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CN102413110A - Communication method and system for multiple registration using session initiation protocol - Google Patents

Communication method and system for multiple registration using session initiation protocol Download PDF

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CN102413110A
CN102413110A CN2010102941444A CN201010294144A CN102413110A CN 102413110 A CN102413110 A CN 102413110A CN 2010102941444 A CN2010102941444 A CN 2010102941444A CN 201010294144 A CN201010294144 A CN 201010294144A CN 102413110 A CN102413110 A CN 102413110A
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sip
relay server
communication
client
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廖经富
林育正
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Chunghwa Telecom Co Ltd
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Chunghwa Telecom Co Ltd
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Abstract

A communication method and system of multiple registration using session initiation protocol includes client end, relay server and multiple SIP servers, among which, the relay server is connected with multiple SIP servers and client end. In addition, the relay server establishes the online with the client in a configuration mode, and registers to the SIP servers in the configuration mode, so that the client directly communicates with the selected SIP server by further selecting at least one of the SIP servers. Therefore, the problems of poor compatibility between the SIP server and the client and poor compatibility between the SIP servers can be solved, and a scheme for saving communication cost 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Communication method and system for multiple registrations using session initiation protocol

技术领域 technical field

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。The present invention relates to a communication method and system using session initiation protocol, in particular to a communication method and system using session initiation protocol for multiple registration.

背景技术 Background technique

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。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。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)。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(SessionInitiation Protocol,SIP)。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。Early voice communications were built on the Public Switched Telephone Network (PSTN) deployed by telecommunication service companies. PSTN is a telephone switching network used for global voice communication. It is currently the largest network in the world, with hundreds of millions of users. And along with the advancement of the Internet, voice communication can also be realized on the Internet, and the most popular technology at present is exactly VoIP (Voice over Internet Protocol, VoIP). To put it simply, VoIP converts the voice analog signal at the sending end into a digital signal, and then transmits it to the receiving end through the Internet, and the receiving end converts the digital signal into a voice analog signal to realize voice communication on the Internet. The most commonly used communication protocol is Session Initiation Protocol (SIP). In addition, there is another IP private branch exchange (IP PBX), which uses digital signals to communicate directly on the Internet.

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。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。On the other hand, due to the development of communication technology, in addition to the above-mentioned public switched telephone network and VoIP, wireless communication technologies such as GSM (Global System for Mobile Communication) mobile phone network and third-generation (3G) mobile phone network have also developed. is quite mature. And the existing communication method using SIP is that the SIP user sends the communication request to the SIP server of the telecommunication service company, and the SIP server forwards the communication request to different telephone networks according to the called number in the communication request, such as public switched telephone Internet, VoIP, etc., to complete the communication connection.

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。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。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。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。However, in an environment with multiple SIP servers, since the multiple SIP servers may belong to different telecommunications service companies, the compatibility between the SIP servers is not good, so it is impossible to set up a SIP trunk between the SIP servers. , and cannot communicate normally. In addition, because the compatibility between the client and the SIP server provided by the telecommunications service company is not high, some clients cannot register with the incompatible SIP server, or the incompatible SIP server cannot set up a SIP trunk with the client. It also causes abnormal communication. Furthermore, the client in the network address translation (Network Address Translation, NAT) environment will also encounter some problems. When the client requests registration from the above SIP server, the NAT server will convert the virtual network address in the enterprise into The physical network address outside the enterprise causes the SIP server to fail to respond to the original client with the registration result, resulting in the inability to register and thus causing abnormal communication. Finally, because the existing communication method using SIP transfers the communication request to different telephone networks in a fixed manner according to the called number, there is no solution for saving communication costs for different dialing numbers of the client.

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。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。To sum up, in the existing communication system, due to poor compatibility or the limitation of the NAT environment, the client cannot register with the SIP server, and there is also a problem of poor compatibility between the SIP servers, and there is no target for customers. Different dialing numbers at different terminals provide a solution to save communication costs. Therefore, there is a great need for a multi-registration communication method and system using SIP to solve the problem of poor compatibility between the SIP server and the client and the poor compatibility between the SIP servers, and can provide different dialing numbers for the client. A plan to save communication costs.

发明内容 Contents of the invention

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。The present invention provides a multi-registration communication method and system using the Session Initiation Protocol to solve the problems of poor compatibility between SIP servers and clients and poor compatibility between SIP servers in the prior art, and can address the problems of clients Different dialing numbers provide a solution to save communication costs.

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。According to one embodiment of the present invention, there is provided a communication method using multiple registrations of the Session Initiation Protocol, comprising the following steps: making the relay server establish a connection with the client; making the relay server register with multiple SIP servers ; making the client use SIP to send a communication request to the relay server; making the relay server select at least one of the plurality of SIP servers and sending the communication request to the selected SIP server; and, making the SIP After checking the content of the SIP packet, the server judges whether to allow the communication request, and transmits the judgment result to the client via the relay server.

