CN103023858A - Method for solving normal call under network address translation (NAT) network environment in session initiation protocol (SIP) network system - Google Patents
Method for solving normal call under network address translation (NAT) network environment in session initiation protocol (SIP) network system Download PDFInfo
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- CN103023858A CN103023858A CN2011102825221A CN201110282522A CN103023858A CN 103023858 A CN103023858 A CN 103023858A CN 2011102825221 A CN2011102825221 A CN 2011102825221A CN 201110282522 A CN201110282522 A CN 201110282522A CN 103023858 A CN103023858 A CN 103023858A
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Abstract
The invention discloses a solving method of session initiation protocol (SIP) call in a network address translation (NAT) environment, which is low in network consumption. In the SIP network system, SIP users in private networks log in an SIP server in a public network through a router, so that terminals in different private networks can mutually communicate normally without occupation of the rated bandwidth of the SIP server under a circumstance of not changing or upgrading the current router. According to the technical solution, by the method disclosed by the invention, normal communication of the SIP terminal is achieved in the normal router NAT environment, and the market demand is met.
Description
Technical field
The present invention relates to the IMS communication field, particularly a kind of calling based on SIP solves NAT network media stream interworking technology, is a kind of SIP call technology that can save in a large number the network bandwidth.
Technical background
Extensive use along with network technology and service, a large amount of network equipments all require to have the independently network address, and the lazy weight of IPv4 address is to satisfy such demand, for solving the predicament of IPv4 address scarcity, the researcher has proposed NAT technology (Network Address Translating is called for short NAT).NAT device generally is positioned at network edge, and whole network is divided into inner and outside two networks.Usually inner and external network uses respectively two class addresses, communicate between inside and outside network equipment, and the address of internal network devices must be converted to the IPv4 address that external network can be identified.Therefore need NAT device all network data messages that advanced it carry out necessary conversion, revise internal network address information wherein, and the data message after will changing is issued down hop or the destination network equipment, thereby guarantee between intra-company and the external network unimpededly, avoid forming " information island ".By the network address translation function of NAT device, the IPv4 address of internal network does not need to be identified by external network, so that private ip v4 address can be reused in LAN, has effectively alleviated the predicament of present IPv4 address scarcity.
Yet the development of network technology is maked rapid progress, new Network, function emerge in an endless stream, although the application of NAT technology and deployment bring benefits such as saving the IPv4 address space, but also destroyed the design concept that the Internet gets " transparently end-to-end " the most substantially simultaneously, increase the complexity of network, hindered professional innovation.
Conversation initialized protocol (Session Initiation Protocol, be called for short SIP) be IEEE subordinate's MMUSIC group 2002 the new application layer network protocol type of proposition, be mainly used in the telephone signaling control of the emerging network business such as the networking telephone, instant messaging, the functions such as session establishment and session control are provided, have been the research of in recent years network technology and use one of focus.Yet the NAT device of having disposed is scarcely supported the network address translation function to this agreement, can't effectively identify, verify and translate the protocol data message, can't realize the application layer routing mechanism forwarding of this protocol requirement, blocked interconnecting between the internal-external network, so that the Session Initiation Protocol data can only be confined to this locality or LAN.This is for can not put up with in the urgent need to the company, tissue, the mechanism that carry out new business or utilize new technology to increase productivity.The existing NAT device disposal ability of upgrading realizes that NAT technology to Session Initiation Protocol is current NAT and the popular problem of the common concern in sip technique field.NAT technology and scheme although the researcher just begins one's study at the beginning of Session Initiation Protocol proposes the Session Initiation Protocol data message, find that there is variety of issue in they, have especially increased unnecessary restriction to autgmentability and the flexibility of Session Initiation Protocol self but analyse in depth.Main purpose of the present invention is exactly by the deep understanding to the Session Initiation Protocol application layer, on the basis of existing SIP passing through NAT technology, designs and Implements a kind of simply and effect, can carry out to the Session Initiation Protocol data upgrading scheme of network address translation.
Summary of the invention
The sip terminal that purpose of the present invention solves the NAT net environment communicates with each other through the public network environment, and not too the bandwidth of temporary sip server realizes.
In order to realize goal of the invention, the technical scheme of employing is as follows:
Common at NAT net environment access Internet network as shown in Figure 1.PC is private network IP address in LAN, carries out NAT network address translation access Internet network by router, and the application on the Internet sees that connecting what come is public network IP address after the router conversion.
SIP in the local area network (LAN) calls out as shown in Figure 2.The SIP software terminal is deployed on the PC, and the calling and called user need to be in a network, and calling and called just can the normal call conversation like this.
