CN100550813C - System and method for multimedia conference of communication between internal and external networks - Google Patents

Abstract

A multimedia conference system and method capable of direct communication between internal and external networks, comprising: more than one network telephone set capable of receiving calls to form an internal network; a calling end located outside the internal network and connected with the internal network via the external network; a network address translation between the calling end and the internal network; a proxy server between the calling end and the internal network, which is logically configured in parallel with the network address conversion to receive the message of the conference establishment protocol sent by the calling end and carry out the connection and transmission of the address to transmit the message to a specific network telephone.

Description

Translated fromChinese

内外网络间通讯的多媒体会议的系统与方法System and method of multimedia conference for communication between internal and external networks

技术领域technical field

本发明关于一种利用互联网通讯的多媒体会议的系统与方法，尤指一种可连接公共网络(Public Network，「外部网络」)与局域网络(Local Area Network，「内部网络」)，使内部网络内的网络话机(Useragent)能对内部网络外的通讯者进行直接通讯的多媒体会议的系统与方法。The present invention relates to a system and method for a multimedia conference using Internet communication, especially to a system and method that can connect a public network (Public Network, "external network") and a local area network (Local Area Network, "internal network"), so that the internal network A system and method for a multimedia conference in which an internal network phone (Useragent) can directly communicate with a communicator outside the internal network.

背景技术Background technique

随着互联网的盛行，在互联网上的应用也呈现多样化的态样，互联网的通讯，从早期的单纯数据传输(Data Transportation)，扩及到利用TCP/IP通讯协议以传输语音资料(Voice)，此即所谓的Voice over IP(VoIP)，User拨出电话后，其模拟的语音讯号即传送到交换机，并经过VoIP网关器(VoIP gateway)将模拟的语音讯号转换为数字的语音讯号，再通过路由器(router)定址传送到通讯另端的VoIP网关器(VoIPgateway)，以将数字语音讯号转为模拟的语音讯号。此种利用互联网来进行语音通讯的应用，不仅从初期的PC(个人计算机)对PC的方式进行通讯，而进展到PC对电话，电话对PC，甚至电话到电话方式来进行。近期，随着数字影像技术的成熟，也成功的利用互联网来做影像讯号(Video)的传递，也使得昔日梦想的视讯沟通得以成真。With the prevalence of the Internet, the applications on the Internet are also diversified. Internet communication has expanded from the early simple data transmission (Data Transportation) to the use of TCP/IP communication protocol to transmit voice data (Voice) , This is the so-called Voice over IP (VoIP). After the user dials out the phone, the analog voice signal is sent to the switch, and the analog voice signal is converted into a digital voice signal by the VoIP gateway (VoIP gateway), and then It is sent to the VoIP gateway (VoIP gateway) at the other end of the communication through the router to convert the digital voice signal into an analog voice signal. This kind of application of using the Internet to carry out voice communication not only communicates from PC (personal computer) to PC in the early stage, but also progresses to PC to phone, phone to PC, and even phone to phone. Recently, with the maturity of digital image technology, the Internet has also been successfully used to transmit image signals (Video), which has also made the dream of video communication come true.

由于互联网通讯，其资料(Data)，语音讯号(Voice)以及影像讯号(Video)可利用共享的TCP/IP封包在互联网上顺畅的同时传输，也因此可将所有互联网的运作与通讯集中在单一的网络架构上进行，以同时进行资料(Data)，语音讯号(Voice)以及影像讯号(Video)的通讯，而不需要分别使用不同的网络系统，因此得以降低整体网络通讯的频宽费率及整体成本。Due to Internet communication, its data (Data), voice signal (Voice) and image signal (Video) can be transmitted smoothly and simultaneously on the Internet using shared TCP/IP packets, so all Internet operations and communications can be concentrated in a single It is carried out on the network structure to simultaneously communicate data (Data), voice signal (Voice) and image signal (Video) without using different network systems respectively, thus reducing the bandwidth rate of the overall network communication and overall cost.

在互联网的通讯中，通讯的双方以特定的IP(Internet Protocal)地址(address)(此指「Public IP」地址)，作为确认通讯双方的来源与目的地的地址，亦即，网络通讯要求每一个通讯端皆需有一个公共IP(Public IP)，以作为网络通讯上辨认目的地址之用，然而，随着企业或单位的内部网络(Intranet)的兴起，每一个内部网络(或称「局域网络」，LAN)内的各个User如皆须有其个别独立的公共IP，将造成互联网所需的公共IP日渐不足的问题，并且，如果内部网络中有20个User(计算机)，而对此20个User提供20个公共IP，也将增加内部网络对外通讯的相关建置成本，因此，RFC 1918协议乃定义了私有IP地址(Private IP address)，以提供给企业或单位内的内部网络自行运用，由于此等私有IP地址，User不须向互联网上游申请，也因此居于内部网络外的互联网并无法看见企业或单位的内部网络User的私有IP地址，简言之，基本上这些User只能和内部网络内的User联机，并无法直接与外部的互联网进行通讯与沟通。In Internet communication, the two parties to the communication use a specific IP (Internet Protocol) address (address) (this refers to the "Public IP" address) as the address to confirm the source and destination of the communication parties, that is, network communication requires each A communication terminal needs to have a public IP (Public IP), which is used to identify the destination address in the network communication. However, with the rise of the internal network (Intranet) of enterprises or units, each internal network (or "local area network" Network", if each User in the LAN) must have its own independent public IP, it will cause the problem that the public IP required by the Internet is increasingly insufficient, and if there are 20 Users (computers) in the internal network, and this 20 users provide 20 public IPs, which will also increase the related construction cost of external communication in the internal network. Therefore, the RFC 1918 protocol defines a private IP address (Private IP address), which is provided to the internal network of the enterprise or unit. Due to these private IP addresses, users do not need to apply to the upstream of the Internet, and therefore the Internet outside the internal network cannot see the private IP addresses of the internal network users of the enterprise or unit. In short, basically these users can only Connect with users in the internal network, and cannot directly communicate with the external Internet.

前述互联网的通讯中，通讯各方必须具有其特定的公共IP地址以进行相互间的通讯与沟通，其主要因互联网的通讯，其关于资料的传送以TCP封包为之，而TCP封包内的文件头信息(Header)里面所包含的来源与目的地的信息(封包传送的起始地址与目的地址)以公共IP为之，因此，互联网上的TCP封包即可依其文件头信息内的公共IP，通过路由器(router)的定址而传送。In the above-mentioned Internet communication, the communication parties must have their specific public IP addresses to communicate with each other, mainly because of the Internet communication, the transmission of data is based on TCP packets, and the files in the TCP packets The source and destination information contained in the header (Header) (the starting address and destination address of the packet transmission) is based on the public IP. Therefore, TCP packets on the Internet can be based on the public IP in the header information of the file. , transmitted through router (router) addressing.

至于内部网络内的User所具有的私有IP(Private IP)，由于公共的互联网无法辨识其所具有的私有IP地址，因此，内部网络内的User如欲与外部的互联网进行连接与通讯，一般的方式，通过「网络地址转换」(Network Address Translation，「NAT」)为之，其作法乃将内部网络内的各User(即client端的计算机)所传送的TCP封包所包含的文件头信息(header)所载的私有IP，加以对映(mapping)与转换为NAT本身的公共IP，使外部的互联网能加以辨认，从而提供内部网络内的各User与外部的互联网进行通讯。As for the private IP (Private IP) of the User in the internal network, since the public Internet cannot identify the private IP address it has, therefore, if the User in the internal network wants to connect and communicate with the external Internet, generally By means of "Network Address Translation" ("NAT"), the method is to convert the file header information (header) contained in the TCP packet transmitted by each User (ie, the computer at the client end) in the internal network to The private IP contained in it is mapped and converted into the public IP of NAT itself, so that the external Internet can be identified, thereby providing each User in the internal network to communicate with the external Internet.

