CN102413110A - Multiple registered communication method and system utilizing session initiation protocol - Google Patents

Multiple registered communication method and system utilizing session initiation protocol Download PDF

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Publication number
CN102413110A
CN102413110A CN 201010294144 CN201010294144A CN102413110A CN 102413110 A CN102413110 A CN 102413110A CN 201010294144 CN201010294144 CN 201010294144 CN 201010294144 A CN201010294144 A CN 201010294144A CN 102413110 A CN102413110 A CN 102413110A
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server
sip
session initiation
communication
client
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CN 201010294144
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Chinese (zh)
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廖经富
林育正
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中华电信股份有限公司
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Abstract

The invention discloses a multiple registered communication method and system utilizing a session initiation protocol. The system comprises a client, a relay server and a plurality of SIP (session initial protocol) servers, wherein the relay server is connected with the plurality of SIP servers and is connected with the client; an in addition, the relay server establishes online with the client through a configuration mode, and registers to the plurality of SIP servers through the configuration mode so as to select at least one of the SIP servers further, so that the client and the SIP servers are communicated directly. So that the problems of poor compatibility of the SIP servers and the client and the poor compatibility among the SIP servers can be solved, and a scheme which can be used for saving communication expenses can be provided aiming at different dialing numbers of the client.

Description

使用会话初始协议的多重注册的通讯方法与系统 Using the Session Initiation Protocol multiple registrations of communication methods and systems

技术领域 FIELD

[0001] 本发明涉及一种使用会话初始协议的通讯方法与系统,特别指涉及一种使用会话初始协议的多重注册的通讯方法与系统。 [0001] The present invention relates to a communication method and system using a Session Initiation Protocol, refers particularly relates to multiple registration using Session Initiation Protocol communication method and system.

背景技术 Background technique

[0002] 早期语音通讯建构在电信服务公司所布建的公共交换电话网络(Public Switched Telephone Network,PSTN)上。 [0002] In the early construction of telecommunications voice communications services company deployment of a public switched telephone network (Public Switched Telephone Network, PSTN) on. PSTN是一种用于全球语音通讯的电话交换网络, 是目前世界上最大的网络,拥有数亿的用户数量。 PSTN is a global voice communication for telephone switching network, is the world's largest network, with hundreds of millions of subscribers. 而随着因特网的进步,语音通讯也可在因特网上实现,目前最普及的技术便是网络电话(Voice over Internet Protocol,VoIP)。 With the advancement of Internet, voice communications can be implemented on the Internet, the most popular technology is VoIP (Voice over Internet Protocol, VoIP). 简单的说,VoIP将送话端的语音模拟信号转成数字信号,再通过因特网传输到收话端,收话端再将数字信号转成语音模拟信号,以实现在因特网上的语音通讯,其中,最常用的通讯协议为会话初始协议(Session Initiation Protocol, SIP) 0此外,另有一种IP用户交换机(IP PBX),利用数字信号在因特网上直接进行通讯。 Briefly, VoIP the sending end of the analog voice signals into digital signals, and then transmitted via the Internet to the receiving terminals at close if end for the digital signals into analog voice signals for voice communication on the Internet, wherein, the most commonly used protocol is the session Initiation protocol (session Initiation protocol, SIP) 0 in addition, another user of an IP switch (IP the PBX), digital signal directly communicate over the Internet.

[0003] 另一方面,由于通讯技术的发达,除了上述的公共交换电话网络、网络电话之外, GSM(Global System for Mobile Communication)移动电话网络、第三代(3G)移动电话网络等无线通信技术也发展的相当成熟。 [0003] On the other hand, due to the development of communication technology, in addition to the public switched telephone network, a telephone network, GSM (Global System for Mobile Communication) mobile telephone network, a third generation (3G) wireless communication such as mobile phone network technology has developed quite mature. 而现有使用SIP的通讯方法是由SIP用户将通讯要求传送至电信服务公司的SIP服务器,该SIP服务器根据通讯要求中的被叫号码将通讯要求转传到不同的电话网络,如公共交换电话网络、网络电话等,以完成通讯连接。 SIP using the existing communication method by the SIP communication request sent to the user's SIP server telecommunications services, the rotation transmitted to the SIP server different telephone network in accordance with communication requirements of the called number to the communications requirement, such as a public switched telephone network, Internet telephony, etc., in order to complete the communication connection.

[0004] 然而,在具有多个SIP服务器的环境中,由于多个SIP服务器可能分别属于不同的电信服务公司,导致SIP服务器之间的兼容性不佳,故SIP服务器之间因无法设立SIP主干(trunk),而无法正常通讯。 [0004] However, in an environment with multiple SIP servers, due to multiple SIP servers may belong to different telecom services company, leading to poor compatibility between the SIP server, SIP server because it can not be established between the SIP trunk (trunk), and can not communicate. 此外,由于客户端与电信服务公司所提供的SIP服务器,兼容性并不高,导致有些客户端并无法向不兼容的SIP服务器注册,或不兼容的SIP服务器无法与客户端设定SIP主干,亦造成通讯异常。 In addition, due to the SIP server client and telecommunication services provided by the company, compatibility is not high, resulting in some clients and not registered with the SIP server is not compatible or incompatible SIP server SIP trunk can not be set with the client, also cause communication error. 再者,在网络地址转换(Network Address Translation,NAT)环境下的客户端也会遭遇一些问题,当客户端向上述SIP服务器请求注册时,由于NAT服务器会将在企业内的虚拟网络地址转换成企业外的实体网络地址,导致SIP服务器无法将注册结果响应至原来的客户端,造成无法注册,因此造成通讯异常。 Furthermore, under the Network Address Translation (Network Address Translation, NAT) environment, the client will face some problems, when a client requests to the SIP registration server, NAT server will be due within the enterprise virtual network address into physical network address outside the enterprise, leading SIP server can not register the results of the response to the original client, the result can not be registered, thus creating a communication error. 最后, 由于现有的使用SIP的通讯方法是根据被叫号码以固定的方式将通讯要求转传至不同的电话网络,并没有针对客户端的不同拨叫号码提供节省通讯费用的方案。 Finally, because of the use of existing SIP communication method is based on the called number in a fixed manner to the onward transmission to different communication requirements of the telephone network, it did not provide cost-saving communication solutions for different dialing numbers clients.

[0005] 综上所述,在现有通讯系统中,由于兼容性不佳或NAT环境的限制,导致客户端无法向SIP服务器注册,而SIP服务器之间亦存在兼容性不佳的问题,且没有针对客户端的不同拨叫号码提供节省通讯费用的方案。 [0005] In summary, the existing communications system, or NAT environment due to poor compatibility constraints, the client can not register with the SIP server, and there is also the problem of poor compatibility between the SIP server, and It did not provide cost-saving communication solutions for different dialing numbers clients. 因此,极需要一种使用SIP的多重注册的通讯方法与系统,以解决SIP服务器与客户端兼容性不佳以及SIP服务器之间兼容性不佳的问题,并可针对客户端的不同拨叫号码提供节省通讯费用的方案。 Therefore, the very need for a communication method and system using multiple registration of SIP to address SIP server and client to poor compatibility and poor compatibility between the SIP server problems, and provide dialing numbers for different clients communication costs savings program.

发明内容 SUMMARY

[0006] 本发明提供一种使用会话初始协议的多重注册的通讯方法与系统,以解决现有技术中SIP服务器与客户端兼容性不佳、SIP服务器之间兼容性不佳的问题,并可针对客户端的不同拨叫号码提供节省通讯费用的方案。 [0006] The present invention provides a use of the Session Initiation Protocol registration of multiple communication method and system to resolve the SIP server and the client poor compatibility, poor compatibility between the SIP server problems of the prior art, and cost-saving communication solutions for different dialing numbers clients.

[0007] 依照本发明的一实施方式,提供一种使用会话初始协议的多重注册的通讯方法, 包括下列步骤:令中继服务器建立与客户端之间的联机;令该中继服务器向多个SIP服务器注册;令该客户端使用SIP将通讯要求传送至该中继服务器;令该中继服务器选择该多个SIP服务器的其中至少一个并将该通讯要求传送至被选择的SIP服务器;以及,令该SIP 服务器检查该SIP的封包内容后,判断是否允许该通讯要求,并将判断结果经由该中继服务器传送至该客户端。 [0007] In accordance with an embodiment of the present invention, there is provided a method of using multiple registered SIP communication, comprising the steps of: establishing connection between the relay server so that the client; enabling the server to a plurality of relay SIP server registration; enabling the client to communication using SIP request sent to the relay server; enabling the plurality of relay servers select the SIP server and wherein at least one of the communication request sent to the selected SIP server; and, after enabling the SIP server checks the contents of the SIP packet, and determines whether to permit the communication requirements, and the determination result is transmitted to the client via the relay server.

