


技术领域technical field
本发明涉及数据通信技术领域,特别涉及一种在包括采用网络地址转换(NAT)技术组网的私有网络在内的网络上实现IP语音业务(VoIP)的方法及系统。The invention relates to the technical field of data communication, in particular to a method and a system for realizing voice over IP service (VoIP) on a network including a private network using network address translation (NAT) technology.
背景技术Background technique
传统的语音业务采用电路交换技术,即通话双方通过公共交换电话网(PSTN)建立一条固定带宽(64kbit/s)的电路,这使得低时延、低失真的实时通信服务质量(QoS)得到保证,但在这种方式下网络带宽利用率低,通讯成本高,而且增值业务的推广比较困难。The traditional voice service adopts circuit switching technology, that is, the two sides of the call establish a fixed bandwidth (64kbit/s) circuit through the public switched telephone network (PSTN), which ensures the real-time communication quality of service (QoS) with low delay and low distortion , but in this way, the network bandwidth utilization rate is low, the communication cost is high, and the promotion of value-added services is difficult.
VoIP(Voice over IP)是指在以IP为网络层协议的计算机网络中进行语音传送的业务,简称IP语音业务。计算机通讯采用分组交换技术,即传输的数据单元都是由标识部分和数据部分封装而成的独立数据包,称之为“分组”(packet),网络中的各通讯节点存储并转发分组。因此,VoIP属于分组语音通信的范畴,是以IP为标志的网络分组化和以多媒体为目标的网络业务综合化两大主流技术融合的结果。分组交换的特点是网络利用率高,通讯成本低,而且IP网络作为开放式网络很容易快速推广新业务。VoIP (Voice over IP) refers to the service of voice transmission in the computer network with IP as the network layer protocol, referred to as IP voice service. Computer communication adopts packet switching technology, that is, the transmitted data units are independent data packets encapsulated by the identification part and the data part, called "packets", and each communication node in the network stores and forwards the packets. Therefore, VoIP belongs to the category of packet voice communication, and is the result of the fusion of two mainstream technologies, network packetization marked by IP and network service integration aimed at multimedia. Packet switching is characterized by high network utilization rate and low communication cost, and IP network, as an open network, is easy to quickly promote new services.
在计算机网络上进行语音通信的研究源自70年代。进入90年代后,特别是近几年,随着Internet在全球范围的兴起和语音编码技术的发展,VoIP取得突破性进展并得到了实际应用。其发展大致分为以下三个阶段:Research on voice communication over computer networks originated in the 1970s. After entering the 1990s, especially in recent years, with the global rise of the Internet and the development of speech coding technology, VoIP has made breakthroughs and has been practically applied. Its development can be roughly divided into the following three stages:
(1)萌芽期:Internet出现后,人们首先尝试了让两台PC机通过IP网络传送语音,这是VoIP最初的一种形式。在这一阶段,人们对VoIP的认识很肤浅,市场也不明确。(1) Emerging stage: After the emergence of the Internet, people first tried to let two PCs transmit voice through the IP network, which is the original form of VoIP. At this stage, people's understanding of VoIP is very superficial, and the market is not clear.
(2)发展期:在VoIP诸多优势的吸引下,很多电信公司认识到仅在PC机之间实现IP电话远远不能满足用户的需要,因此目光投向了众多的PSTN用户,这就导致IP电话网关的出现。IP电话网关将PSTN网络和IP网络连接起来,提供电话到电话,PC到电话,电话到PC等多种VoIP业务形式。这一阶段是IP电话迅速发展的时期。(2) Development period: Attracted by many advantages of VoIP, many telecommunication companies realize that IP telephony between PCs is far from meeting the needs of users, so they turn their attention to many PSTN users, which leads to IP telephony emergence of the gateway. The IP telephone gateway connects the PSTN network and the IP network, and provides various VoIP service forms such as telephone to telephone, PC to telephone, telephone to PC, etc. This stage is a period of rapid development of IP telephony.
(3)成熟期:IP电话应用的发展在未来将进入成熟期,具有通讯采用统一的标准,全球各电信厂家的语音设备可以互通、VoIP话音质量接近传统电话质量等特点。(3) Mature stage: The development of IP telephony applications will enter a mature stage in the future, with the characteristics of unified standards for communication, voice equipment of various telecom manufacturers around the world can communicate with each other, and the voice quality of VoIP is close to that of traditional telephones.
