CN102413111A - Communication method and system utilizing session initial protocol - Google Patents

Communication method and system utilizing session initial protocol Download PDF

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Publication number
CN102413111A
CN102413111A CN 201010294151 CN201010294151A CN102413111A CN 102413111 A CN102413111 A CN 102413111A CN 201010294151 CN201010294151 CN 201010294151 CN 201010294151 A CN201010294151 A CN 201010294151A CN 102413111 A CN102413111 A CN 102413111A
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server
sip
communication
client
session initiation
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CN 201010294151
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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 communication method and system utilizing a session initial 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 through an NAT (network address translation) server; and 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 that the client and the SIP servers are communicated directly further. So, the client can be certified and managed, and the problem of poor compatibility of the SIP servers and the client can be solved.

Description

使用会话初始协议的通讯方法与系统 Communication method and system using the Session Initiation Protocol

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,特别指涉及一种在网络位置转换环境下使用会话初始协议的通讯方法与系统。 [0001] The present invention relates to a communication method and system using a Session Initiation Protocol, communication refers particularly relates to a method and system in the Session Initiation Protocol network location change of surroundings.

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,目前最普及的技术的便是网络电话(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)。 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 session Initiation protocol (session Initiation protocol, SIP). 此外,另有一种设备,IP用户交换机(IP PBX),可利用数字信号在因特网上直接进行通讯。 Further, another kind of devices, IP PBX (IP PBX), a digital signal can be used to communicate directly over the Internet.

[0003] 再者,由于因特网的地址有限,通常不是企业内的每台计算机都会具有一个实体网络地址,所以必须利用网络地址转换(Network Address Translation, NAT)技术,简单的说,当在企业内部进行传输与通讯时,利用虚拟网络地址即可进行传输与通讯。 [0003] Furthermore, due to the limited Internet address, usually each computer within the network is not a business has a physical address, it is necessary to use Network Address Translation (Network Address Translation, NAT) technology, simply put, when in-house when transmission and communications, transport and communications can be carried out using the virtual network address. 当要向企业外部传输与通讯时,先利用NAT服务器将虚拟网络地址与端口转换成实体网络地址与端口,再利用该实体网络地址与端口进行传输与通讯。 When companies want to transfer and external communications, first use NAT server translates the virtual network address and port to the physical network address and port, and then use the physical network address and port transport and communications.

[0004] 然而,目前企业所遭遇到的问题是,由于IP PBX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IP PBX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IP PBX设定SIP主干(trunk),造成IP PBX无法利用其具有数字信号的特性直接进行通讯,而必须利用PSTN模块与SIP服务器兼容的VoIP网关器(VoIP gateway)连接才可使用,如此将容易造成语音质量效果不佳以及存在潜在的障碍风险。 [0004] However, the company encountered a problem, because the SIP server and IP PBX telecommunications services provided by the company, compatibility is not high, resulting in some unable to register with the IP PBX is not compatible with the SIP server, or incompatible Main SIP server SIP can not be set (Trunk) and IP PBX, IP PBX can not cause a digital signal using a characteristic having direct communication, but use the PSTN module is compatible with the SIP server VoIP gateway (VoIP gateway) is connected before use , so the effect is likely to cause poor voice quality and the presence of potential obstacles risk. 此外,虽然有些IP PBX可与SIP服务器进行通讯,但是其采用的方式是IP PBX与SIP服务器彼此信任(trust),导致无法针对特定IP PBX进行认证与管理。 In addition, although some IP PBX can communicate with the SIP server, but the way it is used in IP PBX and SIP server trust (trust), leading to not be authenticated and managed for a specific IP PBX. 再者,在NAT环境下的VoIP也会遭遇一些问题,当VoIP通过VoIP网关器向上述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VoIP网关器,造成无法注册,也因此无法针对特定VoIP进行认证与管理。 Moreover, VoIP in NAT environment will encounter some problems, when the VoIP registration requests to the server via a VoIP SIP gateway, NAT server will be due within the enterprise virtual network address into a physical address outside the enterprise network, SIP server registration can not lead to the result of the response to the original VoIP gateway, cause can not be registered, and therefore can not be authenticated and management for specific VoIP.