[0008] 此外,本发明还提供一种使用会话初始协议的多重注册的通讯系统,包括:中继服务器,架构在因特网上并通过该因特网与客户端连接;以及多个会话初始协议服务器,架构在该因特网上并与该中继服务器连接,其中,该中继服务器通过组态方式以建立与该客户端之间的联机,且该中继服务器通过组态方式向该多个会话初始协议服务器注册,而该客户端通过组态方式以使用会话初始协议将通讯要求传送至该中继服务器,该中继服务器选择该多个会话初始协议服务器的其中至少一个并将该通讯要求传送至被选择的会话初始协议服务器,并且该会话初始协议服务器通过组态方式以检查该会话初始协议的封包内容后,判断是否允许该通讯要求,并将判断结果经由该中继服务器传送至该客户端。 [0008] Further, the present invention also provides a use of the Session Initiation Protocol registration of the multiplex communication system, comprising: a relay server architecture connected to the Internet and the client through the Internet; and a plurality of Session Initiation Protocol server architecture and connected to the Internet and the relay server, wherein the relay server to establish a connection by way of the configuration between the client and the relay server by way of the configuration server to a plurality of session Initiation protocol register, and the client configuration mode by using the session Initiation protocol communication request sent to the relay server, the relay server selects the plurality of SIP servers and wherein at least one of the communication request sent to the selected session Initiation protocol server, and the server through the SIP configuration mode to check the contents of the packet after the session Initiation protocol, whether to permit the communication requirements, and the determination result is transmitted to the client via the relay server.

[0009] 如上所述,相比于现有技术,本发明利用中继服务器一方面建立与客户端之间的联机,另一方面向多个SIP服务器注册,从而通过选择多个SIP服务器的其中至少一个而使客户端与所选择的SIP服务器直接通讯。 [0009] As described above, compared to the prior art, an aspect of the present invention utilizes a relay server establishes a connection between the client, on the other hand to a plurality of SIP registration server, so that by selecting a plurality of SIP servers wherein at least one of the SIP client and server communicate directly selected. 由此解决SIP服务器与客户端兼容性不佳、SIP服务器之间兼容性不佳的问题,并可针对客户端的不同拨叫号码提供节省通讯费用的方案。 Thereby solving the SIP server and client compatibility is poor, poor compatibility between the SIP server problems, and provide cost-saving communication solutions for different dialing numbers clients.

附图说明 BRIEF DESCRIPTION

[0010] 图1为本发明的使用会话初始协议的多重注册的通讯系统的第一实施例的系统架构图; [0010] System Architecture FIG. 1 a first embodiment of the present invention using a session initiation protocol registration multiplex communication system;

[0011] 图2为本发明的使用会话初始协议的多重注册的通讯方法的第一实施例的流程图; [0011] FIG 2 is a flowchart of the first embodiment of the communication method using the present invention, the session initiation protocol registration multiple;

[0012] 图3为本发明的使用会话初始协议的多重注册的通讯系统的第二实施例的系统架构图; [0012] System Architecture FIG. 3 of a second embodiment of the present invention using a session initiation protocol registration multiplex communication system;

[0013] 图4为本发明的使用会话初始协议的多重注册的通讯方法的第二实施例的流程图; A second flowchart of an embodiment of the communication method using the session [0013] FIG. 4 of the present invention multiple initiation protocol registration;

[0014] 图5为本发明的使用会话初始协议的多重注册的通讯系统的第三实施例的系统架构图; [0014] System Architecture FIG. 5 a third embodiment of the present invention using a session initiation protocol registration multiplex communication system;

[0015] 图6为本发明的使用会话初始协议的多重注册的通讯方法的第三实施例的流程图。 Flowchart of a third embodiment of multiple registrations communication method using the Session Initiation Protocol [0015] FIG. 6 of the present invention.

[0016]【主要组件符号说明】 [0016] The main component symbol DESCRIPTION

[0017] 100、300、500 通讯系统 [0017] 100, 300 communications system

[0018] 110 IP PBX [0018] 110 IP PBX

[0019] 120、320NAT 服务器 [0019] 120,320NAT server

[0020] 125、325 路由表 [0020] routing table 125,325

6[0021] 130、330、530 中继服务器 6 [0021] The relay server 130,330,530

[0022] 135、335、535 记录表 [0022] The recording table 135,335,535

[0023] 138、338、538 拨号表 [0023] dial table 138,338,538

[0024] 140、340、540SIP 服务器 [0024] 140,340,540SIP server

[0025] 150、350、550 具有LDAP 的服务器 [0025] with the LDAP server 150,350,550

[0026] 160、360、560 被叫号码端 [0026] the called number end 160,360,560

[0027] 200、400、600 通讯方法 [0027] 200,400,600 communication method

[0028] 310VoIP [0028] 310VoIP

[0029] 315VoIP 网关器 [0029] 315VoIP Gateway

[0030] 510客户端 [0030] Client 510

[0031] S210、S220、S225、S230、S235、S240、S250、S255 步骤 [0031] S210, S220, S225, S230, S235, S240, S250, S255 step

[0032] S260、S270、S280、S290、S410、S420、S425、S430 步骤 [0032] S260, S270, S280, S290, S410, S420, S425, S430 step

[0033] S435、S440、S450、S455、S460、S470、S480、S490 步骤 [0033] S435, S440, S450, S455, S460, S470, S480, S490 step

[0034] S610、S620、S625、S630、S635、S640、S650、S655 步骤 [0034] S610, S620, S625, S630, S635, S640, S650, S655 step

[0035] S660、S67O、S680、S690 步骤 [0035] S660, S67O, S680, S690 step

具体实施方式 detailed description

[0036] 以下通过特定的具体实施例说明本发明的实施方式,本领域技术人员可由本说明书所揭示的内容轻易地了解本发明的其它优点与功效。 [0036] The following examples illustrate embodiments of the present invention by a specific particular, the present art by the art disclosed in the present specification easily understand other advantages and effects of the present invention.

[0037] 第一实施例: [0037] First Embodiment:

[0038] 请参阅图1,为根据本发明的使用会话初始协议的多重注册的通讯系统100的第一实施例的系统架构图。 [0038] Referring to FIG. 1, a system architecture diagram of a first embodiment of the present invention using a session initiation protocol registration multiplex communication system 100 according to.

[0039] 如图1所示,本发明的使用会话初始协议的多重注册的通讯系统100架构在因特网上,包括IP用户交换机(以下称IP PBX) 110、NAT服务器120、中继服务器130、多个SIP 服务器140。 [0039] As shown in FIG. 1, the present invention uses the session initiation protocol registration multiplex communication system 100 over the Internet architecture, including IP private branch exchange (hereinafter referred to as IP PBX) 110, NAT server 120, the relay server 130, multiple SIP server 140. 其中,多个SIP服务器140可为多媒体通讯服务器(Multimedia Communication Server),但并不以此为限,该中继服务器130具有记录表135,用以记录SIP服务器140与IP PBX 110的通讯数据,其中包括通讯时间,但并不以此为限。 Wherein the plurality of SIP server 140 may correspond to a multimedia server (Multimedia Communication Server), but is not limited thereto, the relay server 130 has a recording table 135 for recording communication data with the SIP server 140 of the IP PBX 110, including communication time, but is not limited thereto. 该中继服务器130还具有拨号表(telephony table) 138,用以记录SIP服务器140与IP PBX 110的拨叫号码之间的对应关系。 The relay server 130 further includes a dial table (telephony table) 138, for recording a correspondence between IP PBX 140 and SIP server 110 of the dialed number. NAT服务器120具有路由表(routing table) 125,用以记录经NAT服务器120转换前的地址与端口和经NAT服务器120转换后的地址与端口。 NAT server 120 having a routing table (routing table) 125, for recording the address and port 120 and converted by the NAT address and port of the server before the server 120 converts the NAT. 此外,本实施例中的IP PBX 110与SIP服务器140的数目均为2个,但仅为例示说明,于不同实施例中,该IP PBX 110 与SIP服务器140的数目并不以2个为限。 Further, in the embodiment the number of IP PBX 110 and SIP server 140 of the present embodiment are two, but the description is illustrative only, in various embodiments, the number of the IP PBX 110 and SIP server 140 is not limited to two .

[0040] 在本发明的系统100中,IP PBX 110与NAT服务器120连接,NAT服务器120可将输入的虚拟网络地址与端口予以转换成实体网络地址与端口,并将输入的虚拟网络地址与端口以及转换后的实体网络地址与端口储存于路由表125。 Conversion [0040] In the system of the present invention 100, 120 are connected to the NAT server IP PBX 110, NAT server 120 may be the input port and the virtual network address to be a network address and physical port, and inputs the virtual network address and port and a physical network address stored in the port after the routing conversion table 125. 中继服务器130通过NAT服务器120与IPPBX 110连接。 NAT server 130 via the relay server 120 and IPPBX 110. 多个SIP服务器140则与中继服务器130连接。 A plurality of SIP servers 140 connected to the relay server 130.

