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

A method and system for realizing ip voice service at private network Download PDF

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Publication number
WO2003030463A1
WO2003030463A1 PCT/CN2002/000371 CN0200371W WO03030463A1 WO 2003030463 A1 WO2003030463 A1 WO 2003030463A1 CN 0200371 W CN0200371 W CN 0200371W WO 03030463 A1 WO03030463 A1 WO 03030463A1
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WIPO (PCT)
Prior art keywords
address
server
network
private network
pc
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PCT/CN2002/000371
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French (fr)
Chinese (zh)
Inventor
Haitao Lin
Yiguo Li
Quan Gan
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Huawei Technologies Co., Ltd.
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Filing date
Publication date
Priority to CN01135610.3 priority Critical
Priority to CNB011356103A priority patent/CN1170393C/en
Application filed by Huawei Technologies Co., Ltd. filed Critical Huawei Technologies Co., Ltd.
Publication of WO2003030463A1 publication Critical patent/WO2003030463A1/en

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Classifications

    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M7/00Interconnection arrangements between switching centres
    • H04M7/006Networks other than PSTN/ISDN providing telephone service, e.g. Voice over Internet Protocol (VoIP), including next generation networks with a packet-switched transport layer
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M3/00Automatic or semi-automatic exchanges
    • H04M3/42Systems providing special services or facilities to subscribers
    • H04M3/42314Systems providing special services or facilities to subscribers in private branch exchanges

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

Private network IP voice service method and system FIELD

Technical Field The present invention relates to data communications, and more particularly to methods and systems for IP voice traffic (VoIP) on (NAT) technology network including a private network comprising a network A network address translation. BACKGROUND OF THE INVENTION

Traditional voice service using circuit-switched technology, namely the establishment of a call both fixed bandwidth (64kbii S) circuit through the public switched telephone network (PSTN), which makes low-latency, low distortion of real-time communication quality of service (QoS) is guaranteed, but in this way the network bandwidth utilization is low, high communication costs, and promote value-added services more difficult.

VoIP (Voice over IP) traffic is voice transmission refers to a computer network in the IP network layer protocol, IP for short voice service. Computer communications using packet switching technology, called "packet" (Packet), each communications network node stores and forwards the packet. Therefore, VoIP belongs to the category of packet voice communications, IP-based network packets marked and targeted multimedia network integration of the two main business technology convergence results. Features packet-switched network utilization is high, low-cost communications, and IP network as an open network is vulnerable to rapid adoption of new business.

Research for voice communications over computer networks from the 1970s. Entering the 1990s, especially in recent years, with the development of the global scope of the Internet and the rise of speech coding technology, VoIP has made a breakthrough and has been applied. Its development is roughly divided into the following three stages:

(1) infancy: the emergence of Internet, people first tried to make two PC through voice over IP, which is a form of VoIP original. At this stage, people's understanding of Gen VoIP superficial, the market is not clear.

(2) development period: Attracted by the many advantages of VoIP, many telecommunications companies recognize that implementation of IP telephony only between the PC can not meet the needs of users, and therefore many of the sights PSTN users, which led to IP telephony gateway appears. IP telephony gateway to the PSTN network and IP network connected, providing phone to phone, PC to various forms of business VoIP phones, phone to PC and so on. This stage is a period of rapid development of IP telephony.

(3) maturity: development of IP telephony applications in the future will enter the mature stage, with a standard unified communications, global telecom equipment manufacturers can communicate voice, VoIP voice quality close to the quality of traditional telephone and so on.