兹以实例说明之，当内部网络内的具有私有IP为192.168.1.100的User要对外联机时：Here is an example to illustrate, when a User with a private IP of 192.168.1.100 in the internal network wants to connect to the outside world:

1.该User的网关器(gateway)设定为NAT主机，所以当要连上互联网时，该TCP封包就会被送到NAT主机，此时TCP封包的文件头信息(Header)上所载的来源地址(source IP)为192.168.1.100；1. The gateway of the User is set as a NAT host, so when connecting to the Internet, the TCP packet will be sent to the NAT host. At this time, the header information (Header) of the TCP packet contains the The source address (source IP) is 192.168.1.100;

2.通过NAT主机，她会将User对外通讯的TCP封包上的文件头信息的source IP(192.168.1.100)转换伪装成NAT主机所具有的公共IP ppp0(以拨接情况为例)，因为该TCP封包上的文件头信息内的资料已被改成公共IP，因此，该User所传送的TCP封包即可连接传送到外部的互联网，同时NAT主机会记忆这个TCP封包是由哪一个来源地址的User(私有IP地址为：192.168.1.100)所传送过来；2. Through the NAT host, she will convert the source IP (192.168.1.100) of the file header information on the TCP packet of User's external communication and disguise it as the public IP ppp0 of the NAT host (take dial-up as an example), because the The data in the file header information on the TCP packet has been changed to the public IP, so the TCP packet sent by the user can be connected to the external Internet, and the NAT host will remember which source address the TCP packet is from User (private IP address: 192.168.1.100);

3.互联网传送回来给User的TCP封包，亦由具有公共IP地址的NAT主机来进行接收，此时，NAT主机即会去查询原本记录的路由信息，并将互联网上传送过来的TCP封包的目标IP(object IP)由NAT主机的公共IP ppp0，改回内部网络内的User所原有的私有IP地址192.168.1.100；3. The TCP packet sent back from the Internet to the User is also received by the NAT host with the public IP address. At this time, the NAT host will query the originally recorded routing information and send the TCP packet sent from the Internet to the destination The IP (object IP) is changed from the public IP ppp0 of the NAT host to the original private IP address 192.168.1.100 of the User in the internal network;

4.最后再由NAT主机将该封包传送给原先发送TCP封包的该User。4. Finally, the NAT host sends the packet to the User who originally sent the TCP packet.

内部网络内的User通过NAT主机转换与媒介，以与外部的互联网进行通讯与沟通，此种方式，对一般数据的通讯尚无问题，然而对SIP通讯协议(Session Initiation Protocal)所提供的多媒体会议通讯，由于NAT或/与防火墙的存在，在依SIP通讯协议进行会议的联机呼叫时，即会因NAT或防火墙的存在而使其呼叫与联机无法通过，而造成内部网络与外部互联网无法进行直接的会议通讯。Users in the internal network communicate with the external Internet through NAT host conversion and media. In this way, there is no problem with general data communication. However, multimedia conferences provided by the SIP communication protocol (Session Initiation Protocol) Communication, due to the existence of NAT or/and firewall, when the online call of the conference is made according to the SIP communication protocol, the call and connection cannot pass due to the existence of NAT or firewall, and the internal network and the external Internet cannot be directly connected. conference communications.

所谓SIP通讯协议，其以应用层(application-layer)的控制(讯号)来建立、调整、终止会议的呼叫，以提供一多媒体会议的通讯协议(multimedia session)，同时，其相关的SDP(Session DecriptionProtocal)通讯协议，则提供SIP会议中的会议信息的描述规定。由于SIP通讯协议所提供的通讯品质极佳，且可提供点对点的直接通讯，并且，SIP通讯协议提供相当完整的信息安全服务，例如，拒绝服务的避免(denial-of-service prevention)、User与User间以及代理服务器与User间的认证(authentication)、整合保护(integrity protection)以及加密(encryption)与隐私(privacy)服务。因此，互联网的通讯由原先采用的H.323协议，已广泛变更为采用SIP通讯协议。The so-called SIP communication protocol uses the control (signal) of the application layer (application-layer) to establish, adjust, and terminate the call of the conference to provide a communication protocol (multimedia session) for a multimedia conference. At the same time, its related SDP (Session DecriptionProtocal) communication protocol, which provides the description of the meeting information in the SIP meeting. Because the communication quality provided by the SIP communication protocol is excellent, and it can provide point-to-point direct communication, and the SIP communication protocol provides quite complete information security services, such as denial-of-service prevention (denial-of-service prevention), User and Authentication, integrity protection, encryption and privacy services between users and between proxy servers and users. Therefore, Internet communication has been widely changed from the original H.323 protocol to the SIP communication protocol.

然而，SIP通讯协议虽可以极佳的通讯品质提供点对点的直接通讯以及相当完整的信息安全服务，然而，在面临内部网络为维护一般数据或通讯的信息安全要求所固有的防火墙或NAT(Network AddressTranslation)时，因为防火墙或NAT的阻隔，致SIP的讯息无法通过以进入到内部网络，使得内外网络间在利用SIP通讯协议以建立会议的呼叫与联机，无法达成点对点的直接通讯。However, although the SIP communication protocol can provide point-to-point direct communication and quite complete information security services with excellent communication quality, however, in the face of the inherent firewall or NAT (Network Address Translation) inherent in the internal network to maintain general data or communication information security requirements ), because of the firewall or NAT barrier, the SIP message cannot pass through to enter the internal network, so that the internal and external networks use the SIP communication protocol to establish conference calls and connections, and point-to-point direct communication cannot be achieved.

同时，局域网络为维护网络通讯的信息安全，在对外的通讯上常另有proxy based的防火墙(Firewall)设定，一般而言，防火墙以进入的信息(incoming traffic)所具有的来源(source)、目的地(destination)以及信息型态(traffic type)等参数来决定是否卡住(block)进入的信息流，因此，对为未经认可(un-trusted public domain)的进入信息(incoming traffic)仅在通讯由经认可的内部网络(trusted privatedomain)发起的情形下始能通过，因此，对防火墙而言，SIP通讯协议所提供的多媒体通讯，由于多以未认可的进入呼叫(incoming call)来进行会议的呼叫并建立联机，而在相互配合的应用上存有相互杆格的问题，虽然防火墙可提供SIP通讯协议所需的多端口的动态开关，然对未获认可而进入的信息流却仍存有安全上的问题，更使得外部的互联网直接与局域网络内的各User进行点对点的直接会议通讯有其难以克服的问题。At the same time, in order to maintain the information security of network communication, the local area network often has a proxy based firewall (Firewall) setting for external communication. Generally speaking, the firewall uses the source (source) of incoming traffic. , destination (destination) and information type (traffic type) and other parameters to determine whether to block (block) the incoming information flow, therefore, for incoming information (incoming traffic) that is not authorized (un-trusted public domain) Only when the communication is initiated by the approved internal network (trusted private domain) can it pass through. Therefore, for the firewall, the multimedia communication provided by the SIP communication protocol is mostly received by an unapproved incoming call (incoming call). Make a conference call and establish a connection, but there is a problem of interoperability in the applications that cooperate with each other. Although the firewall can provide the multi-port dynamic switch required by the SIP communication protocol, it does not accept the information flow that has not been approved. There are still security problems, and it is even more difficult to overcome the problem of point-to-point direct conference communication between the external Internet and the users in the local area network.

为解决NAT或防火墙存于内部网络的介于外部网络的入口处，造成SIP通讯无法在外部网络与内部网络的各User间构筑直接的点对点会议通讯，已知的作法中有利用人工配置(manually configure)的方式，将NAT对外的公共IP的地址与多端口(ports)，通过静态的对映(staticmappings)到内部网络内的各User，亦即，NAT或防火墙服务器上建立公共IP与内部网络内的私有IP的静态对应表(static mapping table)，以此方式达成内部网络的各User与外部网络的通讯，然而此一方法不仅要求有固定的IP地址与多端口，并且，因人工配置牵涉相当专业的配置流程(configuration process)，因此，仅能适合于非常小型的局域网络，而有其不便与不实用之处。另外，2004年7月15日公开的美国专利公开案号US2004/0139230A1「具有NAT的网络的SIP服务方法」(SIP Service Method in a Network Having a NAT)亦以相同的方式为之，由于其必须在NAT或防火墙上进行公共IP与私有IP的对映，而使内部网络的User必须于固定的私有IP设定处参与会议，而无法利用SIP通讯协议所支持的名字对应(name mapping)及再定位(redirection)的服务无法，而失去SIP通讯协议所提供的参与会议者能不受其网络位置限制以保有个人移动性(personal mobility)的优点。In order to solve the problem that NAT or firewall exists at the entrance of the internal network between the external network, causing SIP communication to be unable to construct direct point-to-point conference communication between the users of the external network and the internal network, there is a known method of using manual configuration (manually configure), the NAT external public IP address and multiple ports (ports) are statically mapped (staticmappings) to each User in the internal network, that is, the public IP and the internal network are established on the NAT or firewall server The static mapping table of the private IP in the internal network, in this way, the communication between the users of the internal network and the external network is achieved. However, this method not only requires a fixed IP address and multiple ports, but also involves manual configuration. Quite a professional configuration process (configuration process), therefore, only suitable for very small local area networks, and has its inconvenience and impracticality. In addition, the US Patent Publication No. US2004/0139230A1 "SIP Service Method in a Network Having a NAT" published on July 15, 2004 is also done in the same way, because it must The public IP and private IP are mapped on the NAT or firewall, so that the users of the internal network must participate in the conference at the fixed private IP setting, and cannot use the name mapping and re-enactment supported by the SIP communication protocol. The service of location (redirection) cannot, and loses the advantage of personal mobility (personal mobility) that the participants provided by the SIP communication protocol can not be restricted by their network locations.

