CN101552848B - Session method and apparatus based on session initiation protocol - Google Patents
Session method and apparatus based on session initiation protocol Download PDFInfo
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Abstract
The present invention provides a session method and apparatus based on session initiation protocol. The session method is establishing a SIP message channel between two session terminals and using theSIP message to load and transmit media message. According to the invention, a SIP message and a RTP/RTCP message traversing a NAT can be implemented and then a session between a user agent and other user agent of private network can be implemented without function of a NAT gateway supporting a ALG and without setting additional STUN server.
Description
Technical field
The present invention relates to session initiation protocol (SIP, Session Initiation Protocol) technical field, be specifically related to a kind of session method and equipment of dialogue-based initiation protocol.
Background technology
IP phone (VoIP; Voice over Internet Protocol) based on the IP packet switching network; Through the conventional analogue voice signal is carried out digitlization, compression, packing, a series of processing such as encapsulation framing make speech business to carry through IP network.For example, IP phone is exactly the typical application of VoIP.
SIP uses the widest voip technology at present, also is a kind of important communication agreement of 3G (Third Generation) Moblie.Sip message comprises sip request message and sip response message two big classes.The typical networking of SIP is used as shown in Figure 1.Among Fig. 1, registration and acting server (Registrar Server&Proxy Server), user agent (UA, User Agent) 1, UA2 directly are connected on the public network, all use public network IP address; Be installed in soft phone (Softhone) and UA3 on PC (PC) machine all in private network, and link to each other with public network through network address translation (NAT, NetworkAddress Translation) gateway.Soft phone can be regarded as a kind of of UA.
Among Fig. 1, UA1, registration and acting server, UA2 obtain public network IP address, and equal route can reach mutually.Wherein, the public network IP address of UA1 is 172.33.28.33, and the public network IP address of UA2 is 172.33.27.45, and the public network IP address of registration and acting server is 172.33.255.254.The flow process that the public network IP terminal uses public network IP address to register and call out is as shown in Figure 2.Fig. 2 has listed the critical field in each sip message.Wherein mainly may further comprise the steps:
S201～S204, UA1 and UA2 be respectively to registration and acting server registration, and oneself SIP unified resource identifier (URI, Uniform Resource Identifiers) is filled in the Contact header field that SIP registers (REGISTER) request message; After registration and acting server are received the REGISTER request message, the information in the Contact header field (Contact address) is recorded in the database, uses for follow-up calling, and return 200OK message.Shown in Figure 2 mutual in, UA1 that preserves in the database of registration and acting server and the Contact address of UA2 are respectively:
Sip:100172.33.28.33:5060, and sip:200172.33.27.45:5060.
When S205～S206, UA1 need call out UA2 (number of UA2 is 200), send SIP through registration and acting server to UA2 and invite (INVITE) message.In the INVITE shown in Fig. 2; C and m field are respectively and are used to set up required media address and the port numbers of VoIP conversation; Be address 172.33.28.33 and the port numbers 16844 that local terminal is monitored media message; RTP (RTP, Real-time Transport Protocol) or RTCP Real-time Transport Control Protocol (RTCP, Real-time Transport Control Protocol) message will be sent to this address and port in the opposite end; The value of the Contact header field in the INVITE has shown the address that subsequent request will mail to; The media capability collection information that also includes UA1 in the INVITE and supported is like the RTP/AVP 18804 expression speech codings that UA1 supported, to carry out media capability negotiation with UA2.Also include the number 200 of calling out opposite end UA2 in the INVITE shown in Figure 2, after registration and acting server receive above-mentioned INVITE,, thereby INVITE is transmitted to UA2 according to the registered address of the number searching of calling out opposite end UA2 to UA2.
S207～S211, UA2 returns 100Trying or 180Ringing response message through registration and acting server to UA1, and the expression request has been received and has been handled; Behind the UA2 off-hook; Return the sip response message of 200OK to UA1; The value of the Contact header field in this response message has shown the address that subsequent request will mail to, and in this message, equally local terminal is monitored address and the port numbers of media message and the voice coding modes information notification UA1 such as (wherein RTP/AVP 18 represent G.729 speech coding) that chooses; Afterwards, UA1 further returns ACK message to UA2, and UA2 is after receiving the ACK message that UA1 sends, and capability negotiation has just been accomplished, and audio call is able to set up successfully.
S212～S214, then, UA1 and UA2 send the RTP message to the address/port number that monitor the opposite end respectively, to carry out voice communication.Any end is wanted to take out stitches, and all can directly send SIP BYE message to the opposite end according to the value of Contact header field, and receiving after the 200OK that returns the opposite end replys, calls out and just removed.