The SIP of NAT net environment calls out as shown in Figure 3.It is normal that the SIP signaling continues under the NAT environment, but both sides can't hear the other side's voice, reason is exactly the SDP information that the sip terminal on both sides receives, all be the other side's private network IP address and port, both sides can't send the RTP data flow to the other side's private network IP and port, so both sides can't normally hear voice.
The invention provides a kind of under NAT network of network environment, can realize a kind of method of SIP normal talking, when the sip terminal of PC1 sends INVITE to SIPServer, can be with private network IP and the port in the SPD of INVITE the inside, be modified as public network IP and the port of SIPServer, information by sip terminal on the PC2 of sip terminal registration message record, find the public network IP on the PC2, INVITE is sent to sip terminal on the PC2, the SDP that this moment, the sip terminal on the PC2 was seen is SIPServer public network IP and port, sip terminal on the PC2 sends 200 OK response, private network IP and port that IP in the SDP of band and port are PC2, SIPServer receives 200 OK, with private network IP and the port in the SDP, be modified as public network IP and port on the SIPSserver, amended 200 OK are sent to sip terminal on the PC1, the like this sip terminal on the PC1 and the sip terminal on the PC2, respectively rtp streaming is sent on the SIPServer, this moment, SIPServer knew that also sip terminal RTP on PC1 and the PC2 is through IP and port behind the NAT, RTP is sent on separately the NAT IP and port, the such sip terminal on PC1 and the PC2 and can normal talking, but the rtp streaming of this moment is through SIPServer, take the bandwidth of SIPSever, next step processing of the present invention, that the sip terminal on PC1 and the PC2 is sent the NAT IP of RTP and port by reInvite message, notify respectively two sip terminals, like this, the rtp streaming that sip terminal on PCI and the PC2 sends has just no longer passed through SIPServer, but directly mail on separately the NAT device, realize point-to-point RTP communication, two sip terminals can normal talking.
Have as seen upward, the invention provides a kind of a kind of solution based on can normally carrying out SIP under the NAT environment and calling out, following characteristics is arranged.
(1) the SIP normal talking problem of the common NAT net environment of solution
The present invention solves common NAT net environment, does not need special NAT device, and the SIP that solves existing enterprise and the NAT of unit net environment normally passes through function.
(2) bandwidth occupancy is low
The present invention normally passes through at solution NAT net environment SIP, just in the bandwidth that begins most to have taken part SIPServer, get access to the RTP NAT IP and port of SIP to SIPServer after, just send reInvite, rtp streaming is altered course, follow-up conversation will not take the bandwidth of SIPServer, also increase the disposal ability of SIPServer.
Description of drawings
In order to be illustrated more clearly in the embodiment of the invention or technical scheme of the prior art, the below will do simple the introduction to the accompanying drawing of required use in embodiment or the description of the Prior Art, apparently, accompanying drawing in the following describes only is some embodiments of the present invention, for those of ordinary skills, under the prerequisite of not paying creative work, can also obtain according to these accompanying drawings other accompanying drawing.
Pc access Internet figure under the common NAT network of Fig. 1;
The SIP call diagram of Fig. 2 local area network (LAN);
The SIP call diagram of Fig. 3 NAT network environment;
The SIP of Fig. 4 NAT network environment calls out sequential chart;
The SIP of Fig. 5 NAT network environment calls out final mask figure.
Embodiment
The present invention is described further below in conjunction with Fig. 4.
The processing procedure that whole NAT net environment SIP calls out is described below:
1) sip terminal on the PC1 sends Invite message to Route1;
2) Route1 sends to SIPServer on the public network according to destination address with Invite message;
3) SIPServer is modified as the SDP of Invite public network IP and the port of oneself, by the sip terminal registration message on the PC2 before, finds NAT IP and the port of the sip terminal on the PC2, and Invite is sent on the Route2 with the SDP of public network address;
4) Route2 sends to Invite on the sip terminal on the PC2;
5) sip terminal on the PC2 returns 180 (not being with SDP) to Route2;
6) Route2 gives SIPServer with 180;
7) SIPServer issues Route1 with 180 responses;
8) Route1 responds the sip terminal of issuing on the PC1 with 180;
9) sip terminal on the PC2 returns the SDP of the private network IP of 200 karaoke tape PC2 and port to Route2;
10) Route3 sends to SIPServer with 200 OK response;
11) SIPServer replaces to public network IP and the port of oneself with private network IP and port in the SDP of 200 OK, and 200 OK after will replacing again send to Route1;
12) Route1 sends to sip terminal on the PC1 with the SDP of 200 karaoke tape public network IP and port;
13) sip terminal on the PC1 returns ACK to Route1;
14) Route1 sends to SIPServer with ACK;
