲计费方式 进行解析， 得到所要釆取的脉冲计费方式；
步骤 s310 , 用户接入设备根据确定的脉冲计费方式， 在承载此次呼叫的 用户线路上下发计费脉冲进行对呼叫的计费，脉冲下发间隔、脉冲下发间隔内 的脉冲个数、脉冲下发频率以及每脉冲代表的资费值均可由本次呼叫的费率决 定， 而提供公话业务的模拟用户终端可通过检测脉冲来识别计费信息， 以达到 实时计费目的， 其中， 计费脉冲物理上表现为 16/12kHZ、 宽度不小于 50ms 的脉冲信号或者反极脉冲等信号；
步骤 s311 ,软交换控制整个系统进行后续会话处理，并统计本次呼叫费用。 上述 SIP 200 OK信号可通过对 SIP头域和 /或 SIP消息参数进行修改而使 SIP 200 OK信号携带有脉冲计费方式， 其中， 可在该 SIP消息中定义新的消 息媒体类型来涵盖脉冲计费方式信息， 对脉冲计费方式的描述可以参照
H.248协议中 Amet事件包定义的类型， 包含如下几种：
开始自动周期性脉冲 （Enable Meter Type, EM ), 包含有参数： 每时间间 隔下发的脉冲个数（ EM-pulse-count )、事件间隔（ EM-pulse-repetition-interval ); 突发性计费脉冲（ Meter Pulse Burst Type, MPB ), 包含有参数： 一次性下 发 脉 冲 个 数 （ MPB-burst-pulse-count ) 、 脉 冲 之 间 事 件 间 隔 ( MPB-pulse-repetition-interval );
按间隔区分计费类型 （ Phased Meter Type , ΡΜ ), 包含有参数脉冲间隔
( PM-pulse-repetition-interval )、计费间隔内最大脉冲数目（ PM-Maximum pulse count per charge interval, PM-MAX PCCI )、 釆用最大计费脉冲计费间隔数目 ( PM-repetition of Max PCCI , PM-REPX )、 计费间隔最大脉冲数目 ( PM-Minimum pulse per charge interval , PM-MIN PCCI )、 釆用最少计费脉冲 计费间隔数目 （PM-repetition ofMin PCCI, PM-PCN )、 计费间隔（PM-charge interval, PM-CI )、 每个计费间隔时长（ PM-phase duration, PM-PD )。
比如： 在 EM类型下， 可以通过每时间间隔下发的脉冲个数参数获得各脉 冲下发间隔内的脉冲数目信息， 通过事件间隔参数获得脉冲下发间隔信息，从 而通过各脉冲下发间隔内的脉冲数目信息和和脉冲下发间隔信息来实现计费。 具体根据哪些参数如何实现计费的过程均为现有技术， 在此不再详细叙述。
值得说明的是，软交换同用户接入设备之间可以根据运营商的需要而釆用 其他约定的脉冲计费方式定义， 不限于此。
下面参考图 4对本发明实施例的计费系统进行说明， 该计费系统包括： 计费控制装置 41 , 根据模拟用户侧上传的包含呼叫属性信息的 SIP会话 请求信号， 确定所述呼叫的脉冲计费方式， 生成包含该脉冲计费方式的 SIP控 制信号；
计费执行装置 42, 根据所述 SIP控制信号， 对所述呼叫进行脉冲计费。 具体的， 参照图 5的本发明实施例所述的计费控制装置的结构示意图， 该 计费控制装置包括：
脉冲计费方式配置数据库单元 51 , 存储用于对呼叫进行脉冲计费的脉冲 计费方式；
脉冲计费方式查询单元 52, 根据 SIP会话请求信号中的呼叫属性信息， 查询所述脉冲计费方式配置数据库单元，获得包含所述呼叫脉冲计费方式的查 询结果；
计费控制信号单元 53 , 根据所述查询结果获得脉冲计费方式， 生成所述 SIP控制信号。
具体的， 参照图 6的本发明实施例所述的计费执行装置的结构示意图， 该 计费执行装置包括：
解析单元 61 , 解析所述 SIP控制信号中所述呼叫的脉冲计费方式， 得到 所要釆取的脉冲计费方式；
脉冲下发单元 62, 根据所述脉冲计费方式， 在承载所述呼叫的用户线路 上下发计费脉冲；
费用统计单元 63 , 根据所述计费脉冲， 对所述呼叫计费。 当计费控制装置是软交换时，计费执行装置可以是用户接入设备中的综合 接入设备， 为实现对模拟用户的呼叫业务进行脉冲计费， 在上述综合接入设备 和模拟用户终端间可釆用 Z接口藕接， 在综合接入设备和软交换之间可釆用 SIP协议接口藕接， 其中各个设备的功能和相互关系如下述：