Communication protocol IP voice services including voice communication control protocol, voice messaging protocol, conference call control protocols and real-time control protocol. Wherein the communication control protocol i.e. voice telecommunications network call control signaling, including address information, user status information, dual tone multi-frequency (DTMF) signal or the like, generally use the Transmission Control Protocol (TCP) as the transport layer protocol. Voice messaging protocol defines how a voice packet encapsulation, multiplexing and transmission, including how various speech encoding and packet assembling identification, requirements to real-time transmission, only with User Datagram Protocol (UDP) as the transport layer protocol. IP telephone using a Real-time Transport Protocol (RTP) packet adaptation speech data, the RTP timestamp may transmit packets, the packet sequence number and other information, QoS monitoring, and support multiple data streams are combined. In addition, to ensure real-time, also use the Resource Reservation Protocol (RSVP) for the call to conserve network resources. After the current implementation of VoIP are at the beginning of the call, first established call signaling channel, signaling interaction and consultation, to establish a logical channel that is a voice channel, the logical channel address of the packet needs to be transmitted in the call signaling channel.

Private network refers to the internal network to use private IP addresses, such as enterprise networks. Most of today's enterprise networks are implemented Intranet TCP / IP protocol, the user will use DHCP to assign private IP addresses, multiple network segments divided in the private network user data are often beyond the local network segment within each sub private network inter-network transmission. Internal corporate network, branches or headquarters in the same location using a third layer switching technology (also known as IP switching technology, high-speed routing technology, is a use of the third layer protocol information exchange mechanisms to enhance the function of the second layer) over Ethernet network connection, and the branch network to the public network industry generally only exit, when users access the public network, must resort to the NAT function of the proxy server.

I.e. address translation NAT or address of the proxy, for conversion between the private network address and public network address. Private address refers to the internal network (internal LAN) host addresses, and external public address is the address of a local area network (globally unique IP address on the Internet). Internet Assigned Number Authority following three network addresses are reserved for private networks:

10.0.0.0 ~ 10.255.255.255

172.16.0.0 ~ 172.31.255.255

192.168.0.0 ~ 192.168.255.255

That these three are not assigned network addresses on the Internet, but can be used in a private (local area network) internal network. Each private network in the foreseeable future, according to number of hosts, select an appropriate network address. Inner network address different private networks may be the same.

NAT technology is widely used in the private corporate network such as the Internet, to overcome address space limitations and increase security, and solve network problems multiple subnets. Network Address Translation multiplexed by TCP or UDP port number, providing the ability to convert a plurality of inner (private) IP address as an external (public) IP address. About way address translation, it can be divided into several types. Some use the same public addresses, different port numbers to distinguish different connections; some made in a public address pool address, port number acquired in accordance with a certain algorithm.

For a variety of ways of NAT can be concluded from a common characteristic, i.e., when the user wants to use the private network service public network, you need to be a public network address (including the IP address and port number) through NAT, this conversion recording correspondence relationship in an address correspondence table NAT router port, both tables respectively corresponding to the internal private IP address, and source port number sent to the external IP packet, the destination port which is received from an external IP packet number. In the public network node view, this is a private network user's address after NAT public addresses, If you want the user to send private data, send it directly to the public network address, router address by querying the port mapping table storage the correspondence, forwards the data to the private network user.

Briefly, Address Translation (NAT) is to replace the internal IP address and port for the external network IP address and port, and the reverse conversion, the advantages in that NAT technology:

1, provides host access network resources outside the internal network;

2, provides a "privacy (Privacy)" to protect and improve security for internal hosts;

3, to solve the IP address resource issues.

When the application service VoIP private network using an existing VoIP system users will experience the following issues: private network PC initiates a call to the public network PC or gateway, it may occur through a single call or not; the public network PC or telephone ( ) when you dial located in the private network PC, a call can not be established through a gateway. The reason is that: When the PC to initiate a call to the public network when the called PC or Gateway, through the NAT, a private network PC call signaling can be sent correctly to the public network, but the media data portion of the channel included in the call signaling the source address and source port number belonging to the private network, the NAT router can only IP header source address and source port number is converted, can not modify the data portion, the address of the called party can not know the caller media channel, so resulting in a single pass or unreasonable phenomenon. When the public telephone network or PC when a call to the private network PC, called party VoIP server queried (through a gateway) called call channel (or control) address is a private network address, it is not possible to establish a call to the called PC SUMMARY connection

Object of the present invention lies in providing a method and system for implementing IP voice service in networking technologies include NAT private network including a network, the private network from the user can apply the VoIP service.