基于前述NAT/防火墙的限制，两个企业或单位之间(即两个局域网络之间)如欲进行互联网的通讯，特别是利用互联网来进行会议的情形，多数的情形以该等局域网络所具有的公共IP地址来设定通讯端的服务器(server)，以连接欲进行通讯的两个或两个以上的局域网络，而该企业或单位(局域网络)亦仅能以该通讯端的服务器来进行互联网的会议，简言之，该企业或单位内欲参与会议进行的人员，必须集中在设定通讯服务器的会议室中，才有办法与互联网的另一端进行会议，此种要求一方面无法使欲参与会议的人能随时于其工作岗位上来进行会议，而增加不便与成本，另一方面也使会议的召集更加困难。Based on the limitations of the aforementioned NAT/firewall, if two enterprises or units (that is, between two local area networks) want to communicate on the Internet, especially in the case of using the Internet for conferences, most of the cases are based on the local area network. The public IP address of the communication terminal is used to set the server (server) of the communication terminal to connect two or more local area networks that want to communicate, and the enterprise or unit (local area network) can only use the server of the communication terminal. Internet conferences, in short, the personnel who want to participate in the conference in the enterprise or unit must be concentrated in the conference room where the communication server is set up, so as to have a conference with the other end of the Internet. On the one hand, this requirement cannot be used People who want to participate in the meeting can come to the meeting at their working positions at any time, which increases inconvenience and cost, and on the other hand makes it more difficult to convene the meeting.

这种无法与局域网络内的各User进行点对点的互联网会议通讯，而必须集中于局域网络内的特定处以进行互联网会议的限制，实无法满足现代科技社会的要求。为了达到能使企业或单位内的各User皆可于其工作岗位上直接进行互联网的通讯会议，公开技术，请参图1，在具有NAT或防火墙10的内部网络20，另设一VoIP网关器(Gateway)30，以将由互联网(Internet)传来的网络电话直接通过VoIP网关器(Gateway)30以转成普通电话，再由普通电话所具有的多方通讯会议功能，将其转接并直接连接到各个必须参与会议的人员位置，然而此种方式，由于仅利用VoIP来做会议通讯的联络，因此，一方面，其仅能限制应用于语音(Voice)的通讯，至于利用互联网以进行视讯会议(Video Conference)的需求，则无法以此方式达成，简言之，其仅具备利用互联网的通讯来取代传统电话的通讯联机，而达到降低成本的要求；另一方面，由于其通过VoIP Gateway，来连接互联网(Internet)上的网络电话与单位内的话机，因此，未连入内部网络，也未跟内部的IP整合，而无法达到提供多媒体通讯会议以及与内部网络整合的要求。This kind of limitation that cannot carry out peer-to-peer Internet conference communication with each User in the local area network, but must be concentrated in a specific place in the local area network for Internet conference, really cannot meet the requirements of the modern technological society. In order to enable each User in the enterprise or unit to directly conduct Internet communication conferences on their working posts, and to disclose the technology, please refer to Figure 1. In theinternal network 20 with NAT orfirewall 10, another VoIP gateway is set up. (Gateway) 30, to convert the network phone from the Internet (Internet) directly through the VoIP gateway (Gateway) 30 to an ordinary phone, and then transfer it and directly connect it by the multi-party communication conference function that the ordinary phone has. To the position of each person who must participate in the conference. However, this method only uses VoIP for conference communication. Therefore, on the one hand, it can only be used for voice (Voice) communication. As for the use of the Internet for video conferencing (Video Conference) requirements cannot be achieved in this way. In short, it only has the ability to use Internet communication to replace traditional telephone communication connections to meet the requirements of cost reduction; on the other hand, because it uses VoIP Gateway, To connect the VoIP on the Internet (Internet) and the phone in the unit, therefore, it is not connected to the internal network, nor integrated with the internal IP, so it cannot meet the requirements of providing multimedia communication conferences and integrating with the internal network.

因此，发展一种新颖的多媒体会议系统与方法，以达到内部网络的各User可与外部网络进行直接点对点的多媒体会议通讯，并兼顾NAT与防火墙所提供予内部网络的信息安全，乃是有极大价值的。Therefore, it is very important to develop a novel multimedia conferencing system and method so that each user of the internal network can carry out direct point-to-point multimedia conference communication with the external network, and take into account the information security provided by the NAT and the firewall to the internal network. Great value.

发明内容Contents of the invention

因此，本发明的一目的是提供一种内部网络的User可直接与外部网络的通讯者进行直接通讯的多媒体会议系统与方法。Therefore, an object of the present invention is to provide a multimedia conferencing system and method in which the users of the internal network can directly communicate with the correspondents of the external network.

本发明的次一目的是提供能以原有局域网络配置，既不需人工配置设定的介入，也不需改变内部网络的网络设定，即可达成内部网络的User与外部网络进行直接通讯的多媒体会议系统与方法。The second purpose of the present invention is to provide the original local area network configuration without the intervention of manual configuration settings, and without changing the network settings of the internal network, so that the users of the internal network can communicate directly with the external network Multimedia conferencing system and method.

本发明的再一目的是提供内部网络与外部网络的参与会议者，能不受其网络位置限制而保有个人移动性(personal mobility)的直接通讯的多媒体会议系统与方法。Another object of the present invention is to provide a multimedia conferencing system and method for direct communication between internal network and external network participants without being restricted by their network locations and maintaining personal mobility.

本发明的另一目的提供无防火墙/NAT对内外通讯的干扰，同时能顾及信息安全而达成内部网络的User与外部网络进行直接通讯的多媒体会议系统与方法。Another object of the present invention is to provide a multimedia conferencing system and method without firewall/NAT interference to internal and external communication, and at the same time, allowing users of the internal network to communicate directly with external networks in consideration of information security.

为达上述目的，本发明包括一个以上(能接受呼叫)的网络话机(User agents)，以构成内部网络；一呼叫端，其位于内部网络外，并得经外部网络与内部网络连接；一个网络地址转换(NAT)，其介于该呼叫端与该内部网络间；一个代理服务器，其介于该呼叫端与该内部网络间，而逻辑上的与网络地址转换(NAT)平行配置，以负责接收由呼叫端发出的会议建置协议的讯息，并进行地址的连接与传送，以将该讯息传送到特定的网络话机(User agents)。For reaching above-mentioned object, the present invention comprises more than one network phone (User agents) (can accept call), to constitute internal network; A calling end, it is positioned at the outside of internal network, and must be connected with internal network through external network; A network Address Translation (NAT), which is between the calling end and the internal network; a proxy server, which is between the calling end and the internal network, logically configured in parallel with Network Address Translation (NAT), to be responsible Receive the conference establishment protocol message sent by the calling end, and connect and send the address, so as to send the message to specific network phones (User agents).

为达上述目的，本发明的另一特点为：该代理服务器更包括有两个以上的网络介面(Network Interface)，其中至少一个网络介面连接外部网络，以及至少一个网络介面连接内部网络，以负责接收由呼叫端发出的会议建置协议的讯息，并进行地址的连接与传送，以将该讯息传送到特定的网络话机(User agents)。To achieve the above object, another feature of the present invention is: the proxy server further includes more than two network interfaces (Network Interface), wherein at least one network interface is connected to the external network, and at least one network interface is connected to the internal network, to be responsible for Receive the conference establishment protocol message sent by the calling end, and connect and send the address, so as to send the message to specific network phones (User agents).