Attention: because INVITE among the S206 and the Contact header field in the 200OK message among the S210 have been indicated the address that subsequent request mails to; So ACK response message among the S211 and the SIP BYE message among the S213 all will directly be sent out to the opposite end according to the value of Contact.
Be positioned at the soft phone of private network behind the NAT gateway,, following problem can occur if register and call out according to the flow process of Fig. 2:
Soft phone is issued the sip message of registration and acting server when arriving NAT gateway 1; NAT gateway 1 will carry out the NAT conversion; Source address and UDP (UDP with the IP header of this sip message; User Datagram Protocol) source port number of header changes to public network IP address and public network port numbers, and in addition, other content that relates to private network IP address/port number in the sip message is not changed thereupon; So this sip message by 1 forwarding of NAT gateway that registration and acting server are received maybe be as follows, wherein the private network IP address/port number of underscore part is not by 1 change of NAT gateway:
REGISTER?sip：172.33.255.254SIP/2.0
Via：SIP/2.0/UDP?192.168.1.3:5061；branch＝z9hG4bK258e3d43b96
Call-ID：HAnGUeIiNgSlZj124821548y0192.168.1.3
From：<sip：80481001172.33.255.254:5060>；tag＝b961466b
To：<sip：80481001172.33.255.254:5060>
CSeq：1REGISTER
Contact：<sip：80481001192.168.1.3:5061>
Expires：3600
Max-Forwards：70
Content-Length：0
According to general handling process shown in Figure 2, registration and acting server will write down the value of Contact header field: 80481001192.168.1.3:5061.Because registration and acting server do not arrive the route of private net address (192.168.1.3), will cause follow-up sip message can't return to soft phone, and then cause soft phone to set up session with the UA in the public network.For addressing the above problem, adopt following technical scheme in the prior art usually, to realize sip message passing through NAT gateway:
1) the NAT gateway is supported SIP application level gateway (ALG, Application Level Gateway) function, by the NAT gateway private net address/port numbers in the TCP/UDP load is revised as public network address/port numbers external on the NAT gateway.
2) utilize UDP simple traversal network address translater (STUN, Simple Traversalof User Datagram Protocol Through Network Address Translators) agreement to carry out NAT and pass through.This scheme need increase a STUN server in public network, and requires UA to support the STUN client functionality.
Following shortcoming is arranged in the above-mentioned prior art scheme:
1) NAT ALG mode need be discerned sip message through port numbers, and different if the SIP port numbers of UA and ALG are provided with, then ALG can't come into force.Though the default port numbers of SIP is 5060, the port numbers of a lot of soft phones and IP phone all is other port that adopts outside 5060 at present; Secondly, along with increasing of sip message and header field kind, ALG also needs constantly to upgrade thereupon; Once more, during actual networking, need all NAT gateways all to support SIP ALG, can't set up success otherwise possibly cause SIP to call out.
2) utilizing the STUN technology to carry out NAT passes through: this method at first need be in public network extra STUN server of increase; Also require sip terminal to support the STUN client functionality simultaneously; And along with the continuous enhancing of NAT security feature; Existing STUN technology has been difficult to accomplish separately the function of passing through NAT, must combine with other technology and can realize that just NAT passes through.
Summary of the invention
Embodiment of the invention technical problem to be solved provides a kind of session method and equipment based on SIP; The NAT gateway device is supported the ALG function; Extra STUN server need be set yet, can realize user agent and the session between other user agent in the private network.
For solving the problems of the technologies described above, the embodiment of the invention provides scheme following:
The session method of a kind of dialogue-based initiation protocol SIP, the user agent UA in the private network is connected to public network through network address translation NAT gateway, and said session method comprises:
UA sets up the corresponding relation between an IP address/port number and the 2nd IP address/port number; A said IP address/port number is the employed source IP address/source port number of sip message that UA sends; To be the NAT gateway carry out public network IP address/port numbers of obtaining after NAT handles to the source IP address/source port number in the first kind message to said the 2nd IP address/port number, and the purpose IP address/destination slogan of said first kind message is that the registration and IP address/port number, the source IP address/source port number of acting server that are arranged in the public network are a said IP address/port number;
UA utilize sip message to said registration and acting server is registered and is consulted to set up and the session opposite end between session; And the media message that generates in the session process is encapsulated in the sip message of specified type as the sip message body; And the sip message of said specified type being transmitted to said session opposite end through said registration and server, the sip message of said specified type is used to supply the session opposite end to resolve the media message wherein to be carried; Wherein, after said corresponding relation was set up, UA earlier according to said corresponding relation, replaced with the 2nd IP address/port number with the IP address/port number in this sip message header field part, and then this sip message is sent when sending any sip message.