The present invention is a proprietary IP network voice service implemented method, comprising at least the steps of:

a) PC client private network user sends a login request message to the VoIP server to initiate n times TCP / UDP connection to the address of the server, the value n is a private network user PC client and the server to complete a VoIP call to be established call signal so that the number of channels and logical channels overall channel;

b) a public network address of the server address and port number after the connection through the NAT of the received packet are transmitted to the VoIP server and the private network user client PC C) corresponding to the VoIP server to the public network address of the call signaling path and the port number is recorded in the address field of the client PC, the client PC private network user public network address corresponding to the logical channel and port number recorded in the corresponding address field.

The method of the present invention, further comprising the step of:

D) When a private network user's PC client timing mechanism is provided, so that a predetermined time has not exceeded the call message received, repeating steps a), steps b), step c). The method of the present invention, said step b), the address of the server directly sends the public IP address and port number after the corresponding logical channel via the NAT private network user to a client PC.

The method of the present invention, step fans b), the address of the server after the NAT public address of the connection packet received by the port number sent to the VoIP server, the VoIP server corresponding well logical channels reply message network address and port number in the private network user in response to the login request of the client PC transmits to the client PC.

The system of the present invention for realizing the private network IP voice traffic, including at least voice IP (VoIP) server connected through an IP network, PC clients, gatekeeper, a gateway and an exchange device connected via a public switched telephone network and the telephone terminal wherein the PC client for PC users to make voice calls, to achieve the conversion protocol processing and call control and voice data, VoIP server implementation PC client login authentication, in response to a call access request and the call connection control, the gateway is connected to the IP network equipment and public switched telephone network, the gatekeeper provides search function gateway address; characterized in that:

The system further comprises a proxy server address, an IP network connecting system, TCP UDP receives a private network user sent from a client PC after login connection packets, and wherein after the NAT public IP address and port number They were sent to the VoIP server and private network user's PC client;

VoIP server according to the call signaling path corresponding to the address recorded in the address field of the client PC, the client PC private network user channel corresponding to the logical address recorded in the corresponding address field, private network users and client PC VoIP call server implements reservation and call signaling path further Series channel.

Further, the VoIP service server and the servers of the server user. The present invention is by adding a conventional VoIP system in the proxy server address, and to establish a data connection VPN client PC login phase the proxy server address, the call required to complete the channel established between the VoIP server and client PC, comprising call signaling channel, logical channel, it is pre-established in the log phase, and instead of the corresponding PCC after private network address in the source address of the NAT source port number (public address) with each channel, the reserved call signal actually played and logical channel so that the channel (i.e., speech channel) action, application NAT VoIP services in the networking implementation. BRIEF DESCRIPTION OF DRAWINGS

NAT schematic configuration diagram of a network application system of the present invention, the VoIP service.

FIG 2 is a schematic system configuration of an embodiment of the present embodiment of the invention. FIG 3 is a view of an embodiment of the PCC private network login process shown in FIG. Embodiment of the present invention.

The present invention will be described in detail in conjunction with the accompanying drawings.

The present invention is to increase the address of the proxy server on a conventional VoIP system, the need to complete a call path established between private network users VoIP server and client PC, including call signaling channel and a logical channel, by the proxy server address user login stage pre-established, and with each channel instead of the public address and port number of the NAT after the original PCC private network address and port number.

NAT Network System Diagram Referring VoIP service application structure shown in Figure 1, wherein the VoIP service server 10, the client PC and the PC user's telephone subscriber gateway (GW) 11 are connected via an IP network, the system further comprising an address proxy server 12 is also connected via an IP network system. The private network user client PC 13 via the NAT router 14 connected to public network.

