WO2013063959A1 - Flash-based telephone service implementation method and system - Google Patents

Flash-based telephone service implementation method and system Download PDF

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Publication number
WO2013063959A1
WO2013063959A1 PCT/CN2012/078952 CN2012078952W WO2013063959A1 WO 2013063959 A1 WO2013063959 A1 WO 2013063959A1 CN 2012078952 W CN2012078952 W CN 2012078952W WO 2013063959 A1 WO2013063959 A1 WO 2013063959A1
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WO
WIPO (PCT)
Prior art keywords
user
flash
switching network
call
request
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PCT/CN2012/078952
Other languages
French (fr)
Chinese (zh)
Inventor
李刚
贺彬
张治华
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中兴通讯股份有限公司
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Publication of WO2013063959A1 publication Critical patent/WO2013063959A1/en

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Classifications

    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M3/00Automatic or semi-automatic exchanges
    • H04M3/42Systems providing special services or facilities to subscribers
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066Session management
    • H04L65/1069Session establishment or de-establishment
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M7/00Arrangements for interconnection between switching centres
    • H04M7/006Networks other than PSTN/ISDN providing telephone service, e.g. Voice over Internet Protocol (VoIP), including next generation networks with a packet-switched transport layer

Definitions

  • the present invention relates to the field of communications technologies, and in particular, to a method and system for implementing a telephone service based on Flash. Background technique
  • Flash technology can provide text display functions, and audio and video services can be performed based on Flash.
  • the Flash media server is centered, and the typical Flash media servers include: Adobe Flash Media Server and Red5.
  • the embodiment of the invention provides a method and a system for implementing a telephone service based on Flash, which is used to solve the problem that the telephone service according to the Flash in the related art cannot interact with the ordinary telephone user in the switching network, and the scalability is poor.
  • a method for implementing a telephone service based on Flash includes: receiving, by a service side system, an invitation request sent by a switching network;
  • the service side system sends a call arrival request to the Flash user according to the invitation request, and receives an answer request sent by the Flash user;
  • the service side system sends a response message to the session initiation protocol server SIP Server through the switching network according to the received answer request, and establishes a call;
  • the service side system notifies the Flash user that the call is successfully established, and completes the media data transmission between the Flash user and the switching network user.
  • Flash-based telephone service implementation method includes: The service side system receives a call request sent by the Flash user;
  • the service side system sends an invitation request to the session initiation protocol server SIP Server according to the call request; when receiving the successful response returned by the SIP server, the call response information is sent to the Flash user to establish a call;
  • the service side system notifies the switching network user that the call is successfully established, and completes the media data transmission between the Flash user and the switching network user.
  • a service side system provided by an embodiment of the present invention, where the system is configured to:
  • Another service side system provided by the embodiment of the present invention is configured to:
  • FIG. 1 is a schematic diagram of an overall implementation process of a Flash-based telephone answering service according to an embodiment of the present invention
  • FIG. 2 is a schematic structural diagram of a Flash-based telephone service implementation system according to an embodiment of the present invention
  • FIG. 3 is a note of the Flash-based telephone service of the present invention in conjunction with the system of the embodiment shown in FIG. 2. a detailed description of the process of the book;
  • FIG. 4 is a schematic diagram of a call initiation process in a Flash-based telephone service implementation according to an embodiment of the present invention
  • FIG. 5 is a process of receiving a call in a Flash-based telephone service according to an embodiment of the present invention.
  • the embodiment provides a Flash-based telephone service implementation method and system.
  • FIG. 1 is a schematic diagram of an overall implementation process of a Flash-based telephone answering service according to an embodiment of the present invention, including the following steps:
  • S101 The service side system receives an invitation invite request sent by the switching network.
  • Session Initiation Protocol (SIP) server (Server) of the switching network sends an invite request to the service side system.
  • SIP Session Initiation Protocol
  • S102 The service side system sends a call arrival request to the Flash user according to the request, and receives an answer request sent by the Flash user.
  • the service side system notifies the Flash user of the arrival of the call through a reverse invoke ( invoke) request, and simultaneously responds to the SIP Server 180 to try to send information.
  • the Flash user receives the call arrival request sent by the service side system, the user is prompted, and according to the operation of the user clicking the answer button, the service side system sends an answer request requesting to answer the call.
  • the service side system sends a response message to the session initiation protocol server SIP Server through the switching network according to the received answer request, and establishes a call.
  • the service side system sends a standard SIP 200 response according to the answer request to the Sip Server, until the service side system receives the acknowledgement character (ACK) from the calling party, and the call signaling interaction is completed, and the call succeeds. set up.
  • S104 The service side system notifies that the Flash call is successfully established, and completes media data transmission between the Flash user and the switching network user.
  • the process of receiving a call by the Flash user includes: the service side system receives the call request sent by the Flash user; the service side system sends an invitation request to the session initiation protocol server SIP Server according to the call request; Upon receiving the 180Ringing returned by the SIP server, the response message is sent to the Flash user to establish a call; the service side system notifies the switching network user that the call is successfully established, and completes the media data transmission between the Flash user and the switching network user.
  • the service-side system notifies the Flash user that the call is successfully established through a reverse invoke request, and gives a unique release stream and play stream name in the system.
  • the Flash user opens a stream (publish), that is, the local microphone audio is packaged and sent to the service side system in Real Time Messaging Protocal (RTMP) format, and the above play stream named by the service side system is played at the same time.
  • RTMP Real Time Messaging Protocal
  • the service-side system converts the media data data format (such as Nelly ⁇ Speex audio) released by the Flash user into a peer media format (for example, G.711 a-law audio), and sends it to the media port of the other party, and at the same time, from the opposite media.
  • the port receives the media data, converts the peer media data format (for example, G.711 a-law audio) into a media data format recognizable by the Flash user (for example, Nelly ⁇ Speex audio), packages the content in the RTMP format, and sends the data to the Flash user.
  • the Flash user can only be responsible for the interface presentation and the collection and playback of the media data, thereby reducing the core logic and interface coupling degree of the Flash user.
  • the service side system is equivalent to a plurality of softphone objects, and there is no logical connection between the plurality of softphone objects.
  • the process of initiating or receiving a call from a softphone object is handled by the standard sip protocol. Whether it is a call within the system, or an inter-system call or a call over an external exchange network, it is logically consistent, minimizing the need to modify the Flash softphone when the service changes, and the application is highly scalable.
  • the system is easy to expand. Since there is no logical connection between Flash phones, the system capacity can be increased by simply increasing the number of servers.
  • Flash One of the benefits of Flash itself is that 99% of Internet users have installed and are using flash controls, so the terminal side phone software eliminates the need for installation.
  • the service side system completes the Flash user and exchanges the network telephone. Before the telephone service between users, the Flash user needs to complete registration.
  • the specific registration process includes: The service side system receives an invoke registration request sent by the Flash user; the service side system generates a SIP registration request by the registration request, and sends the SIP registration request to the SIP server. ; Notify the Flash user of the registration result information returned by the SIP Server.
  • FIG. 2 is a schematic structural diagram of a Flash-based telephone service implementation system according to an embodiment of the present invention.
  • the system includes: a softswitch system 21, a service side system 22, a Flash Client 231, and an external user agent 24.
  • the monthly service side system is mainly divided into two parts: RTMP adaptation module and user agent class module.
  • the RTMP adaptation module is mainly responsible for interfacing with the Flash user 23 (RTMP-based invoke signaling and media);
  • the user agent class module is mainly responsible for signaling interaction with the softswitch system 21 and performing media exchange with the external user agent 24.
  • the service side system 22 receives the invitation request sent by the softswitch system 21; sends a call arrival request to the Flash user 23 according to the invitation request, and receives the answer request sent by the Flash user 23;
  • the answering request sends a response message to the session initiation protocol server SIP Server through the softswitch network 21 to establish a call; notifies the Flash user 23 that the call is successfully established, and completes the media between the Flash user 23 and the exchange network user (external user agent 24). data transmission.
  • the Flash user 23 implements a telephone service through a terminal interface, and a Flash client 231 is installed on the terminal, and the user 23 initiates and answers the call by using a button operation on the Flash client 231.