Preferably, in the above-mentioned session method, set up said corresponding relation and comprise:
UA is that a said IP address/port number, purpose IP address/destination slogan are first sip request message of the IP address/port number of said registration and acting server to registration and acting server transmission source IP address/source port number, and the source IP address/source port number of said first sip request message is converted into said the 2nd IP address/port number at said NAT gateway place;
UA receives said registration and acting server is directed against the SIP response message that said first sip request message returns, and a said SIP response message carries said the 2nd IP address/port number;
UA extracts said the 2nd IP address/port number from a said SIP response message, set up the corresponding relation between a said IP address/port number and said the 2nd IP address/port number.
Preferably, in the above-mentioned session method, set up said corresponding relation and comprise:
UA is through carrying out alternately with said NAT gateway, obtains said NAT gateway and the source IP address/source port number in the said first kind message carried out the 2nd IP address/port number that obtains after NAT handles;
UA sets up the corresponding relation between a said IP address/port number and said the 2nd IP address/port number according to said the 2nd IP address/port number of obtaining.
Preferably; In the above-mentioned session method; Source IP address/the source port number of said media message is the IP address/port number that UA monitors media message, and the purpose IP address/destination slogan of said media message is the IP address/port number that media message is monitored in said session opposite end.
Preferably, in the above-mentioned session method, said media message comprises live transmission protocol message and RTCP Real-time Transport Control Protocol message.
The embodiment of the invention also provides a kind of user agent UA, is connected to public network through network address translation NAT gateway, and said UA comprises:
Corresponding relation is set up the unit; Be used to set up the corresponding relation between an IP address/port number and the 2nd IP address/port number; A said IP address/port number is the employed source IP address/source port number of sip message that UA sends; To be the NAT gateway carry out public network IP address/port numbers of obtaining after NAT handles to the source IP address/source port number in the first kind message to said the 2nd IP address/port number, and the purpose IP address/destination slogan of said first kind message is that the registration and IP address/port number, the source IP address/source port number of acting server that are arranged in the public network are a said IP address/port number;
The SIP conversation element; Be used to utilize sip message to said registration and acting server is registered and is consulted to set up and the session opposite end between session; And the media message that generates in the session process is encapsulated in the sip message of specified type as the sip message body; And the sip message of said specified type being transmitted to said session opposite end through said registration and server, the sip message of said specified type is used to supply the session opposite end to resolve the media message wherein to be carried; Wherein, after said corresponding relation is set up, when sending any sip message, according to said corresponding relation the IP address/port number in this sip message header field part is replaced with the 2nd IP address/port number earlier, and then this sip message is sent.
Preferably, among the above-mentioned UA, said corresponding relation is set up the unit and is comprised:
Send subelement; Being used for to registration and acting server transmission source IP address/source port number is that a said IP address/port number, purpose IP address/destination slogan are first sip request message of the IP address/port number of said registration and acting server, and the source IP address/source port number of said first sip request message is converted into said the 2nd IP address/port number at said NAT gateway place;
The reception subelement is used to receive said registration and acting server is directed against the SIP response message that said first sip request message returns, and a said SIP response message carries said the 2nd IP address/port number;
Safeguard subelement, be used for extracting said the 2nd IP address/port number, and set up the corresponding relation between a said IP address/port number and said the 2nd IP address/port number from a said SIP response message.
Preferably, among the above-mentioned UA, said corresponding relation is set up the unit and is comprised:
Acquiring unit is used for through carrying out alternately with said NAT gateway, obtains said NAT gateway and the source IP address/source port number in the said first kind message is carried out the 2nd IP address/port number that obtains after NAT handles;
Safeguard subelement, be used for setting up the corresponding relation between a said IP address/port number and said the 2nd IP address/port number according to said the 2nd IP address/port number of obtaining.
Preferably, among the above-mentioned UA, the source IP address/source port number of said media message is the IP address/port number that UA monitors media message, and the purpose IP address/destination slogan of said media message is the IP address/port number that media message is monitored in said session opposite end.
Preferably, among the above-mentioned UA, said media message comprises live transmission protocol message and RTCP Real-time Transport Control Protocol message.