实现IP语音业务的通信协议包括有语音通信控制协议、话音信息传送协议、会议电话控制协议和实时控制协议等。其中,语音通信控制协议即电信网中的呼叫控制信令,包括地址信息、用户状态信息、双音多频(DTMF)信号等,一般采用传输控制协议(TCP)作为传输层协议。话音信息传送协议规定了话音分组如何封装、复用和传送,包括各种语音编码分组如何组装和标识,要求要实时传送,只能用用户数据报协议(UDP)作为传输层协议。IP电话利用实时传输协议(RTP)适配分组语音数据,RTP可传送分组的时间戳、分组序号等信息,支持QoS监视和多数据流合并。另外,为确保实时性,还选用资源预留协议(RSVP)为呼叫保留网络资源。目前VoIP的实现均是在开始呼叫的时候,首先建立呼叫信令通道,进行信令交互和协商以后,建立逻辑通道即语音通道,逻辑通道的地址需要在呼叫信令通道的报文中传送。The communication protocols for implementing IP voice services include voice communication control protocols, voice information transmission protocols, conference call control protocols, and real-time control protocols. Among them, the voice communication control protocol is the call control signaling in the telecommunication network, including address information, user status information, dual-tone multi-frequency (DTMF) signals, etc., and the transmission control protocol (TCP) is generally used as the transport layer protocol. The voice information transmission protocol stipulates how to encapsulate, multiplex and transmit voice packets, including how to assemble and identify various voice coded packets, and requires real-time transmission, so only User Datagram Protocol (UDP) can be used as the transport layer protocol. IP phones use the Real-time Transport Protocol (RTP) to adapt packet voice data. RTP can transmit information such as time stamps and sequence numbers of packets, and supports QoS monitoring and merging of multiple data streams. In addition, in order to ensure real-time performance, the Resource Reservation Protocol (RSVP) is also selected to reserve network resources for calls. At present, the implementation of VoIP is to establish a call signaling channel at the beginning of a call. After signaling interaction and negotiation, a logical channel, that is, a voice channel, is established. The address of the logical channel needs to be transmitted in the message of the call signaling channel.
私有网络是指使用私有IP地址的内部网络,如企业网。当今绝大部分企业网都是实施TCP/IP协议的内联网,用户使用DHCP协议动态分配私有IP地址,在私网上划分多个网段,用户数据往往越过本地网段在私网内的各子网间传送。企业网内部,同一地点的分支或总部使用第三层交换技术(也称IP交换技术、高速路由技术等,是一种利用第三层协议中信息来加强第二层交换功能的机制)通过以太网进行连接,而分支与总部间使用传统路由器通过专线(DDN或Leased line)进行连接。企业网到公网一般有唯一的出口,用户访问公网时,必须借助于代理服务器的NAT功能。A private network refers to an internal network using private IP addresses, such as an enterprise network. Most of today's enterprise networks are intranets that implement the TCP/IP protocol. Users use the DHCP protocol to dynamically allocate private IP addresses and divide multiple network segments on the private network. Internet transmission. Inside the enterprise network, the branches or headquarters at the same location use the third-layer switching technology (also known as IP switching technology, high-speed routing technology, etc., which is a mechanism that uses the information in the third-layer protocol to strengthen the second-layer switching function) through the Ethernet The network is connected, and the branch and the headquarters are connected through a dedicated line (DDN or Leased line) using a traditional router. Generally, there is only one exit from the enterprise network to the public network. When users access the public network, they must rely on the NAT function of the proxy server.
NAT即地址转换或地址代理,用来实现私有网络地址与公有网络地址之间的转换。私有地址是指内部网络(局域网内部)的主机地址,而公有地址是局域网的外部地址(在因特网上的全球唯一的IP地址)。因特网地址分配组织规定以下的三个网络地址段保留用做私有地址:NAT is address translation or address proxy, which is used to realize the translation between private network addresses and public network addresses. The private address refers to the host address of the internal network (inside the local area network), and the public address is the external address of the local area network (the globally unique IP address on the Internet). The Internet Assigned Numbers Organization stipulates that the following three network address segments are reserved for private addresses:
10.0.0.0 ~10.255.255.25510.0.0.0 ~ 10.255.255.255
172.16.0.0 ~172.31.255.255172.16.0.0~172.31.255.255
192.168.0.0~192.168.255.255192.168.0.0~192.168.255.255
也就是说这三个网络地址段不会在因特网上被分配,但可以在一个私有(局域网)内部网络使用。各个私网根据在可预见的将来主机数量的多少,选择一个合适的网络地址。不同私网的内部网络地址可以相同。That is to say, these three network address segments will not be allocated on the Internet, but can be used in a private (local area network) internal network. Each private network selects an appropriate network address according to the number of hosts in the foreseeable future. The internal network addresses of different private networks can be the same.
NAT技术在如企业网的私网上被广泛使用,以克服地址空间的限制和增加安全性,以及解决多个子网的联网问题。网络地址转换通过复用TCP或UDP端口号,提供把多个内部(私网)IP地址转换成外部(公网)IP地址的能力。关于地址转换的方式,又可分为多种类型。有的使用同一公网地址,用不同端口号来区分不同连接;有的在地址池中取得一个公网地址,按照一定算法取得端口号。NAT technology is widely used on private networks such as enterprise networks to overcome the limitation of address space and increase security, and to solve the networking problem of multiple subnets. Network address translation provides the ability to translate multiple internal (private network) IP addresses into external (public network) IP addresses by multiplexing TCP or UDP port numbers. Regarding the way of address translation, it can be divided into many types. Some use the same public network address and use different port numbers to distinguish different connections; some obtain a public network address in the address pool and obtain the port number according to a certain algorithm.