  • the Flash user 23 is further configured to send a registration request to the service side system 22 before the telephone service is implemented by the service side system; the service side system 22 is further configured to receive the registration request sent by the Flash user 23; and generate the SIP registration by using the registration request. Request, and send to the SIP Server; notify the Flash user 23 of the registration result information returned by the SIP Server.
  • the service side system 22 includes two modules, an RTMP adaptation logic module 221 and a SIP user agent class module 222.
  • FIG. 3 is a detailed description of a registration process of a Flash-based telephone service according to an embodiment of the present invention in conjunction with the system shown in FIG. 2, the process comprising the following steps:
  • S301 The callback function of the RTMP adaptation logic module 221 in the service side system 22 receives an invoke registration request from the Flash user 23, where the invoke registration request carries the Flash. Identification information of the household 23.
  • the service side system 22 sends a SIP registration request to the Sip Server.
  • the registration request of the SIP carries the identification information of the Flash user 23 and the address information of the service side system.
  • the Sip Server can route related SIP commands corresponding to the identification information to the service side system.
  • S303 The service side system 22 receives the registration response from the Sip Server.
  • step S304 The service side system 22 determines whether the registration is successful according to the returned registration response. When the registration is successful, the process proceeds to step S305. If the registration is unsuccessful, the process proceeds to step S306.
  • S305 The service side system 22 sends an RTMP adaptation logic module 221 through an interface function.
  • the service side system 22 calls the RTMP adaptation logic module 221 to send the Invoke-RegFail to the Flash user 23 through the interface function, and then proceeds to step S101 to allow the Flash user 23 to initiate the registration request again.
  • FIG. 4 is a process of starting a call in a Flash-based telephone service implementation process according to an embodiment of the present invention, where the process includes the following steps:
  • S401 The Flash user inputs a number in the Flash client, clicks the call button, and sends an Invoke-Call call request to the service side system.
  • S402 The service side system invokes a callback function of the RTMP adaptation logic module to obtain an Invoke-Call call request from the Flash user, and sends a Si command-invitation request to the Sip Server, where the Sip command carries the service-side system.
  • Session description (SDP) information the SDP information includes the address of the service side system, supported media formats (eg, G.711 a-law audio and G.711 u-law audio), and media ports.
  • the media port assignment may be Each time a different port is assigned, multiple calls (Users) can share a Local User Datagram Protocol (UDP) port.
  • UDP Local User Datagram Protocol
  • step S403 The service-side system starts a timer, and determines whether the Sip command -180Ringing from the Sip Server is received within the timing length of the timer. When the determination result is yes, step S404 is performed. When the result of the determination is no, the process proceeds to step S407.
  • S404 The service side system sends an answer message Invoke-alerting to the Flash user through the interface, and the Flash user's client prompts the Flash user peer to ring through the voice and text.
  • S405 Start another timer, and determine whether the Sip command S200 command of the Sip Server is received within the time limit of the timer. When the determination result is yes, proceed to step S406. When the determination result is no, perform the step. S407.
  • S406 Send the SIP command -ACK to the Sip Server, send the Invoke-establish to the Flash user and start the media exchange.
  • the service side system sends the RTMP adaptation logic module by using an interface function.
  • the Flash user Invoke-establish to the Flash user, specifying the name of the playback stream and the distribution stream.
  • the Flash user's client prompts the Flash user that the call has been established through voice and text.
  • the Flash user opens the stream (publish), that is, the local microphone audio is packaged and sent to the service side system in RTMP format, and the stream stream is played (stream.play), which is the above-mentioned playback stream named by the service side system.
  • the service side system initiates media exchange between the parallel Flash user and the call peer.
  • the specific media processing process includes: the service side system obtains an uplink media stream from the Flash user through a callback function of the RTMP adaptation logic module, and decodes the same into Pulse Coding Modulation (PCM) data.
  • PCM Pulse Coding Modulation
  • the decoded PCM is encoded in a negotiated format, such as G.711 a-law audio format encoding, and packaged into a Real-time Transport Protocol (RTP) packet, which is sent to an external user agent at the opposite end of the call.
  • a negotiated format such as G.711 a-law audio format encoding
  • the service side system also receives RTP packets from the external user agent of the opposite end of the call and decodes them into PCM data in a negotiated format.
  • the decoded PCM data is encoded into a media format recognizable by the Flash user, for example, the Speex, Nelly Moser audio format, and the RTMP adaptation logic module is called by the interface function to generate an RTMP data packet to be sent to the Flash user, which is a downlink media stream.
  • FIG. 5 is a process of receiving a call in a Flash-based telephone service implementation according to an embodiment of the present invention, where the process includes the following steps:
  • S501 The service side system receives a Sip command from the Sip Server - an invite (Invite) request.
  • S502 The service side system responds to the Sip Server with the Sip command -180 Ringing.
  • S503 The service side system invokes the RTMP adaptation logic module by using an interface function, and sends an Invoke-incomming request to the Flash user.
  • step S504 Start a timer, determine whether the answer request sent by the Flash user is received within the time limit of the timer, and if the answer request is received, proceed to step S505. If the answer request is received, proceed to step S508.
  • the Flash user's client makes an answer request.
  • S505 The service side system sends a Sip command -200OK to the Sip Server, where the 200 OK carries the SDP information of the service side system.
  • the SDP information includes an address of the service side system, a supported media format (for example, G.711 a law audio and G.711 u law audio), and a media port.
  • the media port can be assigned with different ports each time, or multiple calls can share a local UDP port.
  • S506 The service-side system starts a timer, and determines whether the Sip command-ACK from the Sip Server is received within the timing length of the timer. When the determination result is yes, the process proceeds to step S507. If the determination result is no, the process proceeds to step S508. .
  • the service side system sends an Invoke-establish request to the Flash user, and starts media exchange. Specifically, the service side system calls the RTMP adaptation logic module to send the Invoke-establish to the Flash user through the interface function, and specifies the names of the play stream and the release stream. At this time, the Flash user's client prompts the Flash user to establish a call through voice and text.
  • the Flash user opens the microphone (stream.publish), that is, the local microphone audio is packaged and sent to the service side system in the RTMP format, and the above-mentioned playback stream named by the service side system, that is, the downlink media stream, is played (stream.play). This The service side system initiates media exchange between the parallel Flash user and the call peer.
  • the specific media processing process includes: the service side system obtains an uplink media stream from the Flash user by using a callback function of the RTMP adaptation logic module, and decodes the same into a Primary Code Modulation (PCM) data.
  • PCM Primary Code Modulation
  • the decoded PCM is encoded in a negotiated format, such as G.711 a-law audio format encoding, and packaged into a Real-time Transport Protocol (RTP) packet, which is sent to an external user agent at the opposite end of the call.
  • RTP Real-time Transport Protocol
  • the service side system also receives RTP packets from the external user agent of the opposite end of the call and decodes them into PCM data in a negotiated format.
  • the decoded PCM data is encoded into a media format recognizable by the Flash user, for example, the Speex, Nelly Moser audio format, and the RTMP adaptation logic module is called by the interface function to generate an RTMP data packet to be sent to the Flash user, which is a downlink media stream.
  • each module/unit in the foregoing embodiment may be implemented in the form of hardware, or may use software functions.
  • the form of the module is implemented. The invention is not limited to any specific form of combination of hardware and software.

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Abstract

A Flash-based telephone service implementation method and system. The method includes: a service-side system sending a call arrival request to a Flash user according to a received invitation request sent by an exchange network, receiving an answer request sent by the Flash user, establishing a call, notifying the Flash user that the call is established successfully, and completing the data transmission between the Flash user and an exchange network user.

Description

一种基于 Flash的电话业务实现方法及系统  Method and system for implementing telephone service based on Flash
技术领域 Technical field
本发明涉及通信技术领域,尤其涉及一种基于 Flash的电话业务实现方法 及系统。 背景技术  The present invention relates to the field of communications technologies, and in particular, to a method and system for implementing a telephone service based on Flash. Background technique
Flash技术可以提供文本显示功能,并且基于 Flash可以进行音视频业务。 相关技术中基于 Flash进行电话业务时, 是以 Flash媒体服务器为中心的, 典 型的 Flash媒体服务器包括: Adobe Flash Media Server和 Red5。  Flash technology can provide text display functions, and audio and video services can be performed based on Flash. In the related art, when the telephone service is based on Flash, the Flash media server is centered, and the typical Flash media servers include: Adobe Flash Media Server and Red5.
