CN103023858B - 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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- CN103023858B CN103023858B CN201110282522.1A CN201110282522A CN103023858B CN 103023858 B CN103023858 B CN 103023858B CN 201110282522 A CN201110282522 A CN 201110282522A CN 103023858 B CN103023858 B CN 103023858B
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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 IMS communication field, particularly a kind of calling based on SIP, solving NAT network media stream interworking technology, is a kind of SIP call technology can saving the network bandwidth in a large number.
Technical background
Along with the extensive use of network technology and service, a large amount of network equipment all requires to have the independently network address, and the lazy weight of IPv4 address is to meet such demand, for solving the predicament of IPv4 address scarcity, researcher proposes NAT technology (Network Address Translating is called for short NAT).NAT device is generally positioned at network edge, and whole network is divided into inner and outside two networks.Usual inside and external network use two class addresses respectively, will communicate between the inside and outside network equipment, and the address of internal network devices must be converted to the IPv4 address that external network can identify.Therefore need NAT device all entered its network data message carry out necessary conversion, amendment internal network address information wherein, and the data message after conversion is issued down hop or the destination network equipment, thus guarantee between intra-company and external network unimpeded, avoid the formation of " information island ".By the network address translation function of NAT device, the IPv4 address of internal network does not need by external network identification, private ip v4 address can be reused in LAN, effectively alleviate the predicament of current IPv4 address scarcity.
But 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 IPv4 address space, but also destroy the design concept that the Internet obtains " transparently end-to-end " the most substantially simultaneously, add the complexity of network, hinder the innovation of business.
Conversation initialized protocol (Session Initiation Protocol, be called for short SIP) be the new application layer network protocol type that the MMUSIC group of IEEE subordinate just proposes for 2002, the telephone signaling being mainly used in the emerging network such as the networking telephone, instant messaging business controls, providing the function such as session establishment and session control, is one of investigation and application focus of network technology in recent years.But the NAT device disposed scarcely supports the network address translation function to this agreement, cannot protocol data message effectively be identified, verifies and be translated, the application layer routing mechanism that cannot realize this protocol requirement forwards, block interconnecting between internal-external network, make Session Initiation Protocol data can only be confined to this locality or LAN.This can not put up with in the urgent need to the company carrying out new business or utilize new technology to increase productivity, tissue, mechanism.Upgrade existing NAT device disposal ability, realizing the NAT technology of Session Initiation Protocol is the popular problem that current NAT and sip technique field are paid close attention to jointly.Although researcher proposes from Session Initiation Protocol just begin one's study beginning NAT technology Session Initiation Protocol data message and scheme, but analyse in depth and find that they exist various problem, especially unnecessary restriction is added to the autgmentability of Session Initiation Protocol self and flexibility.Main purpose of the present invention is exactly by the deep understanding to Session Initiation Protocol application layer, on the basis of existing SIP passing through NAT technology, designs and Implements a kind of simple and effect, can carry out the upgrading scheme of network address translation to Session Initiation Protocol data.
Summary of the invention
The sip terminal that object of the present invention solves under NAT network environment communicates with each other through 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:
Commonly under NAT network environment, access Internet network as shown in Figure 1.PC is private network IP address in LAN, by router carry out NAT network address translation access Internet network, the application on Internet see connect come be router conversion after public network IP address.
SIP calling in local area network (LAN) as shown in Figure 2.SIP software terminal is deployed on PC, and calling and called user needs in a network, and such calling and called just can normal call call.
SIP calling under NAT network environment as shown in Figure 3.SIP Signaling Relay is normal under NAT environment, but both sides cannot hear the voice of the other side, reason is exactly the SDP information that the sip terminal on both sides receives, all the other side's private network IP address and port, both sides cannot send RTP data flow, so both sides normally cannot hear voice to the private network IP of the other side and port.