综合接入设备是模拟用户终端接入下一代网络（ Next Generation Net,
NGN )的物理实体，支持用户线信令及 SIP协议，用于在模拟用户终端和 NGN 域间处理电路话音和 NGN话音的相互转换功能， 当接收到模拟用户拨号呼叫 时，产生包含呼叫用户属性信息的 SIP INVITE信号， 并将该 SIP INVITE信号 发至软交换上；
软交换在 NGN中为接入网络的用户提供呼叫控制、路由接续、资源分配、 协议处理、 认证、 计费等主要功能， 当接收到从模拟主叫用户侧的综合接入设 备上传的 SIP INVITE信号后，根据接收到的 SIP INVITE信号中的主、被叫号 码建立主、 被叫间的呼叫连接路由， 并将 SIP INVITE信号发送至远端被叫用 户， 同时， 根据呼叫属性信息， 确定对该次呼叫进行计费的脉冲计费方式； 当 接收到远端被叫用户侧上传的 SIP 200 OK呼叫应答信号后， 将携带有脉冲计 费方式信息的 SIP 200 OK信号下传至模拟主叫用户端的综合接入设备， 控制 综合接入设备向用户线路下发计费脉冲进行统计计费。
当计费控制装置是计费服务器时，计费执行装置可以是用户接入设备中的 接入网关， 为实现对模拟用户的呼叫业务进行脉冲计费， 在上述接入网关和模 拟用户终端间可釆用 Z接口藕接，在接入网关和服务器之间可釆用 SIP协议接 口藕接， 其中各个设备的功能和相互关系如下述：
接入网关是模拟用户终端接入 NGN的物理实体， 支持用户线信令及 SIP 协议， 用于在模拟用户终端和 NGN域间处理电路话音和 NGN话音的相互转 换功能， 当接收到模拟用户拨号呼叫时， 产生包含呼叫用户属性信息的 SIP INVITE信号， 并将该 SIP INVITE信号发至计费服务器上；
计费服务器在 NGN中为接入网络的用户提供计费功能， 当接收到从模拟 主叫用户侧的接入网关上传的 SIP INVITE信号后， 根据接收到的 SIP INVITE 信号中的呼叫属性信息， 确定对该次呼叫进行计费的脉冲计费方式； 当接收到 远端被叫用户侧上传的 SIP 200 OK呼叫应答信号后， 将携带有脉冲计费方式 信息的 SIP 200 OK信号下传至模拟主叫用户端的接入网关， 控制接入网关向 用户线路下发计费脉冲进行统计计费。
上述 SIP 200 OK信号可通过对 SIP头域和 /或 SIP消息参数进行修改而使 SIP 200 OK信号携带有脉冲计费方式，其中，可在该 SIP消息中定义新的消息 媒体类型来涵盖脉冲计费方式信息，对脉冲计费方式的描述可以参照如上所述 H.248协议中 Amet事件包定义的类型。
其中，脉冲计费方式是通过脉冲下发间隔信息、各脉冲下发间隔内的脉冲 数目信息、脉冲下发频率信息和每脉冲代表的资费值的一种或多种的组合来实 现， 且均可由本次呼叫的费率决定， 而具体的， 脉冲下发间隔信息可以根据本 地数据配置，将整个脉冲下发计费时间分为若干个间隔，从而可在不同的计费 间隔釆用不同的计费策略；而提供公话业务的模拟用户终端可通过检测脉冲来 识别计费信息，以达到实时计费目的，其中，计费脉冲物理上表现为 16/12kHZ、 宽度不小于 50ms的脉冲信号或者反极脉冲等信号。
以上所揭露的仅为本发明一种较佳实施例而已，当然不能以此来限定本发 明之权利范围， 因此依本发明权利要求所作的等同变化， 仍属本发明所涵盖的 范围。

Claims

权 利 要 求
1、 一种呼叫计费的方法， 其特征在于， 包括：
根据模拟用户侧上传的包含呼叫属性信息的 SIP请求信号，确定所述呼叫 的脉冲计费方式， 生成包含该脉冲计费方式的 SIP控制信号；
根据所述 SIP控制信号， 对所述呼叫进行脉冲计费。
2、 如权利要求 1所述的呼叫计费的方法， 其特征在于， 确定所述呼叫的 脉冲计费方式包括：
根据所述 SIP请求信号中的呼叫属性信息， 查询脉冲计费方式配置数据 库， 获得包含所述呼叫脉冲计费方式的查询结果；
根据所述查询结果， 生成所述呼叫的脉冲计费方式。
3、如权利要求 1所述的呼叫计费的方法， 其特征在于， 所述根据所述 SIP 控制信号， 对所述呼叫进行脉冲计费的步骤包括：
解析所述 SIP控制信号中所述呼叫的脉冲计费方式， 得到脉冲计费方式； 根据所述脉冲计费方式， 下发计费脉冲；
根据所述计费脉冲， 对所述呼叫进行脉冲计费。