为达上述目的，本发明的又一特点为：该代理服务器逻辑上地更包括一个登录服务器(Registration Server)，使该内部网络内的一个以上的网络话机(User agents)可向该登录服务器做URI(Uniform ResourceIndentifier)的登录，并且，该代理服务器可向该登录服务器取得已登录URI的网络话机(User agents)的联络清单(Contact List)，并依URI定位的方式，将呼叫端传送过来的呼叫，转送到个别网络话机(Useragent)。In order to achieve the above object, another feature of the present invention is: the proxy server logically further includes a registration server (Registration Server), so that more than one network phone (User agents) in the internal network can make a registration to the registration server. URI (Uniform ResourceIndentifier) registration, and the proxy server can obtain the contact list (Contact List) of the network phone (User agents) that has registered the URI from the login server, and send the caller from the calling end according to the URI positioning method Calls are forwarded to individual IP phones (Useragent).

为达上述目的，本发明的更一特点为：该代理服务器以虚拟代理话机(Backend to Backend User Agent，B2BUA)方式，作为该呼叫端与网络话机(User agents)间的对话媒介，以将「呼叫端对网络话机(Useragents)」以及「网络话机(User agents)对呼叫端」两个对话(call legs)予以连接，并且，该虚拟代理话机(B2BUA)并建立「呼叫端」以及「网络话机(User agents)」间的RTP转送机制(RTP RELAY)，而以符合RTP(Real Time Transport Protocal)协议的方式，进行即时的多媒体会议通讯。In order to achieve the above purpose, another feature of the present invention is: the proxy server uses a virtual proxy phone (Backend to Backend User Agent, B2BUA) as the dialogue medium between the calling terminal and the network phone (User agents), so that " The caller connects two dialogues (call legs) to the network phone (Useragents) and the "network phone (User agents) to the caller", and the virtual agent phone (B2BUA) establishes the "caller" and "network phone" (User agents)" between the RTP transfer mechanism (RTP RELAY), and in a manner that conforms to the RTP (Real Time Transport Protocol) protocol, for real-time multimedia conference communication.

通过本发明，可解决内部网络因为NAT或防火墙的存在而无法与外部网络直接进行点对点的多媒体会议的问题，并能以原有的内部网络设定在顾及信息安全的需求前提下，建构一个能直接进行点对点的多媒体会议通讯的系统与方法，并使内外网络的各会议参与者，能不受空间的限制，随时、随处的相互进行直接点对点或多点对多点的语音、视讯等多媒体的会议通讯。Through the present invention, the problem that the internal network cannot directly conduct point-to-point multimedia conferences with the external network due to the existence of NAT or firewalls can be solved, and the original internal network setting can be used under the premise of taking into account the requirements of information security. A system and method for direct point-to-point multimedia conference communication, so that the conference participants in the internal and external networks can conduct direct point-to-point or multipoint-to-multipoint voice, video and other multimedia communication with each other anytime and anywhere without being limited by space. Conference Communications.

下节将叙述本发明的其它特征。实施方式中所举的实施例仅是范例而非限制本发明。更进一步，实施例所举的方法、步骤、系统、装置、配置或其它具有可选择性的部分亦不限制本发明。除此，本发明由权利要求范围所定义。Other features of the present invention will be described in the next section. The examples given in the embodiments are only examples and do not limit the present invention. Furthermore, the methods, steps, systems, devices, configurations or other optional parts mentioned in the embodiments do not limit the present invention. Otherwise, the present invention is defined by the scope of the claims.

附图说明Description of drawings

图1为现有技术利用VoIP网关器(Gateway)将互联网(Internet)传来的网络电话转成普通电话，以构成点对点直接通讯的示意图。FIG. 1 is a schematic diagram of using a VoIP gateway (Gateway) to convert Internet calls from the Internet into ordinary calls in the prior art to form point-to-point direct communication.

图2为SIP通讯协议以一代理服务器作呼叫媒介(call mediation)的会议呼叫程序图。FIG. 2 is a diagram of a conference call procedure in which a proxy server is used as a call mediation by the SIP communication protocol.

图3为本发明的包括一NAT及一与NAT平行建置的符合SIP通讯协议要求的代理服务器的实施例图。FIG. 3 is a diagram of an embodiment of the present invention including a NAT and a proxy server built in parallel with the NAT and meeting the requirements of the SIP communication protocol.

图4为本发明前述的实施例中，该符合SIP通讯协议的代理服务器在为内部网络与外部网络间进行多媒体会议呼叫、联机与通讯进行的示意图。4 is a schematic diagram of the proxy server conforming to the SIP communication protocol performing multimedia conference calling, connection and communication between the internal network and the external network in the foregoing embodiment of the present invention.

图5为本发明实施例的符合SIP通讯协议的代理服务器在内部网络对外部网络进行会议呼叫的程序，以构成内外网络多媒体会议联机的说明图。FIG. 5 is an explanatory diagram of a procedure for a proxy server conforming to the SIP communication protocol to make a conference call to an external network in the internal network to form a multimedia conference connection between the internal and external networks according to an embodiment of the present invention.

图6为本发明实施例的符合SIP通讯协议的代理服务器在外部网络对内部网络进行会议呼叫的程序，以构成内外网络多媒体会议联机的说明图。FIG. 6 is an explanatory diagram of a procedure for a proxy server conforming to the SIP communication protocol to make a conference call from an external network to an internal network to form a multimedia conference connection between the internal and external networks according to an embodiment of the present invention.

图中符号说明：Explanation of symbols in the figure:

10呼叫端10 call terminal

20代理服务器20 proxy servers

30被呼叫端30 called end

100内外网络间直接通讯的多媒体会议系统100 multimedia conferencing system for direct communication between internal and external networks

101至105构成内部局域网络的网络话机(User Agents)101 to 105 constitute the network phone (User Agents) of the internal LAN

110内部网络110 internal network

120网络地址转换(NAT)120 Network Address Translation (NAT)

130代理服务器130 proxy servers

140非对称数字用户回路(ADSL)140 Asymmetric Digital Subscriber Loop (ADSL)

150外部网络150 external network

210内部网络210 internal network

211内部网络User A(Internal Computer)211 Internal network User A (Internal Computer)

212内部网络User B(Internal Computer)212 Internal network User B (Internal Computer)

220外部网络220 external network

221外部网络User D221 External network User D

230代理服务器230 proxy server

310内部网络User(Call Agent)310 internal network User (Call Agent)

320外部网络被呼叫端(Called SIP Terminal)320 External network called terminal (Called SIP Terminal)

330SIP服务器330SIP server

340虚拟代理话机(Backend to Backend User agent，B2BUA)340 virtual agent phone (Backend to Backend User agent, B2BUA)

510外部网络呼叫端(Calling SIP Terminal)510 external network calling terminal (Calling SIP Terminal)

520内部网络User(Call Agent)520 internal network User (Call Agent)

530SIP服务器530SIP server

540虚拟代理话机(Backend to Backend User agent，B2BUA)540 virtual agent phone (Backend to Backend User agent, B2BUA)

具体实施方式Detailed ways

图2说明SIP通讯协议以一代理服务器作呼叫媒介(call mediation)的会议呼叫程序图，用以说明SIP通讯协议如何进行多媒体会议的呼叫与联机，如图2所示，其中，呼叫端10发出(transmit)一会议呼叫(session request message)-INVITE给呼叫媒介的代理服务器20(步骤41)，该代理服务器20然后将呼叫INVITE传送到被呼叫端30(步骤42)。Fig. 2 illustrates SIP communication protocol with a proxy server as a conference call program diagram of call mediation (call mediation), in order to illustrate how SIP communication protocol carries out the calling and connection of multimedia conference, as shown in Fig. 2, wherein, callingterminal 10 sends out (transmit) a conference call (session request message)-INVITE to the proxy server 20 (step 41) of the calling medium, and theproxy server 20 then transmits the calling INVITE to the called end 30 (step 42).

然后，被呼叫端30乃传送一讯息(100Trying)给代理服务器20(步骤43)，该代理服务器20然后将该讯息(100Trying)传送给呼叫端10(步骤44)。并且，该被呼叫端30会传送一呼叫讯号(calling signal)(180Ringing)给代理服务器20(步骤45)，该代理服务器20然后将该讯号(180Ringing)传送给呼叫端10(步骤46)，以告知呼叫端10其已接到呼叫。Then, the calledterminal 30 sends a message (100Trying) to the proxy server 20 (step 43), and theproxy server 20 then transmits the message (100Trying) to the calling terminal 10 (step 44). And, the called terminal 30 will send a calling signal (calling signal) (180Ringing) to the proxy server 20 (step 45), and theproxy server 20 will then transmit the signal (180Ringing) to the calling terminal 10 (step 46), to Inform the callingterminal 10 that it has received the call.