[0041] 此外,在本发明的系统100中,还可选择性地包括具有轻型目录访问协议(Lightweight Directory Access Protocol, LDAP)的服务器(以下称具有LDAP 的服务器)150,其与中继服务器130连接,以进行账号与密码的管理。 [0041] Further, in system 100 of the invention, it may optionally include a server having a Lightweight Directory Access Protocol (Lightweight Directory Access Protocol, LDAP) (hereinafter referred to with the LDAP server) 150 which the relay server 130 connection to manage accounts and passwords. [0042] 再者,在本发明的系统100中,还包括被叫号码端160,与SIP服务器140连接,以进行通讯封包的传送,于本实施例中的被叫号码端160与SIP服务器140的连接关系仅为例示说明,于不同实施例中,被叫号码端160可与其它SIP服务器140连接。 [0042] Further, in system 100 of the invention, the called number further comprises an end 160 connected to the SIP server 140, the communication packet for transmission, in this embodiment 160 the called number end of the SIP server 140 the connection relationship described is only an example, in various embodiments, the called number end 160 may be connected to other SIP servers 140.

[0043] 请参阅图2,为根据本发明的使用会话初始协议的多重注册的通讯方法200的第一实施例的流程图,其中,IP PBX 110、中继服务器130、SIP服务器140通过组态方式进行下列步骤。 [0043] Referring to FIG 2, is a flowchart of a first embodiment of the multiple registrations of the present invention using the session initiation protocol communications method 200, wherein, IP PBX 110, relay server 130, SIP server 140 by configuring manner the following step.

[0044] 如图2所示,在步骤S210中,在因特网上提供IP PBX 110、中继服务器130以及多个SIP服务器140,其中,中继服务器130与多个SIP服务器140连接,并通过NAT服务器120与IP PBX 110连接。 [0044] 2, at step S210, provides IP PBX 110, the relay server 130 and a plurality of SIP server 140 on the Internet, wherein the relay server 130 and a plurality of SIP servers 140 are connected, and through NAT server 120 is connected to the IP PBX 110. 接着进至步骤S220。 Then proceeds to step S220.

[0045] 在步骤S220中,中继服务器130设定与IP PBX 110之间的主干,并向多个SIP服务器140注册,其中,多个SIP服务器140检查该注册的账号及/或密码,并将是否允许该注册的结果传送至中继服务器130。 [0045] In step S220, the relay server 130 is set between the backbone and the 110 IP PBX, SIP servers 140 and a plurality of register, wherein the plurality of SIP register server 140 checks the account and / or password, and whether to allow the registration of the result to the relay server 130. 若允许,则传送允许注册要求,并进至步骤S225 ;若不允许,则传送拒绝注册要求,并结束此程序。 If allowed, the transfer permission registration requirements, and proceeds to step S225; if allowed, is transmitted rejected registration requirements, and this routine ends.

[0046] 在步骤S225中,中继服务器130会监听(listen)是否有通讯要求传送至中继服务器130。 [0046] In step S225, the relay server 130 listens (the listen) whether there is a communication request sent to the relay server 130. 若有,则进至步骤S230 ;若无,则持续执行本步骤S225。 If so, the process proceeds to step S230; if not, then continuing the implementation of this step S225.

[0047] 在步骤S230中,当IP PBX 110使用SIP将通讯要求通过NAT服务器120传送至中继服务器130时,该中继服务器130利用拨号表138选择该多个SIP服务器140的其中至少一个,优选地,中继服务器130根据拨号表138中的SIP服务器140与IP PBX 110的拨叫号码之间的对应关系选择该多个SIP服务器140的其中至少一个。 [0047] In step S230, when the IP PBX 110 communication using SIP to claim 130, the relay server 130 using a dial table 138 selected by the plurality of SIP server 120 to the NAT server wherein at least one of the relay server 140, preferably, the relay server 130 selects at least one of the plurality of SIP server 140 according to the correspondence relation between the dial table 138 in the SIP server 140 and the IP PBX 110 to the dialed number. 此外,中继服务器130变更该SIP的封包内容,优选地,该变更SIP的封包内容是将封包内容中的SIP的标头(header)来源从经NAT服务器120转换前的地址与端口变更为中继服务器130的地址与端口。 Further, the relay server 130 changes the content of the SIP packet, preferably, the change of the packet content is the SIP packet content in a SIP header (header) from the source before the change of address and port NAT server 120 converts the in following the address and port of the server 130. 接着进至步骤S235。 Then proceeds to step S235.

[0048] 在步骤S235中,中继服务器130将该通讯要求传送至被选择的SIP服务器140。 [0048] In step S235, the relay server 130 transmits to the communications requirement of the selected SIP server 140. 接着进至步骤S240。 Then proceeds to step S240.

[0049] 在步骤S240中,SIP服务器140检查该SIP的封包内容,其中,检查该SIP的封包内容包括检查地址与端口、账号、该SIP的网域、被叫号码及/或最大同时通话数量等。 [0049] In step S240, the packet SIP server 140 checks the contents of the SIP, wherein the SIP packet inspection includes inspection port address, account number, the SIP domain, called number and / or maximum number of simultaneous calls Wait. 接着进至步骤S250。 Then proceeds to step S250.

[0050] 在步骤S250中,SIP服务器140根据该检查结果,判断是否允许该通讯要求,并确认被叫号码端160的通讯状况正常后,将是否允许该通讯要求的结果经由中继服务器130 传送至IP PBX 110,其中,当SIP服务器140使用SIP将通讯要求的结果经由中继服务器130传送至IP PBX 110时,中继服务器130变更该SIP的封包内容,优选地,该变更SIP的封包内容是将该封包内容中的该SIP的标头来源从SIP服务器140的地址与端口变更为经NAT服务器120转换前的地址与端口。 [0050] In step S250, SIP server 140 based on the checking result, determines whether to permit the communication request and the called number to confirm the communication terminal 160 of the normal condition, the result as to whether to allow the requested communication via the relay server 130 transmits to IP PBX 110, wherein, when the SIP server 140 using SIP via the communication requirements result of the relay server 130 transmits to the IP PBX 110, the relay server 130 changes the content of the SIP packet, and preferably, the modification of the SIP packet content the first subscript is the source packet content of the SIP address from the SIP server port 140 is changed by the NAT server and the port address 120 before conversion. 若允许该通讯要求,则进至步骤S260 ;若不允许该通讯要求,则进至步骤S255。 If required to allow the communication, the flow advances to step S260; if not allow the communication requirements, the process proceeds to step S255.

[0051] 在步骤S255中,SIP服务器140通过中继服务器130响应IP PBXllO不允许该通讯要求,并结束该通讯要求,接着回到步骤S225。 [0051] In step S255, SIP server 140 through the relay server 130 IP PBXllO does not allow the communication requirements, and ends the communication requirements, and then returns to step S225 in response. 此外,于本发明的不同实施例中,在结束该通讯要求后,亦可选择性地直接结束此程序。 Moreover, different embodiments of the present invention, after the end of the communication requirements, can selectively direct the end of the program.

[0052] 在步骤S260中,SIP服务器140通过中继服务器130响应IP PBXllO允许该通讯要求的结果,且中继服务器130与IP PBX 110建立通讯信道,同时中继服务器130选择使用对应SIP服务器140的账号并与SIP服务器140建立通讯信道,以传送通讯封包至与相对应的SIP服务器140连结的被叫号码端160,且中继服务器130记录建立该通讯信道的时间等通讯数据,以进一步认证与管理IP PBX 110。 [0052] In step S260, the SIP server 140 through the relay server 130 IP PBXllO allows the result to the communication requirements, and the relay server 130 establishes a communication channel in response to the IP PBX 110, while the relay server 130 selects the SIP server 140 using the corresponding account and establishes the SIP server 140 a communication channel to transmit the communication packet to the corresponding SIP server called number 140 coupled to terminal 160 and the relay 130 recording server setup time of the communication channel and other communication data for further authentication and management of IP PBX 110. 接着进至步骤S270。 Then proceeds to step S270.

[0053] 在步骤S270中,当IP PBX 110传送通讯封包至中继服务器130时,中继服务器130记录IP PBX 110使用的实时传输协议(Real-time Transfer Protocol, RTP)的地址与端口。 [0053] In step S270, the transmission when the IP PBX 110 communication packet 130 to the relay server, the relay server 130 IP PBX 110 recording real time transport protocol (Real-time Transfer Protocol, RTP) used address and port. 另一方面,中继服务器130向IP PBX 110传送再邀请(re-invite)要求,并变更IP PBX 110使用的RTP的地址与端口,以使IP PBX 110与SIP服务器140直接通讯。 On the other hand, the relay server 130 transmits the invite IP PBX 110 (re-invite) requirements, and changing the address and port used by RTP IP PBX 110 so that the IP PBX 110 and SIP server 140 communicate directly. 当SIP 服务器140传送通讯封包至中继服务器130时,中继服务器130记录SIP服务器140使用的RTP的地址与端口。 When 130, SIP server 140 using the relay server 130 records an RTP address and port number of the SIP server 140 transmits the communication packet to the relay server. 另一方面,中继服务器130向SIP服务器140传送再邀请要求,并变更SIP服务器140使用的RTP的地址与端口,以使IP PBX 110与该SIP服务器140直接通讯。 On the other hand, the relay server 130 then transmits an invite request 140 to the SIP server, and change the address and port of the SIP server 140 using the RTP, IP PBX 110 so that the SIP server 140 communicate directly. 接着进至步骤S280。 Then proceeds to step S280.