Can find out from the above; Session method and equipment thereof that the embodiment of the invention provides based on SIP; Set about from Session Initiation Protocol itself; Under the situation that does not change existing network element structure, can realize that the NAT of sip message passes through, and sip terminal in the realization private network and the media communication between the sip terminal in the sip terminal in the public network, the sip terminal in the private network and other private network; Present embodiment can be fit to present most networking and use; Need not the NAT gateway and support the ALG function; Extra STUN server or VPN(Virtual Private Network) related network elements need be set yet, can realize that the NAT of media message SIP passes through, simplify networking greatly; And present embodiment can be good at compatible existing Session Initiation Protocol.
Description of drawings
Fig. 1 is the SIP typical case networking sketch map of prior art;
Fig. 2 is the registration call flow process sketch map between the public network UA in the prior art;
Fig. 3 is the flow chart of the described session method based on SIP of the embodiment of the invention;
Fig. 4 is the interaction diagrams of PROBE message and REGISTER message in the embodiment of the invention;
Fig. 5 is the flow chart that SIP calls out in the embodiment of the invention;
Fig. 6 is the transmission sketch map of media message in the embodiment of the invention;
Fig. 7 is the structural representation of UA described in the embodiment of the invention.
Embodiment
Session method and equipment that the embodiment of the invention provides based on SIP; Through existing Session Initiation Protocol being expanded and perfect; Make and to realize sip message passing through NAT gateway based on the SIP self mechanism; And send to the session opposite end through media message being encapsulated in sip message, and then realized that the NAT of media message passes through, make that the UA in the private network can carry out session with the UA in other private network or the public network.Below will combine accompanying drawing the present invention to be done further explanation through specific embodiment.
Please refer to Fig. 3, in the session method based on SIP of the embodiment of the invention, the UA in the private network is connected to public network through the NAT gateway, and said session method may further comprise the steps:
Step 31; UA sets up and safeguards the corresponding relation between an IP address/port number and the 2nd IP address/port number; A said IP address/port number is the employed source IP address/source port number of sip message that UA sends; To be the NAT gateway carry out public network IP address/port numbers of obtaining after NAT handles to the source IP address/source port number in the first kind message to said the 2nd IP address/port number, and the purpose IP address/destination slogan of said first kind message is that the registration and IP address/port number, the source IP address/source port number of acting server that are arranged in the public network are a said IP address/port number;
Be example with applied environment shown in Figure 1 below,, above-mentioned steps and the sip message that adopted be elaborated through the session flow process of soft phone in the private network 1 and the UA3 in the private network 2.Present embodiment is expanded existing Session Initiation Protocol and is strengthened, and has realized that the NAT of sip message and media message passes through.But need to prove that present embodiment only provides a kind of extension mechanism to Session Initiation Protocol, the sip message form that present embodiment adopted does not constitute restriction to protection scope of the present invention.
In order on UA, to set up said corresponding relation, present embodiment has at first defined a kind of new sip request message: PROBE message.UA is before initiating registration and calling out; Need to send PROBE message earlier and give registration and acting server; Source IP address/source port number in the IP header of PROBE message is private network IP address/SIP port numbers of UA, the i.e. employed private network IP of Session Initiation Protocol address/port number.With the soft phone among Fig. 1 is example, and Fig. 4 has provided the interaction flow of concrete PROBE message and REGISTER message and the critical field in each message:
S301; Suppose that the employed IP address/port number of Session Initiation Protocol is 172.33.255.254/5060 on registration and the acting server; The IP address/port number that SIP uses on the soft phone is: 192.168.1.3:5061 (an IP address/port number); Source IP address/the source port number that is the sip message that sends of soft phone is inner private net address 192.168.1.3:5061, and purpose IP address/destination slogan is 172.33.255.254/5060.Soft phone at first sends a PROBE message before registration and acting server initiation registration, and the form after the content of this PROBE message and the encapsulation is as shown in table 1:
Table 1
According to the relevant regulations of SIP, the value of the Via header field in the sip message is used to show the address that this sip message is responded; The value of Contact header field is used to the address that shows that subsequent request will mail to.The value of the Via header field in the present embodiment in the PROBE message is 192.168.1.3:5061, and the value of Contact header field is 80481001192.168.1.3:5061.