之后，被呼叫端30传送一讯息(200OK)给代理服务器20(步骤47)，该代理服务器20然后将该讯息(200OK)传送给呼叫端10(步骤48)，以接受会议的要求；接着，呼叫端10传送确认讯息-ACK给代理服务器20(步骤49)，该代理服务器20然后将ACK传送到被呼叫端30(步骤50)，而建立起会议的联机，在此之后，呼叫端10与被呼叫端20即可进行会议的通讯。Afterwards, the calledterminal 30 sends a message (200 OK) to the proxy server 20 (step 47), and theproxy server 20 then transmits the message (200 OK) to the calling terminal 10 (step 48) to accept the meeting request; then, The callingterminal 10 transmits confirmation message-ACK to the proxy server 20 (step 49), and theproxy server 20 then transmits the ACK to the called terminal 30 (step 50), and sets up the online connection of the conference. After this, the callingterminal 10 and The called terminal 20 can carry out the communication of the meeting.

图3为本发明的一种能进行内外网络间直接通讯的多媒体会议系统的实施例图，如图3所示，本发明的直接通讯的多媒体会议系统100，包括：一个以上(能接受呼叫)的网络话机(User agents)101，102，103，104，105，以构成内部网络110；一呼叫端150，其位于内部网络110外，并得经外部网络与内部网络110连接；一个网络地址转换(NAT)120，其介于该呼叫端150与该内部网络110间；一个代理服务器130，其介于该呼叫端150与该内部网络110间，而逻辑上的与网络地址转换(NAT)120平行配置，以负责接收由呼叫端150发出的会议建置协议的讯息，并进行地址的连接与传送，以将该讯息传送到特定的网络话机(User agents)，并以符合SIP通讯协议的方式进行多媒体会议通讯的呼叫及联机，在完成内外网络间的会议的呼叫与联机后，则以SIP通讯协议定义的「虚拟代理话机」(Backend to Backend Useragent，B2BUA)执行内外网络间(即「内部网络对外部网络」与「外部网络对内部网络」)对话的连接，最后，并以符合RTP(Real TimeTransport Protocal)协议的方式，进行资料的压缩与传送，以达到即时、直接的多媒体通讯。Fig. 3 is the embodiment figure of a kind of multimedia conferencing system capable of direct communication between internal and external networks of the present invention, as shown in Fig. 3, the multimedia conferencing system 100 of direct communication of the present invention comprises: more than one (can accept call) Network phones (User agents) 101, 102, 103, 104, 105 to form the internal network 110; a calling terminal 150, which is located outside the internal network 110 and must be connected to the internal network 110 via the external network; a network address translation (NAT) 120, it is between this calling terminal 150 and this internal network 110; A proxy server 130, it is between this calling terminal 150 and this internal network 110, and logically with Network Address Translation (NAT) 120 Parallel configuration, to be responsible for receiving the message of the conference establishment protocol sent by the calling terminal 150, and to connect and transmit the address, so as to transmit the message to specific network phones (User agents), and in a manner conforming to the SIP communication protocol For multimedia conference communication call and connection, after completing the conference call and connection between the internal and external networks, the "virtual agent phone" (Backend to Backend Useragent, B2BUA) defined by the SIP communication protocol is used to execute the internal and external network (that is, "internal Network-to-external network" and "external network to internal network") dialogue connections, and finally, data compression and transmission are performed in a manner that conforms to the RTP (Real TimeTransport Protocol) protocol to achieve instant and direct multimedia communication.

图4说明本发明前述的实施例中，该符合SIP通讯协议的代理服务器在为内部网络与外部网络间进行多媒体会议呼叫、联机与通讯进行的示意图，如图4所示，内部网络210包含两个网络话机(User agents)(internal computer)：网络话机(User agent)A 211及网络话机(Useragent)B 212，而外部网络220则包含网络话机(User agent)D 221，其中，网络话机(User agent)A 211的IP地址为：192.0.2.101，网络话机(User agent)B 212的IP地址为：192.0.2.103；而网络话机(Useragent)D 221的IP地址为：17.0.0.1，而代理服务器230的私有IP地址：192.0.2.102，公共IP地址为：10.0.0.1。Fig. 4 illustrates that in the aforementioned embodiment of the present invention, the proxy server conforming to the SIP communication protocol performs a schematic diagram of multimedia conference calling, connection and communication between the internal network and the external network. As shown in Fig. 4, the internal network 210 includes two Internet phone (User agents) (internal computer): Internet phone (User agent) A 211 and Internet phone (User agent) B 212, and external network 220 then includes network phone (User agent) D 221, wherein, Internet phone (User agent) D 221 The IP address of agent) A 211 is: 192.0.2.101, the IP address of network phone (User agent) B 212 is: 192.0.2.103; and the IP address of network phone (User agent) D 221 is: 17.0.0.1, and the proxy server 230's private IP address: 192.0.2.102, public IP address: 10.0.0.1.

由于代理服务器230内部网络网络话机(User agent)A 211及网络话机(User agent)B 212对外的呼叫媒介(call mediation)，网络话机(User agent)A 211及网络话机(User agent)B 212以代理服务器230的私有IP地址：192.0.2.102设定(configure)为其对外讯息的目的地址，而从外部网络220传送进来的讯息，则以而代理服务器230的公共IP地址：10.0.0.1为目的地址，由于内部网络210与外部网络220中所有的会议参与者必须事先登记其SIP识别(identity)，即SIP协议所定义的统一来源识别(Uniform Resource Identifier，URI)，到代理服务器230上，因此，该外部网络220传送进来的讯息即可通过代理服务器230以URI(Uniform Resource Identifier)的方式将讯息定位传送到特定的User上。Due to the external call medium (call mediation) of the network phone (User agent) A 211 and the network phone (User agent) B 212 in the proxy server 230, the network phone (User agent) A 211 and the network phone (User agent) B 212 are The private IP address of the proxy server 230: 192.0.2.102 is set (configure) as the destination address of its external messages, and the incoming messages from the external network 220 are for the purpose of the public IP address of the proxy server 230: 10.0.0.1 address, because all conference participants in the internal network 210 and the external network 220 must register its SIP identification (identity) in advance, that is, the unified source identification (Uniform Resource Identifier, URI) defined by the SIP protocol, on the proxy server 230, therefore , the incoming message from the external network 220 can be sent to a specific User through the proxy server 230 in the form of a URI (Uniform Resource Identifier).

代理服务器230负责执行：(1)依SIP通讯协议，进行会议的呼叫与联机；(2)依SIP通讯协议所定义的虚拟代理话机(B2BUA)，进行内部网络210与外部网络220的对话媒介；(3)依RTP(Real TimeTransport Protocal)协议，进行资料的压缩与传送，以提供即时、直接的点对点的多媒体通讯。The proxy server 230 is responsible for: (1) calling and connecting the meeting according to the SIP communication protocol; (2) performing the dialogue medium between the internal network 210 and the external network 220 according to the virtual agent phone (B2BUA) defined by the SIP communication protocol; (3) According to the RTP (Real TimeTransport Protocol) protocol, the data is compressed and transmitted to provide instant and direct point-to-point multimedia communication.

在欲进行内部网络210与外部网络220会议的呼叫与联机前，内部网络210与外部网络220的会议参与者必须事先登记其SIP识别(SIPidentity)，即SIP协议所定义的统一来源识别(Uniform ResourceIdentifier，URI)，到代理服务器230，其中，代理服务器230(执行SIP会议的呼叫与联机时)会于预设端口5060(default port 5060)聆听内外网络两边界面的呼叫，并以执行内部网络210与外部网络220对话媒介的虚拟代理话机(B2BUA)，在预设端口7060(default port 7060)聆听两边界面的呼叫；一旦有从外部网络220而来或由内部网络210发出的到代理服务器230的呼叫，该代理服务器230将做以下的确认：Before calling and connecting to the conference between the internal network 210 and the external network 220, the conference participants of the internal network 210 and the external network 220 must register their SIP identification (SIP identity), which is the uniform source identification (Uniform Resource Identifier) defined by the SIP protocol. , URI), to the proxy server 230, wherein, the proxy server 230 (when performing the call and connection of the SIP conference) will listen to the calls of the internal and external network interfaces at the default port 5060 (default port 5060), and execute the internal network 210 and The virtual agent phone (B2BUA) of the external network 220 dialogue medium listens to the calls of both interfaces at the preset port 7060 (default port 7060); once there is a call from the external network 220 or sent by the internal network 210 to the proxy server 230 , the proxy server 230 will do the following confirmations:

1.关于URI要求(URI request)的内容属公共网络讯息(此即由内部网络210对外部网络220进行会议呼叫)。1. The content of the URI request (URI request) is public network information (that is, the internal network 210 makes a conference call to the external network 220).