对于NAT的多种方式,可以总结出一个共有的特点,即私网用户要使用公网的服务时,需要通过NAT得到一个公网地址(包括IP地址和端口号),这个转换的对应关系记录在NAT路由器的一个地址端口对应表中,表中对应的两者分别为内部私有IP地址,和发往外部的IP报文的源端口号,后者也是从外部收到IP报文的目的端口号。在公网的节点看来,这个私网用户的地址就是NAT转换后的公网地址,如果要向这个私网用户发送数据,就直接发送这个公网地址,路由器通过查询地址端口对应表中存储的对应关系,将数据转发给私网用户。For the various ways of NAT, a common feature can be summarized, that is, when a private network user wants to use public network services, he needs to obtain a public network address (including IP address and port number) through NAT, and the corresponding relationship record of this conversion In an address-port correspondence table of the NAT router, the corresponding two in the table are the internal private IP address and the source port number of the IP message sent to the outside, and the latter is also the destination port of the IP message received from the outside Number. From the perspective of the nodes on the public network, the address of this private network user is the public network address after NAT conversion. If you want to send data to this private network user, you can directly send this public network address. The router queries the address stored in the port corresponding table. The corresponding relationship, and forward the data to the private network user.
简单来说,地址转换技术(NAT)是将内部IP地址和端口替换为外部网络IP地址和端口,以及反向的转换,NAT技术的优点在于:In simple terms, address translation technology (NAT) is to replace internal IP addresses and ports with external network IP addresses and ports, and reverse conversion. The advantages of NAT technology are:
1、提供内部网络的主机访问网外资源;1. Provide hosts in the internal network to access resources outside the network;
2、为内部主机提供了“隐私(Privacy)”保护,提高安全性;2. Provide "Privacy" protection for internal hosts to improve security;
3、解决IP地址资源问题。3. Solve the problem of IP address resources.
当私网用户使用现有的VoIP系统应用VoIP业务时,会遇到下列问题:私网PC向公网PC或网关发起呼叫的时候,可能会出现单通或无法通话;公网PC或电话(通过网关)拨打位于私网中的PC时,通话无法建立。原因就在于:当PC向公网被叫PC或网关发起呼叫的时候,通过NAT转换,私网PC的呼叫信令能正确发送到公网,但由于呼叫信令的数据部分中包含的媒体信道的源地址和源端口号是属于私网的,NAT路由器只能对IP包头的源地址和源端口号进行转换,无法修改数据部分,所以被叫方无法获知主叫方媒体信道的地址,所以造成单通或者不通的现象。当公网PC或电话(通过网关)向私网PC发起呼叫时,被叫方的VoIP服务器查询到的被叫呼叫信道(或控制信道)地址是私网地址,所以无法与被叫PC建立呼叫连接。When private network users use the existing VoIP system to apply VoIP services, they will encounter the following problems: when the private network PC initiates a call to the public network PC or gateway, there may be a single call or no communication; the public network PC or telephone ( When dialing a PC in the private network through the gateway), the call cannot be established. The reason is that when the PC initiates a call to the called PC or gateway on the public network, the call signaling of the private network PC can be correctly sent to the public network through NAT conversion, but due to the media channel contained in the data part of the call signaling The source address and source port number belong to the private network, and the NAT router can only convert the source address and source port number of the IP packet header, but cannot modify the data part, so the called party cannot know the address of the calling party's media channel, so Cause single-pass or no-pass phenomenon. When a public network PC or telephone (through a gateway) initiates a call to a private network PC, the called channel (or control channel) address queried by the called party’s VoIP server is a private network address, so the call cannot be established with the called PC connect.
发明内容Contents of the invention
本发明的目的就在于提供一种在包括以NAT技术组网的私有网络在内的网络上实现IP语音业务的方法及系统,使私网用户可以应用VoIP业务。The purpose of the present invention is to provide a method and system for realizing IP voice service on the network including private network with NAT technology, so that private network users can apply VoIP service.