[0005] 综上所述,现有通讯系统中,如IP PBX或VoIP网关器的客户端由于兼容性不佳或NAT环境的限制,导致客户端无法向SIP服务器注册,且SIP服务器无法对客户端提供认证与管理机制。 [0005] In summary, the existing communications systems, such as client IP PBX or VoIP gateway or NAT compatibility due to poor environmental restrictions, the client can not register with the SIP server, SIP server and the client can not end providing authentication and management mechanism. 因此,极需要一种使用会话初始协议的通讯方法与系统,以解决SIP服务器与客户端兼容性不佳的问题,并能同时对客户端提供认证与管理机制。 Therefore, the very need for a method and system using a communication session initiation protocol to address poor SIP server and client compatibility issues, and can also provide authentication and client management.

发明内容 SUMMARY

[0006] 本发明提供一种在网络位置转换环境下使用会话初始协议的通讯方法与系统,以解决SIP服务器与客户端兼容性不佳的问题,并能同时对客户端提供认证与管理机制。 [0006] The present invention provides a communication method and system using a Session Initiation Protocol network location in the conversion environment, to resolve the SIP server and the client poor compatibility, and can also provide authentication and client management. [0007] 依照本发明的一实施例,提供一种使用会话初始协议的通讯方法,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SIP服务器注册;令该客户端使用SIP将通讯要求通过NAT服务器并经由该中继服务器传送至该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] according to an embodiment of the present invention, there is provided a use of the Session Initiation Protocol communication method, comprising the steps of: establishing connection between the relay server so that the client; enabling the relay server registered to the SIP server; Order the client uses the SIP request and the communication via the relay server transmits to the SIP server via the NAT server; and a rear, enabling the SIP server checks the contents of the SIP packet, and determines whether to permit the communication requirements, and the determination result via the relay server transmits to the client.

[0008] 此外,本发明还提供一种使用会话初始协议的通讯系统,包括:客户端、中继服务器以及SIP服务器,其中,该中继服务器通过NAT服务器与该客户端连接,且与SIP服务器连接;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以向该SIP服务器注册,而该客户端通过组态方式以使用SIP将通讯要求通过NAT 服务器并经由该中继服务器传送至该SIP服务器,并且该SIP服务器通过组态方式以检查该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。 [0008] Further, the present invention also provides a use of the Session Initiation Protocol communication system, comprising: a client, relay server and the SIP server, wherein the relay server connected to the client via the NAT server and the SIP server connection; configuration mode by the relay server to establish a connection between the client and the relay server to register to the SIP server by way of the configuration, and the configuration of the client by way of the communication using the SIP after the request and conveyed through the NAT server to the relay server via the SIP server, the SIP server and configured by way of the SIP to check the contents of the packet, determines whether to permit the communication requirements, and the determination result is transmitted via the relay server to the client.

[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SIP服务器注册,从而使客户端与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 to establish a connection between the server and the client, registered to the SIP server on the other hand, so that the SIP client and server communicate directly. 由此解决SIP服务器与客户端兼容性不佳的问题,并能同时对客户端提供认证与管理机制。 Thereby solving poor SIP server and client compatibility issues, and can also provide authentication and client management.

附图说明 BRIEF DESCRIPTION

[0010] 图1为根据本发明使用会话初始协议的通讯系统的第一实施例的系统架构图; [0010] FIG. 1 is a system architecture diagram of a first embodiment of the communication system using the Session Initiation Protocol of the present invention;

[0011] 图2为根据本发明使用会话初始协议的通讯方法的第一实施例的流程图; [0011] FIG 2 is a flowchart of a first embodiment of a communication method using a session initiation protocol of the present invention;

[0012] 图3为根据本发明使用会话初始协议的通讯系统的第二实施例的系统架构图; [0012] FIG. 3 is a system architecture diagram of a second embodiment of the communication system using the Session Initiation Protocol of the present invention;

[0013] 图4为根据本发明使用会话初始协议的通讯方法的第二实施例的流程图。 [0013] FIG 4 is a flowchart of a second embodiment of the communication method using the Session Initiation Protocol of the present invention.