When the PC Client requests to sign the private network to the VoIP server, TCP or UDP connections initiated several times, equal to the number of connections and VoIP private network PCC to complete the number of channels required to establish a call to the address of the proxy server. These channels connect sequentially numbered 1

(Channel 1), Channel 2 (channel 2),, channel n (channel n), which is a private network source address of PCC address, source port number is the default port number for each desired channel, these address groups ( including IP address and port number) is called PrivateNetAddrl,

PrivateNetAddr2 PrivateNetAddrn. The address belongs to a private network address.

By NAT translation, which is connected proxy server address packet arrives, so that the source address through the address / port conversion, has been converted to a public IP address, called PublicNetAddrl,

PublicNetAddr2 PublicNetAddrn. Address of the proxy server sends these addresses to the client PC and the VoIP server within the private network, the VoIP server public network address corresponding call signaling channel is recorded in the private network address of the client PC field, a private network client PC the public network address corresponding to the logical path recorded in the corresponding address field.

Since the NAT address conversion for recording the address conversion table in which the timing mechanism employed, exceeds the predetermined time does not update the connection, the record will be deleted from the table. It is necessary to set a retransmission timer mechanism, within the predetermined time PrivateNetAddr 1, 2, 3 does not receive the message, it will be automatically sent to a TCP or UDP connection address of the proxy server (AddProxy) again.

For a TCP connection, when you create has a start (SYN) flag, no ACK flag (in addition to the packets of all the TCP packets have the ACK flag). When the interrupt will be terminated (FIN) flag. So for the TCP connection is interrupted and the creation of these rely on to determine the TCP connection.

For UDP packets, the way time estimate may be used. When a connection is not used in some time, it considers the connection has been interrupted. When a new connection occurs, it considers the connection is created. This estimated time for the connection may be configured to provide user.

Through the above steps, the private network in the Internet user login, the call setup before the call is established the desired channel, by a timing mechanism has been reserved for retransmission. A call using a call signaling channel reservation and logical channel when a call occurs. The reserved channel automatically after removing the net users.

When the PC user's private network as the calling application VoIP services, to initiate a call to the VoIP server belongs by reserving the call signaling channel after call set-up, the need to establish a logical channel. At this time, the PCC own logical address call signaling channel ^^ packet address to the VoIP server, the logical channel address is issued at this time has passed after the public network address of the NAT. After establishing logical channel you can make a call smoothly.

PC user's private network as a VoIP service called application by the PC user public network initiated by the user via telephone or PCC GW call, VoIP call signaling server receives a message, find the IP address of the called party, this is the result of address after NAT public IP address, the call signaling messages sent to this address corresponding to the router, the router queries NAT address translation table, find the corresponding private address and forwards the call signaling to the client PC, call signaling channel created. Private network user's PC client call signaling sent back Gen 4 shall contain the logical channel address, this address is a public network address through the NAT. After establishing logical channel you can make a call smoothly.

Aspect of the present invention can be applied to any voice over IP services. Below in connection with FIG. 2, FIG. 3, in a personal number service (ONLY) as an example, it will be further explained and the present invention is applied.

ONLY (One Number Link You) business, with the development of the Internet, in order to meet users increasingly eager to get the go and need to exchange information, based on a blend of traditional telecommunications technology and IP technology, the development of a kind of innovative business. ONLY main business is to provide the user with unique personal service number - - ONLY number to provide a variety of business by the numbers, regardless of user where they can be contacted more quickly to a specific user, the user can be contacted via PC, telephone, voice mail and other means to answer the call.