本发明一种私有网络的IP语音业务的实现方法,至少包括以下步骤:The realization method of the IP voice service of a kind of private network of the present invention, comprises the following steps at least:
a)私网用户PC客户端向VoIP服务器发送登录请求消息后,向地址服务器发起n次TCP/UDP连接,n的取值为私网用户PC客户端与VoIP服务器完成一次通话需要建立的呼叫信令通道和逻辑通道的总的通道数;a) After the private network user PC client sends a login request message to the VoIP server, it initiates n times of TCP/UDP connections to the address server, and the value of n is the call signal that needs to be established to complete a call between the private network user PC client and the VoIP server. The total number of channels of the command channel and the logical channel;
b)地址服务器将所接收的连接报文中的经过NAT转换之后的公网地址和端口号分别发送给VoIP服务器和私网用户PC客户端;b) The address server sends the public network address and port number after NAT conversion in the received connection message to the VoIP server and the private network user PC client respectively;
c)VoIP服务器将对应呼叫信令通道的公网地址和端口号记录在该PC客户端的地址域中,私网用户PC客户端将对应逻辑通道的公网地址和c) The VoIP server records the public network address and port number of the corresponding call signaling channel in the address field of the PC client, and the private network user PC client records the public network address and port number of the corresponding logical channel.
端口号记录在相应的地址域中。The port number is recorded in the corresponding address field.
根据上述本发明方法,进一步包括以下步骤:According to above-mentioned method of the present invention, further comprise the following steps:
d)私网用户的PC客户端设置定时机制,使超过规定时间没有接收到呼d) The PC client of the private network user sets a timing mechanism so that no call is received beyond the specified time.
叫报文时,重复步骤a)、步骤b)、步骤c)。When calling a message, repeat step a), step b), step c).
根据上述本发明方法,所述的步骤b)中,地址服务器直接将对应逻辑通道的经过NAT转换之后的公网地址和端口号发送给私网用户PC客户端。According to the above-mentioned method of the present invention, in the step b), the address server directly sends the public network address and port number of the corresponding logical channel after NAT conversion to the private network user PC client.
根据上述本发明方法,所述的步骤b)中,地址服务器将所接收的连接报文中的经过NAT转换之后的公网地址和端口号发送给VoIP服务器,由VoIP服务器将对应逻辑通道的公网地址和端口号在响应私网用户PC客户端登录请求的应答消息中发送给该PC客户端。According to the above method of the present invention, in step b), the address server sends the public network address and port number after NAT conversion in the received connection message to the VoIP server, and the VoIP server sends the public network address and port number of the corresponding logical channel to the VoIP server. The network address and port number are sent to the PC client in a reply message in response to the login request of the private network user's PC client.
本发明一种实现私有网络IP语音业务的系统,至少包括通过IP网络连接的IP语音业务(VoIP)服务器、PC客户端、网守、网关以及通过公共交换电话网连接的交换设备和电话终端设备,其中PC客户端提供PC用户进行语音通话,实现协议处理和呼叫控制以及语音数据的转换,VoIP服务器实现PC客户端的登录认证、响应呼叫接入请求以及进行呼叫接续的控制,网关是连接IP网络与公共交换电话网的设备,网守提供网关地址的查找功能;其特征在于:The present invention is a system for realizing private network IP voice service, which at least includes IP voice service (VoIP) server connected through IP network, PC client, gatekeeper, gateway, switching equipment and telephone terminal equipment connected through public switched telephone network , where the PC client provides PC users with voice calls, realizes protocol processing, call control, and voice data conversion, and the VoIP server implements login authentication of the PC client, responds to call access requests, and controls call connection. The gateway is connected to the IP network. With the equipment of the public switched telephone network, the gatekeeper provides the search function of the gateway address; it is characterized in that:
该系统进一步包括一地址代理服务器,通过IP网络与系统连接,其接收私网用户PC客户端在登录后发送的TCP/UDP连接报文,并将其中的经过NAT转换之后的公网地址和端口号分别发送给VoIP服务器和私网用户的PC客户端;The system further includes an address proxy server, connected to the system through an IP network, which receives the TCP/UDP connection message sent by the private network user PC client after logging in, and converts the public network address and port after NAT conversion The number is sent to the VoIP server and the PC client of the private network user respectively;
所述的VoIP服务器将对应呼叫信令通道的地址记录在该PC客户端的地址域中,私网用户PC客户端将对应逻辑通道的地址记录在相应的地址域,为私网用户PC客户端与VoIP服务器实现通话预留呼叫信令通道和逻辑通道。The VoIP server records the address of the corresponding call signaling channel in the address domain of the PC client, and the private network user PC client records the address of the corresponding logical channel in the corresponding address domain, which is the private network user PC client and The VoIP server realizes call reservation signaling channel and logical channel.
进一步的,所述的VoIP服务器由业务服务器和用户服务器组成。Further, the VoIP server is composed of a service server and a user server.
本发明通过在现有的VoIP系统中增加地址代理服务器,并在私网PC客户端登录阶段与地址代理服务器建立数据连接,把PC客户端和VoIP服务器之间完成通话所需要建立的通道,包括呼叫信令通道、逻辑通道,在登录阶段就预先建立,并用每个通道在经过NAT转换之后的源地址源端口号(公网地址)代替相应的PCC私网地址,实际起到了预留呼叫信令通道和逻辑通道(即语音通道)的作用,实现VoIP业务在NAT组网方式中的应用。The present invention increases the address proxy server in the existing VoIP system, and establishes a data connection with the address proxy server during the login stage of the private network PC client, and completes the channel that needs to be established between the PC client and the VoIP server, including The call signaling channel and logical channel are pre-established at the login stage, and the source address and source port number (public network address) of each channel after NAT conversion is used to replace the corresponding PCC private network address, which actually plays a role in the reservation of call signaling. The role of the command channel and the logical channel (that is, the voice channel) realizes the application of the VoIP service in the NAT networking mode.