[0014]【主要组件符号说明】 [0014] The main component symbol DESCRIPTION

[0015] 100、300 通讯系统 [0015] 100, 300 communications system

[0016] 110IP PBX [0016] 110IP PBX

[0017] 120、320NAT 服务器 [0017] 120,320NAT server

[0018] 125、325 路由表 [0018] routing table 125,325

[0019] 130、330中继服务器 [0019] The relay servers 130, 330

[0020] 135、335 记录表 [0020] Table records 135,335

[0021] 140、340SIP 服务器 [0021] 140,340SIP server

[0022] 150,350具有轻型目录访问协议的服务器 [0022] 150, 350 server with Lightweight Directory Access Protocol

[0023] 160、360被叫号码端 [0023] 160, 360, called number end

[0024] 310VoIP [0024] 310VoIP

[0025] 315VoIP 网关器 [0025] 315VoIP Gateway

[0026] 200、400 通讯方法 [0026] 200, 400 communication method

[0027] S210、S220、S221、S230、S240、S250、S251 步骤 [0027] S210, S220, S221, S230, S240, S250, S251 step

[0028] S260、S270、S280、S290、S410、S420、S421 步骤 [0028] S260, S270, S280, S290, S410, S420, S421 step

[0029] S430、S440、S450、S451、S460、S470、S480、S490 步骤 [0029] S430, S440, S450, S451, S460, S470, S480, S490 step

具体实施方式[0030] 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。 DETAILED DESCRIPTION [0030] Hereinafter, an embodiment of the present invention by certain specific embodiments, those skilled in the art may be disclosed in the present specification easily understand other advantages and effects of the present invention.

[0031] 第一实施例: [0031] First Embodiment:

[0032] 请参阅图1,为根据本发明的使用会话初始协议的通讯系统100的第一实施例的系统架构图。 [0032] Referring to FIG. 1, a system architecture diagram of a first embodiment of the present invention using a session initiation protocol communication system 100 according to.

[0033] 如图1所示,本发明的使用会话初始协议的通讯系统100架构在因特网上,包括IP 用户交换机(IP PBX) 110、NAT服务器120、中继服务器130、SIP服务器140。 [0033] As shown in FIG. 1, the present invention uses the session initiation protocol architecture in a communication system on the Internet 100, 110, NAT server including IP PBX (IP PBX) 120, relay server 130, SIP server 140. 其中,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,NAT服务器120具有路由表(routing table) 125,用以记录经NAT服务器转换前的地址与端口与经NAT服务器转换后的地址与端口。 Wherein, the SIP server 140 may be a multimedia communication 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 the communication time but not limited to, the NAT server 120 having a routing table (routing table) 125, for recording address and port before and after conversion by the NAT server address and port and converted by the NAT server. 此外,本实施例中的IP PBX数目为2个,但仅为例示说明,于本发明的不同实施例中,该IP PBX的数目并不以2个为限。 Further, the present embodiment the number of IP PBX as in Example 2, but only illustrative description, different embodiments of the present invention, the number of the IP PBX is not limited to two.

[0034] 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 [0034] 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. SIP服务器140与中继服务器130连接。 SIP server 140 is connected with the relay server 130.

[0035] 此外,在本发明的系统100中,进一步具有轻型目录访问协议(Lightweight Directory Access Protocol, 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。 Server [0035] Further, in system 100 of the invention, further having a Lightweight Directory Access Protocol (Lightweight Directory Access Protocol, LDAP) 150, which is connected with the relay server 130, to manage the account and password.

[0036] 再者,在本发明的系统100中,进一步具有被叫号码端160,其与SIP服务器140连接,以进行通讯封包的传送。 [0036] Further, in system 100 of the invention, the called number having a further end 160, which is connected to the SIP server 140, the communication packet for transmission.

[0037] 请参阅图2,为根据本发明的使用会话初始协议的通讯方法200的第一实施例的流程图,其中,中继服务器130、IP PBX 110、SIP服务器140通过组态方式进行下列步骤。 [0037] Referring to FIG 2, a flowchart of a first embodiment according to the communication method of the present invention using a session initiation protocol 200, wherein the relay server 130, IP PBX 110, SIP server 140 by configuring the following manner step.