[0054] 在步骤S280中,当IP PBX 110与SIP服务器140结束通讯时,IPPBX 110传送结束通讯要求至中继服务器130,且中继服务器130记录结束该通讯信道的时间等通讯数据, 以进一步认证与管理IP PBX 110。 [0054] In step S280, the IP PBX 110 when the SIP server 140 and the communication ends, the communication ends in claim IPPBX 110 transmits to the relay server 130 and relay server 130 ends the recording of the communication channel, data communication time, to further certification and management IP PBX 110. 接着进至步骤S290。 Then proceeds to step S290.

[0055] 在步骤S290中,中继服务器130传送该结束通讯要求至SIP服务器140并结束该通讯信道,且将建立该通讯信道与结束该通讯信道的通讯数据进行处理以认证与管理IP PBX 110,其处理可例如为计算建立该通讯信道的时间与结束该通讯信道的时间,以计算通讯费用等,但并不以此为限。 [0055] In step S290, the relay server 130 transmits the end of the communication request to the SIP server 140 and ends the communication channel, and the establishment of the communication channel and communication data end of the communication channel is processed to authentication and management of IP PBX 110 , which may be, for example, calculate the processing time of the establishment of a communication channel of the communication channel and the end time to calculate the communication fee and the like, but is not limited thereto.

[0056] 第二实施例: [0056] Second Example:

[0057] 请参阅图3,为根据本发明的使用会话初始协议的多重注册的通讯系统300的第二实施例的系统架构图。 [0057] Please refer to FIG. 3, a system architecture diagram of a second embodiment of the present invention using a session initiation protocol registration multiplex communication system 300 according to. 本实施例与第一实施例的主要差异在于本实施例以VoIP与VoIP 网关器取代第一实施例的IP PBX0而于本实施例中,主要的应用环境与步骤与第一实施例相同,故于相同的部分不另为文赘述的。 The main difference between the present embodiment and the first embodiment in that the embodiment of the present embodiment and the embodiment VoIP to VoIP gateways substituted IP PBX0 first embodiment and the present embodiment, the main application environment and the same steps as the first embodiment, so to the same parts is not repeated for the text of the other.

[0058] 如图3所示,本发明的使用会话初始协议的多重注册的系统300架构在因特网上, 包括网络电话(VoIP) 310、VoIP网关器315、NAT服务器320、中继服务器330、多个SIP服务器340,其中,VoIP 310与VoIP网关器315连接,且VoIP网关器315与NAT服务器320 连接,NAT服务器320可将输入的虚拟网络地址与端口予以转换成实体网络地址与端口,并将输入的虚拟网络地址与端口以及转换后的实体网络地址与端口储存于路由表325。 [0058] As shown, the present invention uses the session initiation protocol register multiple system architecture 300 over the Internet, Internet telephony (VoIP) 310, VoIP gateway 315, NAT server 3203, the relay server 330, multiple SIP server 340, wherein, VoIP gateway 310 and VoIP connections 315, 315 and the VoIP gateway server 320 is connected to the NAT, NAT server 320 can be entered virtual network address and port to be converted into a physical address and a network port, and the input port and the virtual network address and the physical network address and port in the routing conversion table 325 is stored. 中继服务器330通过NAT服务器320与VoIP网关器315连接,且中继服务器330具有记录表335与拨号表338。 NAT server 330 is connected via the relay server 320 and VoIP gateway 315, the relay server 330 and the table 335 and a recording dial table 338. 多个SIP服务器340与中继服务器330连接。 A plurality of SIP servers 340 connected to the relay server 330. 此外,本实施例中的VoIP 310、VoIP网关器315与SIP服务器340的数目均为例示说明,于本发明的不同实施例中, 该VoIP 310、VoIP网关器315与SIP服务器340的数目并不以此为限。 In addition, VoIP 310 in the present embodiment, VoIP gateway 315 and the number of the SIP server 340 are described as an example is shown, different embodiments of the present invention, the number of VoIP 310, VoIP gateway 315 and SIP server 340 is not limited thereto.

[0059] 此外,在本发明的系统300中,可选择性地包括具有LDAP的服务器350,与中继服务器330连接,以进行账号与密码的管理。 [0059] Further, in the system 300 of the present invention, it may optionally include an LDAP server 350, connected to the relay server 330, to manage the account and password.

[0060] 再者,在本发明的系统300中,可选择性地包括被叫号码端360,被叫号码端360与SIP服务器340连接,以进行通讯封包的传送,于本实施例中的被叫号码端360与SIP服务器340的连接关系仅为例示说明,于本发明的不同实施例中,被叫号码端360可与其它SIP 服务器340连接。 [0060] Further, in the system 300 according to the present invention may optionally include an end 360 the called number, the called number end 340 is connected to the SIP server 360, the communication packet for transmission, the present embodiment is the embodiment call number 360 and the end 340 of the SIP server connection relationship described is only an example, in various embodiments of the present invention, the called number end 360 may be connected to other SIP servers 340.

[0061] 请参阅图4,为根据本发明的使用会话初始协议的多重注册的通讯方法400的第二实施例的流程图,其中,VoIP网关器315、中继服务器330、SIP服务器340通过组态方式进行下列步骤。 [0061] Please refer to FIG. 4, is a flowchart of a second embodiment of the communication method using multiple registration sessions of the present invention the SIP 400, which, VoIP gateways 315, relay server 330, SIP server 340 through a group states the following step manner.

[0062] 如图4所示,在步骤S410中,在因特网上提供VoIP 310、VoIP网关器315、中继服务器330以及多个SIP服务器340,其中,VoIP 310与VoIP网关器315连接,且中继服务器330与多个SIP服务器340连接,并通过NAT服务器320与VoIP网关器315连接。 [0062] As shown in FIG. 4, in step S410, 310 provided on the Internet VoIP, VoIP gateway 315, the relay server 330 and a plurality of SIP servers 340, wherein, VoIP gateway 310 and VoIP connection 315, and in following the SIP server 330 connected to multiple servers 340 and 320 connected by NAT server 315 of VoIP gateways. 接着进至步骤S420。 Then proceeds to step S420.

[0063] 在步骤S420中,VoIP网关器315向中继服务器330注册,且中继服务器330向多个SIP服务器340注册,其中,多个SIP服务器340检查该注册的账号及/或密码,并将是否允许该注册的结果传送至中继服务器330。 [0063] In step S420, VoIP gateway 315 330 on to the relay server, the relay server 330 and the register 340 to a plurality of SIP servers, wherein the plurality of SIP register server 340 checks the account and / or password, and the result of the registration is allowed is transmitted to the relay server 330. 若允许,则传送允许注册,并进至步骤S425 ; 若不允许,则传送拒绝注册要求,并结束此程序。 If allowed, the transfer permission registration, and proceeds to step S425 of; if allowed, is transmitted rejected registration requirements, and this routine ends.

[0064] 在步骤S425中,中继服务器330会监听是否有通讯要求传送至中继服务器330。 [0064] In step S425, the relay server 330 may monitor whether there claims correspond to the relay server 330. 若有,则进至步骤S430 ;若无,则持续执行本步骤S425。 If so, the process proceeds to step S430; if not, then continuing the implementation of this step S425.

[0065] 在步骤S430中,当VoIP网关器315使用SIP将通讯要求通过NAT服务器320传送至中继服务器330时,该中继服务器330利用拨号表338选择该多个SIP服务器340的其中至少一个,优选地,中继服务器330根据拨号表338中的SIP服务器340与VoIP网关器315的拨叫号码之间的对应关系选择该多个SIP服务器340的其中至少一个;此外,中继服务器330变更该SIP的封包内容,优选地,该变更SIP的封包内容是将封包内容中的该SIP的标头来源从经NAT服务器320转换前的地址与端口变更为中继服务器330的地址与端口。 [0065] In step S430, when the VoIP gateway 315 using SIP 338 will select the communication requirements of the SIP server 340 through a plurality of NAT server 320 transmits to the relay server 330, the relay server 330 using a dial-up table wherein the at least one preferably, the relay server 330 selects the plurality of SIP server 340, wherein the at least one correspondence between the dial table 338 in the SIP server 340 and VoIP gateway 315 of the dialed number; in addition, the relay server 330 changes the SIP packet content, preferably, the content of the change is the SIP packet header of the source packet content from the SIP address and port before the change by the NAT server 320 to the relay server 330 converts the address and port. 接着进至步骤S435。 Then proceeds to step S435.

[0066] 在步骤S435中,中继服务器330将该通讯要求传送至被选择的SIP服务器340。 [0066] In step S435, the relay server 330 transmits to the communications requirement of the selected SIP server 340. 接着进至步骤S440。 Then proceeds to step S440.

