WO2003030463A1

WO2003030463A1 - A method and system for realizing ip voice service at private network

Info

Publication number: WO2003030463A1
Application number: PCT/CN2002/000371
Authority: WO
Inventors: Haitao Lin; Yiguo Li; Quan Gan
Original assignee: Huawei Technologies Co., Ltd.
Priority date: 2001-10-04
Filing date: 2002-05-30
Publication date: 2003-04-10
Also published as: CN1170393C; CN1411220A

Abstract

The invention discloses a method and system for realizing IP voice service at network including private network established with NAT technology. Through increasing the number of address proxy servers and establishing TCP/UDP data link with address proxy server during PC client logon in private network, all channels required for communication between PC client and VoIP server are set up in advance during logon, which include call signaling channels and logical channels. The PCC private network addresses are replaced with source port of source address (public network address) obtained by apply the NAT conversion to each channel, in fact it serves as reservation of call signaling channel and logical channel (or voice channel), so as to realize application of VoIP service in network established with NAT technology.

Description

Method and system for implementing IP voice service in private network

The present invention relates to the field of data communication technology, and in particular, to a method and system for implementing an IP voice service (VoIP) on a network including a private network using a network address translation (NAT) technology. Background of the invention

The traditional voice service uses circuit switching technology, that is, the two parties of the call establish a fixed bandwidth (64kbii _S ) circuit through the public switched telephone network (PSTN), which guarantees low-latency and low-distortion real-time communication service quality (QoS). However, in this way, the network bandwidth utilization is low, communication costs are high, and the promotion of value-added services is difficult.

VoIP (Voice over IP) refers to the service of voice transmission in a computer network using IP as the network layer protocol, referred to as the IP voice service. Computer communication uses packet switching technology, which is called "packet", and each communication node in the network stores and forwards packets. Therefore, VoIP belongs to the category of packet voice communication, which is the result of the integration of two mainstream technologies: network packetization marked by IP and network service integration targeted at multimedia. The characteristics of packet switching are high network utilization, low communication costs, and IP networks as open networks make it easy to quickly promote new services.

Research on voice communications over computer networks dates back to the 1970s. After entering the 1990s, especially in recent years, with the rise of the Internet worldwide and the development of speech coding technology, VoIP has made breakthrough progress and has been practically applied. Its development is roughly divided into the following three stages:

(1) Germination period: After the Internet appeared, people first tried to let two PCs pass IP networks carry voice, which was an original form of VoIP. At this stage, people's understanding of VoIP is superficial and the market is unclear.

(2) Development period: attracted by the many advantages of VoIP, many telecommunications companies realize that IP phones between PCs are far from meeting the needs of users, so they are focusing on many PSTN users, which leads to IP phones. The emergence of the gateway. The IP phone gateway connects the PSTN network and the IP network, and provides a variety of VoIP service forms such as phone to phone, PC to phone, and phone to PC. This stage is a period of rapid development of IP phones.

(3) Maturity period: The development of IP telephony applications will enter a mature period in the future, with unified standards for communication, voice equipment of telecommunications manufacturers around the world can communicate with each other, and VoIP voice quality is close to traditional telephone quality.

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, that is, call control signaling in a telecommunications network, includes address information, user status information, and dual-tone multi-frequency (DTMF) signals. Generally, Transmission Control Protocol (TCP) is used as a transport layer protocol. The voice information transfer protocol specifies how voice packets are encapsulated, multiplexed, and transmitted, including how various voice-encoded packets are assembled and identified. It is required to be transmitted in real time, and only User Datagram Protocol (UDP) can be used as the transport layer protocol. IP phones use real-time transmission protocol (RTP) to adapt to packet voice data. RTP can transmit information such as packet timestamps and packet sequence numbers, and supports QoS monitoring and multi-data stream merging. In addition, to ensure real-time performance, a Resource Reservation Protocol (RSVP) is also selected to reserve network resources for calls. At present, when VoIP is implemented, a call signaling channel is first established. 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.