S302; This PROBE message arrives NAT gateway 1; And carry out network address translation (nat) by NAT gateway 1 and handle: at first; NAT gateway 1 distributes the public network IP address/port numbers of an outside according to the source IP address/source port number and the purpose IP address/destination slogan of this PROBE message, and this public network IP address/port numbers is corresponding with the source IP address/source port number and the purpose IP address/destination slogan of this PROBE message; Then; Source IP address/the source port number of this PROBE message is converted into the public network IP address/port numbers (the 2nd IP address/port number) of outside by the private network IP address/port number (an IP address/port number) of inside; And then with registration and acting server in this PROBE forwards to public network, and preserve above-mentioned corresponding relation.When supposing this PROBE message through NAT gateway 1, NAT gateway 1 replaces with 2.2.2.30:12288 with the source IP address/source port number of PROBE message by 192.168.1.3:5061, and in nat translation table, preserves corresponding relation as shown in table 2.
| Inner private network IP address | The inside end slogan | Purpose IP address | The destination slogan | Outside public network IP address | The outer end slogan | Protocol type |
| 192.168.1.3 | 5061 | 172.33.255.254 | 5060 | 2.2.2.30 | 12288 | UDP |
Table 2
Table 3
S303, registration and acting server are replied the 200OK response message to this PROBE message, and the form of response message is as shown in table 4:
Table 4
In the 200OK response message shown in the table 4, Via header field is: Via:SIP/2.0/UDP192.168.1.3:5061; Received=2.2.2.30;
Re-port=12288Via header field to the RFC3261 regulation in the present embodiment should be used as expansion, has increased the re-port parameter:
Regulation according to RFC3261; If the source IP address of the sip request message of receiving is different with the IP address of Via header field; Then in the Via header field of response message, add a received parameter; This received parameter has indicated the IP address of really sending sip request message, and will send to the response message of this sip request message on the IP address that the received parameter indicated.Present embodiment is not in the IP address of the source IP address of the sip request message of receiving and Via header field not simultaneously; Further on the basis of RFC3261, increase a re-port parameter; This re-port parameter and received parameter are together; Accurately having indicated the IP address/port number of sending sip request message, be exactly 2.2.2.30:12288 in the present embodiment, and this response message also will be sent out to 2.2.2.30:12288.
S304; After NAT gateway 1 is received the IP message that this 200OK replys; Search the nat translation table of self preserving, find the occurrence that is complementary, then corresponding N AT conversion is done in " purpose IP address/destination slogan "; Convert the IP address/port number 192.168.1.3:5061 of inner private network into, and then issue 192.168.1.3:5061.200OK response message after the NAT conversion is as shown in table 5:
Table 5
After soft phone is received this 200OK response message; According to wherein the received parameter and the re-port parameter of Via header field; Just know that the corresponding public network address of self 192.168.1.3:5061 is 2.2.2.30:12288, thereby can set up a mapping table, as shown in table 6.The mapping table (table 2) of this mapping table and NAT gateway is consistent, has also just reached the purpose of obtaining public network IP address/port numbers that NAT gateway 1 shone upon for private network IP address/port number 192.168.1.3:5061.Can also and receive corresponding response message through regular transmission PROBE message in the present embodiment, safeguard said mapping table, reach the purpose of regular update corresponding relation according to the value of Via header field in the response message.
| Inner private network IP address | The inside end slogan | Purpose IP address | The destination slogan | Outside public network IP address | The outer end slogan | Protocol type |
| 192.168.1.3 | 5061 | 172.33.255.254 | 5060 | 2.2.2.30 | 12288 | UDP |
Table 6
S305, after generating said mapping table, soft phone sends the sip request message of REGISTER and registers to registration and acting server.Here; Soft phone generates REGISTER message according to existing normal flow; And then, this REGISTER message header field private network IP address/port number is partly all replaced with public network IP address/port numbers according to above-mentioned mapping table, again REGISTER message is sent.Concrete, Via header field is replaced with 2.2.2.30:12288 by 192.168.1.3:5061, Contact header field is replaced with 804810012.2.2.30:12288 by 80481001192.168.1.3:5061.Also possibly include other header field in the sip message, like From header field, Remote-party-ID header field etc., the private network IP address/port number in these header fields also will replace with public network IP address/port numbers.The REGISTER message that final soft phone sent is as shown in table 7:
Table 7
S306～S308; After REGISTER message arrives NAT gateway 1; NAT gateway 1 is handled according to normal N AT handling process, can find to have had at present a mapping item to be complementary with this REGISTER message, therefore can not distribute new address/port number to 192.168.1.3:5061 again; But carry out the NAT conversion, and then REGISTER message is sent to registration and acting server according to this mapping item.Therefore, registration and the REGISTER message that receives of acting server are as shown in table 8:
Table 8
At this moment; In the REGISTER message that registration and acting server are received; Source IP address/source port number in IP header and the UDP header is " 2.2.2.30/12288 ", and is identical with the Via header field of REGISTER message, and IP address in registration and the acting server record Contact header field and port numbers are as the log-on message of soft phone at this moment; So that soft phone is registered, and return the 200OK response message.Because the source IP address/source port number of REGISTER message is identical with the IP address/port number of Via header field, can not add received and re-port parameter in this response message.