[0067] 在步骤S440中,SIP服务器340检查该SIP的封包内容,其中,检查该SIP的封包内容包括检查地址与端口、账号、该SIP的网域、被叫号码及/或最大同时通话数量等。 [0067] In step S440, the packet SIP server 340 checks the contents of the SIP, wherein the SIP packet inspection includes inspection port address, account number, the SIP domain, called number and / or maximum number of simultaneous calls Wait. 接着进至步骤S450。 Then proceeds to step S450.

[0068] 在步骤S450中,SIP服务器340根据该检查结果,判断是否允许该通讯要求,并确认被叫号码端360的通讯状况正常后,将是否允许该通讯要求的结果经由中继服务器330 传送至该VoIP网关器315,其中,当SIP服务器340使用SIP将通讯要求的结果经由中继服务器330传送至VoIP网关器315时,中继服务器330变更该SIP的封包内容,优选地,该变更SIP的封包内容是将该封包内容中的该SIP的标头来源从SIP服务器340的地址与端口变更为经该NAT服务器320转换前的地址与端口。 [0068] In step S450, SIP server 340 based on the checking result, determines whether to permit the communication request and the called number to confirm the end 360 correspond to normal conditions after the result whether to allow the requested communication via the relay server 330 transmits the VoIP gateway to 315, wherein, when the SIP server 340 using SIP communication requirements will result 315, the relay server 330 changes the content of the SIP packet via the relay server 330 transmits to the VoIP gateway, preferably, the change SIP the packet content is the standard source of the packet header contents of the SIP address from the SIP server port 340 is changed by the NAT server address and port 320 before conversion. 若允许该通讯要求,则进至步骤S460 ; 若不允许该通讯要求,则进至步骤S455。 If required to allow the communication, the flow advances to step S460; if not allow the communication requirements, the process proceeds to step S455.

[0069] 在步骤S455中,SIP服务器340通过中继服务器330响应VoIP网关器315不允许该通讯要求,并结束该通讯要求,接着回到步骤S425。 [0069] In step S455, SIP server through the relay server response 340330 VoIP gateway 315 does not allow the communication requirements, and ends the communication requirements, and then returns to step S425. 此外,于本发明的不同实施例中,在结束该通讯要求后,亦可选择性地直接结束此程序。 Moreover, different embodiments of the present invention, after the end of the communication requirements, can selectively direct the end of the program.

[0070] 在步骤S460中,SIP服务器340通过中继服务器330响应该VoIP网关器315允许该通讯要求的结果,且中继服务器330与VoIP网关器315建立通讯信道,同时中继服务器330选择使用对应SIP服务器340的账号并与SIP服务器340建立通讯信道,以传送通讯封包至与相对应的SIP服务器340连结的被叫号码端360,且中继服务器330记录建立该通讯信道的时间等通讯数据,以进一步认证与管理VoIP网关器315。 [0070] In step S460, SIP server 340 through the relay server 330 in response to the VoIP gateway 315 allows the result to the communication requirements, the relay server 330 and the gateway 315 to establish a VoIP communication channel, while choosing the relay server 330 corresponding account SIP server 340 and the SIP server to establish a communication channel 340 to transmit the communication packet corresponding to the SIP server called number 340 coupled to terminal 360 and the relay 330 records the server to establish the communication channel time communication data to to further authenticate and manage VoIP gateway 315. 接着进至步骤S470。 Then proceeds to step S470. [0071] 在步骤S470中,当VoIP网关器315传送通讯封包至中继服务器330时,中继服务器330记录VoIP网关器315使用的RTP的地址与端口。 [0071] In step S470, when the address and port VoIP gateway 315 transmits the communication packet to the relay server 330, the relay server 330 records the VoIP gateway 315 using the RTP. 另一方面,中继服务器330向VoIP 网关器315传送再邀请要求,并变更VoIP网关器315使用的RTP的地址与端口,以使VoIP 网关器315与SIP服务器340直接通讯。 On the other hand, the relay server 330 re-transmits an invite request 315 to the VoIP gateway, and changing address and port VoIP gateway 315 uses the RTP, VoIP gateway 315 so that the SIP server 340 communicate directly. 当SIP服务器340传送通讯封包至中继服务器330时,中继服务器330记录SIP服务器340使用的RTP的地址与端口。 Address and port when the SIP server 340 transmits the communication packet to the relay server 330, the relay server 330 records the SIP server 340 using the RTP. 另一方面,中继服务器330向SIP服务器340传送再邀请要求,并变更SIP服务器340使用的RTP的地址与端口,以使VoIP网关器315与SIP服务器340直接通讯。 On the other hand, the relay server 330 transmits invite SIP server 340 to request and change RTP address and port used by the SIP server 340, VoIP gateway 315 so that the SIP server 340 communicate directly. 接着进至步骤S480。 Then proceeds to step S480.

[0072] 在步骤S480中,当VoIP网关器315与SIP服务器340结束通讯时,VoIP网关器315传送结束通讯要求至中继服务器330,且中继服务器330记录结束该通讯信道的时间等通讯数据,以进一步认证与管理VoIP网关器315。 [0072] In step S480, when the VoIP gateway 315 and SIP server 340 ends communications, VoIP gateway 315 transmits the communication request to the end of the relay server 330, and the end time of the communication channel and other communication data relay server 330 records to further authenticate and manage VoIP gateway 315. 接着进至步骤S490。 Then proceeds to step S490.

[0073] 在步骤S490中,中继服务器330传送该结束通讯要求至SIP服务器340,并结束该通讯信道,且将建立该通讯信道与结束该通讯信道的通讯数据进行处理以认证与管理VoIP 网关器315。 [0073] In step S490, the relay server 330 transmits the end of the communication request to the SIP server 340, and ends the communication channel, and the establishment of the communication channel and communication data end of the communication channel is processed to authentication and management of VoIP gateways 315. 其处理可例如为计算建立该通讯信道的时间与结束该通讯信道的时间,以计算通讯费用等,但并不以此为限。 Which process may be established, for example, the communication channel for the computing time and end time of the communication channel to calculate the communication fee and the like, but is not limited thereto.

[0074] 在上述的实施例中,IP PBX与VoIP网关器可统称为客户端,且中继服务器设定与IP PBX之间的主干以及VoIP网关器向中继服务器注册,可统称为中继服务器建立与客户端之间的联机。 [0074] In the above embodiment, the VoIP gateway IP PBX may be collectively referred to as the client, and the server is set between the relay and IP PBX trunk and VoIP gateways registered to the relay server, the relay may be referred to as establish a connection between the server and the client.

[0075] 第三实施例: [0075] Third Embodiment:

[0076] 请参阅图5,为根据本发明的使用会话初始协议的多重注册的通讯系统500的第三实施例的系统架构图。 [0076] Referring to FIG. 5, is a system architecture diagram according to the present invention using the session initiation protocol registration multiplex communication system 500 of the third embodiment. 本实施例与第一、二实施例的主要差异在于本实施例不具有NAT 服务器与路由表。 The present embodiment and the first embodiment, the main difference between the two embodiments is that the present embodiment does not have a NAT server in the routing table. 而于本实施例中,主要的应用环境与步骤与第一、二实施例相同,故于相同的部分不另为文赘述的。 And in the present embodiment, the main application environment and the first step, two cases of the same embodiment, it is the same as the other portions are not repeated in the text.

[0077] 如图5所示,本发明的使用会话初始协议的多重注册的系统500架构在因特网上, 包括中继服务器530以及多个SIP服务器540,其中,中继服务器530与客户端510连接,且中继服务器530具有记录表535与拨号表538。 [0077] As shown in FIG. 5, using the session initiation protocol of the present invention, multiple registration system 500 over the Internet architecture, including the relay server 530 and a plurality of SIP server 540, wherein the relay server 530 and the client terminal 510 is connected , the relay server 530 and the table 535 and a recording dial table 538. 多个SIP服务器540与中继服务器530连接。 A plurality of SIP servers 540 connected to the relay server 530. 此外,本实施例中的客户端510与SIP服务器540的数目均为例示说明,于本发明的不同实施例中,该客户端510与SIP服务器540的数目并不以此为限。 Further, in the embodiment according to the present embodiment the client end 510 and the number of the SIP server 540 are shown as an example described in the various embodiments of the present invention, the number of the client 510 and the SIP server 540 side is not limited thereto.

[0078] 此外,在本发明的系统500中,可选择性地包括具有LDAP的服务器550,具有LDAP 的服务器550与中继服务器530连接,以进行账号与密码的管理。 [0078] Further, in system 500 of the present invention, may optionally include an LDAP server 550, the LDAP server 550 has the relay server 530 is connected to the account and password management.

[0079] 再者,在本发明的系统500中,可选择性地包括被叫号码端560,与SIP服务器540 连接,以进行通讯封包的传送,于本实施例中的被叫号码端560与SIP服务器540的连接关系仅为例示说明,于本发明的不同实施例中,被叫号码端560可与其它SIP服务器540连接。 [0079] Further, in system 500 of the present invention, the called number may optionally include an end 560 connected to the SIP server 540, the communication packet for transmission, called number to the terminal 560 of the present embodiment and connection relationship SIP server 540 described are illustrative only, different embodiments of the present invention, the called number end 560 may be connected to other SIP servers 540.