A private network is an internal network that uses a private IP address, such as an enterprise network. Most of today's corporate networks are intranets that implement the TCP / IP protocol. Users use DHCP to dynamically assign private IP addresses and divide multiple network segments on the private network. User data often crosses the local network segment. Transfer between subnets in the private network. Within the corporate network, the branch or headquarters at the same location uses Layer 3 switching technology (also known as IP switching technology, high-speed routing technology, etc., which is a mechanism that uses the information in the Layer 3 protocol to strengthen the Layer 2 switching function) through Ethernet. Networks are connected, and the branch network usually has a unique exit from the public network. When users access the public network, they must rely on the NAT function of the proxy server.

NAT is the address translation or address proxy, which is used to implement 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 address allocation organization specifies that the following three network address segments are reserved for private addresses:

10.0.0.0 ~ 10.255.255.255

172.16.0.0 ~ 172.31.255.255

192.168.0.0 ~ 192.168.255.255

In other words, these three network address segments will not be allocated on the Internet, but can be used on a private (local area network) internal network. Each private network selects an appropriate network address based on the number of hosts in the foreseeable future. The internal network addresses of different private networks can be the same.

NAT technology is widely used on private networks such as corporate networks to overcome the limitation of address space and increase security, as well as to solve the networking problem of multiple subnets. Network address translation provides the ability to convert multiple internal (private network) IP addresses to external (public network) IP addresses by multiplexing TCP or UDP port numbers. There are many types of address translation methods. 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.

For the multiple methods of NAT, a common feature can be summarized. When private network users want to use the services of the public network, they need to obtain a public network address (including the IP address and port number) through NAT. Corresponds to an address port on the NAT router In the table, the corresponding two in the table are the internal private IP address and the source port number of the IP packet sent to the outside, which is also the destination port number of the IP packet received from the outside. From the perspective of 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 directly send this public network address. The router stores the address in the address-port correspondence table by querying it. The corresponding relationship is forwarded to the private network user.

To put it simply, address translation technology (NAT) is to replace the internal IP address and port with the external network IP address and port, and reverse translation. The advantages of NAT technology are:

1. Provide hosts on the internal network to access resources outside the network;

2. Provide "Privacy" protection for internal hosts to improve security;

3. Solve the problem of IP address resources.

When private network users use the existing VoIP system to apply VoIP services, they will encounter the following problems: When a private network PC initiates a call to a public network PC or gateway, a single-way or unavailable call may occur; the public network PC or phone ( When a PC on the private network is dialed through the gateway, the call cannot be established. The reason is: 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. The source address and source port number belong to the private network. The NAT router can only translate the source address and source port number of the IP packet header. The data part cannot be modified. Therefore, the called party cannot know the address of the media channel of the calling party. Causes a single-pass or no-pass phenomenon. When a public network PC or phone initiates a call (via a gateway) 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 a call cannot be established with the called PC Connection Summary

The purpose of the present invention is to provide a method and system for implementing IP voice services on a network including a private network networked with NAT technology, so that private network users can apply VoIP business.

A method for implementing an IP voice service in a private network according to the present invention includes at least the following steps:

a) After the private network user PC client sends a login request message to the VoIP server, it initiates n TCP / UDP connections to the address server. The value of n is the call letter that the private network user PC client needs to establish to complete a call with the VoIP server. Let the total number of channels and logical channels;

b) the address server sends the public network address and port number in the received connection packet after NAT conversion to the VoIP server and the private network user PC client c) the VoIP server sends the public network address and The port number is recorded 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 in the corresponding address field.

According to the method of the present invention, the method further includes the following steps:

d) The PC client of the private network user sets a timing mechanism so that when no call message is received within the prescribed time, repeat steps a), b), and c). According to the method of the present invention, in step b), the address server directly sends the public network address and port number of the corresponding logical channel after NAT conversion to the PC client of the private network user.

According to the 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 packet to the VoIP server, and the VoIP server sends the public address of the corresponding logical channel. The network address and port number are sent to the PC client in a response message in response to the PC client login request from the private network user.