4、 如权利要求 1所述的呼叫计费的方法， 其特征在于， 所述 SIP控制信 号是通过 SIP头域和 /或 SIP消息参数携带所述脉冲计费方式的 SIP消息。
5、 如权利要求 1、 2、 3或 4所述的呼叫计费的方法， 其特征在于， 所述 脉冲计费方式中至少包括脉冲下发间隔信息，各脉冲下发间隔内的脉冲数目信 息， 脉冲下发频率信息， 每脉冲代表的资费值中的一种或任意组合。
6、 一种计费系统， 其特征在于， 该计费系统包括：
计费控制装置， 根据模拟用户侧上传的包含呼叫属性信息的 SIP请求信 号，确定所述呼叫的脉冲计费方式，生成包含该脉冲计费方式的 SIP控制信号； 计费执行装置， 根据所述 SIP控制信号， 对所述呼叫的进行脉冲计费。
7、 如权利要求 6所述的计费系统， 其特征在于， 所述计费控制装置包括： 脉冲计费方式配置数据库单元，存储用于对呼叫进行脉冲计费的脉冲计费 方式；
脉冲计费方式查询单元，根据所述 SIP请求信号中的呼叫属性信息， 查询 所述脉冲计费方式配置数据库单元，获得包含所述呼叫脉冲计费方式的查询结 果；
计费控制信号单元， 根据所述查询结果获得脉冲计费方式， 生成所述 SIP 控制信号。
8、 如权利要求 6所述的计费系统， 其特征在于， 所述计费执行装置包括： 解析单元， 解析所述 SIP控制信号中所述呼叫的脉冲计费方式，得到脉冲 计费方式；
脉冲下发单元， 根据所述脉冲计费方式， 下发计费脉冲；
费用统计单元， 根据所述计费脉冲， 对所述呼叫进行脉冲计费。
9、 如权利要求 6所述的计费系统， 其特征在于， 所述 SIP控制信号是通 过 SIP头域和 /或 SIP消息参数携带所述脉冲计费方式的 SIP消息。
10、 如权利要求 6、 7、 8或 9所述的计费系统， 其特征在于， 所述脉冲计 费方式中至少包括脉冲下发间隔信息， 各脉冲下发间隔内的脉冲数目信息，脉 冲下发频率信息， 每脉冲代表的资费值中的一种或任意组合。
11、 如权利要求 6、 7、 8或 9所述的计费系统， 其特征在于， 所述计费执 行装置是用户接入设备。
12、 如权利要求 11所述的计费系统， 其特征在于， 所述用户接入设备是 综合接入设备或接入网关。
13、 如权利要求 6、 7、 8或 9所述的计费系统， 其特征在于， 所述计费控 制装置是软交换或计费服务器。
14、 一种计费控制装置， 其特征在于， 包括：
脉冲计费方式配置数据库单元，存储用于对呼叫进行脉冲计费的脉冲计费 方式；
脉冲计费方式查询单元，根据 SIP请求信号中的呼叫属性信息， 查询所述 脉冲计费方式配置数据库单元， 获得包含所述呼叫脉冲计费方式的查询结果； 计费控制信号单元， 根据所述查询结果获得脉冲计费方式， 生成 SIP控制 信号。
15、 如权利要求 14所述的计费控制装置， 其特征在于， 所述脉冲计费方 式中至少包括脉冲下发间隔信息，各脉冲下发间隔内的脉冲数目信息， 脉冲下 发频率信息， 每脉冲代表的资费值中的一种或任意组合。
16、 如权利要求 14所述的计费控制装置， 其特征在于， 所述计费控制装 置是软交换或计费服务器。
17、 一种计费执行装置， 其特征在于， 包括：
解析单元，解析 SIP控制信号中呼叫的脉冲计费方式，得到脉冲计费方式； 脉冲下发单元， 根据所述脉冲计费方式， 下发计费脉冲；
费用统计单元， 根据所述计费脉冲， 对所述呼叫进行脉冲计费。
18、 如权利要求 17所述的计费执行装置， 其特征在于，
所述计费执行装置是用户接入设备；
所述用户接入设备是综合接入设备或接入网关。
PCT/CN2007/070669 2006-09-29 2007-09-11 Procédé de taxation d'appels et système et dispositif de taxation WO2008040216A1 (fr)

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CN101605038B (zh) * 2008-06-12 2012-03-28 朗讯科技公司 基于sip消息体的计费方法和系统
CN102014104B (zh) * 2009-09-04 2014-10-22 中兴通讯股份有限公司 在ip多媒体子系统中实现脉冲计费业务的方法及系统

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