2.关于URI要求(URI request)的内容属内部网络讯息，且所经过者为公共路径(此即由外部网络220对内部网络210进行会议呼叫)。2. The content of the URI request (URI request) is an internal network message, and what passes through is a public path (that is, the conference call is made from the external network 220 to the internal network 210).

如果上述的其中之一为真，则代理服务器230即会将呼叫予以定址(route)传送，以接着执行虚拟代理话机(B2BUA)的对话媒介功能，由虚拟代理话机(B2BUA)功能对该呼叫加上其联络IP(contact IP)以记录其来源路径后，再将该呼叫传送到目的地，而完成会议通讯的呼叫与联机，此外，执行虚拟代理话机(B2BUA)功能时，代理服务器230会修改SDP(Session Description Protocal)参数的字段(SDP parameter’sOrigin)以及连接字段(Connection filed)来通知代理服务器230中的RTP功能，进行RTP Relay-对通讯各方开放其RTP端口，使其等能进行即时的会议通讯。If one of the above is true, the proxy server 230 will route the call to then perform the dialogue mediation function of the virtual agent unit (B2BUA), which adds to the call by the virtual agent unit (B2BUA) function. After uploading its contact IP (contact IP) to record its source path, the call is sent to the destination, and the call and connection of the conference communication are completed. In addition, when the virtual agent phone (B2BUA) function is executed, the proxy server 230 will modify the The field (SDP parameter'sOrigin) and the connection field (Connection filed) of the SDP (Session Description Protocol) parameter are used to notify the RTP function in the proxy server 230 to carry out RTP Relay-open its RTP port to the communication parties so that it can be carried out Instant conference communication.

更详细而言，当符合SIP通讯协议的代理服务器230，依据SIP通讯协议建立多媒体会议的呼叫与联机，而将内部网络210与外部网络220予以串连后，该SIP的讯号将被定址到虚拟代理话机(B2BUA)，如果该SIP讯号内部网络210的网络话机(User agent)对外的呼叫(outgoing call)，虚拟代理话机(B2BUA)将终止该SIP讯号，并产生一新的讯号(改变SIP讯号参数及SDP参数)而传送到外部网络220，反之亦然，即将外部网络220对内的呼叫(incoming call)传送到内部网络210的网络话机(User agent)，因此，通过虚拟代理话机(B2BUA)可将「内对外」以及「外对内」两个对话(call leg，dialogue)予以连接，在虚拟代理话机(B2BUA)的对话连接完成后，则由RTP(Real timeTransport Protocal)Relay来进行即时的对话通讯。In more detail, when the proxy server 230 conforming to the SIP communication protocol establishes a multimedia conference call and connection according to the SIP communication protocol, and connects the internal network 210 and the external network 220 in series, the SIP signal will be addressed to the virtual Agent phone (B2BUA), if the network phone (User agent) of the internal network 210 of the SIP signal calls out (outgoing call), the virtual agent phone (B2BUA) will terminate the SIP signal and generate a new signal (change SIP signal parameters and SDP parameters) to the external network 220, and vice versa, the incoming call (incoming call) of the external network 220 is transmitted to the network phone (User agent) of the internal network 210, therefore, through the virtual agent phone (B2BUA) The two dialogues (call leg, dialogue) of "internal and external" and "external and internal" can be connected. After the dialogue connection of the virtual agent phone (B2BUA) is completed, the RTP (Real timeTransport Protocol) Relay will perform real-time communication Conversation newsletter.

图5本发明实施例的符合SIP通讯协议的代理服务器在内部网络对外部网络进行会议呼叫的程序，以构成内外网络多媒体会议联机的说明图，如图5所示，为便于细部流程的说明，图5中乃将图4所述的本发明实施例中代理服务器负责执行的「会议的呼叫与联机工作」及「媒介内部网络与外部网络的对话」两项工作分别以「SIP服务器」及「虚拟代理话机」(B2BUA)来表示。The program of the proxy server conforming to the SIP communication protocol of the embodiment of the present invention in the internal network carries out the program of the conference call to the external network, to form an explanatory diagram of the online multimedia conference connection of the internal and external network, as shown in Figure 5, for the convenience of the description of the detailed process, In Fig. 5, in the embodiment of the present invention described in Fig. 4, the two tasks of "conference call and online work" and "dialogue between media internal network and external network" that the proxy server is responsible for performing are respectively represented by "SIP server" and " Virtual agent phone" (B2BUA) to represent.

首先，内部网络内的网络话机(User agent)310对外部网络的「被呼叫端」(called SIP terminal)320进行呼叫(invite)，而该呼叫(INVITE)则被定址传送到内部网络对外的SIP服务器330(步骤401)，而SIP服务器330在加入其记录路径的via后即将该呼叫传送到虚拟代理话机(B2BUA)340(步骤402)，虚拟代理话机(B2BUA)340则接着对URI要求(URI Request)进行DNS(Domain Name Server)的确认检查，以取得该呼叫的目的地的IP地址，之后则将先前的via文件头信息(header)移除，并在修改联络地址(contact address)及SDP参数后，加上其自身的via文件头信息(header)，然后，该呼叫(INVITE)即可被定址(route)到URI要求(URI Request)所对应目的地的IP地址，在经过一或多个代理服务器后(未示出)，该呼叫(INVITE)即被传送到被呼叫端320(步骤403)。First, the network phone (User agent) 310 in the internal network makes a call (invite) to the "called SIP terminal" (called SIP terminal) 320 in the external network, and the call (INVITE) is addressed and transmitted to the external SIP terminal of the internal network. Server 330 (step 401), andSIP server 330 is about to transfer this call to virtual agent phone (B2BUA) 340 (step 402) after adding the via of its record path, and virtual agent phone (B2BUA) 340 then requests URI (URI Request) to perform a DNS (Domain Name Server) confirmation check to obtain the IP address of the destination of the call, and then remove the previous via file header information (header), and modify the contact address (contact address) and SDP After the parameters, add its own via file header information (header), then, the call (INVITE) can be addressed (route) to the IP address of the destination corresponding to the URI request (URI Request), after one or more After a proxy server (not shown), the call (INVITE) is sent to the called end 320 (step 403).

之后，被呼叫端320先会传送一个讯息(100Trying)给虚拟代理话机(B2BUA)340(步骤404)，后被呼叫端320在加上其自身的联络IP地址后响应呼叫(180Ringing)，该被呼叫端320的响应即会传送到虚拟代理话机(B2BUA)(步骤405)，虚拟代理话机(B2BUA)在移除掉该响应(180Ringing)的via文件头信息(header)后，并且将被呼叫端320的联络IP地址改为其联络IP地址，且插入SIP服务器330的via文件头信息(header)及该User 310的via文件头信息(header)后，将该响应(180Ringing)送到SIP服务器330，而SIP服务器330再将该响应(180Ringing)传至User 310(步骤406)；而被呼叫端320的接受联机的响应(200OK)亦循此路径完成(步骤407，408)。After that, the called terminal 320 will first send a message (100Trying) to the virtual agent phone (B2BUA) 340 (step 404), and then the called terminal 320 will respond to the call (180Ringing) after adding its own contact IP address. The response of the callingterminal 320 will be transmitted to the virtual agent phone (B2BUA) (step 405). After the virtual agent phone (B2BUA) removes the via file header information (header) of the response (180Ringing), and will be The contact IP address of 320 is changed to its contact IP address, and after inserting the via file header information (header) of theSIP server 330 and the via file header information (header) of theUser 310, the response (180Ringing) is sent to theSIP server 330 , and theSIP server 330 transmits the response (180Ringing) to the User 310 (step 406); and the response (200OK) of the called terminal 320 to accept the connection is also completed along this path (steps 407, 408).

在构成网络话机(User agent)310与被呼叫端320的联机后，网络话机(User agent)310的联机确认(ACK)即会直接传送到虚拟代理话机(B2BUA)340(步骤409)，并由虚拟代理话机(B2BUA)340直接将联机确认(ACK)传送到被呼叫端320(步骤410)，当会议的呼叫与联机确立后，而呼叫端网络话机(User agent)310与被呼叫端320相互知悉对方的联络IP地址后，双方即直接通过虚拟代理话机(B2BUA)340与RTP Relay进行即时的会议通讯。After forming the connection between the network phone (User agent) 310 and the calledterminal 320, the online confirmation (ACK) of the network phone (User agent) 310 will be directly sent to the virtual agent phone (B2BUA) 340 (step 409), and the The virtual agent phone (B2BUA) 340 directly transmits the connection acknowledgment (ACK) to the called terminal 320 (step 410). After knowing the contact IP address of the other party, the two parties will directly conduct real-time conference communication with RTP Relay through the virtual agent phone (B2BUA) 340.