目前, 基于 Flash进行电话业务时, 通话双方只能为 Flash用户, Flash 用户不能和交换网络中的普通电话用户进行互通,并且各个 Flash用户之间的 数据交换依赖于 Flash媒体服务器, 不便于扩展。 发明内容  At present, when the telephone service is based on Flash, the two parties can only be Flash users, and the Flash users cannot communicate with ordinary telephone users in the switching network, and the data exchange between the Flash users depends on the Flash media server, which is inconvenient to expand. Summary of the invention
本发明实施例提供一种基于 Flash的电话业务实现方法及系统,用以解决 相关技术中基于 Flash进行电话业务时,不能与交换网络中的普通电话用户进 行交互, 并且扩展性差的问题。  The embodiment of the invention provides a method and a system for implementing a telephone service based on Flash, which is used to solve the problem that the telephone service according to the Flash in the related art cannot interact with the ordinary telephone user in the switching network, and the scalability is poor.
本发明实施例提供的一种基于 Flash的电话业务实现方法, 包括: 服务侧系统接收交换网络发送的邀请请求;  A method for implementing a telephone service based on Flash according to an embodiment of the present invention includes: receiving, by a service side system, an invitation request sent by a switching network;
服务侧系统根据该邀请请求向 Flash用户发送呼叫到达请求,并接收 Flash 用户发送的接听请求;  The service side system sends a call arrival request to the Flash user according to the invitation request, and receives an answer request sent by the Flash user;
服务侧系统根据接收到的该接听请求, 通过交换网络向会话发起协议服 务器 SIP Server发送应答信息, 建立呼叫;  The service side system sends a response message to the session initiation protocol server SIP Server through the switching network according to the received answer request, and establishes a call;
服务侧系统通知 Flash用户呼叫建立成功, 并完成 Flash用户与交换网络 用户之间的媒体数据传输。  The service side system notifies the Flash user that the call is successfully established, and completes the media data transmission between the Flash user and the switching network user.
本发明实施例提供的另一种基于 Flash的电话业务实现方法, 包括: 服务侧系统接收 Flash用户发送的呼叫请求; Another Flash-based telephone service implementation method provided by the embodiment of the present invention includes: The service side system receives a call request sent by the Flash user;
服务侧系统根据该呼叫请求, 向会话发起协议服务器 SIP Server发送邀 请请求; 当接收到 SIP Server返回的成功应答时,向 Flash用户发送被叫应答信息, 建立呼叫;  The service side system sends an invitation request to the session initiation protocol server SIP Server according to the call request; when receiving the successful response returned by the SIP server, the call response information is sent to the Flash user to establish a call;
服务侧系统通知交换网络用户呼叫建立成功,并完成 Flash用户与交换网 络用户之间的媒体数据传输。  The service side system notifies the switching network user that the call is successfully established, and completes the media data transmission between the Flash user and the switching network user.
本发明实施例提供的一种服务侧系统, 所述系统设置为:  A service side system provided by an embodiment of the present invention, where the system is configured to:
根据接收到的来自交换网络用户的邀请请求,向 Flash用户发送呼叫到达 请求, 并接收 Flash用户发送的接听请求, 根据接收到的该接听请求, 通过交 换网络向会话发起协议服务器 SIP Server发送应答信息,建立呼叫,通知 Flash 用户呼叫建立成功, 并完成 Flash用户与交换网络用户之间的媒体数据传输。 本发明实施例提供的另一种服务侧系统, 设置为:  Sending a call arrival request to the Flash user according to the received invitation request from the exchange network user, and receiving an answer request sent by the Flash user, and sending a response message to the session initiation protocol server SIP Server through the switching network according to the received answer request. Establish a call, notify the Flash user that the call is successfully established, and complete the media data transmission between the Flash user and the switching network user. Another service side system provided by the embodiment of the present invention is configured to:
接收 Flash用户发送的呼叫请求;根据该呼叫请求, 向会话启动协议服务 器 SIP Server发送邀请请求; 当接收到 SIP Server返回的成功应答时,向 Flash 用户发送被叫应答信息, 建立呼叫; 通知交换网络用户呼叫建立成功, 并完 成 Flash用户与交换网络用户之间的媒体数据传输。  Receiving a call request sent by the Flash user; sending an invitation request to the session initiation protocol server SIP Server according to the call request; sending a called response message to the Flash user, establishing a call when receiving the successful response returned by the SIP server; notifying the switching network The user call is successfully established, and media data transmission between the Flash user and the switching network user is completed.
由于在本发明实施例中无论是系统内部的通话, 还是系统外部的, 逻辑 上都是一样的, 因此最大程度减少了业务变化时需要修改 Flash软电话的可 能, 应用扩展性强, 由于各 Flash用户间无逻辑联系, 通过简单的增加服务器 的数量即可增加系统容量。  In the embodiment of the present invention, whether the call inside the system or the outside of the system is logically the same, the possibility of modifying the Flash soft phone when the service change is minimized is minimized, and the application is highly scalable, because each Flash There is no logical connection between users, and system capacity can be increased by simply increasing the number of servers.
附图概述 BRIEF abstract
图 1为本发明实施例提供的一种基于 Flash的电话接听业务的总体实现流 程;  FIG. 1 is a schematic diagram of an overall implementation process of a Flash-based telephone answering service according to an embodiment of the present invention;
图 2为本发明的一实施例提供的基于 Flash的电话业务实现系统的结构示 意图;  2 is a schematic structural diagram of a Flash-based telephone service implementation system according to an embodiment of the present invention;
图 3为结合图 2所示实施例的系统对本发明的基于 Flash的电话业务的注 册过程进行的详细说明; 3 is a note of the Flash-based telephone service of the present invention in conjunction with the system of the embodiment shown in FIG. 2. a detailed description of the process of the book;
图 4为本发明的一实施例提供的基于 Flash的电话业务实现过程中起呼的 过程;  FIG. 4 is a schematic diagram of a call initiation process in a Flash-based telephone service implementation according to an embodiment of the present invention; FIG.
图 5为本发明的一实施例提供的基于 Flash的电话业务实现过程中接听的 过程。  FIG. 5 is a process of receiving a call in a Flash-based telephone service according to an embodiment of the present invention.
本发明的较佳实施方式 Preferred embodiment of the invention
下文中将结合附图对本发明的实施例进行详细说明。 需要说明的是, 在 不冲突的情况下, 本申请中的实施例及实施例中的特征可以相互任意组合。  Embodiments of the present invention will be described in detail below with reference to the accompanying drawings. It should be noted that, in the case of no conflict, the features in the embodiments and the embodiments in the present application may be arbitrarily combined with each other.
为了实现 Flash用户与交换网的普通电话用户之间的电话互通, 并提高 In order to realize the intercommunication between the Flash user and the ordinary telephone user of the switching network, and improve
Flash用户的扩展性, 本实施例提供了一种基于 Flash的电话业务实现方法及 系统。 The scalability of the Flash user, the embodiment provides a Flash-based telephone service implementation method and system.
图 1为本发明的一实施例提供的一种基于 Flash的电话接听业务的总体实 现流程, 包括以下步骤:  FIG. 1 is a schematic diagram of an overall implementation process of a Flash-based telephone answering service according to an embodiment of the present invention, including the following steps:
S101 : 服务侧系统接收交换网络发送的邀请 invite请求。  S101: The service side system receives an invitation invite request sent by the switching network.
当交换网络有来电时, 交换网络的会话发起协议 (Session Initiation Protocol, SIP )服务器( Server ) , 向服务侧系统发送 invite请求。  When there is an incoming call on the switching network, the Session Initiation Protocol (SIP) server (Server) of the switching network sends an invite request to the service side system.
S102: 服务侧系统根据该请求向 Flash用户发送呼叫达到请求, 并接收 Flash用户发送的接听请求。  S102: The service side system sends a call arrival request to the Flash user according to the request, and receives an answer request sent by the Flash user.