附图说明Description of drawings
图1为本发明VoIP业务的NAT组网应用系统结构示意图。FIG. 1 is a schematic structural diagram of a NAT networking application system for VoIP services in the present invention.
图2为本发明一实施例系统结构示意图。Fig. 2 is a schematic diagram of the system structure of an embodiment of the present invention.
图3为图2所示实施例的私网PCC的登录流程。FIG. 3 is a login process of the private network PCC in the embodiment shown in FIG. 2 .
具体实施方式Detailed ways
为使本发明的目的、技术方案及有益效果更加清楚明白,以下参照附图并举实施例,对本发明进行进一步详细说明。In order to make the object, technical solution and beneficial effects of the present invention more clear, the present invention will be further described in detail below with reference to the accompanying drawings and examples.
本发明是在现有VoIP系统上增加一地址代理服务器,把完成一次通话需要在私网用户PC客户端和VoIP服务器之间建立的通道,包括呼叫信令通道和逻辑通道,通过地址代理服务器在用户登录阶段就预先建立,并且用每个通道在经过NAT转换之后的公网地址及端口号代替原来PCC的私网地址及端口号。The present invention adds an address proxy server to the existing VoIP system, and completes a call that needs to be established between the private network user PC client and the VoIP server, including call signaling channels and logical channels, through the address proxy server. The user login stage is established in advance, and the original private network address and port number of PCC are replaced with the public network address and port number of each channel after NAT conversion.
参见图1所示的VoIP业务的NAT组网应用系统结构示意图,其中VoIP业务服务器10、PC用户的PC客户端和电话用户的网关(GW)11均通过IP网络连接,该系统进一步包括一地址代理服务器12,亦通过IP网络与系统连接。私网用户的PC客户端13通过NAT路由器14接入公网。Referring to the schematic structural diagram of the NAT networking application system for VoIP services shown in Figure 1, wherein the
当私网的PC客户端向VoIP服务器发出登录请求后,向地址代理服务器发起若干次TCP或UDP连接,连接的次数等于私网PCC与VoIP完成一次通话所需要建立的通道数。这些连接按顺序编号为通道1(channel 1)、通道2(channel 2)、......、通道n(channel n),它们的源地址是私网PCC的地址,源端口号分别是所需通道的默认端口号,将这几组地址(包括IP地址和端口号)称为PrivateNetAddr1、PrivateNetAddr2、......、PrivateNetAddrn。这些地址属于私网地址。When the PC client of the private network sends a login request to the VoIP server, it initiates several TCP or UDP connections to the address proxy server. The number of connections is equal to the number of channels that need to be established for a call between the private network PCC and VoIP. These connections are numbered in sequence as channel 1 (channel 1), channel 2 (channel 2), ..., channel n (channel n), their source addresses are the addresses of the private network PCC, and the source port numbers are respectively The default port number of the required channel, these groups of addresses (including IP addresses and port numbers) are called PrivateNetAddr1, PrivateNetAddr2, . . . , PrivateNetAddrn. These addresses are private network addresses.
通过NAT转换后,这些连接报文到达地址代理服务器,此时源地址经过地址/端口转换,已经转换为公网地址,称为PubicNetAddr1、PublicNetAddr2、......、PublicNetAddrn。地址代理服务器将这些地址发送给VoIP服务器和私网内的PC客户端,VoIP服务器将对应呼叫信令通道的公网地址记录在该私网PC客户端的地址域中,私网内的PC客户端将对应逻辑通道的公网地址记录在相应的地址域中。After NAT conversion, these connection packets reach the address proxy server. At this time, the source address has been converted into a public network address after address/port conversion, which is called PublicNetAddr1, PublicNetAddr2, ..., PublicNetAddrn. The address proxy server sends these addresses to the VoIP server and the PC client in the private network, and the VoIP server records the public network address corresponding to the call signaling channel in the address field of the PC client in the private network, and the PC client in the private network Record the public network address corresponding to the logical channel in the corresponding address field.
由于NAT对于在其地址转换表中的地址转换记录采用了定时机制,超过规定的时间没有更新连接,就会将记录从表中删去。所以需要设置一个定时重发机制,在规定的时间内PrivateNetAddr 1、2、3没有接收到报文,就需要自动向地址代理服务器(AddProxy)重新发送TCP或UDP连接。Because NAT adopts a timing mechanism for the address translation records in its address translation table, if the connection is not updated within the specified time, the records will be deleted from the table. Therefore, it is necessary to set up a timing retransmission mechanism. If PrivateNetAddr 1, 2, and 3 do not receive a message within the specified time, it needs to automatically resend the TCP or UDP connection to the address proxy server (AddProxy).