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。In addition, the present invention also provides a multi-registration communication system using the session initiation protocol, including: a relay server, which is built on the Internet and connected to the client through the Internet; and a plurality of session initiation protocol servers, which are built on the Internet and connect with the relay server, wherein the relay server establishes connection with the client through the configuration method, and the relay server registers with the plurality of session initiation protocol servers through the configuration method, and The client is configured to use the session initiation protocol to transmit the communication request to the relay server, and the relay server selects at least one of the plurality of session initiation protocol servers and transmits the communication request to the selected session initiation protocol A protocol server, and the SIP server checks the packet content of the SIP through configuration, judges whether to allow the communication request, and transmits the judgment result to the client via the relay server.

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。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。As mentioned above, compared with the prior art, the present invention utilizes the relay server to establish connection with the client on the one hand, and to register with multiple SIP servers on the other hand, so that by selecting at least one of the multiple SIP servers, Enables the client to communicate directly with the selected SIP server. In this way, the problem of poor compatibility between the SIP server and the client and between SIP servers is solved, and a solution for saving communication costs can be provided for different dial numbers of the client.

附图说明 Description of drawings

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;FIG. 1 is a system architecture diagram of the first embodiment of the communication system using multiple registrations of the session initiation protocol of the present invention;

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;FIG. 2 is a flow chart of the first embodiment of the communication method using multiple registrations of the session initiation protocol of the present invention;

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;FIG. 3 is a system architecture diagram of a second embodiment of a communication system using multiple registrations of the session initiation protocol of the present invention;

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;FIG. 4 is a flowchart of a second embodiment of the communication method using multiple registrations of the session initiation protocol of the present invention;

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;FIG. 5 is a system architecture diagram of a third embodiment of a communication system using multiple registrations of the session initiation protocol of the present invention;

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。FIG. 6 is a flow chart of a third embodiment of the communication method using multiple registrations of the Session Initiation Protocol of the present invention.

【主要组件符号说明】[Description of main component symbols]

100、300、500通讯系统100, 300, 500 communication system

110 IP PBX110 IP PBX

120、320NAT服务器120, 320NAT server

125、325路由表125, 325 routing table

130、330、530中继服务器130, 330, 530 relay server

135、335、535记录表135, 335, 535 record form

138、338、538拨号表138, 338, 538 dial table

140、340、540SIP服务器140, 340, 540SIP server

150、350、550具有LDAP的服务器150, 350, 550 servers with LDAP

160、360、560被叫号码端160, 360, 560 called number terminal

200、400、600通讯方法200, 400, 600 communication methods

310VoIP310VoIP

315VoIP网关器315VoIP Gateway

510客户端510 client

S210、S220、S225、S230、S235、S240、S250、S255步骤S210, S220, S225, S230, S235, S240, S250, S255 steps

S260、S270、S280、S290、S410、S420、S425、S430步骤S260, S270, S280, S290, S410, S420, S425, S430 steps

S435、S440、S450、S455、S460、S470、S480、S490步骤S435, S440, S450, S455, S460, S470, S480, S490 steps

S610、S620、S625、S630、S635、S640、S650、S655步骤S610, S620, S625, S630, S635, S640, S650, S655 steps

S660、S670、S680、S690步骤S660, S670, S680, S690 steps

具体实施方式 Detailed ways

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。The implementation of the present invention is described below through specific examples, and those skilled in the art can easily understand other advantages and effects of the present invention from the content disclosed in this specification.

第一实施例:First embodiment:

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。Please refer to FIG. 1 , which is a system architecture diagram of a first embodiment of a communication system 100 using SIP multiple registration according to the present invention.

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。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。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。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。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。As shown in Figure 1, the multiple registration communication system 100 architecture of the present invention using session initiation protocol is on the Internet, including IP private branch exchange (hereinafter referred to as IP PBX) 110, NAT server 120, relay server 130, a plurality of SIP servers 140. Wherein, a plurality of SIP servers 140 may be multimedia communication servers (Multimedia Communication Server), but not limited thereto, the relay server 130 has a recording table 135 for recording the communication data between the SIP server 140 and the IP PBX 110, This includes, but is not limited to, communication time. The relay server 130 also has a dialing table (telephony table) 138 for recording the correspondence between the dialing numbers of the SIP server 140 and the IP PBX 110. NAT server 120 has routing table (routing table) 125, in order to record the address and port before NAT server 120 conversion and the address and port after NAT server 120 conversion. In addition, the number of IP PBX 110 and SIP server 140 in this embodiment is 2, but this is only for illustration. In different embodiments, the number of IP PBX 110 and SIP server 140 is not limited to 2 .