[0038] 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SIP服务器140,其中,中继服务器130与SIP服务器140连接,并通过NAT服务器120与IP PBX 110连接。 [0038] 2, at step S210, provides IP PBX 110, relay server 130 and SIP server 140 on the Internet, wherein the relay server 130 connected to the SIP server 140, and server 120 through the IP NAT PBX 110 is connected. 接着进至步骤S220。 Then proceeds to step S220.

[0039] 在步骤S220中,中继服务器130设定与IP PBX 110之间的主干,并向SIP服务器140注册,其中,SIP服务器140检查账号及/或密码,并将是否允许该注册的结果传送至中继服务器130。 [0039] In step S220, the relay server 130 is set between the backbone and the 110 IP PBX, SIP server 140 to register, wherein, the SIP server 140 checks the account and / or password, and whether to permit the registration of the result 130 to the relay server. 若允许,则传送允许注册要求,并进至步骤S221 ;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 S221; if allowed, is transmitted rejected registration requirements, and this routine ends.

[0040] 在步骤S221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。 [0040] In step S221, 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 ;若没有,则持续监听。 If so, the process proceeds to step S230; if not, then continuously monitoring.

[0041] 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,中继服务器130会将通讯要求传送至SIP服务器140,其中,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。 [0041] In step S230, when the IP PBX 110 by using SIP to the communications requirement NAT server 120 transmits to the relay server 130, the relay server 130 will correspond to the SIP server 140 transmits claims, wherein the relay server 130 changes the SIP packet content is, preferably, change the SIP packet to the packet content is the content of the SIP header (header) from the source before the change of address and port NAT server 120 converts the address and port of the relay server 130 . 接着进至步骤S240。 Then proceeds to step S240.

[0042] 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。 [0042] 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.

[0043] 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。 [0043] 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S251。 If required to allow the communication, the flow advances to step S260; if not allow the communication requirements, the process proceeds to step S251.

[0044] 在步骤S251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S221。 [0044] In step S251, the SIP server 140 through the relay server 130 IP PBXllO does not allow the communication requirements, and ends the communication requirements, then returns to step S221 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.

[0045] 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。 [0045] 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 SIP account server 140 and establishes a communication channel to transmit the communication packet to the called number end 160, and a relay server 130 records the time of the establishment of the communication channel, data communication and the like, for further authentication and management of IP PBX 110. 接着进至步骤S270。 Then proceeds to step S270.

[0046] 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。 [0046] 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,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, changing 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,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, the SIP server address and port changes using RTP 140 so that the IP PBX 110 and SIP server 140 communicate directly. 接着进至步骤S280。 Then proceeds to step S280.

[0047] 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。 [0047] 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.

[0048] 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。 [0048] 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 for example be calculated settling time of the communication channel of the communication channel and end time to calculate the cost of communications, but are not limited thereto.

[0049] 第二实施例: [0049] Second Example:

[0050] 请参阅图3,为根据本发明的使用会话初始协议的通讯系统300的第二实施例的系统架构图。 [0050] Please refer to FIG. 3, a system architecture diagram of a second embodiment of the present invention using a session initiation protocol 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 PBX。 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 gateway IP PBX substituted first embodiment. 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。 And in the present embodiment, the main application environment and the same steps as the first embodiment, so the same part of the text is not repeated in the other.

[0051] 如图3所示,本发明的系统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,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。 [0051] As shown in FIG. 3, the system architecture 300 of the present invention over the Internet, Internet telephony (VoIP) 310, VoIP gateway 315, NAT server 320, the relay server 330, SIP server 340, wherein, VoIP 310 and VoIP gateway 315 is connected, and the VoIP gateway server 320 is connected to the NAT 315, the NAT server 320 may be an input port and the virtual network address to be converted into a physical address and a network port, and inputs the virtual network address and port, and after conversion physical network address stored in the routing table 325 port. 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。 NAT server 330 is connected via the relay server 320 and VoIP gateway 315, server 330 and the relay 335 has a recording sheet. SIP服务器340与中继服务器330连接。 SIP server 340 is connected to the relay server 330. 此外,本实施例中的VoIP与VoIP网关器的数目仅为例示说明,于本发明的不同实施例中,该VoIP与VoIP网关器的数目并不以此为限。 Further, in the embodiment the number of VoIP to VoIP gateway according to the present embodiment described are illustrative only, different embodiments of the present invention, the number of VoIP to VoIP gateways are not limited thereto.