对于TCP连接,在创建时具有开始(SYN)标志,没有ACK标志(除了这个报文之外所有的TCP报文都有ACK标志)。在中断时会有终止(FIN)标志。因此对于TCP连接是依靠这些来判断TCP连接的中断和创建。For a TCP connection, it has a start (SYN) flag when it is created, and there is no ACK flag (all TCP packets except this message have an ACK flag). There will be a termination (FIN) flag when interrupted. Therefore, for TCP connections, these are used to judge the interruption and creation of TCP connections.
对于UDP报文,可以采用时间估计的方式。当某个连接在若干时间没有用时,就认为这个连接已经中断了。在一个新的连接发生时,就认为这个连接创建了。这种用于估算连接的时间提供用户可以配置。For UDP packets, a time estimation method may be used. A connection is considered broken when it has not been used for a certain amount of time. A connection is considered created when a new connection occurs. This time for estimating connections is provided user configurable.
通过上述的步骤,私网用户在上网登录时,在呼叫建立之前就建立了通话所需的通道,并由定时重发机制一直预留。呼叫发生时利用预留的呼叫信令通道和逻辑通道进行通话。用户下网后该预留通道自动拆除。Through the above steps, when the private network user logs in on the Internet, the channel required for the call is established before the call is established, and the channel is always reserved by the timing retransmission mechanism. When a call occurs, the reserved call signaling channel and logical channel are used for conversation. After the user logs off the network, the reserved channel is automatically removed.
私网的PC用户作为主叫应用VoIP业务时,通过预留的呼叫信令通道向所属的VoIP服务器发起呼叫,呼叫建立以后,需要建立逻辑通道。此时PCC将自己的逻辑通道地址通过呼叫信令报文地址发送给VoIP服务器,这时发出的逻辑通道地址是已经经过NAT转换后的公网地址。建立逻辑通道之后就可以顺利进行通话了。When a PC user on the private network serves as the calling party to apply VoIP services, he initiates a call to the VoIP server through the reserved call signaling channel. After the call is established, a logical channel needs to be established. At this time, the PCC sends its own logical channel address to the VoIP server through the address of the call signaling message, and the logical channel address sent at this time is the public network address after NAT conversion. After the logical channel is established, the call can proceed smoothly.
私网的PC用户作为被叫应用VoIP业务时,由公网的PC用户通过PCC或电话用户通过GW发起呼叫,VoIP服务器收到呼叫信令消息,查找到被叫的IP地址,此地址是经过NAT转换后的公网地址,将呼叫信令报文发送到此地址对应的路由器,路由器查询NAT地址转换表,找到相对应的私网地址,将呼叫信令转发到PC客户端,呼叫信令通道建立。私网用户的PC客户端回送的呼叫信令报文应包含逻辑通道地址,此地址是经过NAT转换后的公网地址。建立逻辑通道之后就可以顺利进行通话了。When a PC user on the private network is used as the called party to apply VoIP services, the PC user on the public network initiates a call through the PCC or the telephone user through the GW, and the VoIP server receives the call signaling message and finds the called IP address. The public network address after NAT conversion sends the call signaling packet to the router corresponding to this address. The router queries the NAT address translation table, finds the corresponding private network address, and forwards the call signaling to the PC client. The channel is established. The call signaling message sent back by the PC client of the private network user should contain the logical channel address, which is the public network address after NAT conversion. After the logical channel is established, the call can proceed smoothly.
本发明技术方案可以应用于任何IP语音业务。以下结合图2、图3所示,以个人号码业务(ONLY)为例,进一步说明本发明内容及应用。The technical scheme of the invention can be applied to any IP voice service. The content and application of the present invention will be further described below in conjunction with FIG. 2 and FIG. 3 , taking the personal number service (ONLY) as an example.
ONLY(One Number Link You)业务是随着互联网的发展,为满足用户越来越迫切地希望随时随地相互交流和获取信息的需要,在融合了传统电信技术和IP技术的基础上,开发的一种创新性业务。ONLY业务主要是为使用者提供唯一的个人业务号码--ONLY号码,通过该号码提供多种业务,无论用户身处何地都可以更快捷地联络到指定用户,被联络的用户可以通过PC、电话、语音信箱等多种方式接听呼叫。ONLY (One Number Link You) service is developed on the basis of traditional telecommunication technology and IP technology in order to meet the needs of users who are more and more eager to communicate with each other and obtain information anytime and anywhere with the development of the Internet. an innovative business. ONLY service is mainly to provide users with a unique personal business number -- ONLY number, through which a variety of services can be provided, no matter where the user is located, the designated user can be contacted more quickly, and the contacted user can use PC, Answer the call in various ways such as telephone and voice mail.