[0080] 请参阅图6,为根据本发明的使用会话初始协议的多重注册的通讯方法600的第三实施例的流程图,其中,客户端510、中继服务器530、SIP服务器540通过组态方式进行下列步骤。 [0080] Please refer to FIG. 6 is a flowchart of a third embodiment of the communication method using multiple registration session initiation protocol of the present invention 600, in which the client 510, relay server 530, SIP server 540 by configuring manner the following step.

[0081] 如图6所示,在步骤S610中,在因特网上提供中继服务器530以及多个SIP服务器540,其中,中继服务器530分别与客户端510以及多个SIP服务器540连接。 [0081] As shown in FIG 6, in step S610, the relay server 530, and providing a plurality of SIP server 540 on the Internet, wherein the relay server 530 and the client 510 respectively, and a plurality of SIP server 540 is connected. 接着进至步骤S620。 Then proceeds to step S620. [0082] 在步骤S620中,中继服务器530建立与该客户端510之间的联机,且中继服务器530向多个SIP服务器540注册,其中,多个SIP服务器540检查该注册的账号及/或密码, 并将是否允许该注册的结果传送至中继服务器530。 [0082] In step S620, the relay server 530 and establishes a connection between the client 510, and the relay server 530 to a plurality of SIP register server 540, wherein the plurality of SIP register server 540 checks the account and / or password, and whether to allow the results of the registration to the relay server 530. 若允许,则传送允许注册,并进至步骤S625若不允许,则传送拒绝注册要求,并结束此程序。 If allowed, the transfer is allowed to register and proceeds to step S625 if not allowed, then transferred to refuse registration requirements, and end this program.

[0083] 在步骤S625中,中继服务器530会监听是否有通讯要求传送至中继服务器530。 [0083] In step S625, the relay server 530 may monitor whether there claims correspond to the relay server 530. 若有,则进至步骤S630 ;若无,则持续执行本步骤S625。 If so, the process proceeds to step S630; if not, then continuing the implementation of this step S625.

[0084] 在步骤S630中,当客户端510使用SIP将通讯要求传送至中继服务器530时,该中继服务器530利用拨号表538选择该多个SIP服务器540的其中至少一个,优选地,中继服务器530根据拨号表538中的SIP服务器540与客户端510的拨叫号码之间的对应关系选择该多个SIP服务器540的其中至少一个。 When [0084] In step S630, the client 510 when using SIP request sent to the communication server 530 to the relay, the relay server 530 using a dial-up table 538 to select the plurality of SIP server 540 wherein the at least one, preferably, in the relay server 530 selects a plurality of SIP server 540 according to the correspondence between the dial table 538 in the SIP server 540 and the client 510 of the dialed number, wherein the at least one. 接着进至步骤S635。 Then proceeds to step S635.

[0085] 在步骤S635中,中继服务器530将该通讯要求传送至被选择的SIP服务器540。 [0085] In step S635, the relay server 530 transmits to the communications requirement of the selected SIP server 540. 接着进至步骤S640。 Then proceeds to step S640.

[0086] 在步骤S640中,SIP服务器540检查该SIP的封包内容,其中,检查该SIP的封包内容包括检查地址与端口、账号、该SIP的网域、被叫号码及/或最大同时通话数量等。 [0086] In step S640, the packet SIP server 540 checks the contents of the SIP, wherein the SIP packet inspection includes inspection port address, account number, the SIP domain, called number and / or maximum number of simultaneous calls Wait. 接着进至步骤S650。 Then proceeds to step S650.

[0087] 在步骤S650中,SIP服务器540根据该检查结果,判断是否允许该通讯要求,并确认被叫号码端560的通讯状况正常后,将是否允许该通讯要求的结果经由中继服务器530 传送至该客户端510。 [0087] In step S650, SIP server 540 based on the checking result, determines whether to permit the communication request and the called number to confirm the end 560 correspond to normal conditions after the result whether to allow the requested communication via the relay server 530 transmits to the client 510. 若允许该通讯要求,则进至步骤S660 ;若不允许该通讯要求,则进至步骤S655。 If required to allow the communication, the flow advances to step S660; if not allow the communication requirements, the process proceeds to step S655.

[0088] 在步骤S655中,SIP服务器540通过中继服务器530响应客户端510不允许该通讯要求,并结束该通讯要求,接着回到步骤S625。 [0088] In step S655, SIP server 540 through the relay server 530 in response to the client 510 does not allow the communication requirements, and ends the communication requirements, and then returns to step S625. 此外,于本发明的不同实施例中,在结束该通讯要求后,亦可选择性地直接结束此程序。 Moreover, different embodiments of the present invention, after the end of the communication requirements, can selectively direct the end of the program.

[0089] 在步骤S660中,SIP服务器540通过中继服务器530响应该客户端510允许该通讯要求的结果,且中继服务器530与客户端510建立通讯信道,同时中继服务器530选择使用对应SIP服务器540的账号并与SIP服务器540建立通讯信道,以传送通讯封包至与相对应的SIP服务器540连结的被叫号码端560,且中继服务器530记录建立该通讯信道的时间等通讯数据,以进一步认证与管理客户端510。 [0089] In step S660, the SIP server 540 of the client 510 allows the result to the communication requirements, and the relay server 530 to establish a communication channel with the client 510 through the relay server 530 in response, the relay server 530 while choosing the corresponding SIP server 540 account and establishes the SIP server 540 a communication channel to transmit the communication packet to the called number corresponding to the SIP server 540 coupled to terminal 560 and the relay 530 recording server setup time of the communication channel and other communication data to further authentication and management client 510. 接着进至步骤S670。 Then proceeds to step S670.

[0090] 在步骤S670中,当客户端510传送通讯封包至中继服务器530时,中继服务器530 记录客户端510使用的RTP的地址与端口。 [0090] In step S670, when the relay 510 transmits the communication packet to the client server 530, the relay server 530 records the client 510 uses the port address and the RTP. 另一方面,中继服务器530向客户端510传送再邀请要求,并变更客户端510使用的RTP的地址与端口,以使客户端510与SIP服务器540 直接通讯。 On the other hand, the relay server 530 to the client transfer 510 invite requirements, and changes of address and RTP port used by the client 510, so that the client 510 directly communicate with the SIP server 540. 当SIP服务器540传送通讯封包至中继服务器530时,中继服务器530记录SIP 服务器540使用的RTP的地址与端口。 When the SIP server 540 transmits the communication packet to the relay server 530, the relay server address and port 530 records the SIP server 540 using the RTP. 另一方面,中继服务器530向SIP服务器540传送再邀请要求,并变更SIP服务器540使用的RTP的地址与端口,以使客户端510与SIP服务器540直接通讯。 On the other hand, the relay server 530 to the SIP server 540 transmit invite requirements, and change of address with the SIP server port 540 using the RTP so that the client 510 to communicate directly with the SIP server 540. 接着进至步骤S680。 Then proceeds to step S680.

[0091] 在步骤S680中,当客户端510与SIP服务器540结束通讯时,客户端510传送结束通讯要求至中继服务器530,且中继服务器530记录结束该通讯信道的时间等通讯数据, 以进一步认证与管理客户端510。 [0091] In step S680, when the end of the SIP client 510 and the server 540 when the communication ends, the client 510 transmits the communication request to the relay server 530, the relay server 530 and the recording end time of the communication channel, data communication, etc., to further authentication and management client 510. 接着进至步骤S690。 Then proceeds to step S690.

[0092] 在步骤S690中,中继服务器530传送该结束通讯要求至SIP服务器540,并结束该通讯信道,且将建立该通讯信道与结束该通讯信道的通讯数据进行处理以认证与管理客户端510。 [0092] In step S690, the relay server 530 transmits the end of the communication request to the SIP server 540, and ends the communication channel, and the establishment of the communication channel and communication data end of the communication channel is processed to authentication and management of a client 510. 其处理可例如为计算建立该通讯信道的时间与结束该通讯信道的时间,以计算通讯费用等,但并不以此为限。 Which process may be established, for example, the communication channel for the computing time and end time of the communication channel to calculate the communication fee and the like, but is not limited thereto.