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。中继服务器130通过NAT服务器120与IPPBX 110连接。多个SIP服务器140则与中继服务器130连接。In the system 100 of the present invention, the IP PBX 110 is connected to the NAT server 120, and the NAT server 120 can convert the input virtual network address and port into a physical network address and port, and convert the input virtual network address and port and the converted The physical network addresses and ports of the physical network are stored in the routing table 125 . The relay server 130 is connected to the IPPBX 110 through the NAT server 120. A plurality of SIP servers 140 are connected to the relay server 130 .

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。In addition, in the system 100 of the present invention, also can optionally include the server (hereinafter referred to as the server with LDAP) 150 with Lightweight Directory Access Protocol (Lightweight Directory Access Protocol, LDAP), it is connected with relay server 130, with Manage account and password.

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。Furthermore, in the system 100 of the present invention, the called number terminal 160 is also included, which is connected to the SIP server 140 to transmit the communication packet. The connection relationship between the called number terminal 160 and the SIP server 140 in this embodiment For illustrative purposes only, in different embodiments, the called number terminal 160 can be connected to other SIP servers 140 .

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。Please refer to Fig. 2, which is a flow chart of the first embodiment of the communication method 200 using the multiple registration of the session initiation protocol according to the present invention, wherein, the IP PBX 110, the relay server 130, and the SIP server 140 perform the following configurations step.

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。接着进至步骤S220。As shown in Figure 2, in step S210, provide IP PBX 110, relay server 130 and a plurality of SIP servers 140 on the Internet, wherein, relay server 130 is connected with a plurality of SIP servers 140, and through NAT server 120 and IP PBX 110 connection. Then proceed to step S220.

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。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。In step S220, the relay server 130 sets the backbone with the IP PBX 110, and registers with a plurality of SIP servers 140, wherein the plurality of SIP servers 140 check the registered account number and/or password, and determine whether to allow The result of this registration is transmitted to the relay server 130 . If it is allowed, then send the permission registration request, and go to step S225; if not, send the denial registration request, and end this procedure.

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。若有,则进至步骤S230;若无,则持续执行本步骤S225。In step S225 , the relay server 130 will listen to whether there is a communication request to be sent to the relay server 130 . If yes, proceed to step S230; if not, continue to execute step S225.

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。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。接着进至步骤S235。In step S230, when the IP PBX 110 uses SIP to transmit the communication request to the relay server 130 through the NAT server 120, the relay server 130 uses the dial table 138 to select at least one of the plurality of SIP servers 140, preferably, The relay server 130 selects at least one of the plurality of SIP servers 140 according to the correspondence between the SIP server 140 in the dial table 138 and the dialing number of the IP PBX 110. In addition, the relay server 130 changes the package content of the SIP. Preferably, the change of the package content of the SIP is to change the source of the SIP header (header) in the package content from the address and port before the conversion by the NAT server 120 to the The address and port of the server 130. Then proceed to step S235.

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。接着进至步骤S240。In step S235 , the relay server 130 transmits the communication request to the selected SIP server 140 . Then proceed to step S240.

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。接着进至步骤S250。In step S240, the SIP server 140 checks the content of the SIP packet, wherein checking the content of the SIP packet includes checking the address and port, account, domain of the SIP, called number, and/or the maximum number of simultaneous calls. Then proceed to step S250.

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。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。In step S250, the SIP server 140 judges whether to allow the communication request according to the inspection result, and after confirming that the communication status of the called number terminal 160 is normal, the result of whether to allow the communication request is sent to the IP PBX via the relay server 130 110, wherein, when the SIP server 140 uses SIP to transmit the result of the communication request to the IP PBX 110 via the relay server 130, the relay server 130 changes the package content of the SIP, preferably, the package content of the changed SIP is the The source of the SIP header in the packet content is changed from the address and port of the SIP server 140 to the address and port before being translated by the NAT server 120 . If the communication request is allowed, go to step S260; if not, go to step S255.

在步骤S255中,SIP服务器140通过中继服务器130响应IP PBX110不允许该通讯要求,并结束该通讯要求,接着回到步骤S225。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。In step S255, SIP server 140 responds that IP PBX 110 does not allow this communication request by relay server 130, and ends this communication request, then returns to step S225. In addition, in different embodiments of the present invention, after the communication request is terminated, the program can also be selectively terminated directly.

在步骤S260中,SIP服务器140通过中继服务器130响应IP PBX110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。接着进至步骤S270。In step S260, the SIP server 140 responds to the result that the IP PBX 110 allows the communication request through the relay server 130, and the relay server 130 establishes a communication channel with the IP PBX 110, and the relay server 130 selects to use the account number of the corresponding SIP server 140 and Establish a communication channel with the SIP server 140 to transmit the communication packet to the called number terminal 160 connected to the corresponding SIP server 140, and the relay server 130 records communication data such as the time of establishing the communication channel to further authenticate and manage IP PBX 110. Then proceed to step S270.