如网络话机(User agent)310要中断会议的联机，则传送一个会议中断讯息(BYE)到虚拟代理话机(B2BUA)340(步骤411)，然后虚拟代理话机(B2BUA)340再将会议中断讯息(BYE)传送到被呼叫端320(步骤412)，而被呼叫端320再将接受会议中断的讯息(200OK)，传送到虚拟代理话机(B2BUA)340(步骤413)，并由虚拟代理话机(B2BUA)340将接受会议中断的讯息(200OK)回传网络话机给(Useragent)310(步骤414)，而完成会议的中断。If the network phone (User agent) 310 wants to interrupt the connection of the meeting, then send a meeting interruption message (BYE) to the virtual agent phone (B2BUA) 340 (step 411), and then the virtual agent phone (B2BUA) 340 will then send the meeting interruption message ( BYE) is transmitted to the called terminal 320 (step 412), and the called terminal 320 then transmits the message (200 OK) of accepting the interruption of the meeting to the virtual agent phone (B2BUA) 340 (step 413), and the virtual agent phone (B2BUA ) 340 returns the message (200 OK) of accepting the interruption of the conference to the Internet phone (Useragent) 310 (step 414 ), and completes the interruption of the conference.

图6本发明实施例的符合SIP通讯协议的代理服务器在外部网络对内部网络进行会议呼叫的程序，以构成内外网络多媒体会议联机的说明图，如图6所示，为便于细部流程的说明，图6中乃将图4所述的本发明实施例中代理服务器负责执行的「会议的呼叫与联机工作」及「媒介内部网络与外部网络的对话」两项工作分别以「SIP服务器」及「虚拟代理话机」(B2 BUA)来表示。Fig. 6 is the procedure for the proxy server conforming to the SIP communication protocol of the embodiment of the present invention to carry out a conference call program to the internal network in the external network, to form an explanatory diagram of the online multimedia conference connection of the internal and external network, as shown in Fig. 6, for the convenience of the description of the detailed process, In Fig. 6, in the embodiment of the present invention described in Fig. 4, the two tasks of "conference call and online work" and "dialogue between media internal network and external network" that the proxy server is responsible for performing are respectively represented by "SIP server" and " Virtual agent phone" (B2 BUA) to represent.

首先，外部网络的呼叫端(calling SIP terminal)510对内部网络的网络话机(User agent)520进行呼叫(INVITE)，而该呼叫(INVITE)则被定址传送到SIP服务器530(步骤601)，其中，SIP服务器530的IP地址被设定为该DNS(Domain Name Server)的外部网络入口，SIP服务器530在接收呼叫后，即对该呼叫进行位置检查(location look-up)，并将呼叫端510所传送的呼叫讯息中的URI要求(URI Request)改为网络话机(User agent)520的IP地址，并在加入其记录路径的via后，即将该呼叫传送到虚拟代理话机(B2BUA)540(步骤602)，虚拟代理话机(B2BUA)540会移除先前讯息中的via文件头信息(header)，并在修改联络地址(contact address)及SDP参数后，加上其自身的via文件头信息(header)，然后，将该呼叫(INVITE)定址(route)到该呼叫讯息的URI要求(URI Request)所对应的网络话机(User agent)520的IP地址，而将该呼叫传送到网络话机(User agent)520(步骤603)。First, the calling end (calling SIP terminal) 510 of the external network calls (INVITE) the network phone (User agent) 520 of the internal network, and the calling (INVITE) is then addressed and transmitted to the SIP server 530 (step 601), wherein , the IP address of theSIP server 530 is set as the external network entrance of the DNS (Domain Name Server). The URI request (URI Request) in the sent call message changes the IP address of network phone (User agent) 520 into, and after adding the via of its record path, be about to send this call to virtual agent phone (B2BUA) 540 (step 602), the virtual agent phone (B2BUA) 540 will remove the via file header information (header) in the previous message, and after modifying the contact address (contact address) and SDP parameters, add its own via file header information (header ), then, the call (INVITE) is addressed (route) to the IP address of the network phone (User agent) 520 corresponding to the URI request (URI Request) of the call message, and the call is sent to the network phone (User agent) ) 520 (step 603).

网络话机(User agent)520会先传送一个讯息(100Trying)给虚拟代理话机(B2BUA)540(步骤604)，然后再将呼叫响应(180Ringing)传送到虚拟代理话机(B2BUA)540(步骤605)，虚拟代理话机(B2BUA)540会移除掉该呼叫响应(180Ringing)的via文件头信息(header)，并在插入SIP服务器530的via文件头信息(header)及该呼叫端510的via文件头信息(header)后，将该呼叫响应(180Ringing)送到SIP服务器530，而SIP服务器530再将该呼叫响应(180Ringing)传至呼叫端510(步骤606)，而网络话机(User agent)520接受联机的响应(200OK)亦循此路径完成(步骤607，608)。The network phone (User agent) 520 will first send a message (100Trying) to the virtual agent phone (B2BUA) 540 (step 604), and then send the call response (180Ringing) to the virtual agent phone (B2BUA) 540 (step 605), The virtual agent phone (B2BUA) 540 will remove the via file header information (header) of the call response (180Ringing), and insert the via file header information (header) of theSIP server 530 and the via file header information of the calling terminal 510 (header), the call response (180Ringing) is sent to theSIP server 530, and theSIP server 530 sends the call response (180Ringing) to the calling terminal 510 (step 606), and the network phone (User agent) 520 accepts the connection The response (200OK) is also completed along this path (steps 607, 608).

在构成呼叫端510与网络话机(User agent)520的联机后，呼叫端510的联机确认(ACK)即会直接传送到虚拟代理话机(B2BUA)540(步骤609)，并由虚拟代理话机(B2BUA)540直接将联机确认(ACK)传送到User 520(步骤610)，当会议的呼叫与联机确立后，而呼叫端510与网络话机(User agent)520相互知悉对方的联络IP地址后，双方即直接通过虚拟代理话机(B2BUA)540与RTP Relay进行即时的会议通讯。After forming the online connection between the callingterminal 510 and the network phone (User agent) 520, the online confirmation (ACK) of the callingterminal 510 will be directly sent to the virtual agent phone (B2BUA) 540 (step 609), and the virtual agent phone (B2BUA) ) 540 directly transmits the online confirmation (ACK) to User 520 (step 610). After the conference call and online connection are established, and after the callingterminal 510 and the network phone (User agent) 520 know each other's contact IP address, both parties immediately Real-time conference communication with RTP Relay directly through the virtual agent phone (B2BUA) 540.

如呼叫端510要中断会议的联机，则传送一个会议中断讯息(BYE)到虚拟代理话机(B2BUA)540(步骤611)，然后虚拟代理话机(B2BUA)540再将会议中断讯息(BYE)传送到网络话机(User agent)520(步骤612)，而网络话机(User agent)520再将接受会议中断的讯息(200OK)，传送到虚拟代理话机(B2BUA)540(步骤613)，并由虚拟代理话机(B2BUA)540将接受会议中断的讯息(200OK)回传给呼叫端510(步骤614)，而完成会议的中断。If the callingterminal 510 wants to interrupt the connection of the conference, then send a conference interruption message (BYE) to the virtual agent phone (B2BUA) 540 (step 611), and then the virtual agent phone (B2BUA) 540 sends the conference interruption message (BYE) to The network phone (User agent) 520 (step 612), and the network phone (User agent) 520 will accept the message (200OK) that the meeting is interrupted again, transmits to the virtual agent phone (B2BUA) 540 (step 613), and by the virtual agent phone (B2BUA) 540 returns the message (200 OK) of accepting the interruption of the conference to the calling terminal 510 (step 614), and completes the interruption of the conference.

应注意的是上述实施例仅是范例的而非限制本发明。更进一步，实施例所举的制程、步骤、材料、尺度、结构或其它具有可选择性的部分亦不限制本发明。除此，本发明由权利要求范围所定义。It should be noted that the above-mentioned embodiments are only exemplary rather than limiting the present invention. Furthermore, the manufacturing processes, steps, materials, dimensions, structures or other optional parts mentioned in the embodiments do not limit the present invention. Otherwise, the present invention is defined by the scope of the claims.