[0052] 此外,在本发明的系统300中,进一步具有轻型目录访问协议的服务器350,其与中继服务器330连接,以进行账号与密码的管理。 Server 350 [0052] Further, in the system 300 of the present invention, further having a Lightweight Directory Access Protocol, which is connected with the relay server 330 to manage the account and password.

[0053] 再者,在本发明的系统300中,进一步具有被叫号码端360,其与SIP服务器340 连接,以进行通讯封包的传送。 [0053] Further, in the system 300 of the present invention, the called number having a further end 360 which is connected with the SIP server 340, the communication packet for transmission. 请参阅图4,为根据本发明的使用会话初始协议的通讯方法400的第二实施例的流程图,其中,中继服务器330、VoIP网关器315、SIP服务器340通过组态方式进行下列步骤。 Please refer to FIG. 4, is a flowchart of a second embodiment of the communication method of the present invention is a session initiation protocol 400, wherein the relay server 330, VoIP gateway 315, SIP server 340 configuration mode by the following steps.

[0054] 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SIP服务器340连接,并通过NAT服务器320与VoIP网关器315连接。 [0054] As shown in FIG. 4, in step S410, 310 provided on the Internet VoIP, VoIP gateway 315, a SIP server and the relay server 330; 340, wherein, VoIP gateway 310 and VoIP connection 315, and the relay server 330 connected to the SIP server 340, and server 320 is connected through a NAT gateway 315 and VoIP. 接着进至步骤S420。 Then proceeds to step S420.

[0055] 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SIP 服务器340注册,其中,SIP服务器340检查账号及/或密码,并将是否允许该注册的结果传送至中继服务器330。 [0055] In step S420, VoIP gateway 315 330 on to the relay server, the relay server 330 and the register 340 to the SIP server, wherein, the SIP server 340 checks the account and / or password, and whether to permit the registration of the result 330 to the relay server. 若允许,则传送允许注册,并进至步骤S421 ;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 S421; if allowed, is transmitted rejected registration requirements, and this routine ends.

[0056] 在步骤S421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。 [0056] In step S421, the relay server 330 may monitor whether there claims correspond to the relay server 330. 若有,则进至步骤S430 ;若没有,则持续监听。 If so, the process proceeds to step S430; if not, then continuously monitoring.

[0057] 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,中继服务器330会将通讯要求传送至SIP服务器340,其中,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。 [0057] In step S430, when the VoIP gateway 315 by using SIP to the communications requirement NAT server 330, the relay server 330 will correspond to the SIP request sent to the server 340,320 transmits the relay server, wherein the relay server 330 changing the content of the SIP packets, preferably the content will be changed SIP packet to the source packet header of the 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S440。 Then proceeds to step S440.

[0058] 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。 [0058] 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.

[0059] 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。 [0059] 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S451。 If required to allow the communication, the flow advances to step S460; if not allow the communication requirements, the process proceeds to step S451.

[0060] 在步骤S451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S421。 [0060] In step S451, the 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 S421. 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.

[0061] 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。 [0061] 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 to the called number end 360, and the settling time of the communication channel and other communication data relay 330 records server for further authentication and management of VoIP gateways 315. 接着进至步骤S470。 Then proceeds to step S470.

[0062] 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。 [0062] 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,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, changing the address and port of the VoIP gateway 315 using RTP such that the VoIP gateway 315 and 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,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 the change of 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.

[0063] 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。 [0063] 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.

[0064] 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。 [0064] 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。 It can establish the communication channel, for example, calculate the time and end time of the communication channel, in order to calculate the cost of communications, but are not limited thereto.