服务侧系统通过反向的调用 ( invoke )请求通知 Flash用户有呼叫到达, 同时回应给 SIP Server 180 trying信息。当 Flash用户接收到服务侧系统发送的 呼叫达到请求后, 提示用户, 并根据用户点击接听按钮的操作, 向服务侧系 统发送接听请求, 要求接听该来电。  The service side system notifies the Flash user of the arrival of the call through a reverse invoke ( invoke) request, and simultaneously responds to the SIP Server 180 to try to send information. When the Flash user receives the call arrival request sent by the service side system, the user is prompted, and according to the operation of the user clicking the answer button, the service side system sends an answer request requesting to answer the call.
S103 : 服务侧系统根据接收到的该接听请求, 通过交换网络向会话启动 协议服务器 SIP Server发送应答信息, 建立呼叫。  S103: The service side system sends a response message to the session initiation protocol server SIP Server through the switching network according to the received answer request, and establishes a call.
在本发明实施例中服务侧系统根据该接听请求生成标准的 SIP 200应答 发送给 Sip Server, 直到服务侧系统收到来自主叫的确认字符(ACK ) , 至 此, 呼叫信令交互完成, 呼叫成功建立。 S104: 服务侧系统通知 Flash呼叫建立成功, 并完成 Flash用户与交换网 络用户之间的媒体数据传输。 In the embodiment of the present invention, the service side system sends a standard SIP 200 response according to the answer request to the Sip Server, until the service side system receives the acknowledgement character (ACK) from the calling party, and the call signaling interaction is completed, and the call succeeds. set up. S104: The service side system notifies that the Flash call is successfully established, and completes media data transmission between the Flash user and the switching network user.
上述为 Flash用户的接听过程, 对于 Flash用户的主动发起呼叫的过程包 括:服务侧系统接收 Flash用户发送的呼叫请求;服务侧系统根据该呼叫请求, 向会话启动协议服务器 SIP Server发送邀请请求; 当接收到 SIP Server返回的 180Ringing时, 向 Flash用户发送应答信息, 建立呼叫; 服务侧系统通知交换 网络用户呼叫建立成功,并完成 Flash用户与交换网络用户之间的媒体数据传 输。  The process of receiving a call by the Flash user includes: the service side system receives the call request sent by the Flash user; the service side system sends an invitation request to the session initiation protocol server SIP Server according to the call request; Upon receiving the 180Ringing returned by the SIP server, the response message is sent to the Flash user to establish a call; the service side system notifies the switching network user that the call is successfully established, and completes the media data transmission between the Flash user and the switching network user.
服务侧系统通过反向的 invoke请求通知 Flash用户呼叫建立成功, 并给 出一个系统中独一无二的发布流和播放流名称。 Flash用户打开麦克风 (stream publish), 即将本地麦克风音频以实时消息协议(Real Time Messaging Protocal , RTMP )格式打包发送给服务侧系统, 同时播放 (stream play)由服务 侧系统命名的上述播放流。  The service-side system notifies the Flash user that the call is successfully established through a reverse invoke request, and gives a unique release stream and play stream name in the system. The Flash user opens a stream (publish), that is, the local microphone audio is packaged and sent to the service side system in Real Time Messaging Protocal (RTMP) format, and the above play stream named by the service side system is played at the same time.
服务侧系统将 Flash用户发布的媒体数据数据格式(例如 Nelly\Speex音 频)转换成对端的媒体格式 (例如 G.711 a律音频), 并发送到对方媒体端口, 与此同时,从对端媒体端口接收媒体数据, 将对端媒体数据格式(例如 G.711 a律音频 )转换成 Flash用户可识别的媒体数据格式(例如 Nelly\Speex音频 ) , 按 RTMP格式打包, 发送给 Flash用户。  The service-side system converts the media data data format (such as Nelly\Speex audio) released by the Flash user into a peer media format (for example, G.711 a-law audio), and sends it to the media port of the other party, and at the same time, from the opposite media. The port receives the media data, converts the peer media data format (for example, G.711 a-law audio) into a media data format recognizable by the Flash user (for example, Nelly\Speex audio), packages the content in the RTMP format, and sends the data to the Flash user.
由于在本发明实施例中将软电话的核心逻辑放在服务侧系统实现, Flash 用户可以只负责界面呈现以及媒体数据的釆集和播放, 因此可以降低 Flash 用户的核心逻辑和界面耦合度。 另外, 服务侧系统相当于多个软电话对象, 多个软电话对象间无逻辑联系。 软电话对象发起或者接听电话的流程均^^ 于标准的 sip协议。无论是在本系统内通话,或是系统间通话或者是跨外部交 换网络的通话, 在逻辑上一致, 最大程度减少了业务变化时需要修改 Flash 软电话的可能, 应用扩展性强。 并且系统扩容方便, 由于各 Flash电话间无逻 辑联系, 通过简单的增加服务器的数量即可增加系统容量。  Since the core logic of the softphone is implemented in the service side system in the embodiment of the present invention, the Flash user can only be responsible for the interface presentation and the collection and playback of the media data, thereby reducing the core logic and interface coupling degree of the Flash user. In addition, the service side system is equivalent to a plurality of softphone objects, and there is no logical connection between the plurality of softphone objects. The process of initiating or receiving a call from a softphone object is handled by the standard sip protocol. Whether it is a call within the system, or an inter-system call or a call over an external exchange network, it is logically consistent, minimizing the need to modify the Flash softphone when the service changes, and the application is highly scalable. Moreover, the system is easy to expand. Since there is no logical connection between Flash phones, the system capacity can be increased by simply increasing the number of servers.
而 Flash本身的一个好处是 99%的互联网用户,已经安装并正在使用 flash 控件, 因此终端侧电话软件中就免去了安装的需求。  One of the benefits of Flash itself is that 99% of Internet users have installed and are using flash controls, so the terminal side phone software eliminates the need for installation.
另夕卜,在本发明实施例中,服务侧系统在完成 Flash用户和交换网络电话 用户之间的电话业务之前, Flash用户需要完成注册, 具体的注册过程包括: 服务侧系统接收 Flash用户发送的 invoke注册请求; 服务侧系统将所述注册 请求生成 SIP注册请求, 并发送给 SIP Server; 将 SIP Server返回的注册结果 信息通知 Flash用户。 In addition, in the embodiment of the present invention, the service side system completes the Flash user and exchanges the network telephone. Before the telephone service between users, the Flash user needs to complete registration. The specific registration process includes: The service side system receives an invoke registration request sent by the Flash user; the service side system generates a SIP registration request by the registration request, and sends the SIP registration request to the SIP server. ; Notify the Flash user of the registration result information returned by the SIP Server.
图 2为本发明的一实施例提供的基于 Flash的电话业务实现系统的结构 示意图, 该系统包括: 软交换系统 21、 服务侧系统 22、 Flash Client231和外 部用户代理 24。 月良务侧系统主要划分为两个部分: RTMP适配模块和用户代 理类模块。 RTMP适配模块主要负责与 Flash用户 23接口 (基于 RTMP的 invoke信令和媒体); 用户代理类模块主要负责与软交换系统 21进行信令交 互, 与外部用户代理 24进行媒体交换。  2 is a schematic structural diagram of a Flash-based telephone service implementation system according to an embodiment of the present invention. The system includes: a softswitch system 21, a service side system 22, a Flash Client 231, and an external user agent 24. The monthly service side system is mainly divided into two parts: RTMP adaptation module and user agent class module. The RTMP adaptation module is mainly responsible for interfacing with the Flash user 23 (RTMP-based invoke signaling and media); the user agent class module is mainly responsible for signaling interaction with the softswitch system 21 and performing media exchange with the external user agent 24.
具体的, 以接听业务为例, 服务侧系统 22接收软交换系统 21发送的邀 请请求; 根据该邀请请求向 Flash用户 23发送呼叫到达请求, 并接收 Flash 用户 23发送的接听请求; 根据接收到的该接听请求, 通过软交换网络 21向 会话启动协议服务器 SIP Server发送应答信息,建立呼叫; 通知 Flash用户 23 呼叫建立成功, 并完成 Flash用户 23与交换网络用户 (外部用户代理 24 )之 间的媒体数据传输。  Specifically, taking the answering service as an example, the service side system 22 receives the invitation request sent by the softswitch system 21; sends a call arrival request to the Flash user 23 according to the invitation request, and receives the answer request sent by the Flash user 23; The answering request sends a response message to the session initiation protocol server SIP Server through the softswitch network 21 to establish a call; notifies the Flash user 23 that the call is successfully established, and completes the media between the Flash user 23 and the exchange network user (external user agent 24). data transmission.