A system for implementing a private network IP voice service includes at least an IP voice service (VoIP) server, a PC client, a gatekeeper, a gateway, and a communication terminal connected through an IP network. Switching equipment and telephone terminal equipment connected through the public switched telephone network. The PC client provides PC users for voice calls, protocol processing and call control, and conversion of voice data. The VoIP server implements PC client login authentication and responds to call access. Request and control of call connection. The gateway is a device that connects the IP network with the public switched telephone network. The gatekeeper provides the function of finding the gateway address. It is characterized by:

The system further includes an address proxy server, which is connected to the system through the IP network, and receives a TCP UDP connection packet sent by a private client PC client after login, and converts the public network address and port number thereof after NAT conversion. Send to VoIP server and PC client of private network user respectively;

The VoIP server records the address of the corresponding call signaling channel in the address field of the PC client, and the private network user PC client records the address of the corresponding logical channel in the corresponding address field. The VoIP server implements the reserved call signaling channel and the reversion channel during the call.

Further, the VoIP server is composed of a service server and a user server. By adding an address proxy server to the existing VoIP system and establishing a data connection with the address proxy server during the login phase of the private network PC client, the present invention includes a channel needed to establish a call between the PC client and the VoIP server, including Call signaling channels and logical channels are pre-established during the login phase, and the corresponding PCC private network address is replaced by the source address source port number (public network address) of each channel after NAT conversion, which actually serves to reserve the call information. The role of the channel and logical channel (that is, the voice channel), to achieve the application of VoIP services in NAT networking. Brief description of the drawings

FIG. 1 is a schematic structural diagram of a NAT networking application system of a VoIP service according to the present invention.

FIG. 2 is a schematic diagram of a system structure according to an embodiment of the present invention. FIG. 3 is a login flow of the private network PCC in the embodiment shown in FIG. 2. Mode of Carrying Out the Invention

The present invention is described in detail below with reference to the drawings.

The present invention adds an address proxy server to the existing VoIP system. The channel that needs to be established between the PC client of the private network user and the VoIP server to complete a call includes call signaling channels and logical channels. The user login phase is established in advance, and the public network address and port number of each channel after NAT conversion are used to replace the original PCC private network address and port number.

Referring to the schematic diagram of the NAT networking application system structure of the VoIP service shown in FIG. 1, where the VoIP service server 10, the PC client's PC client, and the telephone user's gateway (GW) 11 are all connected through an IP network, the system further includes an address The proxy server 12 is also connected to the system through an IP network. The PC client 13 of the private network user accesses the public network through the NAT router 14.

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 PCC and VoIP need to establish a call to complete. These connections are numbered sequentially as channel 1

(channel 1), channel 2 (channel 2), channel n (channel n), their source address is the address of the private network PCC, and the source port numbers are the default port numbers of the required channels, and these groups of addresses ( Including IP address and port number) called PrivateNetAddrl,

PrivateNetAddr2 PrivateNetAddrn. These addresses are private addresses.

After the NAT conversion, these connection packets reach the address proxy server. At this time, the source address has been converted to a public network address through address / port conversion, which is called PublicNetAddrl,

PublicNetAddr2 PublicNetAddrn. The address proxy server sends these addresses to the VoIP server and the PC client in the private network. The VoIP server will communicate the corresponding call signaling. The public network address of the channel is recorded in the address field of the PC client of the private network, and the PC client in the private network records the public network address of the corresponding logical channel in the corresponding address field.

Because NAT uses a timing mechanism for the address translation records in its address translation table, if the connection is not updated within the prescribed time, the record will be deleted from the table. Therefore, a regular retransmission mechanism needs to be set up. PrivateNetAddr 1, 2, 3 does not receive a message within a specified time, and it needs to automatically resend the TCP or UDP connection to the address proxy server (AddProxy).

For a TCP connection, it has a start (SYN) flag when it is created, and no ACK flag (all TCP packets except this message have an ACK flag). There will be a FIN (FIN) flag during the interrupt. Therefore, for TCP connections, these are used to determine the interruption and creation of TCP connections.

For UDP packets, time estimation can be used. When a connection is not used for a certain amount of time, it is considered that the connection has been broken. When a new connection occurs, the connection is considered to have been created. This time for estimating the connection is provided by the user and can be configured.

Through the above steps, when a private network user logs on to the Internet, the channel required for the call is established before the call is established, and it is always reserved by the timing retransmission mechanism. When a call occurs, the reserved call signaling channel and logical channel are used for conversation. This reserved channel is automatically removed after the user goes offline.

When a PC user on a private network uses the VoIP service as a calling party, a call is initiated to the VoIP server to which the VoIP server belongs through a 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 call signaling address, and the logical channel address issued at this time is the public network address that has been NAT-translated. After the logical channel is established, the conversation can proceed smoothly.