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-timeTransfer Protocol,RTP)的地址与端口。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。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。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。接着进至步骤S280。In step S270, when the IP PBX 110 transmits the communication packet to the relay server 130, the relay server 130 records the address and port of the Real-time Transfer Protocol (RTP) used by the IP PBX 110. On the other hand, the relay server 130 sends a re-invite request to the IP PBX 110, and changes the address and port of the RTP used by the IP PBX 110, so that the IP PBX 110 communicates directly with the SIP server 140. When the SIP server 140 transmits the communication packet to the relay server 130 , the relay server 130 records the address and port of the RTP used by the SIP server 140 . On the other hand, the relay server 130 sends a re-invitation request to the SIP server 140, and changes the address and port of the RTP used by the SIP server 140, so that the IP PBX 110 communicates directly with the SIP server 140. Then go to step S280.

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。接着进至步骤S290。In step S280, when the IP PBX 110 ends the communication with the SIP server 140, the IPPBX 110 sends a communication end request to the relay server 130, and the relay server 130 records communication data such as the time when the communication channel ends for further authentication and management IP PBX 110. Then proceed to step S290.

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。In step S290, the relay server 130 transmits the end communication request to the SIP server 140 and ends the communication channel, and processes the communication data of establishing the communication channel and ending the communication channel to authenticate and manage the IP PBX 110. For example, it can be used to calculate the time of establishing the communication channel and the time of ending the communication channel, so as to calculate the communication fee, etc., but it is not limited thereto.

第二实施例:Second embodiment:

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。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 PBX。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。Please refer to FIG. 3 , which is a system architecture diagram of a second embodiment of a communication system 300 using SIP multiple registration according to the present invention. The main difference between this embodiment and the first embodiment is that this embodiment replaces the IP PBX of the first embodiment with VoIP and VoIP gateway. In this embodiment, the main application environment and steps are the same as those in the first embodiment, so the same parts will not be repeated here.

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。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。多个SIP服务器340与中继服务器330连接。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。As shown in FIG. 3 , the system 300 of multiple registrations using the Session Initiation Protocol of the present invention is built on the Internet, including VoIP (VoIP) 310, VoIP gateway 315, NAT server 320, relay server 330, and multiple SIP servers 340, wherein, VoIP 310 is connected with VoIP gateway 315, and VoIP gateway 315 is connected with NAT server 320, and NAT server 320 can convert the input virtual network address and port into physical network address and port, and convert the input virtual network address and port The network address and port and the translated physical network address and port are stored in the routing table 325 . The relay server 330 is connected to the VoIP gateway 315 through the NAT server 320 , and the relay server 330 has a record table 335 and a dial table 338 . A plurality of SIP servers 340 are connected to the relay server 330 . In addition, the numbers of the VoIP 310, the VoIP gateway 315 and the SIP server 340 in this embodiment are all examples, and in different embodiments of the present invention, the numbers of the VoIP 310, the VoIP gateway 315 and the SIP server 340 are not the same This is the limit.

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。In addition, in the system 300 of the present invention, an LDAP server 350 may optionally be included to connect with the relay server 330 for account and password management.

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。Furthermore, in the system 300 of the present invention, the called number terminal 360 may optionally be included, and the called number terminal 360 is connected to the SIP server 340 to transmit communication packets. In this embodiment, the called number terminal The connection relationship between 360 and the SIP server 340 is only for illustration. In different embodiments of the present invention, the called number terminal 360 can be connected to other SIP servers 340 .

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。Please refer to FIG. 4 , which is a flowchart of a second embodiment of a communication method 400 using multiple registrations of the Session Initiation Protocol according to the present invention, wherein the VoIP gateway 315, the relay server 330, and the SIP server 340 are configured in a configuration manner. Follow the steps below.

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。接着进至步骤S420。As shown in Figure 4, in step S410, VoIP 310, VoIP gateway device 315, relay server 330 and a plurality of SIP servers 340 are provided on the Internet, wherein, VoIP 310 is connected with VoIP gateway device 315, and relay server 330 It is connected to a plurality of SIP servers 340 and is connected to a VoIP gateway 315 through a NAT server 320 . Then go to step S420.

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。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。In step S420, the VoIP gateway 315 registers with the relay server 330, and the relay server 330 registers with a plurality of SIP servers 340, wherein the plurality of SIP servers 340 check the registered account number and/or password, and determine whether to allow The result of this registration is transmitted to the relay server 330 . If yes, then send permission to register, and go to step S425; if not, send reject registration request, and end this procedure.