Like this, through above-mentioned S301～S308, the soft phone in the private network 1 successfully has been registered on registration and the acting server, and the log-on message of soft phone on registration and acting server is: 804810012.2.2.30:12288.The INVITE of follow-up this soft phone of calling will be sent to NAT gateway 1 (2.2.2.30:12288) by registration and acting server; Then, undertaken sending to the soft phone in the private network 1 after the NAT conversion, thereby realized that sip message directly visits private network from public network by NAT gateway 1.
Can find out that above-mentioned S301～S308 has got through the SIP layer, realize sip message passing through NAT gateway, therefore can utilize sip message to carry media message, between public network, private network, transmit medium stream information.For this reason, present embodiment has also defined another kind of new sip request message and has been used to carry media message: STRAM message.Utilize SIP tunnel of SIP STRAM message constructing, realize that the NAT of media message passes through.
With applied environment shown in Figure 1 is example, and UA3 (SIP:40010.0.0.2:5060) is connected to public network through NAT gateway 2 and NAT gateway 1 respectively with among soft phone (SIP:80481001192.168.1.3:5061) is in different private networks separately.Suppose that UA3 and soft phone all have been registered on registration and the acting server through the method for introducing among above-mentioned S301～S308.Wherein, the mapping item that generated of two NAT gateways is as shown in table 9:
| The NAT gateway | Inner private network IP address | The inside end slogan | Purpose IP address | The destination slogan | Outside public network IP address | The outer end slogan | |
| NAT gateway | |||||||
| 1 | 192.168.1.3 | 5061 | 172.33.255.254 | 5060 | 2.2.2.30 | 12288 | |
| NAT gateway | |||||||
| 2 | 10.0.0.2 | 5060 | 172.33.255.254 | 5060 | 202.106.0.20 | 12345 | UDP |
Table 9
Soft phone and the UA3 log-on message on registration and acting server is as shown in table 10:
| Sequence number | Telephone number | SIP? |
| 1 | 80481001 | SIP：804810012.2.2.30:12288 |
| 2 | 400 | SIP：400202.106.0.20:12345 |
Table 10
In the present embodiment,, when initiating SIP calling (INVITE) request, will require to use new STREAM message to encapsulate the media message that comprises RTP and RTCP message because UA3 and soft phone all are in after the NAT gateway.In the present embodiment; UA is through designated parameter " STREAM " in the Support of sip message header field; Representing that this UA supports the STREAM encapsulation, and through in Require, carrying parameter " STREAM ", show in the private network that lays oneself open to after the NAT gateway; Require the session both sides to use STREAM message that follow-up media message is encapsulated, concrete calling interaction flow is as shown in Figure 5:
S401～S402, the negotiation of INVITE request message: soft phone and UA3 are in the NAT back in the present embodiment, and all support the expansion to SIP.Soft phone is when calling out UA3, and the telephone number 400 generation INVITE request messages according to UA3 are transmitted to UA3 through registration and acting server with this INVITE request message.This INVITE includes Session Description Protocol (SDP, SessionDescription Protocol) message body, and the media capability collection information that wherein includes UA3 and supported is used for carrying out media capability negotiation with the session opposite end; All fill in STREAM in the Supported of this INVITE and the Require, represent that respectively this soft phone supports STREAM to encapsulate and follow-up needs use SIP STREAM message encapsulation RTP and RTCP message; This INVITE request message sends to registration and acting server through NAT gateway 1; Registration and acting server are again according to the log-on message of UA3; This INVITE request message is sent to NAT gateway 2 (202.106.0.20:12345), be forwarded to UA3 by NAT gateway 2 again.