具体的该 Flash用户 23通过终端界面实现电话业务, 在该终端上安装有 Flash客户端 231 , 用户 23发起和接听呼叫都是通过在 Flash客户端 231上的 按键操作实现的。 Flash用户 23在通过服务侧系统实现电话业务之前, 还用 于向服务侧系统 22发送注册请求; 服务侧系统 22,还用于接收 Flash用户 23 发送的注册请求; 将所述注册请求生成 SIP注册请求, 并发送给 SIP Server; 将 SIP Server返回的注册结果信息通知 Flash用户 23。  Specifically, the Flash user 23 implements a telephone service through a terminal interface, and a Flash client 231 is installed on the terminal, and the user 23 initiates and answers the call by using a button operation on the Flash client 231. The Flash user 23 is further configured to send a registration request to the service side system 22 before the telephone service is implemented by the service side system; the service side system 22 is further configured to receive the registration request sent by the Flash user 23; and generate the SIP registration by using the registration request. Request, and send to the SIP Server; notify the Flash user 23 of the registration result information returned by the SIP Server.
该服务侧系统 22为了实现 Flash用户 23与外部用户代理 24之间的软电 话业务, 其包括两个模块, 分别为 RTMP适配逻辑模块 221和 SIP用户代理 类模块 222。  In order to implement the soft phone service between the Flash user 23 and the external user agent 24, the service side system 22 includes two modules, an RTMP adaptation logic module 221 and a SIP user agent class module 222.
图 3为结合图 2所示实施例的系统对本发明实施例的基于 Flash的电话业 务的注册过程进行的详细说明, 该过程包括以下步骤:  FIG. 3 is a detailed description of a registration process of a Flash-based telephone service according to an embodiment of the present invention in conjunction with the system shown in FIG. 2, the process comprising the following steps:
S301 : 服务侧系统 22中的 RTMP适配逻辑模块 221的回调函数收到来 自 Flash用户 23的 invoke注册请求, 其中该 invoke注册请求中携带 Flash用 户 23的标识信息。 S301: The callback function of the RTMP adaptation logic module 221 in the service side system 22 receives an invoke registration request from the Flash user 23, where the invoke registration request carries the Flash. Identification information of the household 23.
S302: 服务侧系统 22发送 SIP的注册 ( register )请求到 Sip Server。 其 中该 SIP的注册 register请求中携带 Flash用户 23的标识信息和服务侧系统的 地址信息。  S302: The service side system 22 sends a SIP registration request to the Sip Server. The registration request of the SIP carries the identification information of the Flash user 23 and the address information of the service side system.
以便 Sip Server可以将对应标识信息的相关 SIP命令路由到该服务侧系 统。  Therefore, the Sip Server can route related SIP commands corresponding to the identification information to the service side system.
S303: 服务侧系统 22接收来自 Sip Server的注册响应。  S303: The service side system 22 receives the registration response from the Sip Server.
S304: 服务侧系统 22根据返回的注册响应, 判断是否注册成功, 当注册 成功时, 进行步骤 S305 , 当注册不成功时, 进行步骤 S306。  S304: The service side system 22 determines whether the registration is successful according to the returned registration response. When the registration is successful, the process proceeds to step S305. If the registration is unsuccessful, the process proceeds to step S306.
S305: 服务侧系统 22通过接口函数调用 RTMP适配逻辑模块 221发送 S305: The service side system 22 sends an RTMP adaptation logic module 221 through an interface function.
Invoke-RegOK到 Flash用户 23 , 注册流程结束。 Invoke-RegOK to Flash user 23, the registration process ends.
S306: 服务侧系统 22通过接口函数调用 RTMP适配逻辑模块 221发送 Invoke-RegFail到 Flash用户 23 ,之后进行步骤 S101 ,允许 Flash用户 23再次 发起注册请求。  S306: The service side system 22 calls the RTMP adaptation logic module 221 to send the Invoke-RegFail to the Flash user 23 through the interface function, and then proceeds to step S101 to allow the Flash user 23 to initiate the registration request again.
Flash用户注册完成后, 即可进行起呼和接听的操作了。  After the Flash user registration is completed, the call and answer operations can be performed.
图 4为本发明的一实施例提供的基于 Flash的电话业务实现过程中起呼 的过程, 该过程包括以下步骤:  FIG. 4 is a process of starting a call in a Flash-based telephone service implementation process according to an embodiment of the present invention, where the process includes the following steps:
S401 : Flash用户在 Flash客户端输入号码, 点击呼叫按钮, 向服务侧系 统发出 Invoke-Call呼叫请求。  S401: The Flash user inputs a number in the Flash client, clicks the call button, and sends an Invoke-Call call request to the service side system.
S402: 服务侧系统调用 RTMP适配逻辑模块的回调函数获得来自 Flash 用户的 Invoke-Call呼叫请求,并发送 Si 命令 -邀请( invite )请求到 Sip Server„ 其中该 Sip命令中携带该服务侧系统的会话描述(SDP )信息, 该 SDP 信息中包含本服务侧系统的地址、 支持的媒体格式(例如 G.711 a律音频和 G.711 u律音频 )、 媒体端口。 其中媒体端口的分配可以是每次分配不同的端 口, 也可以多个呼叫 ( Call )共用一个本地用户数据包协议( User Datagram Protocol, UDP )端口。  S402: The service side system invokes a callback function of the RTMP adaptation logic module to obtain an Invoke-Call call request from the Flash user, and sends a Si command-invitation request to the Sip Server, where the Sip command carries the service-side system. Session description (SDP) information, the SDP information includes the address of the service side system, supported media formats (eg, G.711 a-law audio and G.711 u-law audio), and media ports. The media port assignment may be Each time a different port is assigned, multiple calls (Users) can share a Local User Datagram Protocol (UDP) port.
S403: 服务侧系统启动定时器, 判断是否在定时器的定时长度内接收到 来自 Sip Server的 Sip命令 -180Ringing, 当判断结果为是时, 进行步骤 S404, 当判断结果为否时, 进行步骤 S407。 S403: The service-side system starts a timer, and determines whether the Sip command -180Ringing from the Sip Server is received within the timing length of the timer. When the determination result is yes, step S404 is performed. When the result of the determination is no, the process proceeds to step S407.
S404: 服务侧系统通过接口调用 RTMP适配逻辑模块向 Flash用户发送 应答信息 Invoke-alerting, 此时 Flash用户的客户端通过声音和文字提示 Flash 用户对端正在振铃。  S404: The service side system sends an answer message Invoke-alerting to the Flash user through the interface, and the Flash user's client prompts the Flash user peer to ring through the voice and text.
S405: 启动另一定时器, 并判断在该定时器的定时时间长度内是否接收 到 Sip Server的 Sip命令 -200OK, 当判断结果为是时, 进行步骤 S406, 当判 断结果为否时, 进行步骤 S407。  S405: Start another timer, and determine whether the Sip command S200 command of the Sip Server is received within the time limit of the timer. When the determination result is yes, proceed to step S406. When the determination result is no, perform the step. S407.
S406: 发送 SIP命令 -ACK到 Sip Server, 发送 Invoke-establish到 Flash 用户并启动媒体交换。  S406: Send the SIP command -ACK to the Sip Server, send the Invoke-establish to the Flash user and start the media exchange.
具体的, 服务侧系统通过接口函数调用 RTMP 适配逻辑模块发送 Specifically, the service side system sends the RTMP adaptation logic module by using an interface function.