ONLY系统由ONLY业务服务器(ONLY Server)、ONLY用户服务器(User Server)、PC客户端(PCC)、交互式语音应答设备(IVR)、数据库、Web服务器等构成,其中ONLY业务系统的VoIP服务器功能是由ONLYServer和User Server共同完成。ONLY Server的主要功能有:响应节点呼叫接入请求;与User Server交互得到ONLY号码的地址翻译;与网守(GK)交互得到被叫电话号码的路由;根据解析的目的地址中转呼叫信令;完成对网关(MG)的控制功能;与IVR交互,支持透传双音多频(DTMF)号码;记录简单计费信息并向User Server发送。User Server的主要功能有:呼叫策略选择(根据用户设置的呼叫策略将虚拟ONLY号码翻译为真实电话号码或PCC的IP地址);PCC登录认证、呼出授权、状态维护;语音信息(VM)通告等。ONLY system consists of ONLY business server (ONLY Server), ONLY user server (User Server), PC client (PCC), interactive voice response device (IVR), database, Web server, etc., among which the VoIP server function of ONLY business system It is done jointly by ONLYServer and User Server. The main functions of the ONLY Server are: responding to node call access requests; interacting with the User Server to obtain the address translation of the ONLY number; interacting with the gatekeeper (GK) to obtain the route of the called phone number; relaying call signaling according to the resolved destination address; Complete the control function of the gateway (MG); interact with the IVR, support transparent transmission of dual-tone multi-frequency (DTMF) numbers; record simple billing information and send it to the User Server. The main functions of User Server include: call strategy selection (translate the virtual ONLY number into a real phone number or PCC IP address according to the call strategy set by the user); PCC login authentication, outgoing call authorization, status maintenance; voice message (VM) notification, etc. .
和其他的VoIP系统一样,ONLY业务在私网环境应用中也遇到了相同的问题。参见图2所示,为采用本发明技术方案的ONLY业务的NAT组网应用系统结构示意图。其中除ONLY系统原有的设备外,新增加一地址代理服务器12,通过IP网络接入系统,并对原有用户服务器和PC客户端做出相应的改动,使PCC在向User Server登录后,向地址代理服务器发起UDP连接,地址代理服务器将其中的经过NAT转换的公网地址发送给UserServer,并由User Server在返回PCC的登录请求响应中将经过NAT转换的公网地址传送给PCC,User Server和PCC分别记录该等经过NAT转换的公网地址,为私网PCC与ONLY Server建立呼叫预留需要的通道。Like other VoIP systems, the ONLY service also encounters the same problem in the application of the private network environment. Referring to FIG. 2 , it is a schematic structural diagram of the NAT networking application system of the ONLY service adopting the technical solution of the present invention. In addition to the original equipment of the ONLY system, a new
私网PCC的登录流程参见图3所示,其中,私网中ONLY号码为16655551234的PCC 13向User Server 102发起登录请求消息(Login_Req)后,需要向地址代理服务器12发起三个UDP连接。将这三个UDP连接按顺序编号为UDP1、UDP2、UDP3,它们的源地址是私网PCC 13的地址,UDP1、2、3的源端口号分别是MGCP、RTP、RTCP的默认端口号。将这三组地址(包括IP地址和端口号)称为PrivateNetAddr1、PrivateNetAddr2、PrivateNetAddr3。这三组地址属于私网地址。The login process of the private network PCC is shown in Figure 3, wherein, after the
通过NAT转换后,上述三个UDP报文到达地址代理服务器12,此时源地址经过地址/端口转换,已经变为了公网地址,将其称为PublicNetAddr1、PublicNetAddr2、PublicNetAddr3。地址代理服务器12将这三组公网地址发送给User Server 102,User Server 102将PublicNetAddr1记录在PCC16655551234的地址域,并在登录请求响应消息(Login Ack)中将PublicNetAddr2和PublicNetAddr3发送给PCC,PCC记录这两个地址。After NAT conversion, the above three UDP packets arrive at the
由于NAT对于UDP连接在地址转换表中的地址转换记录采用了定时机制,超过规定的时间没有更新连接,就会将记录从表中删去。所以在没有通话发生的时候,PCC也必须与地址代理服务器12保持UDP1、2、3的连接,所以需要设置一个特殊的定时重发机制,在规定的时间内PrivateNetAddr1、2、3没有接收到报文,就自动向地址代理服务器12重复发送UDP报文。Because NAT adopts a timing mechanism for the address translation records of UDP connections in the address translation table, if the connection is not updated within the specified time, the records will be deleted from the table. Therefore, when there is no call, the PCC must also maintain UDP1, 2, and 3 connections with the
经过上述的登录过程,通过地址代理服务器12就预先建立起了私网PCC与ONLY Server之间呼叫需要建立的通道。Through the above-mentioned login process, the channel that calls need to be established between the private network PCC and the ONLY Server has been established in advance by the
当私网的PCC 16655551234作为主叫,发起呼叫的时候,建立呼叫信令通道,由于在路由器的地址转换表中保持有呼叫信令通道的私网和公网地址对应关系,所以这个呼叫信令实际是使用了上述预留的通道。PCC 13从源地址PrivateNetAddr1发送MGCP数据到ONLY Server 101,ONLY Server 101向User Server 102查询呼叫控制策略,向GK查询被叫网关或在User Server得到被叫ONLY号码所属ONLY Server的地址,开始建立逻辑通道。此时PCC 16655551234将自己的RTP和RTCP地址发送给被叫方,按照协议规定,应该是PrivateNetAddr2、PrivateNetAddr3。但如果送出私网地址,会造成对方的语音数据无法正确发送给PCC 16655551234,所以应该将PrivateNetAddr2、PrivateNetAddr3替换为PublicNetAddr2、PublicNetAddr3,这样就可以利用上述预留的逻辑通道,使通话顺利进行。When the PCC 16655551234 of the private network acts as the calling party and initiates a call, the call signaling channel is established. Since the address translation table of the router maintains the corresponding relationship between the private network and the public network address of the call signaling channel, the call signaling channel In fact, the channel reserved above is used.