Claims (36)

1. the multimedia conference system of a direct communication between internal-external network is characterized in that, comprising:

The network phone of energy call accepted more than one, to constitute internal network, the network phone of this call accepted is defined as the network phone;

One calling terminal, it is positioned at outside this internal network, and is connected with this internal network through external network;

A network address translation, it is between between this calling terminal and this internal network;

An acting server, it is between between this calling terminal and this internal network, and in logic with the network address translation configured in parallel, build the message of putting agreement to be responsible for the receiving meeting of sending by calling terminal, the go forward side by side connection and the transmission of row address are sent to described network phone this meeting is built the message of putting agreement.

2. multimedia conference system as claimed in claim 1 is characterized in that, this acting server is selected the network phone to the calling that calling terminal proposed, and this acting server is as the interface between it, to connect the both sides of conference communications.

3. multimedia conference system as claimed in claim 1 is characterized in that, this acting server is to meet the mode of SIP communications protocol, to build the calling of putting conference communications and online.

4. multimedia conference system as claimed in claim 2 is characterized in that this acting server has plural network interface, and wherein at least one network interface connects external network, and at least one network interface connects internal network.

5. multimedia conference system as claimed in claim 4 is characterized in that, the network interface of this connection external network has independently public IP, and the network interface of this connection internal network then has the IP that internal network defines the net territory.

6. multimedia conference system as claimed in claim 3 is characterized in that, this acting server is transferred to the network phone with the calling that calling terminal sends.

7. multimedia conference system as claimed in claim 5 is characterized in that, this acting server more comprises a logon server in logic, and this more than one network phone is done the login of URI to logon server.

8. multimedia conference system as claimed in claim 7, it is characterized in that this acting server can be obtained the contact inventory of the network phone of logining URI to this logon server, and according to the mode of URI location, with the calling that calling terminal sends, be transferred to the network phone.

9. multimedia conference system as claimed in claim 8 is characterized in that, this acting server is in virtual protocol phone mode, as the dialogue media between this calling terminal and network phone.

10. multimedia conference system as claimed in claim 9 is characterized in that, two dialogues are connected the virtual protocol phone of this acting server to calling terminal to network phone and network phone with calling terminal.

11. multimedia conference system as claimed in claim 10 is characterized in that, the virtual protocol phone of this acting server is set up the RTP forwarding mechanism between calling terminal and network phone.

12. multimedia conference system as claimed in claim 11 is characterized in that, this acting server carries out instant multimedia conferencing communication to meet the mode of Real-time Transport Protocol.

13. the multimedia conference system that can carry out direct communication between internal-external network is characterized in that, comprising:

The network phone of energy call accepted more than one, to constitute internal network, the network phone of this call accepted is defined as the network phone;

An acting server, it is connected with the network phone by this internal network, and this acting server receives the meeting of being sent by network phone in the internal network and builds the message of putting agreement, the connection of the row address of going forward side by side and transmission as the interface of external network;

One called side, it is connected with this acting server by external network;

A network address translation, it is between between this called side and this internal network, and this network address translation is put with parallel in logic the building of this acting server.

14. multimedia conference system as claimed in claim 13 is characterized in that, the calling that this acting server is proposed the network phone, and by the DNS inquiry, to connect called side, this acting server is as the interface between the both sides of conference communications.

15. multimedia conference system as claimed in claim 13 is characterized in that, this acting server is to meet the mode of SIP communications protocol, to build the calling of putting conference communications and online.

16. multimedia conference system as claimed in claim 14 is characterized in that, this acting server has plural network interface, and wherein at least one network interface connects external network, and at least one network interface connects internal network.

17. multimedia conference system as claimed in claim 16 is characterized in that, the network interface of this connection external network has independently public IP, and the network interface of this connection internal network then has the IP that internal network defines the net territory.

18. multimedia conference system as claimed in claim 15 is characterized in that, this acting server is transferred to called side with the calling that the network phone sends.

19. multimedia conference system as claimed in claim 17 is characterized in that, this acting server more comprises a logon server in logic, and this more than one network phone can be done the login of URI to logon server.

20. multimedia conference system as claimed in claim 19 is characterized in that, this acting server is in virtual protocol phone mode, as the dialogue media between this network phone and called side.

21. multimedia conference system as claimed in claim 20 is characterized in that, two dialogues are connected the virtual protocol phone of this acting server to the network phone to called side and called side with the network phone.

22. multimedia conference system as claimed in claim 21 is characterized in that, the virtual protocol phone of this acting server is set up the RTP forwarding mechanism between network phone and called side.

23. multimedia conference system as claimed in claim 22 is characterized in that, this acting server carries out instant multimedia conferencing communication to meet the mode of Real-time Transport Protocol.

24. the method that can carry out the multimedia conferencing of direct communication between internal-external network is characterized in that, comprising:

(1) setting one meets the acting server of Session Initiation Protocol, this acting server is arranged between internal network and external network, this acting server and network address translation be configured in parallel in logic, the message of putting agreement is built in the meeting that calling terminal transmitted that reception is connected with internal network through external network, this message is confirmed in check, and the metered call end sends the source path of call forwarding message;

(2) the network phone being set a meeting builds and puts message and store access path;

(3) according to the access path of the network phone after selecting in external network calling terminal and the internal network, carry out multimedia conference communications.

25. the method for multimedia conferencing as claimed in claim 24 is characterized in that, this step (2) more comprises: (2-1) logon server that comprised in addition in logic in acting server of the network phone of internal network carries out the login of URI.

26. the method for multimedia conferencing as claimed in claim 25, it is characterized in that, this step (2-1), more comprise: (2-2) acting server is obtained the contact inventory of the network phone of logining URI to this logon server, and according to the mode of URI location, with the calling that calling terminal sends, be transferred to the network phone.

27. the method for multimedia conferencing as claimed in claim 26, it is characterized in that, should (3) step more comprise: (3-1), carry out the multimedia conference communications through between the network phone after the selection in external network calling terminal and the internal network with the virtual protocol phone media that engages in the dialogue.

28. the method for multimedia conferencing as claimed in claim 27 is characterized in that, should (3-1) step carries out the network phone to the connection to two dialogues of network phone of calling terminal and calling terminal with the virtual protocol phone.

29. the method for multimedia conferencing as claimed in claim 28 is characterized in that, should (3-1) step more comprise: (3-2) the virtual protocol phone carries out the foundation of the RTP forwarding mechanism between network phone and calling terminal.

30. the method for multimedia conferencing as claimed in claim 29 is characterized in that, should (3-2) step in a mode that meets Real-time Transport Protocol, carry out the instant multimedia conferencing communication of communication each side.

31. the method that can carry out the multimedia conferencing of direct communication between internal-external network is characterized in that, comprising:

(1) setting one meets the acting server of Session Initiation Protocol, this acting server is arranged between an internal network and an external network, this acting server and network address translation be configured in parallel in logic, the message of putting agreement is built in the meeting that reception is sent by network phone in the internal network, this message is confirmed in check, and record network phone sends the source path of calling out INVITE;

(2) this acting server calling INVITE that the network phone is sent by the DNS inquiry, is stored in the outer called side of internal network with connection, and builds at this meeting and to put message and store access path;

(3) according to the access path of the network phone of the internal network called side outer, carry out multimedia conference communications with being stored in internal network.

32. the method for multimedia conferencing as claimed in claim 31 is characterized in that, this step (1) more comprises: (1-1) logon server that comprised in addition in logic in acting server of the network phone of internal network carries out the login of URI.

33. the method for multimedia conferencing as claimed in claim 32, it is characterized in that, should (3) step more comprise: (3-1) with the virtual protocol phone as the dialogue media, the multimedia conference communications between the network phone that carries out internal network and internal network called side outward.

34. the method for multimedia conferencing as claimed in claim 33 is characterized in that, should (3-1) step carries out the network phone to the connection to two dialogues of network phone of called side and called side with the virtual protocol phone.

35. the method for multimedia conferencing as claimed in claim 34 is characterized in that, should (3-1) step more comprise: (3-2) carry out the foundation of the RTP forwarding mechanism between network phone and called side with the virtual protocol phone.

36. the method for multimedia conferencing as claimed in claim 35 is characterized in that, should (3-2) step in a mode that meets Real-time Transport Protocol, carry out the instant multimedia conferencing communication of communication each side.

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