Invoke-establish到 Flash用户, 指定播放流和发布流的名称, 此时 Flash用户 的客户端通过声音和文本提示 Flash用户通话已经建立。 Flash用户打开麦克 风 (stream.publish), 即将本地麦克风音频以 RTMP格式打包发送给服务侧系 统, 同时播放 (stream.play)由服务侧系统命名的上述播放流即下行媒体流。 此 时服务侧系统启动并行的 Flash用户与通话对端的媒体交换。 Invoke-establish to the Flash user, specifying the name of the playback stream and the distribution stream. At this point, the Flash user's client prompts the Flash user that the call has been established through voice and text. The Flash user opens the stream (publish), that is, the local microphone audio is packaged and sent to the service side system in RTMP format, and the stream stream is played (stream.play), which is the above-mentioned playback stream named by the service side system. At this time, the service side system initiates media exchange between the parallel Flash user and the call peer.
其中, 该具体的媒体处理过程包括: 服务侧系统通过 RTMP适配逻辑模 块的回调函数得到来自 Flash用户的上行媒体流, 将其解码为脉冲编码调制 ( Pulse Coding Modulation, PCM )数据。 将解码出来的 PCM按协商格式编 码, 例如 G.711 a律音频格式编码, 打包成实时传送协议 ( Real-time Transport Protocol, RTP )数据包, 发送给通话对端的外部用户代理。  The specific media processing process includes: the service side system obtains an uplink media stream from the Flash user through a callback function of the RTMP adaptation logic module, and decodes the same into Pulse Coding Modulation (PCM) data. The decoded PCM is encoded in a negotiated format, such as G.711 a-law audio format encoding, and packaged into a Real-time Transport Protocol (RTP) packet, which is sent to an external user agent at the opposite end of the call.
服务侧系统还接收来自通话对端的外部用户代理的 RTP数据包,按协商 格式解码成 PCM数据。 将解码出来的 PCM数据编码成 Flash用户可识别的 媒体格式, 例如 Speex、 Nelly Moser音频格式, 通过接口函数调用 RTMP适 配逻辑模块生成 RTMP数据包发送给 Flash用户, 此为下行媒体流。  The service side system also receives RTP packets from the external user agent of the opposite end of the call and decodes them into PCM data in a negotiated format. The decoded PCM data is encoded into a media format recognizable by the Flash user, for example, the Speex, Nelly Moser audio format, and the RTMP adaptation logic module is called by the interface function to generate an RTMP data packet to be sent to the Flash user, which is a downlink media stream.
S407:如果 Flash用户拒绝接听,则需要向 Sip Server发送表示错误的 4xx 命令;如果呼叫建立成功, Flash用户按挂断按钮则需要向 Sip Server发送 bye 命令, 与之相反如果收到来自 Sip Server的 bye命令, 则需要向 Sip Server回 应 200OK。 最后, 发送 Invoke-CallClose给 Flash用户, 通知 Flash用户重置 为空闲状态。 S407: If the Flash user refuses to answer, the 4xx command indicating the error needs to be sent to the Sip Server; if the call is successfully established, the Flash user needs to send the bye command to the Sip Server by pressing the hang up button, instead, if it receives the Sip Server from the Sip Server The bye command, you need to respond to the Sip Server 200OK. Finally, send Invoke-CallClose to the Flash user, notify the Flash user to reset Is idle.
图 5为本发明的一实施例提供的基于 Flash的电话业务实现过程中接听 的过程, 该过程包括以下步骤:  FIG. 5 is a process of receiving a call in a Flash-based telephone service implementation according to an embodiment of the present invention, where the process includes the following steps:
S501 : 服务侧系统接收来自 Sip Server的 Sip命令 -邀请 ( Invite )请求。 S502: 服务侧系统向 Sip Server回应 Sip命令 -180 Ringing。  S501: The service side system receives a Sip command from the Sip Server - an invite (Invite) request. S502: The service side system responds to the Sip Server with the Sip command -180 Ringing.
S503: 服务侧系统通过接口函数调用 RTMP适配逻辑模块, 发送呼叫到 达 ( Invoke-incomming )请求到 Flash用户。  S503: The service side system invokes the RTMP adaptation logic module by using an interface function, and sends an Invoke-incomming request to the Flash user.
S504: 启动定时器, 判断是否在该定时器的定时时间长度内接收到 Flash 用户发送的接听请求, 如果接收到接听请求, 则进行步骤 S505 , 如果诶有接 收到接听请求, 进行步骤 S508。  S504: Start a timer, determine whether the answer request sent by the Flash user is received within the time limit of the timer, and if the answer request is received, proceed to step S505. If the answer request is received, proceed to step S508.
如果 Flash用户点接听按钮, 该 Flash用户的客户端发出接听请求。  If the Flash user clicks the button, the Flash user's client makes an answer request.
如果 Flash用户此时用户点击挂断按钮, 该 Flash用户的客户端发出挂断 ( invoke-hangu )请求。  If the Flash user clicks the hang up button at this time, the Flash user's client issues an hang-up request.
S505: 服务侧系统发送 Sip命令 -200OK到 Sip Server, 其中在该 200OK 中携带有服务侧系统的 SDP信息。  S505: The service side system sends a Sip command -200OK to the Sip Server, where the 200 OK carries the SDP information of the service side system.
其中该 SDP信息包括服务侧系统的地址、支持的媒体格式(例如 G.711 a 律音频和 G.711 u律音频) 、 媒体端口。 其中媒体端口的分配可以是每次分 配不同的端口, 也可以多个 Call共用一个本地 UDP端口。  The SDP information includes an address of the service side system, a supported media format (for example, G.711 a law audio and G.711 u law audio), and a media port. The media port can be assigned with different ports each time, or multiple calls can share a local UDP port.
S506: 服务侧系统启动定时器, 判断是否在定时器的定时长度内接收到 来自 Sip Server的 Sip命令 -ACK, 当判断结果为是时, 进行步骤 S507 , 当判 断结果为否时, 进行步骤 S508。  S506: The service-side system starts a timer, and determines whether the Sip command-ACK from the Sip Server is received within the timing length of the timer. When the determination result is yes, the process proceeds to step S507. If the determination result is no, the process proceeds to step S508. .
S507: 服务侧系统发送呼叫建立 ( Invoke-establish )请求到 Flash用户, 并启动媒体交换。 具体的, 服务侧系统通过接口函数调用 RTMP 适配逻辑模块发送 Invoke-establish到 Flash用户, 指定播放流和发布流的名称, 此时 Flash用户 的客户端通过声音和文本提示 Flash用户通话已经建立。 Flash用户打开麦克 风 (stream.publish), 即将本地麦克风音频以 RTMP格式打包发送给服务侧系 统, 同时播放 (stream.play)由服务侧系统命名的上述播放流即下行媒体流。 此 时服务侧系统启动并行的 Flash用户与通话对端的媒体交换。 S507: The service side system sends an Invoke-establish request to the Flash user, and starts media exchange. Specifically, the service side system calls the RTMP adaptation logic module to send the Invoke-establish to the Flash user through the interface function, and specifies the names of the play stream and the release stream. At this time, the Flash user's client prompts the Flash user to establish a call through voice and text. The Flash user opens the microphone (stream.publish), that is, the local microphone audio is packaged and sent to the service side system in the RTMP format, and the above-mentioned playback stream named by the service side system, that is, the downlink media stream, is played (stream.play). This The service side system initiates media exchange between the parallel Flash user and the call peer.
其中, 该具体的媒体处理过程包括: 服务侧系统通过 RTMP适配逻辑模 块的回调函数得到来自 Flash用户的上行媒体流, 将其解码为原代码调制 ( Primary Code Modulation, PCM )数据。 将解码出来的 PCM按协商格式编 码, 例如 G.711 a律音频格式编码, 打包成实时传送协议 ( Real-time Transport Protocol, RTP )数据包, 发送给通话对端的外部用户代理。  The specific media processing process includes: the service side system obtains an uplink media stream from the Flash user by using a callback function of the RTMP adaptation logic module, and decodes the same into a Primary Code Modulation (PCM) data. The decoded PCM is encoded in a negotiated format, such as G.711 a-law audio format encoding, and packaged into a Real-time Transport Protocol (RTP) packet, which is sent to an external user agent at the opposite end of the call.