[0065] 承前所述,举例而言,在步骤S410中的VoIP网关器315的地址为192. 168. 1. 1, NAT服务器320的地址为10. 254. 254. 1,中继服务器330的地址为61. 219. 12. 36以及SIP 服务器340的地址为203. 66. 96. 148。 [0065] The front bearing, for example, address of VoIP gateway 315 in step S410 is 192. 168. 1.1, NAT server 320 for address 254. 254. 10. 1 of the relay server 330 61. 219. the address is the address of the SIP server 340, and 12.36 to 203. 66. 96.148. 接着进至步骤S420。 Then proceeds to step S420.

[0066] 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SIP 服务器340注册。 [0066] In step S420, VoIP gateway 315 330 on to the relay server and the relay server 330 340 register with the SIP server. 接着进至步骤S421。 Then proceeds to step S421.

[0067] 在步骤S421中,当中继服务器430收到VoIP网关器315使用SIP传送的通讯要求时,则进至步骤S430。 [0067] In step S421, when the relay server 430 receives VoIP gateway 315 uses the transmitted SIP communication requirements, the process proceeds to step S430.

[0068] 在步骤S430中,中继服务器330将SIP的封包内容中的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,也就是将该SIP的封包内容中的该SIP的标头来源从192. 168. 1. 1 : 12345变更为61. 219. 12. 36力4321。 [0068] In step S430, the relay server 330 to the content source of a packet header in the SIP from SIP address and port NAT server before the change by the relay server 320 converts the address of the port 330, i.e. the source SIP headers in the packet content from the SIP 192. 168. 1.1: changed to 61. 219. 12345 4321 12.36 force. 接着进至步骤S440。 Then proceeds to step S440.

[0069] 在步骤S440中,SIP服务器340检查该SIP的封包内容。 [0069] In step S440, SIP server 340 checks the contents of the packet of the SIP. 接着进至步骤S450。 Then proceeds to step S450.

[0070] 在步骤S450中,中继服务器330将该SIP的封包内容中的该SIP的标头来源从SIP 服务器340的地址与端口变更为经NAT服务器320转换前的地址与端口,也就是将该SIP 的封包内容中的该SIP的标头来源从203. 66. 96. 148 :54321变更为192. 168. 1. 1 :123450 接着进至步骤S460。 [0070] In step S450, the header of the source packet content relay server 330 in the SIP SIP address from the SIP server 340 changes the port address and port of the NAT server 320 premenstrual conversion, that is, the contents of the source packet header in the SIP from SIP 203. 66. 96.148: changed to 192. 168. 1.1 54321: 123450 followed by performing step S460.

[0071] 在步骤S460中,SIP服务器340通过中继服务器330响应VoIP网关器315允许该通讯要求的结果。 [0071] In step S460, SIP server 340 through the relay server 330 in response to VoIP gateway 315 allows the result to the communication requirements. 接着进至步骤S470。 Then proceeds to step S470.

[0072] 在步骤S470中,中继服务器330变更VoIP网关器315使用的RTP的地址与端口并变更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,也就是将VoIP网关器315使用的RTP的地址与端口从61. 219. 12. 36 :54321变更为203. 66. 96. 148力4321,并将SIP服务器340使用的RTP的地址与端口从61. 219. 12. 36 : 54321 变更为10. 254. 254. 1 :54321。 [0072] In step S470, the relay server 330 changes the address port for RTP VoIP gateway 315 and is used to change the RTP address and port used by the SIP server 340, VoIP gateway 315 so that the SIP server 340 communicate directly, i.e. the VoIP gateway 315 using an RTP address and port from the 61. 219. 12.36: changed to 203. 66. 54321 4321 96.148 force, and the SIP server 340 using the RTP address and port 61 from 219. 12.36: 54321 10. 254. 254. change 1: 54321. 接着进至步骤S480。 Then proceeds to step S480.

[0073] 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。 [0073] 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. 接着进至步骤S490。 Then proceeds to step S490.

[0074] 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。 [0074] In step S490, the relay server 330 transmits the communication request to the SIP server end 340, and ends the communication channel.