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。若有,则进至步骤S430;若无,则持续执行本步骤S425。In step S425 , the relay server 330 monitors whether there is a communication request sent to the relay server 330 . If yes, proceed to step S430; if not, continue to execute step S425.

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。接着进至步骤S435。In step S430, when the VoIP gateway 315 uses SIP to transmit the communication request to the relay server 330 through the NAT server 320, the relay server 330 uses the dial table 338 to select at least one of the plurality of SIP servers 340, preferably , the relay server 330 selects at least one of the plurality of SIP servers 340 according to the corresponding relationship between the SIP server 340 in the dial table 338 and the dialing number of the VoIP gateway 315; in addition, the relay server 330 changes the SIP Packet content, preferably, changing the SIP packet content is to change the source of the SIP header in the packet content from the address and port before being converted by the NAT server 320 to the address and port of the relay server 330 . Then proceed to step S435.

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。接着进至步骤S440。In step S435 , the relay server 330 transmits the communication request to the selected SIP server 340 . Then proceed to step S440.

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。接着进至步骤S450。In step S440, the SIP server 340 checks the packet content of the SIP, wherein checking the packet content of the SIP includes checking address and port, account, domain of the SIP, called number and/or maximum number of simultaneous calls, etc. Then proceed to step S450.

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。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。In step S450, the SIP server 340 judges whether to allow the communication request based on the inspection result, and after confirming that the communication status of the called number terminal 360 is normal, sends the result of whether the communication request is allowed to the VoIP via the relay server 330 Gateway 315, wherein, when the SIP server 340 uses SIP to transmit the result of the communication request to the VoIP gateway 315 via the relay server 330, the relay server 330 changes the package content of the SIP, preferably, the package content of the changed SIP The source of the SIP header in the packet content is changed from the address and port of the SIP server 340 to the address and port before being translated by the NAT server 320 . If the communication request is allowed, go to step S460; if the communication request is not allowed, go to step S455.

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。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。In step S455, the SIP server 340 responds through the relay server 330 that the VoIP gateway 315 does not allow the communication request, and ends the communication request, and then returns to step S425. In addition, in different embodiments of the present invention, after the communication request is terminated, the program can also be selectively terminated directly.

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。接着进至步骤S470。In step S460, the SIP server 340 responds to the result that the VoIP gateway 315 allows the communication request through the relay server 330, and the relay server 330 establishes a communication channel with the VoIP gateway 315, and the relay server 330 selects to use the corresponding SIP server 340's account number and establishes a communication channel with the SIP server 340 to transmit the communication packet to the called number terminal 360 connected to the corresponding SIP server 340, and the relay server 330 records communication data such as the time of establishing the communication channel for further Authenticate and manage the VoIP gateway 315 . Then proceed to step S470.

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。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。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。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。接着进至步骤S480。In step S470 , when the VoIP gateway 315 transmits the communication packet to the relay server 330 , the relay server 330 records the address and port of the RTP used by the VoIP gateway 315 . On the other hand, the relay server 330 sends a re-invitation request to the VoIP gateway 315 and changes the RTP address and port used by the VoIP gateway 315 so that the VoIP gateway 315 communicates directly with the SIP server 340 . When the SIP server 340 transmits the communication packet to the relay server 330 , the relay server 330 records the address and port of the RTP used by the SIP server 340 . On the other hand, the relay server 330 sends a re-invitation request to the SIP server 340 , and changes the RTP address and port used by the SIP server 340 so that the VoIP gateway 315 communicates directly with the SIP server 340 . Then go to step S480.

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。接着进至步骤S490。In step S480, when the VoIP gateway 315 ends the communication with the SIP server 340, the VoIP gateway 315 transmits a communication termination request to the relay server 330, and the relay server 330 records the communication data such as the time of ending the communication channel for further Authenticate and manage the VoIP gateway 315 . Then go to step S490.

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。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。In step S490 , the relay server 330 sends the end communication request to the SIP server 340 , ends the communication channel, and processes the communication data of establishing the communication channel and ending the communication channel to authenticate and manage the VoIP gateway 315 . The processing can be, for example, calculating the time of establishing the communication channel and the time of terminating the communication channel, so as to calculate the communication fee, etc., but it is not limited thereto.

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。In the above-mentioned embodiment, the IP PBX and the VoIP gateway can be collectively referred to as the client, and the backbone between the relay server setting and the IP PBX and the VoIP gateway registering with the relay server can be collectively referred to as the relay server establishing and connection between clients.

第三实施例:Third embodiment:

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。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。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。Please refer to FIG. 5 , which is a system architecture diagram of a third embodiment of a communication system 500 using SIP multiple registration according to the present invention. The main difference between this embodiment and the first and second embodiments is that this embodiment does not have a NAT server and routing table. In this embodiment, the main application environment and steps are the same as those in the first and second embodiments, so the same parts will not be repeated here.