[0093] 举例而言,请再次参阅图5,客户端510欲拨打室内电话,如0212345678,至被叫号码端560,则当通讯要求传送至中继服务器530时,中继服务器530利用拨号表538中的SIP服务器540与客户端510的拨叫号码之间的对应关系选择通讯费用较低廉的SIP服务器540。 [0093] For example, referring to FIG 5 again, the client want to place calls to 510, such as 0212345678, called number to terminal 560, then when the communication request sent to the relay server 530, the relay server 530 using a dial-up table 538 in the SIP server 540 correspondence between dialing numbers 510 selected communication costs cheaper SIP server 540 and the client. 相似地,客户端510欲拨打移动电话,如0912345678,至被叫号码端560,则当通讯要求传送至中继服务器530时,中继服务器530利用拨号表538中的SIP服务器540与客户端510的拨叫号码之间的对应关系选择通讯费用较低廉的SIP服务器540。 Similarly, the client 510 to be a mobile telephone call, such as 0912345678, called number to terminal 560, then when the communication request sent to the relay server 530, the relay server 530 using a dial-up table 538 in the SIP server 540 and the client 510 correspondence between the dialed number to select communication costs cheaper SIP server 540. 因此,中继服务器中的拨号表可针对不同拨叫号码提供通讯费用较低廉的SIP服务器,以节省客户端的通讯费用。 Therefore, the relay server in the dial-up tables to provide communication costs cheaper SIP server for different dialing numbers, in order to save communication costs the client. 综上所述,本发明利用中继服务器一方面建立与客户端之间的联机,另一方面向多个SIP服务器注册,从而通过选择多个SIP服务器的其中至少一个而使客户端与所选择的SIP服务器直接通讯。 In summary, the present invention utilizes a relay server establishes a connection between the aspect of the client, on the other hand to a plurality of SIP registration server, so that by selecting a plurality of SIP servers wherein at least one of the client and the selected the SIP server communicate directly. 由此解决SIP服务器与客户端兼容性不佳、SIP服务器之间兼容性不佳的问题,并可针对客户端的不同拨叫号码提供节省通讯费用的方案。 Thereby solving the SIP server and client compatibility is poor, poor compatibility between the SIP server problems, and provide cost-saving communication solutions for different dialing numbers clients.

[0094] 上述实施例仅例示性说明本发明的原理及其功效,而非用于限制本发明,任何本领域技术人员均可在不违背本发明的精神及范畴下,对上述实施例进行修饰与改变。 [0094] The above-described embodiments are only illustrative of the principles and effect of the present invention and is not intended to limit the present invention, anyone skilled in the art may be made without departing from the spirit and scope of the invention, the above-described embodiments can be modified and change. 此外, 在上述实施例中的组件的数量仅为例示性说明,亦非用于限制本发明。 Further, the number of components in the above embodiments described are exemplary only, nor intended to limit the present invention. 因此,本发明的权利保护范围,应如权利要求书所列。 Accordingly, the scope of rights of the present invention, as listed in a claim should book.

Claims (30)