[0075] 在上述的实施例中,该IP PBX与VoIP网关器可统称为客户端,且该中继服务器设定与该IP PBX之间的主干与该VoIP网关器向该中继服务器注册,可统称为该中继服务器建立与该客户端之间的联机。 [0075] In the above embodiments, the IP PBX gateways with VoIP clients may be collectively referred to, and the relay server is set to the relay server and register the IP PBX trunk between the VoIP gateway, that may be collectively referred to establish connection between the relay server and the client.

[0076] 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。 [0076] 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 (25)

  1. 1. 一种使用会话初始协议的通讯方法,其特征在于,包括: 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通过网络地址转换服务器并经由该中继服务器传送至该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。 CLAIMS 1. A communication method using the Session Initiation Protocol, wherein, comprising: a command to establish a connection between the relay server and the client; enabling the relay server registered with the SIP server; enabling the client uses Session Initiation converting the communication protocol required by the network address of the server and transmitted to the SIP server via the relay server; and after enabling the SIP session server checks the contents of the original packet protocol, determines whether to allow the communication requirements, and determines 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通过网络地址转换服务器与该客户端连接;以及该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 use of SIP communication method according to claim 1, wherein: the client schema on the Internet; the relay server on the Internet architecture and the client server connection network address translation ; and 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通讯方法,其特征在于,进一步包括: 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 SIP communication method according to claim 1, characterized in that, further comprising: when the SIP server permits the communication requirements, enabling the response to the Session Initiation Protocol server through the relay server the client allows the result to the communication requirements, and to make the relay server to establish a communication channel with the client terminal, and enabling the relay server choose to use the account to be a session Initiation protocol server and establishes a communication channel with the session Initiation protocol server.
  4. 4.根据权利要求1所述的使用会话初始协议的通讯方法,其特征在于,进一步包括: 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。 The use of 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 relay server In response to the client does not allow the communication requirements, and the end of the communication requirements.
  5. 5.根据权利要求1所述的使用会话初始协议的通讯方法,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。 The use of SIP communication method according to claim 1, wherein, when the relay server to register with the SIP server, enabling the SIP server checks the account and / or password, and whether to allow the registration of the results transmitted to the relay server.
  6. 6.根据权利要求1所述的使用会话初始协议的通讯方法,其特征在于,当该客户端使用会话初始协议将通讯要求通过该网络地址转换服务器并经由该中继服务器传送至该会话初始协议服务器时,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 claim 1 using SIP claim, wherein, when the client uses the session initiation protocol via the communications requirement of the network address translation server and transmitted to the SIP server via the relay server, 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。 The use of SIP communication method according to claim 6, wherein the change of the SIP packet content is the source of the packet header of the content of the Session Initiation Protocol, via the network address from change the address and port for the conversion before the conversion server address and port relay server.
  8. 8.根据权利要求1所述的使用会话初始协议的通讯方法,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。 8. The communication method of claim 1 using the Session Initiation Protocol according to claim, wherein the Session Initiation Protocol server checks the domain of the SIP packet content inspection comprises address and port, account number, the Session Initiation Protocol, called number and / or maximum number of simultaneous calls.
  9. 9.根据权利要求3所述的使用会话初始协议的通讯方法,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 Session Initiation Protocol communication method according to claim 3, characterized in that, further comprising: when the client and server ends the session initiation protocol communications, enabling the client terminal end transmits to the relay communication requirements 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 of claim 9 using the Session Initiation Protocol according to claim, characterized in that to enable 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通讯方法,其特征在于,该通讯数据为通讯时间。 11. The use of the Session Initiation Protocol communication method according to claim 10, wherein the communication data communication time.
  12. 12.根据权利要求3所述的使用会话初始协议的通讯方法,其特征在于,进一步包括: 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。 When the real-time communication client transmits the packet to the relay server, the relay server so that the client uses to record: 12. SIP communication method according to claim 3, characterized in that, further comprising address and port transport protocol; and enabling the relay server to the client and then transfer the invitation requirements, and changes in real-time transport protocol used by the client's address and port, so that the client and the server communicate directly session Initiation protocol .
