WO2003030463A1 - Procede et systeme pour la prestation d'un service vocal ip sur un reseau prive - Google Patents

Procede et systeme pour la prestation d'un service vocal ip sur un reseau prive 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
client
server
private network
network
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PCT/CN2002/000371
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English (en)
Chinese (zh)
Inventor
Haitao Lin
Yiguo Li
Quan Gan
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Huawei Technologies Co., Ltd.
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Publication date
Application filed by Huawei Technologies Co., Ltd. filed Critical Huawei Technologies Co., Ltd.
Publication of WO2003030463A1 publication Critical patent/WO2003030463A1/fr

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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M7/00—Arrangements for interconnection between switching centres
    • H04M7/006—Networks other than PSTN/ISDN providing telephone service, e.g. Voice over Internet Protocol (VoIP), including next generation networks with a packet-switched transport layer
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M3/00—Automatic or semi-automatic exchanges
    • H04M3/42—Systems providing special services or facilities to subscribers
    • H04M3/42314—Systems providing special services or facilities to subscribers in private branch exchanges

Definitions

  • 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.
  • VoIP IP voice service
  • NAT network address translation
  • 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).
  • PSTN public switched telephone network
  • QoS real-time communication service quality
  • VoIP Voice over IP
  • IP IP over IP
  • 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.
  • 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.
  • 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.
  • 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.
  • TCP Transmission Control 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.
  • a Resource Reservation Protocol (RSVP) is also selected to reserve network resources for calls.
  • VoIP 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.
  • 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:
  • 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.
  • 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.
  • 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.
  • NAT For the multiple methods of NAT, a common feature can be summarized.
  • a public network address including the IP address and port number
  • 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.
  • 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.
  • NAT address translation technology
  • 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 includes at least the following steps:
  • n TCP / UDP connections 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.
  • 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
  • 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.
  • the method further includes the following steps:
  • step b) 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).
  • 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.
  • 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.
  • VoIP IP voice service
  • 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.
  • the VoIP server is composed of a service server and a user server.
  • 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.
  • 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.
  • 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.
  • 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.
  • the PC client of the private network 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
  • PrivateNetAddr2 PrivateNetAddrn. These addresses are private addresses.
  • 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.
  • 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).
  • TCP connections 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.
  • 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.
  • the channel required for the call is established before the call is established, and it is always reserved by the timing retransmission mechanism.
  • the reserved call signaling channel and logical channel are used for conversation. This reserved channel is automatically removed after the user goes offline.
  • 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.
  • 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.
  • 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.
  • GK gatekeeper
  • 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.
  • 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.
  • 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.
  • the above three UDP packets reach the address proxy server 12.
  • 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).
  • Login—Ack login request response message
  • 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.
  • the PCC 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.
  • the address proxy server 12 has previously established a channel to be established between the private network PCC and the ONLY Server call.
  • PCC 16655551234 of the private network 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.
  • PrivateNetAddr2 PrivateNetAddr3.
  • PrivateNetAddr2 PrivateNetAddr3
  • PublicNetAddr2 PublicNetAddr3
  • PCC 16655551234 of the private network 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.
  • 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.
  • 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.
  • 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.
  • the solution of the present invention is simple and easy to implement, and has good practicability and reliability.

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  • Engineering & Computer Science (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Signal Processing (AREA)
  • Data Exchanges In Wide-Area Networks (AREA)

Abstract

L'invention concerne un procédé et un système de prestation de service vocal IP sur un réseau, notamment un réseau privé, mis en oeuvre à l'aide de la technologie NAT. Par l'augmentation du nombre de serveurs mandataires d'adresses et par l'établissement d'une liaison de données TCP/UDP avec le serveur mandataire d'adresses pendant la procédure de demande de connexion du client du PC, toutes les voies requises pour la communication entre le client PC et le serveur VoIP sont déterminées à l'avance pendant la procédure de demande de connexion, les voies concernées étant les voies de signalisation et les voies logiques. Les adresses du réseau privé PCC sont remplacées par un port source d'adresses sources (adresse réseau public) obtenu par l'application de la conversion NAT à chaque voie, ce qui sert en fait de réservation de voie de signalisation d'appel et de voie logique (ou voie vocale), de sorte que le service VoIP soit réalisé dans un réseau mis en oeuvre par la technologie NAT.
PCT/CN2002/000371 2001-10-04 2002-05-30 Procede et systeme pour la prestation d'un service vocal ip sur un reseau prive WO2003030463A1 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
CNB011356103A CN1170393C (zh) 2001-10-04 2001-10-04 私有网络的ip语音业务实现方法及系统
CN01135610.3 2001-10-04

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US8605728B2 (en) 2003-09-02 2013-12-10 Huawei Technologies Co., Ltd. Method of implementing traversal of multimedia protocols through network address translation device
WO2016201966A1 (fr) * 2015-06-18 2016-12-22 中兴通讯股份有限公司 Procédé, appareil et système de traversée de nat
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US8605728B2 (en) 2003-09-02 2013-12-10 Huawei Technologies Co., Ltd. Method of implementing traversal of multimedia protocols through network address translation device
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WO2016201966A1 (fr) * 2015-06-18 2016-12-22 中兴通讯股份有限公司 Procédé, appareil et système de traversée de nat
CN106331187A (zh) * 2015-06-18 2017-01-11 中兴通讯股份有限公司 一种nat穿透方法、装置及系统
CN114363117A (zh) * 2021-12-13 2022-04-15 武汉船舶通信研究所（中国船舶重工集团公司第七二二研究所） 一种无线子网异构互联方法和系统
CN114363117B (zh) * 2021-12-13 2023-11-28 武汉船舶通信研究所（中国船舶重工集团公司第七二二研究所） 一种无线子网异构互联方法和系统

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