ONLY ONLY service system consists of a server (ONLY Server), ONLY Subscriber Server (User Server), PC Client (PCC), interactive voice response component devices (IVR), databases, Web servers, which ONLY business systems VoIP server functionality It is jointly carried out by ONLY Server and User Server. The main function ONLY Server are: a call access request to the responding node; cross fork with User S erver obtained ONLY numbers of the address translation; gatekeeper (GK) routes obtained interaction of the called telephone number; transit call signal according to the destination address resolution make; complete control of the gateway (MG); and interact with the IVR, supports transparent transmission Han-tone multi-frequency (DTMF) number; record simple billing information is sent to the User Server. The main function of User Server include: call strategy selection (based on call policy set by the user to translate the virtual ONLY number is a real phone number or IP address of the PCC); PCC login authentication, outbound authorization, state maintenance; voice message (VM) circulars . Like other VoIP systems, ONLY business in private network environments also encountered the same problem. Referring to FIG, 2 is a schematic structural diagram using ONLY service aspect of the present invention NAT networking applications. Which in addition to the original equipment ONLY system, adding a new address of the proxy server 12 through the IP network access system, and the original server and PC client user to make the appropriate changes to the PCC after Login User Server, UDP connections to initiate a proxy server address, wherein the address of the proxy server address sent through the public network to the User server for NAT, returned by the User server PCC login request public network address in the response transmitted through the NAT to the PCC, User Server PCC and those were recorded through the public address NAT translation, the establishment of channels need to call to reserve a private network and PCC ONLY Server.

PCC private network logon process shown in Figure 3, wherein the private network 16655551234 ONLY number after initiation of the PCC 13 is a login request message (Login-Req) to the User Server 102, to the address proxy server needs to initiate a three UDP connections 12 . UDP connections these three numbered sequentially UDP1, UDP2, UDP3, their source address is the address of the private network PCC 13, UDP1, source port number 2, 3 respectively MGCP, RTP, RTCP default port number. These three sets of address (including the IP address and port number) is called PrivateNetAddrl, PrivateNetAddr2, PrivateNetAddr3. The three belong to the group address private network address.

After the NAT through the three UDP packet arrives address of the proxy server 12, so that the source address through the address / port translation, has become a public address, which is referred to PublicNetAddrK PublicNetAddr2, PublicNetAddr3. Address of the proxy server 12 public addresses these three groups is sent to User Server 102, User Server 102 will PublicNetAddrl 16655551234 recorded in the address field of the PCC, and in response to the login request message (Login- Ack) will be sent to the PCC PublicNetAddr2 and PublicNetAddr3, PCC record these two addresses. Since UDP connections for the NAT address conversion table in the address conversion using the recorded timing mechanism, exceeds a predetermined time does not update the connection, the record will be deleted from the table. Therefore, in the absence of a call occurs, the PCC must be kept with the address of the proxy server 12 UDP1, connection 2, 3, it is necessary to set a special timing of retransmission mechanism, PrivateNetAddrl, 2, 3 does not receive the message within a predetermined period of time Wen, 12 will automatically repeat sending a UDP packet to the address of the proxy server.

After the above login process, by the proxy server address 12 on pre-established channels between the private network and PCC ONLY Server calls need to be established.

When the private network 16655551234 PCC as a calling party initiating the call, the call establishment signaling path, since the call signaling channel is held private network and public network address in the address conversion table of correspondence between the router, so the call signaling actually using the above reserved channel. PCC 13 sent from the source address to data PrivateNetAddrl MGCP ONLY Server 101, ONLY Server 101 queries the User Server 102 call control strategy, to the gateway or the called GK queries obtained address of the called number belongs ONLY ONLY Server in User Server, begins to establish a logical aisle. At this PCC 16655551234 send their RTP and RTCP address to the called party, in accordance with the agreement should be PrivateNetAddr2, PrivateNetAddr3. But if you send a private network address can cause other side of the voice data can not be sent correctly to the PCC 16655551234, so it should be PrivateNetAddr2, PrivateNetAddr3 replaced PublicNetAddr2, PublicNetAddr3, so that you can take advantage of the reserved logical channels above the call smoothly.