15) SIPServer is transmitted to Route2 with ACK;
16) Route2 sends to sip terminal on the PC2 with ACK message;
17) sip terminal on this moment PC1 and PC2 sends a SIPServer with rtp streaming, and SIPServer does the forwarding of rtp streaming, and such two sip terminals can normal talking;
18) SIPServer receives the rtp streaming that the sip terminal on the PC2 sends out through Route2, thereby has got access to IP and the port of the NAT network of the upper sip terminal transmission of PC2 rtp streaming use;
19) SIPServer SDP that the upper sip terminal of PC2 is brought is modified as the upper sip terminal of PC2 and sends NAT network IP and the port that rtp streaming uses, and sends reInvite message to Route1;
20) Route1 sends to sip terminal on the PC1 with reInvite message;
21) sip terminal on the PC1 sends 200 OK message to Route1;
22) Route1 sends to SIPServer with 200 OK message;
23) SIPServer responds ACK to Route1;
24) Route1 sends to sip terminal on the PC1 with ACK;
25) at this moment, the sip terminal on the PC1 alters course rtp streaming, sends on the Route2, is exactly NAT network IP and port that the upper sip terminal of PC2 sends RTP;
26) SIPServer receives the rtp streaming that the sip terminal on the PC1 sends out through Route1, thereby has got access to IP and the port of the NAT network of the upper sip terminal transmission of PC1 rtp streaming use;
27) SIPServer SDP that the upper sip terminal of PC1 is brought is modified as the upper sip terminal of PC1 and sends NAT network IP and the port that rtp streaming uses, and sends reInvite message to Route2;
28) Route2 sends to sip terminal on the PC1 with reInvite message;
29) sip terminal on the PC2 sends 200 OK message to Route2;
30) Route2 sends to SIPServer with 200 OK message;
31) SIPServer responds ACK to Route2;
32) Route2 sends to sip terminal on the PC2 with ACK;
33) at this moment, the sip terminal on the PC2 alters course rtp streaming, sends on the Route1, is exactly NAT network IP and port that the upper sip terminal of PC1 sends RTP;
34) this time PC1 and PC2 rtp streaming made point-to-point pattern into, with reference to figure 5.
Claims (4)
1. one kind solves the method that SIP under the NAT environment calls out, its feature with, comprising:
By the registration of sip terminal, SipServer gets access to the NAT IP port of sip terminal.
2. method according to claim 1 is characterized in that, also comprises:
The IP of RTP and IP and the port that port is SipServer in the SDP of SipServer modification sip terminal.
3. method according to claim 2 is characterized in that:
SipServer gets access to NAT IP and the port that sip terminal sends RTP, by reInvite message, revises IP and port in the SDP, changes NAT IP and the port of sip terminal into, sends to the sip terminal of opposite end.
4. method according to claim 3 is characterized in that:
SipServer gets access to NAT IP and the port that another sip terminal sends RTP, by reInvite message, revises IP and port in the SDP, changes NAT IP and the port of sip terminal into, sends to the sip terminal of opposite end.
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| CN201110282522.1A CN103023858B (en) | 2011-09-20 | 2011-09-20 | Method for solving normal call under network address translation (NAT) network environment in session initiation protocol (SIP) network system |
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| CN201110282522.1A CN103023858B (en) | 2011-09-20 | 2011-09-20 | Method for solving normal call under network address translation (NAT) network environment in session initiation protocol (SIP) network system |
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Cited By (1)
| Publication number | Priority date | Publication date | Assignee | Title |
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| CN103916382A (en) * | 2013-12-25 | 2014-07-09 | 三亚中兴软件有限责任公司 | NAT through method based on SIP media capacity re-negotiation, proxy server and system |
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| US20070121596A1 (en) * | 2005-08-09 | 2007-05-31 | Sipera Systems, Inc. | System and method for providing network level and nodal level vulnerability protection in VoIP networks |
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| US20070121596A1 (en) * | 2005-08-09 | 2007-05-31 | Sipera Systems, Inc. | System and method for providing network level and nodal level vulnerability protection in VoIP networks |
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Cited By (2)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN103916382A (en) * | 2013-12-25 | 2014-07-09 | 三亚中兴软件有限责任公司 | NAT through method based on SIP media capacity re-negotiation, proxy server and system |
| CN103916382B (en) * | 2013-12-25 | 2018-05-01 | 三亚中兴软件有限责任公司 | NAT through method, proxy server and system based on SIP media ability re-negotiations |
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| CN103023858B (en) | 2015-07-22 |
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Address after: Room 306, area 2, building 1, Fanshan Venture Center, Panyu energy saving science and Technology Park, 832 Yingbin Road, Donghuan street, Panyu District, Guangzhou, Guangdong 510000 Patentee after: Jiadu Technology Group Co.,Ltd. Address before: No.4, Jiangong Road, Tianhe Software Park, Guangzhou, Guangdong 510665 Patentee before: PCI-SUNTEKTECH Co.,Ltd. |