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。多个SIP服务器540与中继服务器530连接。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。As shown in FIG. 5 , the system 500 of multiple registration using the session initiation protocol of the present invention is built on the Internet, including a relay server 530 and a plurality of SIP servers 540, wherein the relay server 530 is connected to the client 510, and the The relay server 530 has a recording table 535 and a dialing table 538 . A plurality of SIP servers 540 are connected to the relay server 530 . In addition, the numbers of the clients 510 and the SIP servers 540 in this embodiment are just examples, and in different embodiments of the present invention, the numbers of the clients 510 and the SIP servers 540 are not limited thereto.

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。In addition, the system 500 of the present invention may optionally include a server 550 with LDAP, and the server 550 with LDAP is connected to the relay server 530 for account and password management.

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。Furthermore, in the system 500 of the present invention, the called number terminal 560 may optionally be included to connect with the SIP server 540 to transmit communication packets. In this embodiment, the called number terminal 560 and the SIP server 540 The connection relationship is only for illustration, and in different embodiments of the present invention, the called number terminal 560 can be connected with other SIP servers 540 .

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。Please refer to FIG. 6 , which is a flowchart of a third embodiment of a communication method 600 using multiple registrations of the Session Initiation Protocol according to the present invention, wherein the client 510, the relay server 530, and the SIP server 540 perform the following configurations: step.

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。接着进至步骤S620。As shown in FIG. 6 , in step S610 , a relay server 530 and multiple SIP servers 540 are provided on the Internet, wherein the relay server 530 is connected to the client 510 and the multiple SIP servers 540 respectively. Then go to step S620.

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。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。In step S620, the relay server 530 establishes a connection with the client 510, and the relay server 530 registers with a plurality of SIP servers 540, wherein the plurality of SIP servers 540 check the registered account number and/or password, And the result of whether to allow the registration is sent to the relay server 530 . If allowed, then send permission to register, and proceed to step S625; if not allowed, then send reject registration request, and end this procedure.

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。若有,则进至步骤S630;若无,则持续执行本步骤S625。In step S625 , the relay server 530 monitors whether there is a communication request sent to the relay server 530 . If yes, proceed to step S630; if not, continue to execute step S625.

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。接着进至步骤S635。In step S630, when the client 510 uses SIP to transmit the communication request to the relay server 530, the relay server 530 uses the dial table 538 to select at least one of the plurality of SIP servers 540, preferably, the relay server 530 At least one of the plurality of SIP servers 540 is selected according to the corresponding relationship between the SIP server 540 in the dial table 538 and the dialing number of the client 510 . Then proceed to step S635.

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。接着进至步骤S640。In step S635 , the relay server 530 transmits the communication request to the selected SIP server 540 . Then go to step S640.

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。接着进至步骤S650。In step S640, the SIP server 540 checks the content of the SIP packet, wherein checking the content of the SIP packet includes checking the address and port, account, domain of the SIP, called number and/or the maximum number of simultaneous calls. Then proceed to step S650.

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。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。In step S650, the SIP server 540 judges whether to allow the communication request based on the inspection result, and after confirming that the communication status of the called number terminal 560 is normal, the result of whether to allow the communication request is sent to the client via the relay server 530 Terminal 510. If the communication request is allowed, go to step S660; if the communication request is not allowed, go to step S655.

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。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。In step S655, the SIP server 540 responds to the client 510 through the relay server 530 that the communication request is not allowed, and ends the communication request, and then returns to step S625. In addition, in different embodiments of the present invention, after the communication request is terminated, the program can also be selectively terminated directly.

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。接着进至步骤S670。In step S660, the SIP server 540 responds to the client 510 through the relay server 530 to allow the communication request, and the relay server 530 establishes a communication channel with the client 510, and the relay server 530 selects to use the corresponding SIP server 540. account and establish a communication channel with the SIP server 540 to transmit the communication packet to the called number terminal 560 connected to the corresponding SIP server 540, and the relay server 530 records communication data such as the time of establishing the communication channel to further authenticate and Management client 510 . Then proceed to step S670.

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。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。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。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。接着进至步骤S680。In step S670 , when the client 510 transmits the communication packet to the relay server 530 , the relay server 530 records the address and port of the RTP used by the client 510 . On the other hand, the relay server 530 sends a re-invitation request to the client 510 and changes the RTP address and port used by the client 510 so that the client 510 communicates directly with the SIP server 540 . When the SIP server 540 transmits the communication packet to the relay server 530 , the relay server 530 records the address and port of the RTP used by the SIP server 540 . On the other hand, the relay server 530 sends a re-invitation request to the SIP server 540 and changes the RTP address and port used by the SIP server 540 so that the client 510 communicates directly with the SIP server 540 . Then go to step S680.