服务侧系统还接收来自通话对端的外部用户代理的 RTP数据包,按协商 格式解码成 PCM数据。 将解码出来的 PCM数据编码成 Flash用户可识别的 媒体格式, 例如 Speex、 Nelly Moser音频格式, 通过接口函数调用 RTMP适 配逻辑模块生成 RTMP数据包发送给 Flash用户, 此为下行媒体流。  The service side system also receives RTP packets from the external user agent of the opposite end of the call and decodes them into PCM data in a negotiated format. The decoded PCM data is encoded into a media format recognizable by the Flash user, for example, the Speex, Nelly Moser audio format, and the RTMP adaptation logic module is called by the interface function to generate an RTMP data packet to be sent to the Flash user, which is a downlink media stream.
S508:如果 Flash用户拒绝接听,则需要向 Sip Server发送表示错误的 4xx 命令;如果呼叫建立成功, Flash用户按挂断按钮则需要向 Sip Server发送 bye 命令, 与之相反如果收到来自 Sip Server的 bye命令, 则需要向 Sip Server回 应 200OK。 最后, 发送 Invoke-CallClose给 Flash用户, 通知 Flash用户重置 为空闲状态。  S508: If the Flash user refuses to answer, the 4xx command indicating the error needs to be sent to the Sip Server; if the call is successfully established, the Flash user needs to send the bye command to the Sip Server by pressing the hang up button, instead, if it receives the Sip Server from the Sip Server The bye command, you need to respond to the Sip Server 200OK. Finally, send Invoke-CallClose to the Flash user, informing the Flash user to reset to idle state.
由于在本发明实施例中无论是系统内部的通话, 还是系统外部的, 逻辑 上都是一样的, 因此最大程度减少了业务变化时需要修改 Flash软电话的可 能, 应用扩展性强, 由于各 Flash用户间无逻辑联系, 通过简单的增加服务器 的数量即可增加系统容量。  In the embodiment of the present invention, whether the call inside the system or the outside of the system is logically the same, the possibility of modifying the Flash soft phone when the service change is minimized is minimized, and the application is highly scalable, because each Flash There is no logical connection between users, and system capacity can be increased by simply increasing the number of servers.
本领域普通技术人员可以理解上述方法中的全部或部分步骤可通过程序 来指令相关硬件完成, 所述程序可以存储于计算机可读存储介质中, 如只读 存储器、 磁盘或光盘等。 可选地, 上述实施例的全部或部分步骤也可以使用 一个或多个集成电路来实现, 相应地, 上述实施例中的各模块 /单元可以釆用 硬件的形式实现, 也可以釆用软件功能模块的形式实现。 本发明不限制于任 何特定形式的硬件和软件的结合。 One of ordinary skill in the art will appreciate that all or a portion of the above steps may be accomplished by a program instructing the associated hardware, such as a read-only memory, a magnetic disk, or an optical disk. Optionally, all or part of the steps of the foregoing embodiments may also be implemented by using one or more integrated circuits. Accordingly, each module/unit in the foregoing embodiment may be implemented in the form of hardware, or may use software functions. The form of the module is implemented. The invention is not limited to any specific form of combination of hardware and software.
需要说明的是, 本发明还可有其他多种实施例, 在不背离本发明精神及 的改变和变形, 但这些相应的改变和变形都应属于本发明所附的权利要求的 保护范围。 It should be noted that the present invention may have various other embodiments without departing from the spirit of the present invention. Changes and modifications are intended to be included within the scope of the appended claims.
工业实用性 本发明实施例的方案最大程度减少了业务变化时需要修改 Flash软电话 的可能, 应用扩展性强, 兵器通过简单的增加服务器的数量即可增加系统容 量。 INDUSTRIAL APPLICABILITY The solution of the embodiment of the present invention minimizes the possibility of modifying the Flash softphone when the service changes, and the application has strong scalability, and the weapon can increase the system capacity by simply increasing the number of servers.

Claims

权 利 要 求 书 Claim
1、 一种基于 Flash的电话业务实现方法, 所述方法包括: A method for implementing a telephone service based on Flash, the method comprising:
服务侧系统接收交换网络发送的邀请请求;  The service side system receives an invitation request sent by the switching network;
所述服务侧系统根据所述邀请请求向 Flash用户发送呼叫到达请求,并接 收所述 Flash用户发送的接听请求;  The service side system sends a call arrival request to the Flash user according to the invitation request, and receives an answer request sent by the Flash user;
所述服务侧系统根据所述接听请求, 通过所述交换网络向会话发起协议 ( SIP )服务器发送应答信息, 建立呼叫;  The service side system sends a response message to a Session Initiation Protocol (SIP) server through the switching network according to the answering request, and establishes a call;
所述月良务侧系统通知所述 Flash用户呼叫建立成功, 并完成所述 Flash用 户与交换网络用户之间的媒体数据传输。  The monthly service side system notifies the Flash user that the call is successfully established, and completes media data transmission between the Flash user and the switching network user.
2、 如权利要求 1所述的方法, 其中, 所述服务侧系统接收交换网络发送 的邀请请求的步骤之前, 所述方法还包括:  2. The method according to claim 1, wherein before the step of the service side system receiving the invitation request sent by the switching network, the method further includes:
所述服务侧系统接收所述 Flash用户发送的注册请求;  Receiving, by the service side system, a registration request sent by the Flash user;
所述服务侧系统将所述注册请求生成 SIP注册请求, 并发送给所述 SIP 服务器;  The service side system generates a SIP registration request by the registration request, and sends the SIP registration request to the SIP server;
将所述 SIP服务器返回的注册结果信息通知所述 Flash用户。  Notifying the Flash user of the registration result information returned by the SIP server.
3、 如权利要求 1所述的方法, 其中, 所述完成 Flash用户与交换网络用 户之间的媒体数据传输的步骤包括:  3. The method according to claim 1, wherein the step of completing media data transmission between the Flash user and the switching network user comprises:
接收来自所述 Flash用户的上行媒体流,将所述上行媒体流解码为脉冲编 码调制 (PCM )数据, 并将所述 PCM数据按照与所述交换网络用户之间协 商的编码格式编码后, 发送给所述交换网络用户;  Receiving an uplink media stream from the Flash user, decoding the uplink media stream into pulse code modulation (PCM) data, and encoding the PCM data according to an encoding format negotiated with the user of the switching network, and transmitting To the switching network user;
接收来自所述交换网用户发送的实时传送协议(RTP )数据包, 按照与 所述 Flash用户协商的格式解码为 PCM数据, 并将所述 PCM数据编码为所 述 Flash用户可识别的媒体格式, 生成实时消息协议(RTMP )数据包发送给 所述 Flash用户。  Receiving a Real Time Transport Protocol (RTP) data packet sent from the user of the switching network, decoding into PCM data according to a format negotiated with the Flash user, and encoding the PCM data into a media format recognizable by the Flash user, A Real Time Messaging Protocol (RTMP) data packet is generated and sent to the Flash user.
4、 一种基于 Flash的电话业务实现方法, 所述方法包括:  4. A method for implementing a telephone service based on Flash, the method comprising:
服务侧系统接收 Flash用户发送的呼叫请求; 服务侧系统根据所述呼叫请求, 向会话发起协议(SIP )服务器发送邀请 请求; The service side system receives a call request sent by the Flash user; The service side system sends an invitation request to a Session Initiation Protocol (SIP) server according to the call request;
当接收到所述 SIP服务器返回的成功应答时, 向所述 Flash用户发送被 叫应答信息, 建立呼叫;  Sending a call response message to the Flash user to establish a call when receiving a successful response returned by the SIP server;
所述服务侧系统通知交换网络用户呼叫建立成功,并完成所述 Flash用户 与所述交换网络用户之间的媒体数据传输。  The service side system notifies the switching network user that the call setup is successful, and completes the media data transmission between the Flash user and the exchange network user.
5、 如权利要求 4所述的方法, 其中, 所述服务侧系统接收 Flash用户发 送的呼叫请求的步骤之前, 所述方法还包括:  The method of claim 4, wherein before the step of the service side system receiving the call request sent by the Flash user, the method further includes:
所述服务侧系统接收所述 Flash用户发送的注册请求;  Receiving, by the service side system, a registration request sent by the Flash user;
所述服务侧系统将所述注册请求生成 SIP注册请求, 并发送给所述 SIP 服务器;  The service side system generates a SIP registration request by the registration request, and sends the SIP registration request to the SIP server;
将所述 SIP服务器返回的注册结果信息通知所述 Flash用户。  Notifying the Flash user of the registration result information returned by the SIP server.