  1. 1. 一种使用会话初始协议的多重注册的通讯方法,其特征在于:令中继服务器建立与客户端之间的联机;令该中继服务器向多个会话初始协议服务器注册;令该客户端使用会话初始协议将通讯要求传送至该中继服务器;令该中继服务器选择该多个会话初始协议服务器的其中至少一个,并将该通讯要求传送至被选择的会话初始协议服务器;以及令该会话初始协议服务器检查该会话初始协议的封包内容后,判断是否允许该通讯要求,并将判断结果经由该中继服务器传送至该客户端。 1. A method of using the Session Initiation Protocol registration of multiple communication method, comprising: command relay server establishes a connection between the client; enabling the relay server to a plurality of Session Initiation Protocol registration server; enabling the client using the session Initiation protocol communication request sent to the relay server; enabling the relay server selects the plurality of session Initiation protocol server wherein at least one, and the communication request sent to the SIP server is selected; and enabling the the session Initiation protocol SIP server checks the contents of the packet, determines whether to permit the communication requirements, and the determination result is transmitted to the client via the relay server.
  2. 2.根据权利要求1所述的使用会话初始协议的多重注册的通讯方法,其特征在于,该客户端架构在因特网上,该中继服务器架构在该因特网上并与该客户端连接,该多个会话初始协议服务器架构在该因特网上并与该中继服务器连接。 The multiple registrations using Session Initiation Protocol communication method according to claim 1, wherein the client on the Internet architecture, the architecture and the relay server connected to the client on the Internet, the multi- a session Initiation protocol server architecture and connected to the relay server on the Internet.
  3. 3.根据权利要求1所述的使用会话初始协议的多重注册的通讯方法,其特征在于,该中继服务器利用拨号表选择该多个会话初始协议服务器的其中至少一个。 The use of multiple registrations SIP communication method according to claim 1, wherein the relay server using the selected plurality of dial SIP server wherein at least one.
  4. 4.根据权利要求3所述的使用会话初始协议的多重注册的通讯方法,其特征在于,该中继服务器根据该拨号表中的会话初始协议服务器与客户端的拨叫号码之间的对应关系选择该多个会话初始协议服务器的其中至少一个。 The use of multiple registrations SIP communication method according to claim 3, wherein the relay server according to the correspondence table between the dial Session Initiation Protocol server and the client dialing number selection the plurality of session Initiation protocol server, wherein the at least one.
  5. 5.根据权利要求1所述的使用会话初始协议的多重注册的通讯方法,其特征在于,该客户端使用该会话初始协议将该通讯要求通过网络地址转换服务器传送至该中继服务器。 The use of multiple registrations SIP communication method according to claim 1, wherein the client uses the session initiation protocol via the communication requirements of network address translation server transmits to the relay server.
  6. 6.根据权利要求5所述的使用会话初始协议的多重注册的通讯方法,其特征在于,当该客户端使用会话初始协议将该通讯要求通过该网络地址转换服务器传送至该中继服务器时,令该中继服务器变更该会话初始协议的封包内容。 The use of multiple SIP registration communication method according to claim 5, wherein, when the client uses the Session Initiation Protocol server is transmitted to the communications requirement of the relay server via the network address conversion, enabling the relay server to change the packet contents of the session Initiation protocol.
  7. 7.根据权利要求6所述的使用会话初始协议的多重注册的通讯方法,其特征在于,该变更会话初始协议的封包内容是将该封包内容中的该会话初始协议的标头来源,从经该网络地址转换服务器转换前的地址与端口变更为该中继服务器的地址与端口。 7. The communication method of multiple registrations using SIP according to claim 6, characterized in that the change of the SIP packet content is the source of the packet header of the content of the Session Initiation Protocol, via the the network address translation and port address change before the server converted to the relay server address and port.
  8. 8.根据权利要求1所述的使用会话初始协议的多重注册的通讯方法,其特征在于,还包括:当该会话初始协议服务器允许该通讯要求时,则令该会话初始协议服务器通过该中继服务器响应该客户端允许该通讯要求的结果,并令该中继服务器与该客户端建立通讯信道,且令该中继服务器选择使用对应该会话初始协议服务器的账号并与该会话初始协议服务器建立通讯信道。 The use of multiple registrations SIP communication method according to claim 1, characterized in that, further comprising: when the SIP server permits the communication requirements, enabling the Session Initiation Protocol server through the relay this allows the server responds to the client the result of the communication requirements, and to make the relay server to establish a communication channel with the client, server and enabling the relay to be selected using the SIP account server and establishes the session Initiation protocol server communications channel.
  9. 9.根据权利要求8所述的使用会话初始协议的多重注册的通讯方法,其特征在于,进一步包括:当该客户端与该会话初始协议服务器结束通讯时,令该客户端传送结束通讯要求至该中继服务器;令该中继服务器传送该结束通讯要求至该会话初始协议服务器;以及令该中继服务器结束该通讯信道。 9. The use of multiple registrations SIP communication method according to claim 8, characterized in that, further comprising: when the client and server ends the session initiation protocol communications, enabling the client terminal end transmits to the communications requirements the relay server; enabling the relay server transmits the communication request to the end of the session Initiation protocol server; and enabling the relay server ends the communication channel.
  10. 10.根据权利要求9所述的使用会话初始协议的多重注册的通讯方法,其特征在于,令该中继服务器记录建立该通讯信道与结束该通讯信道的通讯数据。 10. The use of multiple SIP registration communication method according to claim 9, wherein enabling the establishment of the communication relay server records the communication data channel with the end of the communication channel.
  11. 11.根据权利要求10所述的使用会话初始协议的多重注册的通讯方法,其特征在于,该通讯数据为通讯时间。 11. The use of multiple SIP registration communication method according to claim 10, wherein the communication data communication time.
  12. 12.根据权利要求1所述的使用会话初始协议的多重注册的通讯方法,其特征在于,还包括:当该会话初始协议服务器不允许该通讯要求的结果,则令该会话初始协议服务器通过该中继服务器响应该客户端不允许该通讯要求,且结束该通讯要求。 12. The use of multiple registrations SIP communication method according to claim 1, characterized in that, further comprising: when the SIP server does not allow the result of the communication requirements, then enabling the Session Initiation Protocol server through the the relay server responds to the client does not allow the communication requirements, and the end of the communication requirements.
  13. 13.根据权利要求1所述的使用会话初始协议的多重注册的通讯方法,其特征在于,当该中继服务器向该会话初始协议服务器注册时,令该会话初始协议服务器检查该注册的账号及/或密码,并将是否允许该注册的结果传送至该中继服务器。 Multiple registration communication method according to claim SIP use of claim 1, wherein, when the relay server to the SIP register server, enabling the SIP registration server checks the account and / or password, and whether to permit the registration result is transmitted to the relay server.
  14. 14.根据权利要求1所述的使用会话初始协议的多重注册的通讯方法,其特征在于,该会话初始协议服务器检查该会话初始协议的封包内容包括检查地址与端口、账号、该会话初始协议的网域、被叫号码及/或最大同时通话数量。 14. The use of multiple registrations SIP communication method according to claim 1, wherein the Session Initiation Protocol server checks the contents of the SIP packet includes checking the address and port, account number, the Session Initiation Protocol domain, called number and / or the maximum number of calls simultaneously.
  15. 15.根据权利要求14所述的使用会话初始协议的多重注册的通讯方法,其特征在于, 还包括:当该客户端传送通讯封包至该中继服务器时,令该中继服务器记录该客户端使用的实时传输协议的地址与端口;以及令该中继服务器向该客户端传送再邀请要求,并变更该客户端使用的实时传输协议的地址与端口,以使该客户端与该会话初始协议服务器直接通讯。 15. The use of the Session Initiation Protocol registration of multiple communication method according to claim 14, characterized in that, further comprising: when the UE sends the packet to the communication relay server, the relay server so that the client record address and port using the real-time transport protocol; and enabling the relay server to the client and then transmits the invitation request and the changed time transport protocol used by the client address and port, such that the client and the session Initiation protocol server communicate directly.
  16. 16.根据权利要求15所述的使用会话初始协议的多重注册的通讯方法,其特征在于, 还包括:当该会话初始协议服务器传送该通讯封包至该中继服务器时,令该中继服务器记录该会话初始协议服务器使用的实时传输协议的地址与端口;以及令该中继服务器向该会话初始协议服务器传送再邀请要求,并变更该会话初始协议服务器使用的实时传输协议的地址与端口,以使该客户端与该会话初始协议服务器直接通讯。 16. The use of the Session Initiation Protocol registration of multiple communication method according to claim 15, characterized in that, further comprising: when the SIP server transmits the communication packet to the relay server, the relay server so that the recording address and port of the session initiation protocol real-time transport protocol server for use; and enabling the relay server transmits to the server a session initiation protocol invite request and changes the session initiation protocol real-time transport protocol server address and port used to the customers of the session Initiation protocol server communicate directly end.
  17. 17.根据权利要求1所述的使用会话初始协议的多重注册的通讯方法,其特征在于,该客户端为网络电话网关器及/或IP用户交换机。 17. The use of multiple registrations SIP communication method according to claim 1, wherein the client is a VoIP gateway and / or IP PBX.
  18. 18.根据权利要求17所述的使用会话初始协议的多重注册的通讯方法,其特征在于, 当该客户端为网络电话网关器时,该中继服务器建立与该客户端之间的联机是令该客户端向该中继服务器注册。 18. The use of multiple SIP registration communication method according to claim 17, wherein, when the client is a VoIP gateway, it establishes a connection between the relay server and the client is to make the client registers to the relay server.
  19. 19.根据权利要求17所述的使用会话初始协议的多重注册的通讯方法,其特征在于, 当该客户端为IP用户交换机时,该中继服务器建立与该客户端之间的联机是令该中继服务器设定与该客户端之间的主干。 19. The use of multiple SIP registration communication method according to claim 17, wherein, when the user switches to IP client, establishes a connection between the relay server and the client is the enabling the trunk set between the relay server and the client.
  20. 20.根据权利要求1所述的使用会话初始协议的多重注册的通讯方法,其特征在于,该会话初始协议服务器为多媒体通讯服务器。 20. The use of multiple registrations SIP communication method according to claim 1, wherein the Session Initiation Protocol server for multimedia communication server.
  21. 21. 一种使用会话初始协议的多重注册的通讯系统,其特征在于,包括:中继服务器,架构在因特网上并通过该因特网与客户端连接;以及多个会话初始协议服务器,架构在该因特网上并与该中继服务器连接,其中,该中继服务器通过组态方式以建立与该客户端之间的联机,且该中继服务器通过组态方式向该多个会话初始协议服务器注册,而该客户端通过组态方式以使用会话初始协议将通讯要求传送至该中继服务器,该中继服务器选择该多个会话初始协议服务器的其中至少一个并将该通讯要求传送至被选择的会话初始协议服务器,并且该会话初始协议服务器通过组态方式以检查该会话初始协议的封包内容后,判断是否允许该通讯要求,并将判断结果经由该中继服务器传送至该客户端。 21. A method of using the Session Initiation Protocol registration of the multiplex communication system, characterized by comprising: a relay server architecture connected to the Internet and the client through the Internet; and a plurality of Session Initiation Protocol server, the Internet Architecture and connected to the relay server, wherein the relay server to establish a connection by way of the configuration between the client and the relay server to register a plurality of session Initiation protocol server through the configuration mode, and the client configuration mode by using the session Initiation protocol communication request sent to the relay server, the relay server selects the plurality of SIP servers and wherein at least one of the communication request sent to the selected session Initiation server protocol, and the session Initiation protocol server through configuration mode to check the contents of the packet after the session Initiation protocol, whether to permit the communication requirements, and the determination result is transmitted to the client via the relay server.
  22. 22.根据权利要求21所述的使用会话初始协议的多重注册的通讯系统,其特征在于, 还包括:网络地址转换服务器,架构在该因特网上并与该客户端连接,且与该中继服务器连接, 其中,该客户端通过组态方式以使用该会话初始协议将该通讯要求通过该网络地址转换服务器传送至该中继服务器。 22. The use of the Session Initiation Protocol as claimed in claim 21, multiple registrations of the communication system, characterized by, further comprising: a network address translation server on the Internet architecture and connected to the client, and the relay server connection, wherein the client by using the configuration mode the communications requirement of the session Initiation protocol server to the relay server transmits through the network address translation.
  23. 23.根据权利要求21所述的使用会话初始协议的多重注册的通讯系统,其特征在于, 该客户端为网络电话网关器及/或IP用户交换机中的至少其中一个。 23. The use of multiple registrations SIP communication system according to claim 21, wherein the at least one client of the telephone network gateway and / or user IP switch.
  24. 24.根据权利要求21所述的使用会话初始协议的多重注册的通讯系统,其特征在于, 该会话初始协议服务器为多媒体通讯服务器。 24. The use of multiple registrations SIP communication system according to claim 21, wherein the Session Initiation Protocol server for multimedia communication server.
  25. 25.根据权利要求22所述的使用会话初始协议的多重注册的通讯系统,其特征在于, 该中继服务器通过组态方式以变更该会话初始协议的封包内容。 25. The use of multiple registrations SIP communication system according to claim 22, wherein the relay server by configuring the contents of the packet to change the manner Session Initiation Protocol.
  26. 26.根据权利要求25所述的使用会话初始协议的多重注册的通讯系统,其特征在于, 该中继服务器通过组态方式以变更该会话初始协议的封包内容,是将该封包内容中的该会话初始协议的标头来源从经该网络地址转换服务器转换前的地址与端口变更为该中继服务器的地址与端口。 26. The use of the Session Initiation Protocol registration of the 25 multiplex communication system as claimed in claim, wherein the relay server is changed by the configuration mode to the Session Initiation Protocol packet content, that the content of the packet SIP header from sources via the network address translation with port address before the change of addresses and ports that the server converts the relay server.
  27. 27.根据权利要求21所述的使用会话初始协议的多重注册的通讯系统,其特征在于, 还包括:具有轻型目录访问协议的服务器,架构在该因特网上并与该中继服务器连接,以进行账号与密码的管理。 27. The use of the Session Initiation Protocol as claimed in claim 21, multiple registrations of the communication system, characterized by, further comprising: a server having a Lightweight Directory Access Protocol, and Internet architecture on which the relay server is connected to, for account and password management.
  28. 28.根据权利要求21所述的使用会话初始协议的多重注册的通讯系统,其特征在于, 该中继服务器具有记录表,用以记录该客户端与该会话初始协议服务器之间的通讯数据。 28. The use of multiple registrations SIP communication system according to claim 21, wherein the relay server having a record table for recording communication data between the client and the SIP server.
  29. 29.根据权利要求28所述的使用会话初始协议的多重注册的通讯系统,其特征在于, 该记录表用以记录该客户端与该会话初始协议服务器之间的通讯时间。 29. The use of multiple registrations SIP communication system according to claim 28, characterized in that, the recording sheet for recording the communication time between the client and the SIP server.
  30. 30.根据权利要求21所述的使用会话初始协议的多重注册的通讯系统,其特征在于, 该中继服务器具有拨号表,用以记录会话初始协议服务器与客户端的拨叫号码之间的对应关系。 30. The use of the Session Initiation Protocol registration of the multiple communication system according to claim 21, wherein the relay server has a dial table for correspondence between the dialed number recording Session Initiation Protocol server and the client .
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Citations (2)

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CN1889541A (en) * 2005-06-28 2007-01-03 北京寰龙技术有限公司 System for supporting multi ITSP based on SIP and realizing method
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Patent Citations (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN1889541A (en) * 2005-06-28 2007-01-03 北京寰龙技术有限公司 System for supporting multi ITSP based on SIP and realizing method
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