When a PC user in a private network uses the VoIP service as a called party, a PC user in a public network initiates a call through a PCC or a telephone user through a GW, and the VoIP server receives a call signaling message. Find the called IP address. This address is the public network address after NAT conversion. Send the call signaling packet to the router corresponding to this address. The router queries the NAT address translation table to find the corresponding private network address. The call signaling is forwarded to the PC client, and the call signaling channel is established. The call signaling returned by the PC client of the private network user should include the logical channel address, which is the public network address after NAT translation. After the logical channel is established, the conversation can proceed smoothly.

The technical solution of the present invention can be applied to any IP voice service. The content and application of the present invention will be further described below with reference to FIG. 2 and FIG. 3 and taking the personal number service (ONLY) as an example.

The ONLY (One Number Link You) service is developed along with the development of the Internet. In order to meet the needs of users who are more and more eager to communicate and obtain information with each other anytime, anywhere, it is developed based on the integration of traditional telecommunications technology and IP technology. Kind of innovative business. ONLY business is mainly to provide users with a unique personal business number-ONLY number, through which a variety of services are provided, no matter where the user is located, he can more quickly contact the designated user, the contacted user can use the PC, Answer calls by phone, voice mail, etc.

The ONLY system consists of the 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 the ONLY business system It is completed by ONLY Server and User Server. The main functions of ONLY Server are: responding to the call access request of the node; obtaining the address translation of the ONLY number with the User Server fork; interacting with the gatekeeper (GK) to obtain the route of the called phone number; forwarding the call letter according to the resolved destination address Order; Complete the control function of the gateway (MG); Interact with the IVR, support the transparent transmission of Han tone multi-frequency (DTMF) numbers; Record simple billing information and send it to the User Server. User Server's main functions are: Call policy selection (translate virtual ONLY number to real phone number or PCC IP address according to the call policy set by the user); PCC login authentication, outgoing call authorization, status maintenance; voice message (VM) announcement, etc. . Like other VoIP systems, ONLY services have encountered the same problems in private network environment applications. Referring to FIG. 2, it is a schematic structural diagram of a NAT networking application system of an ONLY service adopting the technical solution of the present invention. In addition to the original equipment of the ONLY system, an address proxy server 12 is newly added to access the system through the IP network, and corresponding changes are made to the original user server and PC client, so that the PCC logs in to the User Server. Initiate a UDP connection to the address proxy server. The address proxy server sends the NAT-translated public network address to the User Server, and the User Server transmits the NAT-translated public network address to the PCC in the response to the PCC login request. User Server and PCC record these NAT-translated public network addresses, respectively, and reserve channels needed for the private network PCC and ONLY server to establish a call.

The private network PCC login process is shown in Figure 3. Among them, the PCC 13 with the ONLY number of 16655551234 in the private network initiates a login request message (Login_Req) to the User Server 102, and needs to initiate three UDP connections to the address proxy server 12. . The three UDP connections are sequentially numbered as UDP1, UDP2, and UDP3. Their source addresses are the addresses of the private network PCC 13, and the source port numbers of UDP1, 2, and 3 are the default port numbers of MGCP, RTP, and RTCP. These three sets of addresses (including IP address and port number) are called PrivateNetAddrl, PrivateNetAddr2, PrivateNetAddr3. These three groups of addresses belong to private network addresses.

After the NAT conversion, the above three UDP packets reach the address proxy server 12. At this time, the source address has been converted to a public network address through address / port conversion, which is called PublicNetAddrK PublicNetAddr2, PublicNetAddr3. The address proxy server 12 sends these three sets of public network addresses to User Server 102. User Server 102 records PublicNetAddrl in the address field of PCC 16655551234, and sends PublicNetAddr2 and PublicNetAddr3 to the PCC in a login request response message (Login—Ack). The PCC records these two addresses. Because NAT uses a timing mechanism for the address translation records in the address translation table for UDP connections, if the connection is not updated within the prescribed time, the records will be deleted from the table. Therefore, when no call occurs, the PCC must also maintain a UDP 1, 2, 3 connection with the address proxy server 12, so a special timer retransmission mechanism needs to be set up. PrivateNetAddrl, 2, 3 do not receive the report within the specified time. Send the UDP packet to the address proxy server 12 automatically and repeatedly.