. When a private network 16655551234 PCC as a called, the PC user's public network GW a PCC or telephone user initiates a call, ONLY Server 101 receives the call signaling, the first address queries to PCC 16655551234 User Server 102, to find to PublicNetAddrl, MGCP data will be sent to PublicNetAddrl, query router NAT address translation table to find the corresponding PrivateNetAddrl, MGCP data will be sent to the PCC 16655551234. When the call signaling channel is established, PCC 16655551234 MGCP loopback data should include RTP and RTCP addresses, in accordance with the agreement should be PrivateNetAddr2, PrivateNetAddr3. However, if feeding a private network address, the calling party will cause the voice data is not transmitted correctly to the PCC 16655551234, it should be PrivateNetAddr2, PrivateNetAddr3 replaced PublicNetAddr2, PublicNetAddr3, thus using the above-described logical channel reserved for voice data transmission.

Aspect of the present invention is applicable to any system VoIP service. Such as interoperability testing IP telephony gateway (Master, Refmer, Expert), and with a H.323 protocol implementation IP telephony applications perform on a PC. This applied in a private network environment also encountered the same problem with the ONLY business. The introduction of the above aspect of the present invention to the application, except that: H.225 call signaling protocol used to establish the channel using TCP. Then establishes H.245 control channel, and finally to establish a logical channel. But the basic idea and ONLY call setup is to have in common. After the improvements according to this embodiment, it can be smoothly in a private network call environment, by the basic functional verification and testing a large call volume.

In summary, the embodiment of the present invention is simple, with good usability and reliability.

Claims

Claims
1, a proprietary IP network voice service realization method, characterized in that the method comprises at least the steps of:
a) PC client private network user sends a login request message to the VoIP server to initiate n times TCP / UDP connection to the address of the server, the value n is a private network user PC client and the server to complete a VoIP call to be established call signal so that the number of channels and logical channels overall channel;
After the public address (NAT) and port number are sent to the VoIP private network server and client PC user;
c) VoIP call signaling channel corresponding to the server public network address and port number recorded in the address field of the client PC, the client PC private network user corresponding public IP address and port number of the logical channels is recorded in the corresponding address fields in.
2. The method of claim 1 IP implement voice service claim, characterized by further comprising the step of:
D) When a private network user's PC client timing mechanism is provided, so that a predetermined time has not exceeded the call message received, repeating steps a), steps b), step c).
3, according to an implementation method of claim 1 IP voice service claim, wherein: said step b), the address server directly transmit the public network address after the passage through the corresponding logical port number and the NAT to private network users PC client.
4. The method of claim 1 IP implement voice service claim, wherein: said step b), the address of the NAT server after connecting the public network address in the received packet and the port number transmits to the VoIP server, the VoIP server to the corresponding logical channel public network address and port number sent to the client PC reply message in response to private network client PC user login request.
5. A system for implementing a private network IP voice services, including at least IP voice service (VoIP) server connected via an IP network, PC client, gatekeepers, gateways and switching equipment connected through the public switched telephone network and telephone terminal equipment wherein the PC client for PC users to make voice calls, and the call control protocol processing to achieve, and the voice data conversion,
VoIP server implements client PC login authentication, the access request in response to the call connection and call control, the gateway is connected to the IP network and the public switched telephone network equipment, the gatekeeper provides the address to find the gateway; wherein:
The system further comprises a proxy server address, an IP network connecting system, TCP UDP receives a private network user sent from a client PC after login connection packets, and wherein after the NAT public IP address and port number They were sent to the VoIP server and private network user's PC client;
VoIP server according to the call signaling path corresponding to the address recorded in the address field of the client PC, the client PC private network user channel corresponding to the logical address recorded in the corresponding address field, private network users and client PC VoIP call server implements reservation and call signaling path further Series channel.
6, IP voice service system according to claim according to claim 3, wherein: Further, the VoIP server to the user by the service server and servers.
PCT/CN2002/000371 2001-10-04 2002-05-30 A method and system for realizing ip voice service at private network WO2003030463A1 (en)

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