The invention provides a kind of under NAT network of network environment, a kind of method of SIP normal talking can be realized, when the sip terminal of PC1 sends INVITE to SIPServer, can by the private network IP in the SPD inside INVITE and port, be modified as public network IP and the port of SIPServer, by the information of sip terminal on the PC2 of sip terminal registration message record, find the public network IP on PC2, INVITE is sent to the sip terminal on PC2, the SDP that sip terminal now on PC2 is seen is SIPServer public network IP and port, sip terminal on PC2 sends 200 OK responses, IP in the SDP of band and port are private network IP and the port of PC2, SIPServer receives 200 OK, by the private network IP in SDP and port, be modified as the public network IP on SIPSserver and port, amended 200 OK are sent to the sip terminal on PC1, sip terminal on such PC1 and the sip terminal on PC2, respectively rtp streaming is sent on SIPServer, now SIPServer is also aware of IP after NAT of sip terminal RTP on PC1 and PC2 and port, RTP is sent on respective NAT IP and port, sip terminal on such PC1 and PC2 and can normal talking, but rtp streaming is now through SIPServer's, take the bandwidth of SIPSever, next step process of the present invention, the sip terminal on PC1 and PC2 is sent the NAT IP of RTP and port by reInvite message, notify two sip terminals respectively, like this, the rtp streaming that sip terminal on PCI and PC2 sends is just no longer through SIPServer, but directly mail on respective NAT device, realize point-to-point RTP to communicate, two sip terminals can normal talking.
Have upper visible, the invention provides a kind of a kind of solution based on SIP calling normally can be carried out under NAT environment, have following characteristics.
(1) the SIP normal talking problem under common NAT network environment is solved
The present invention does not need special NAT device under solving common NAT network environment, solves SIP under existing enterprise and unit NAT network environment normal through function.
(2) bandwidth occupancy is low
The present invention under solution NAT network environment SIP normal through, just starting most the bandwidth occupying part SIPServer, after the RTP NAT IP getting SIP to SIPServer and port, just send reInvite, rtp streaming is altered course, follow-up call will not take the bandwidth of SIPServer, too increases the disposal ability of SIPServer.
Accompanying drawing explanation
In order to be illustrated more clearly in the embodiment of the present invention or technical scheme of the prior art, be briefly described to the accompanying drawing used required in embodiment or description of the prior art below, apparently, accompanying drawing in the following describes is only some embodiments of the present invention, for those of ordinary skill in the art, under the prerequisite not paying creative work, other accompanying drawing can also be obtained according to these accompanying drawings.
Under the common NAT network of Fig. 1, pc access Internet schemes;
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
Below in conjunction with Fig. 4, the present invention is described further.
Under whole NAT network environment, the processing procedure of SIP calling is described below:
1) sip terminal on PC1 sends Invite message to Route1;
2) Invite message to be sent to the SIPServer on public network by Route1 according to destination address;
3) SDP of Invite is modified as public network IP and the port of oneself by SIPServer, by the sip terminal registration message on PC2 before, finds NAT IP and the port of the sip terminal on PC2, is sent on Route2 by the SDP of Invite with public network address;
4) Invite is sent on the sip terminal on PC2 by Route2;
5) sip terminal on PC2 returns 180 (not being with SDP) to Route2;
6) Route2 gives SIPServer by 180;
7) 180 responses are issued Route1 by SIPServer;
8) Route1 responds 180 the sip terminal issued on PC1;
9) SDP of private network IP and port that the sip terminal on PC2 returns 200 karaoke tape PC2 is to Route2;
10) 200 OK responses are sent to SIPServer by Route3;
11) SIPServer is by the private network IP in the SDP of 200 OK and port, replaces to oneself public network IP and port, then 200 OK after replacing are sent to Route1;
12) SDP of 200 karaoke tape public network IP and port is sent to the sip terminal on PC1 by Route1;
13) sip terminal on PC1 returns ACK to Route1;
14) ACK is sent to SIPServer by Route1;
15) ACK is transmitted to Route2 by SIPServer;
16) ACK message is sent to the sip terminal on PC2 by Route2;
17) rtp streaming is sent the forwarding that SIPServer, a SIPServer do rtp streaming by the sip terminal now on PC1 and PC2, and such two sip terminals can normal talking;
18) SIPServer receives the rtp streaming that the sip terminal on PC2 sends out through Route2, thus has got IP and port that sip terminal on PC2 sends the NAT network that rtp streaming uses;
19) SDP that sip terminal on PC2 is brought by SIPServer is modified as NAT network IP and the port that sip terminal on PC2 sends rtp streaming use, sends reInvite message to Route1;
20) reInvite message is sent to the sip terminal on PC1 by Route1;
21) sip terminal on PC1 sends 200 OK message to Route1;
22) 200 OK message are sent to SIPServer by Route1;
23) SIPServer responds ACK to Route1;
24) ACK is sent to the sip terminal on PC1 by Route1;
25) now, rtp streaming alters course by the sip terminal on PC1, is sent on Route2, is exactly NAT network IP and port that on PC2, sip terminal sends RTP;
26) SIPServer receives the rtp streaming that the sip terminal on PC1 sends out through Route1, thus has got IP and port that sip terminal on PC1 sends the NAT network that rtp streaming uses;
27) SDP that sip terminal on PC1 is brought by SIPServer is modified as NAT network IP and the port that sip terminal on PC1 sends rtp streaming use, sends reInvite message to Route2;
28) reInvite message is sent to the sip terminal on PC1 by Route2;
29) sip terminal on PC2 sends 200 OK message to Route2;
30) 200 OK message are sent to SIPServer by Route2;
31) SIPServer responds ACK to Route2;
32) ACK is sent to the sip terminal on PC2 by Route2;
33) now, rtp streaming alters course by the sip terminal on PC2, is sent on Route1, is exactly NAT network IP and port that on PC1, sip terminal sends RTP;
34) this time PC1 and PC2 rtp streaming made point-to-point pattern into, with reference to figure 5.