After the above login process, the address proxy server 12 has previously established a channel to be established between the private network PCC and the ONLY Server call.

When PCC 16655551234 of the private network is used as the calling party to initiate a call, a call signaling channel is established. Since the correspondence between the private network and the public network address of the call signaling channel is maintained in the address translation table of the router, this call signaling Actually, the above reserved channel is used. PCC 13 sends MGCP data from the source address PrivateNetAddrl to ONLY Server 101. ONLY Server 101 queries User Server 102 for the call control policy, queries the called gateway of the GK, or obtains the address of the ONLY Server to which the called ONLY number belongs from the User Server, and begins to establish logic. aisle. At this time, PCC 16655551234 sends its own RTP and RTCP addresses to the called party. According to the agreement, it should be PrivateNetAddr2, PrivateNetAddr3. However, if the private network address is sent, the other party's voice data cannot be sent to PCC 16655551234 correctly, so PrivateNetAddr2, PrivateNetAddr3 should be replaced with PublicNetAddr2, PublicNetAddr3, so that the reserved logical channel can be used to make the call proceed smoothly.

When PCC 16655551234 of the private network is called and the call is initiated by the PCC of the PC user of the public network or the GW of the phone user, ONLY Server 101 receives the call signaling and first queries User Server 102 for the address of PCC 16655551234 and finds Go to PublicNetAddrl and send MGCP data to PublicNetAddrl. The router queries the NAT address translation table, finds the corresponding PrivateNetAddrl, and sends the MGCP data to PCC 16655551234. After the call signaling channel is established, the data returned by PCC 16655551234 to the MGCP should include the RTP and RTCP addresses. According to the protocol, it should be PrivateNetAddr2, PrivateNetAddr3. However, if the private network address is sent, the caller's voice data will not be sent to PCC 16655551234 correctly. Therefore, PrivateNetAddr2 and PrivateNetAddr3 should be replaced with PublicNetAddr2 and PublicNetAddr3. In this way, the reserved logical channels are used for voice data transmission.

The technical solution of the present invention can be applied to any VoIP service system. For example, to test the interoperability of the IP phone gateway (Master, Refmer, Expert), an H.323 protocol is used to implement an IP phone application on a PC. This application also encountered the same problems in the private network environment as the ONLY business. The technical solution of the present invention is introduced into the above application with the following differences: The call signaling uses the H.225 protocol and the TCP is used to establish a channel. Then establish the H.245 control channel, and finally establish a logical channel. But the basic idea of call establishment and ONLY have something in common. After improving according to this scheme, it can make a smooth call even in the private network environment, and passed the basic function verification and large call volume test.

In summary, the solution of the present invention is simple and easy to implement, and has good practicability and reliability.

Claims

Claim

1. A method for implementing IP voice services in a private network, characterized in that the method includes at least the following steps:

(NAT) after the public network address and port number are sent to the VoIP server and the private network user PC client;

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 in the corresponding address field in.

2. The method for implementing an IP voice service according to claim 1, further comprising the following steps:

d) The PC client of the private network user sets a timing mechanism so that when no call message is received within the prescribed time, repeat steps a), b), and c).

3. The method for implementing an IP voice service according to claim 1, wherein 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. Web client PC client.

4. The method for implementing an IP voice service according to claim 1, wherein in the step b), the address server passes the NAT in the received connection packet through NAT. The converted public network address and port number are sent to the VoIP server, and the VoIP server sends the public network address and port number of the corresponding logical channel to the PC client in a response message in response to the PC client login request from the private network user.

5. A system for implementing a private network IP voice service, including at least an IP voice service (VoIP) server, PC client, gatekeeper, gateway connected through an IP network, and switching equipment and telephone terminal equipment connected through a public switched telephone network Among them, the PC client provides PC users for voice calls, to implement protocol processing and call control, and to convert voice data.

The VoIP server implements the login authentication of the PC client, responds to call access requests, and controls call continuity. The gateway is a device that connects the IP network with the public switched telephone network. The gatekeeper provides a gateway address lookup function. Its characteristics are:

6. The IP voice service system according to claim 3, further comprising: a VoIP server consisting of a service server and a user server.