  13. 13.根据权利要求12所述的使用会话初始协议的通讯方法,其特征在于,进一步包括: 当该会话初始协议服务器传送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。 13. The communication method of claim 12 using the Session Initiation Protocol according to claim, characterized in that, further comprising: when the SIP server transmits the communication packet to the relay server, the relay server so that the recording Session Initiation Protocol real time transport protocol address and port of the server used; and the relay server so that the server transmits to the SIP rE-INVITE request and changes the session initiation protocol real-time transport protocol server address and port used to make the client direct communication with the session Initiation protocol server.
  14. 14.根据权利要求1所述的使用会话初始协议的通讯方法,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。 14. The use of Session Initiation Protocol communication method according to claim 1, wherein the client is a VoIP gateway and / or IP PBX.
  15. 15.根据权利要求14所述的使用会话初始协议的通讯方法,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。 15. The communication method of claim 14 using the Session Initiation Protocol according to claim, wherein, when the client is a VoIP gateway, the relay server establishes a connection between the client and the client is to make Sign up to the relay server.
  16. 16.根据权利要求14所述的使用会话初始协议的通讯方法,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。 16. The use of Session Initiation Protocol communication method according to claim 14, wherein, when the client is a user IP switch, the relay server establishes a connection between the client and the server is to make the relay the trunk is set between the client.
  17. 17.根据权利要求1所述的使用会话初始协议的通讯方法,其特征在于,该会话初始协议服务器为多媒体通讯服务器。 17. The use of Session Initiation Protocol communication method according to claim 1, wherein the Session Initiation Protocol server for multimedia communication server.
  18. 18. 一种使用会话初始协议的通讯系统,其特征在于,包括: 客户端,架构在因特网上; 中继服务器,架构在该因特网上并通过网络地址转换服务器与该客户端连接;以及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以向该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通过该网络地址转换服务器并经由该中继服务器传送至该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。 18. A method of using the Session Initiation Protocol communication system comprising: a client, on the Internet architecture; relay server, the client server architecture connected to the Internet and network address translation; and Session Initiation protocol server architecture 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 to the session by configuring a manner register Initiation protocol server, and the client configuration mode by using SIP via the communication requirements of the network address translation server and transmitted to the SIP server via the relay server, and the server through the SIP configuration after the way to check the contents of the SIP packet, it is determined whether to permit the communication requirements, and the determination result is transmitted to the client via the relay server.
  19. 19.根据权利要求18所述的使用会话初始协议的通讯系统,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。 19. The use of the SIP communication system according to claim 18, wherein the at least one client of the telephone network gateway and / or user IP switch.
  20. 20.根据权利要求18所述的使用会话初始协议的通讯系统,其特征在于,该会话初始协议服务器为多媒体通讯服务器。 20. The use of the SIP communication system according to claim 18, wherein the Session Initiation Protocol server for multimedia communication server.
  21. 21.根据权利要求18至20中任意一项所述的使用会话初始协议的通讯系统,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。 21.18 to 20 using any one of the SIP communication system according to claim, wherein the relay server by configuring the contents of the packet to change the manner Session Initiation Protocol.
  22. 22.根据权利要求21所述的使用会话初始协议的通讯系统,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,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□。 22. The use of the SIP communication system according to claim 21, wherein the relay server by configuring the contents of the packet to change the manner Session Initiation Protocol, the packet content is the Session Initiation Protocol header from a source via the network address translation with port address before the change of address of the server that converts end □ relay server.
  23. 23.根据权利要求18所述的使用会话初始协议的通讯系统,其特征在于,进一步包括: 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。 23. The use of the SIP communication system according to claim 18, characterized in that, further comprising: a server having a Lightweight Directory Access Protocol, the Internet architecture and connected to the relay server to perform username and password management.
  24. 24.根据权利要求18所述的使用会话初始协议的通讯系统,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。 24. The use of the SIP communication system according to claim 18, wherein the relay server having a record table for recording the data communication between the client and the SIP server.
  25. 25.根据权利要求M所述的使用会话初始协议的通讯系统,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。 25. The use of the SIP communication system according to claim M, wherein the recording sheet for recording the communication time between the client and the SIP server.
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Patent Citations (3)

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CN1717913A (en) * 2003-08-06 2006-01-04 松下电器产业株式会社 Relay server, relay server service management method, service providing system, and program
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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