As shown in Figure 5, soft phone generates the INVITE request message, and wherein SDP has partly comprised the address and the port numbers of soft telephone monitoring media message, is respectively 16844 in 192.168.1.3 and the m field in the o field, that is, and and 192.168.1.3:16844; The IP header of this INVITE request message and the source IP address of UDP header and source port number are 192.168.1.3:5061; Before sending this INVITE request message; Soft phone is according to the mapping table of setting up among Fig. 4; The private network IP address/port number of INVITE request message header field part is replaced with external public network IP address/port numbers, that is, the 192.168.1.3:5061 in Via and the Contact header field is replaced with 2.2.2.30:12288; Then, again this INVITE request message is sent to registration and acting server; This INVITE request message will at first arrive NAT gateway 1, carry out NAT by NAT gateway 1 according to table 9 and handle, and source IP address and source port number replaced with 2.2.2.30:12288 by 192.168.1.3:5061, and then send to registration and acting server; Registration and acting server send to NAT gateway 2 (202.106.0.20:12345) according to the log-on message of UA3 in the table 10 with this INVITE request message; Carry out NAT by NAT gateway 2 according to table 9 again and handle, wherein purpose IP address and destination slogan replaced with 10.0.0.2:5060 by 202.106.0.20:12345, this INVITE request message sends to UA3 the most at last.
S403～S404, UA3 return 100Trying or 180Ringing response message, and the expression request is. and receive and handle.
S405～S408, UA3 return the response message of 200OK, and the key content of this message please refer to Fig. 5, and wherein SDP partly includes G.729 the encoding and decoding agreement of consulting to confirm monitors media message with UA3 address and port numbers: contents such as 10.0.0.2:16384.Through the forwarding of registration and acting server, the soft phone of the final arrival of this 200OK response message; Soft phone returns ACK message to registration and acting server, and registration and acting server are given UA3 with this ACK forwards.UA3 is after receiving this ACK message, and the capability negotiation between soft phone and the UA3 is accomplished, and session is set up successfully.Mutual address of media message and port are: 192.168.1.3:16844 <-10.0.0.2:16384.
Be example with soft phone below, introduce the STREAM encapsulation (UA3 adopts identical encapsulation process) of media message.
After session is set up successfully; Soft phone generates media message according to normal flow; And media message is encapsulated in the IP message; So the IP header of this media message and the purpose IP address/destination slogan in the UDP header are the address/port number that UA3 monitors media message: 10.0.0.2:16384, source IP address/source port number are the address/port number of soft telephone monitoring media message: 192.168.1.3:16844.Table 11 shows the encapsulating structure of media message (RTP message), and the RTP message comprises RTP head and voice payload.
Table 11
Then, with the message body of the media message shown in the table 11 as SIP STREAM message, be encapsulated in the SIP STREAM message, and then STREAM message is carried out the IP encapsulation, the structure of the STREAM message after the encapsulation is as shown in table 12:
| The IP header | UDP header | Sip message |
Table 12
The IP header in the table 12 and the structure of UDP header are as shown in table 13, and the form of the sip message in the table 12 is as shown in table 14:
Table 13
Table 14
Wherein, The Via of sip message and Contact header field are all according to the mapping table of previous foundation in the table 14; Original private network IP address/port number is replaced with public network IP address/port numbers, that is, the 192.168.1.3:5061 in Via and the Contact header field is replaced with 2.2.2.30:12288.
Media message between soft phone and the UA3 transmits through STREAM message, and idiographic flow please refer to shown in Figure 6:
S501～S502; Mutual address of media message and port numbers between soft phone basis and the UA3; Generate media message according to normal flow; The IP header of this media message and the source IP address/source port number in the UDP header are the IP address/port number 192.168.1.3:16844 of soft telephone monitoring media message, and purpose IP address/destination slogan is the address/port number 10.0.0.2:16384 that UA3 monitors media message; Then, this media message is encapsulated in the STREAM message as the sip message body, and is public network IP address/port numbers according to the private network IP address/port number of the corresponding relation replacement STREAM message header field part of setting up in advance; Then, this STREAM message is sent to registration and acting server; Include the number of session opposite end UA3 in this STREAM message, thereby registration and acting server can be sent to UA3 with this STREAM message according to the number searching of the UA3 log-on message to UA3.Can find out that the pass-through mode of STERAM message is similar with INVITE, the NAT in the repeating process handles and repeats no more here.UA3 is after receiving this STREAM message; It is carried out decapsulation; Obtain sip message body wherein; The purpose IP address/destination slogan of the media message that obtains after the decapsulation is private network IP address and the port numbers 10.0.0.2:16384 that UA3 monitors media message, so UA3 can carry out handled to media message according to existing normal process flow process, obtains voice messaging wherein.
S503～S508, UA3 returns the 200OK response message to registration and acting server; And follow-up UA3 can adopt the handling process among similar S501～S502, utilizes STREAM message to send media message, thereby realized interactive voice two-way between soft phone and the UA3.The transmission of RTCP message and RTP message are similar, repeat no more.