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。接着进至步骤S690。In step S680, when the client 510 ends the communication with the SIP server 540, the client 510 transmits a communication end request to the relay server 530, and the relay server 530 records the communication data such as the time of ending the communication channel, so as to further authenticate and communicate with the SIP server. Management client 510 . Then go to step S690.

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。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。In step S690 , the relay server 530 sends the end communication request to the SIP server 540 to end the communication channel, and process the communication data of establishing the communication channel and ending the communication channel to authenticate and manage the client 510 . The processing can be, for example, calculating the time of establishing the communication channel and the time of terminating the communication channel, so as to calculate the communication fee, etc., but it is not limited thereto.

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。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。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。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。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。For example, please refer to FIG. 5 again. If the client 510 wants to make an indoor call, such as 0212345678, to the called number terminal 560, then when the communication request is sent to the relay server 530, the relay server 530 uses the number in the dial table 538. The corresponding relationship between the SIP server 540 and the dialing number of the client 510 selects the SIP server 540 with a lower communication fee. Similarly, if the client 510 wants to dial a mobile phone, such as 0912345678, to the called number 560, then when the communication request is sent to the relay server 530, the relay server 530 uses the SIP server 540 in the dial table 538 to contact the client 510 The corresponding relationship between the dialed numbers selects the SIP server 540 with relatively low communication costs. Therefore, the dial table in the relay server can provide a SIP server with lower communication cost for different dial numbers, so as to save the communication cost of the client. In summary, the present invention utilizes the relay server to establish a connection with the client on the one hand, and registers with multiple SIP servers on the other hand, so that the client can communicate with the selected SIP server by selecting at least one of the multiple SIP servers. SIP server direct communication. In this way, the problem of poor compatibility between the SIP server and the client and between SIP servers is solved, and a solution for saving communication costs can be provided for different dial numbers of the client.

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。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。因此,本发明的权利保护范围,应如权利要求书所列。The above-mentioned embodiments are only illustrative to illustrate the principles and functions of the present invention, and are not intended to limit the present invention. Anyone skilled in the art can modify and change the above-mentioned embodiments without departing from the spirit and scope of the present invention. In addition, the number of components in the above embodiments is only for illustration, and is not intended to limit the present invention. Therefore, the protection scope of the present invention should be listed in the claims.

Claims (30)