6、 如权利要求 4所述的方法, 其中, 所述完成 Flash用户与交换网络用 户之间的媒体数据传输的步骤包括:  The method of claim 4, wherein the step of completing media data transmission between the Flash user and the switching network user comprises:
接收来自所述 Flash用户的上行媒体流,将所述上行媒体流解码为原代码 调制 (PCM )数据, 并将所述 PCM数据按照与所述交换网络用户之间协商 的编码格式编码后, 发送给所述交换网络用户;  Receiving an uplink media stream from the Flash user, decoding the uplink media stream into original code modulation (PCM) data, and encoding the PCM data according to an encoding format negotiated with the user of the switching network, and transmitting To the switching network user;
接收来自所述交换网用户发送的实时传送协议(RTP )数据包, 按照与 所述 Flash用户协商的格式解码为 PCM数据, 并将所述 PCM数据编码为所 述 Flash用户可识别的媒体格式, 生成实时消息协议(RTMP )数据包发送给 所述 Flash用户。  Receiving a Real Time Transport Protocol (RTP) data packet sent from the user of the switching network, decoding into PCM data according to a format negotiated with the Flash user, and encoding the PCM data into a media format recognizable by the Flash user, A Real Time Messaging Protocol (RTMP) data packet is generated and sent to the Flash user.
7、 如权利要求 4-6任一项所述的方法, 其中, 所述成功应答为 200 OK。 7. The method of any of claims 4-6, wherein the success response is 200 OK.
8、 一种服务侧系统, 所述系统设置为: 8. A service side system, the system being set to:
根据接收到的来自交换网络用户的邀请请求,向 Flash用户发送呼叫到达 请求, 并接收所述 Flash用户发送的接听请求, 根据接收到的该接听请求, 通 过交换网络向会话发起协议(SIP )服务器发送应答信息, 建立呼叫, 通知所 述 Flash用户呼叫建立成功, 并完成所述 Flash用户与交换网络用户之间的媒 体数据传输。 Receiving a call arrival request to the Flash user according to the received invitation request from the switching network user, and receiving an answer request sent by the Flash user, according to the received answer request, to the Session Initiation Protocol (SIP) server through the switching network Sending a response message, establishing a call, notifying the Flash user that the call is successfully established, and completing media data transmission between the Flash user and the switching network user.
9、 如权利要求 8所述的系统, 其还设置为: 9. The system of claim 8 further configured to:
接收所述 Flash用户发送的注册请求;将所述注册请求生成 SIP注册请求, 并发送给所述 SIP服务器;将 SIP服务器返回的注册结果信息通知所述 Flash 用户。  Receiving a registration request sent by the Flash user; generating a SIP registration request by the registration request, and sending the SIP registration request to the SIP server; and notifying the Flash user of the registration result information returned by the SIP server.
10、 如权利要求 8 所述的系统, 其是设置为以如下方式完成所述 Flash 用户与交换网络用户之间的媒体数据传输:接收来自所述 Flash用户的上行媒 体流, 将其解码为脉冲编码调制 (PCM )数据, 并将所述 PCM数据按照与 所述交换网络用户之间协商的编码格式编码后, 发送给所述交换网络用户; 接收来自所述交换网用户发送的实时传送协议(RTP )数据包, 按照与所述 Flash用户协商的格式解码为 PCM数据,并将所述 PCM数据编码为所述 Flash 用户可识别的媒体格式,生成实时消息协议( RTMP )数据包发送给所述 Flash 用户。  10. The system of claim 8 configured to perform media data transmission between the Flash user and a switched network user in the following manner: receiving an upstream media stream from the Flash user, decoding it into a pulse Coding and modulating (PCM) data, and encoding the PCM data according to an encoding format negotiated with the user of the switching network, and transmitting to the switching network user; receiving a real-time transmission protocol sent by the user of the switching network ( An RTP packet is decoded into PCM data according to a format negotiated with the Flash user, and the PCM data is encoded into a media format recognizable by the Flash user, and a Real Time Message Protocol (RTMP) data packet is generated and sent to the Flash user.
11、 如权利要求 10所述的系统, 其包括 RTMP适配模块和用户代理类 模块, 其中, 所述 RTMP适配模块是设置为与所述 Flash用户进行信令和媒 体交互, 所述用户代理类模块设置为与所述 SIP服务器进行信令交互, 与所 述交换网络用户进行媒体交互。  11. The system of claim 10, comprising an RTMP adaptation module and a user agent class module, wherein the RTMP adaptation module is configured to perform signaling and media interaction with the Flash user, the user agent The class module is configured to perform signaling interaction with the SIP server to perform media interaction with the switching network user.
12、 一种服务侧系统, 其设置为:  12. A service side system, which is configured to:
接收 Flash用户发送的呼叫请求;根据该呼叫请求,向会话启动协议( SIP ) 服务器发送邀请请求; 当接收到所述 SIP服务器返回的成功应答时, 向所述 Flash用户发送被叫应答信息, 建立呼叫; 通知交换网络用户呼叫建立成功, 并完成所述 Flash用户与所述交换网络用户之间的媒体数据传输。  Receiving a call request sent by the Flash user; sending an invitation request to the Session Initiation Protocol (SIP) server according to the call request; sending a called response message to the Flash user when receiving the successful response returned by the SIP server, establishing Calling; notifying the switching network user that the call is successfully established, and completing media data transmission between the Flash user and the switching network user.
13、 如权利要求 12所述的系统, 其还设置为:  13. The system of claim 12 further configured to:
接收 Flash用户发送的注册请求; 将所述注册请求生成 SIP注册请求, 并 发送给 SIP服务器; 将所述 SIP服务器返回的注册结果信息通知 Flash用户。  Receiving a registration request sent by the Flash user; generating a SIP registration request by the registration request, and sending the request to the SIP server; and notifying the Flash user of the registration result information returned by the SIP server.
14、 如权利要求 12所述的系统, 其是设置为以如下方式完成所述 Flash 用户与所述交换网络用户之间的媒体数据传输:  14. The system of claim 12 arranged to perform media data transfer between the Flash user and the switching network user in the following manner:
接收来自所述 Flash用户的上行媒体流,将其解码为脉冲编码调制( PCM ) 数据,并将解码后的 PCM数据按照与所述交换网络用户之间协商的编码格式 编码后, 发送给所述交换网络用户; 接收来自所述交换网用户发送的实时消 息协议(RTP )数据包, 按照与所述 Flash用户协商的格式解码为 PCM数据, 并将所述 PCM数据编码为所述 Flash用户可识别的媒体格式, 生成路由选择 表维护协议( RTMP )数据包发送给所述 Flash用户。 Receiving an upstream media stream from the Flash user, decoding it into Pulse Code Modulation (PCM) data, and decoding the decoded PCM data according to an encoding format negotiated with the user of the switching network After encoding, sending to the switching network user; receiving a Real Time Message Protocol (RTP) data packet sent from the switching network user, decoding into PCM data according to a format negotiated with the Flash user, and encoding the PCM data A routing table maintenance protocol (RTMP) data packet is generated for the Flash user identifiable media format and sent to the Flash user.
15、 如权利要求 14所述的系统, 其包括 RTMP适配模块和用户代理类 模块, 其中, 所述 RTMP适配模块是设置为与所述 Flash用户进行信令和媒 体交互, 所述用户代理类模块设置为与所述 SIP服务器进行信令交互, 与所 述交换网络用户进行媒体交互。  15. The system of claim 14, comprising an RTMP adaptation module and a user agent class module, wherein the RTMP adaptation module is configured to perform signaling and media interaction with the Flash user, the user agent The class module is configured to perform signaling interaction with the SIP server to perform media interaction with the switching network user.
16、如权利要求 12-14任一项所述的系统,其中,所述成功应答为 200 OK。  The system of any of claims 12-14, wherein the success response is 200 OK.
PCT/CN2012/078952 2011-10-31 2012-07-20 Flash-based telephone service implementation method and system WO2013063959A1 (en)

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