Claims (1)
1. solve a method for SIP calling under NAT environment, it is characterized in that, comprising:
Sip terminal on PC1 sends Invite message to Route1;
Invite message to be sent to the SIPServer on public network by Route1 according to destination address;
The SDP of Invite is modified as public network IP and the port of SIPServer by SIPServer, by the sip terminal registration message on PC2, finds NAT IP and the port of the sip terminal on PC2, is sent on Route2 by the SDP of Invite with public network address;
Invite is sent on the sip terminal on PC2 by Route2;
Sip terminal on PC2 returns 180 to Route2;
Route2 gives SIPServer by 180;
180 responses are issued Route1 by SIPServer;
Route1 responds 180 the sip terminal issued on PC1;
The SDP of private network IP and port that the sip terminal on PC2 returns 200OK band PC2 is to Route2;
200OK response is sent to SIPServer by Route2;
SIPServer, by the private network IP in the SDP of 200OK and port, replaces to public network IP and the port of SIPServer, then the 200OK after replacing is sent to Route1;
The SDP of 200OK with public network IP and port is sent to the sip terminal on PC1 by Route1;
Sip terminal on PC1 returns ACK to Route1;
ACK is sent to SIPServer by Route1;
ACK is transmitted to Route2 by SIPServer;
ACK message is sent to the sip terminal on PC2 by Route2;
Rtp streaming is sent to SIPServer by the sip terminal on PC1 and PC2, and SIPServer does the forwarding of rtp streaming, makes the sip terminal normal talking on PC1 and PC2;
SIPServer receives the rtp streaming that the sip terminal on PC2 sends out through Route2, gets IP and port that sip terminal on PC2 sends the NAT network that rtp streaming uses;
The SDP that sip terminal on PC2 is brought by SIPServer is modified as NAT network IP and the port that sip terminal on PC2 sends rtp streaming use, sends reInvite message to Route1;
ReInvite message is sent to the sip terminal on PC1 by Route1;
Sip terminal on PC1 sends 200OK message to Route1;
200OK message is sent to SIPServer by Route1;
SIPServer responds ACK to Route1;
ACK is sent to the sip terminal on PC1 by Route1;
Rtp streaming alters course by the sip terminal on PC1, is sent on Route2;
SIPServer receives the rtp streaming that the sip terminal on PC1 sends out through Route1, thus gets IP and port that sip terminal on PC1 sends the NAT network that rtp streaming uses;
The SDP that sip terminal on PC1 is brought by SIPServer is modified as NAT network IP and the port that sip terminal on PC1 sends rtp streaming use, sends reInvite message to Route2;
ReInvite message is sent to the sip terminal on PC1 by Route2;
Sip terminal on PC2 sends 200OK message to Route2;
200OK message is sent to SIPServer by Route2;
SIPServer responds ACK to Route2;
ACK is sent to the sip terminal on PC2 by Route2;
Rtp streaming alters course by the sip terminal on PC2, is sent on Route1;
Make the rtp streaming of PC1 and PC2 into ad hoc mode.
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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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| 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 |
Citations (1)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN101834874A (en) * | 2010-05-21 | 2010-09-15 | 四川长虹电器股份有限公司 | Multimedia network communication method capable of penetrating firewall |
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| WO2007019583A2 (en) * | 2005-08-09 | 2007-02-15 | Sipera Systems, Inc. | System and method for providing network level and nodal level vulnerability protection in voip networks |
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| CN101834874A (en) * | 2010-05-21 | 2010-09-15 | 四川长虹电器股份有限公司 | Multimedia network communication method capable of penetrating firewall |
Non-Patent Citations (1)
| Title |
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| P2P穿越NAT的几种实现方式研究;刘镇瑜等;《军事通信技术》;20090925;第30卷(第3期);第3节,图3 * |
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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. |
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