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:1. A communication method using multiple registrations of the Session Initiation Protocol, characterized in that: 令中继服务器建立与客户端之间的联机;Make the relay server establish a connection with the client; 令该中继服务器向多个会话初始协议服务器注册;causing the relay server to register with a plurality of session initiation protocol servers; 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;causing the client to send a communication request to the relay server using a session initiation protocol; 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causing the relay server to select at least one of the plurality of SIP servers, and transmit the communication request to the selected SIP server; and 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。After the SIP server checks the packet content of the SIP, it judges whether to allow the communication request, and transmits the judgment result to the client via the relay server. 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。2. The communication method using multiple registration of session initiation protocol according to claim 1, characterized in that, the client is built on the Internet, the relay server is built on the Internet and connected to the client, and the multiple A session initiation protocol server is built on the Internet and connected with the relay server. 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。3. The communication method using multiple registrations of SIP as claimed in claim 1, wherein the relay server uses a dial table to select at least one of the plurality of SIP servers. 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。4. The communication method using multiple registrations of SIP according to claim 3, characterized in that, the relay server selects according to the corresponding relationship between the SIP server and the dialing number of the client in the dialing table. At least one of the plurality of SIP servers. 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。5 . The communication method of multiple registrations using SIP as claimed in claim 1 , wherein the client uses the SIP to transmit the communication request to the relay server through the NAT 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。6. The communication method of multiple registrations using the Session Initiation Protocol according to claim 5, wherein when the client uses the Session Initiation Protocol to transmit the communication request to the relay server through the NAT server, Instructing the relay server to change the packet content of the SIP. 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 for multiple registrations using the Session Initiation Protocol according to claim 6, wherein the change of the packet content of the Session Initiation Protocol is to obtain the source of the header of the Session Initiation Protocol in the packet content from the The address and port before conversion by the NAT server are changed to the address and port of the relay server. 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:8. The communication method for multiple registrations using the Session Initiation Protocol according to claim 1, further comprising: 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。When the SIP server allows the communication request, the SIP server responds to the client through the relay server to allow the communication request, and the relay server establishes a communication channel with the client, and The relay server is made to choose to use the account corresponding to the SIP server and establish a communication channel with the SIP server. 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:9. The communication method for multiple registrations using the Session Initiation Protocol according to claim 8, further comprising: 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;When the client ends the communication with the SIP server, the client sends a communication end request to the relay server; 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causing the relay server to send the end communication request to the SIP server; and 令该中继服务器结束该通讯信道。Instruct the relay server to end the communication channel. 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 communication method using multiple registrations of the Session Initiation Protocol according to claim 9 , wherein the relay server is made to record the communication data of establishing the communication channel and terminating 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 communication method using multiple registrations of the Session Initiation Protocol according to claim 10, wherein the communication data is communication time. 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:12. The communication method for multiple registrations using the Session Initiation Protocol according to claim 1, further comprising: 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。When the SIP server does not allow the result of the communication request, the SIP server responds to the client through the relay server that the communication request is not allowed, and ends the communication request. 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。13. The communication method using multiple registration of SIP according to claim 1, characterized in that, when the relay server registers with the SIP server, the SIP server is ordered to check the registered account number and /or password, and whether to allow the result of the registration to be sent to the relay server. 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 communication method using multiple registration of SIP according to claim 1, characterized in that, the SIP server checks the package content of the SIP including checking the address and port, account number, and the SIP Domain, called number and/or maximum number of simultaneous calls. 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:15. The communication method for multiple registrations using the Session Initiation Protocol according to claim 14, further comprising: 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When the client transmits a communication packet to the relay server, causing the relay server to record the address and port of the real-time transport protocol used by the client; and 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。Make the relay server send a re-invitation request to the client, and change the address and port of the real-time transport protocol used by the client, so that the client communicates directly with the SIP server. 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:16. The communication method for multiple registrations using the Session Initiation Protocol according to claim 15, further comprising: 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When the SIP server transmits the communication packet to the relay server, causing the relay server to record the address and port of the RTP used by the SIP server; and 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。The relay server sends a re-invitation request to the SIP server, and changes the RTP address and port used by the SIP server, so that the client communicates directly with the SIP server. 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 communication method using multiple registrations of the Session Initiation Protocol according to claim 1, wherein the client is an Internet telephony gateway and/or an IP subscriber exchange. 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 communication method using multiple registrations of SIP according to claim 17, characterized in that, when the client is an Internet telephony gateway, the relay server establishes a connection with the client by ordering The client registers with the relay server. 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 communication method using multiple registrations of the Session Initiation Protocol according to claim 17, wherein when the client is an IP PBX, the relay server establishes a connection with the client to make the The relay server sets up the backbone with the client. 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 communication method using multiple registration of SIP as claimed in claim 1, wherein the SIP server is a multimedia communication server. 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:21. A communication system using multiple registrations of session initiation protocol, characterized in that it comprises: 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The relay server is built on the Internet and connected with the client through the Internet; and 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,A plurality of session initiation protocol servers are constructed on the Internet and connected to the relay server, 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。Wherein, the relay server establishes the connection with the client through the configuration method, and the relay server registers with the plurality of session initiation protocol servers through the configuration method, and the client uses the configuration method to use The session initiation protocol transmits the communication request to the relay server, the relay server selects at least one of the plurality of session initiation protocol servers and transmits the communication request to the selected session initiation protocol server, and the session initiation protocol server After checking the package content of the session initiation protocol through the configuration method, it is judged whether to allow the communication request, and the judgment result is sent to the client terminal via the relay server. 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:22. The communication system using multiple registration of session initiation protocol according to claim 21, further comprising: 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。A network address translation server, configured on the Internet and connected to the client, and connected to the relay server, wherein the client uses the session initiation protocol to pass the communication request through the network address translation server in a configuration manner sent to the relay server. 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 multi-registration communication system using Session Initiation Protocol according to claim 21, wherein the client is at least one of a VoIP gateway and/or an IP subscriber exchange. 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 communication system using multiple registrations of SIP as claimed in claim 21, wherein the SIP server is a multimedia communication server. 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 communication system using multiple registrations of the Session Initiation Protocol according to claim 22, wherein the relay server changes the packet content of the Session Initiation Protocol through configuration. 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 communication system using multiple registrations of the Session Initiation Protocol according to claim 25, wherein the relay server changes the packet content of the Session Initiation Protocol through a configuration method, such that the packet content in the packet content The source of the SIP header is changed from the address and port before being translated by the NAT server to the address and port of the relay server. 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:27. The communication system using multiple registrations of the Session Initiation Protocol according to claim 21, further comprising: 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。A server with a light directory access protocol is built on the Internet and connected to the relay server to manage account numbers and passwords. 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 communication system using multiple registrations of SIP as claimed in claim 21, wherein the relay server has a recording table for recording communication data between the client and the SIP server. 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 communication system using multiple registrations of SIP according to claim 28, wherein the recording table is used to record the communication time between the client and the SIP server. 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 communication system using multiple registrations of SIP according to claim 21, wherein the relay server has a dial table for recording the corresponding relationship between the SIP server and the dialing numbers of the client .
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Application publication date: 20120411