当私网的PCC 16655551234作为被叫,由公网的PC用户的PCC或电话用户的GW发起呼叫的时候,ONLY Server 101接收到呼叫信令,首先向User Server 102查询PCC 16655551234的地址,查找到PublicNetAddr1,将MGCP数据发送到PublicNetAddr1,路由器查询NAT的地址转换表,找到相对应的PrivateNetAddr1,将MGCP数据发送到PCC 16655551234。当呼叫信令通道建立之后,PCC16655551234回送MGCP的数据应包含RTP和RTCP地址,按照协议规定,应该是PrivateNetAddr2、PrivateNetAddr3。但如果送出私网地址,会造成主叫方的语音数据无法正确发送给PCC16655551234,所以应该将PrivateNetAddr2、PrivateNetAddr3替换为PublicNetAddr2、PublicNetAddr3,这样就利用了上述预留的逻辑通道进行语音数据的传输。When the PCC 16655551234 on the private network is called, and the PCC of the PC user on the public network or the GW of the telephone user initiates a call,
本发明技术方案可应用于任何的VoIP业务系统。如为测试IP电话网关(Master、Refiner、Expert)的互通性,用H.323协议实现的一个在PC上进行IP电话的应用。这个应用在私网环境下也遇到了与ONLY业务同样的问题。将本发明技术方案引入到上述应用中,所不同的是:呼叫信令使用H.225协议,用TCP建立通道。然后建立H.245的控制通道,最后建立一个逻辑通道。但呼叫建立的基本思想和ONLY是有共同之处的。按照这个方案进行改进以后,使其在私网环境下也能顺利通话,通过了基本功能验证和大呼叫量测试。The technical scheme of the invention can be applied to any VoIP service system. For example, in order to test the interoperability of IP telephone gateways (Master, Refiner, Expert), an application for IP telephony on a PC is implemented using the H.323 protocol. This application also encountered the same problem as the ONLY business in the private network environment. The technical scheme of the present invention is introduced into the above-mentioned application, the difference is that: the call signaling uses the H.225 protocol, and TCP is used to establish the channel. Then establish the control channel of H.245, and finally establish a logical channel. But the basic idea of calling and ONLY have something in common. After improving according to this plan, it can make calls smoothly in the private network environment, and passed the basic function verification and large call volume test.
综上所述,本发明方案简便易行,具有很好的实用性和可靠性。In summary, the solution of the present invention is simple and easy to implement, and has good practicability and reliability.
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| CNB011356103ACN1170393C (en) | 2001-10-04 | 2001-10-04 | Method and system for realizing IP voice service in private network |
| PCT/CN2002/000371WO2003030463A1 (en) | 2001-10-04 | 2002-05-30 | A method and system for realizing ip voice service at private network |
| Application Number | Priority Date | Filing Date | Title |
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| CNB011356103ACN1170393C (en) | 2001-10-04 | 2001-10-04 | Method and system for realizing IP voice service in private network |
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| CN1411220Atrue CN1411220A (en) | 2003-04-16 |
| CN1170393C CN1170393C (en) | 2004-10-06 |
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| CNB011356103AExpired - Fee RelatedCN1170393C (en) | 2001-10-04 | 2001-10-04 | Method and system for realizing IP voice service in private network |
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| CN (1) | CN1170393C (en) |
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| C06 | Publication | ||
| PB01 | Publication | ||
| C10 | Entry into substantive examination | ||
| SE01 | Entry into force of request for substantive examination | ||
| C14 | Grant of patent or utility model | ||
| GR01 | Patent grant | ||
| C17 | Cessation of patent right | ||
| CF01 | Termination of patent right due to non-payment of annual fee | Granted publication date:20041006 Termination date:20111004 |