The principle that the above-mentioned SIP of utilization STREAM message is transmitted media message is: utilize the sip message this " tunnel " that not influenced by NAT; The media message that follow-up needs are mutual (RTP and RTCP message) is as the sip message body; Be encapsulated in the sip message of specified type and go, after the sip message of specified type is received in the session opposite end, carry out decapsulation like this; Can extract the content in the sip message body, just can handle then according to existing normal process flow process.
Two kinds of new sip request message: PROBE and STREAM message have been defined among the above embodiment.UA in the private network at first sends SIP PROBE message to registration and acting server; Registration and acting server are then to the PROBE message response time; Return public network IP address/port numbers that this PROBE message obtains after NAT gateway NAT handles, then UA be able to set up and safeguard said corresponding relation, and according to said corresponding relation; The private network IP address/port number of header field part in the sip message of follow-up transmission is replaced with public network IP address/port numbers, pass through with the NAT that realizes sip message.And; In the foregoing description; UA also is encapsulated in the media message that generates in the session process in the STREAM message and sends, and the session opposite end receives after the STREAM message can obtain media message after the decapsulation, and according to the normal handling flow process media message is handled.
NAT gateway in the present embodiment is just accomplished the NAT translation function of inner private net address/port numbers to outside public network address/port numbers; Therefore need not improve in the present embodiment, adopt the prior NAT gateway can realize present embodiment the prior NAT gateway.
Compare with existing Session Initiation Protocol; Processing to sip message in the present embodiment has 2 differences: 1) UA is after setting up said corresponding relation; Before sending sip message, can be according to the private network IP address/port number of Via and Contact header field in the said corresponding relation replacement sip message; 2) registration and acting server also can return one and be used to identify the port numbers of sending this sip request message in the IP address of the source IP address of the sip request message of receiving and Via header field not simultaneously when this sip request message is responded.Except this 2 point, present embodiment is consistent to the processing mode and the prior art of sip message, with existing Session Initiation Protocol good compatibility is arranged.
In the foregoing description; UA sets up and safeguards that said corresponding relation can also adopt alternate manner: for example; Can adopt the mode of manual configuration; Confirm that at first UA goes up SIP employed private network IP address/port number and NAT gateway the source IP address/source port number in the first kind message is carried out public network IP address/port numbers of obtaining after NAT handles, the purpose IP address/destination slogan of said first kind message is that the registration and IP address/port number, the source IP address/source port number of acting server that are arranged in the public network are a said IP address/port number; Then, on UA, dispose this corresponding relation again; Again for example; Can also be: UA be through carrying out alternately with the NAT gateway; Obtain the NAT gateway and the source IP address/source port number in the said first kind message is carried out the 2nd IP address/port number that obtains after NAT handles; UA sets up and safeguards said corresponding relation according to said the 2nd IP address/port number of obtaining then.
Can find out from the above; Present embodiment is set about from Session Initiation Protocol itself; Under the situation that does not change existing network element structure; The NAT that can realize sip message passes through, and sip terminal in the realization private network and the media communication between the sip terminal in the sip terminal in the public network, the sip terminal in the private network and other private network; Present embodiment can be fit to present most networking and use; Need not the NAT gateway and support ALG function or STUN function; Also need not newly-increased any server or VPN(Virtual Private Network) related network elements, can realize that the NAT of media message SIP passes through, simplified networking greatly; And present embodiment can be good at compatible existing Session Initiation Protocol.
Based on above-mentioned SIP session method, present embodiment also provides a kind of user agent UA.Be connected to public network through the NAT gateway, as shown in Figure 7, this UA comprises:
Corresponding relation is set up the unit; Be used to set up the corresponding relation between an IP address/port number and the 2nd IP address/port number; A said IP address/port number is that UA goes up the employed private network IP of session initiation protocol SIP address/port number, and said the 2nd IP address/port number is that the NAT gateway is public network IP address/port numbers that a said IP address/port number is shone upon;
The SIP conversation element; Be used to utilize sip message to registration and acting server is registered and is consulted to set up and the session opposite end between session; And the media message that generates in the session process is encapsulated in as the sip message body in the sip message of specified type and sends to the session opposite end, the sip message of said specified type is used to supply the session opposite end to resolve the media message wherein to be carried; Wherein, after said corresponding relation is set up, when sending any sip message, according to said corresponding relation the IP address/port number in this sip message header field part is replaced with the 2nd IP address/port number earlier, and then this sip message is sent.
Preferably, above-mentioned corresponding relation is set up the unit and can be comprised:
Send subelement; Being used for to registration and acting server transmission source IP address/source port number is first sip request message of a said IP address/port number, and the source IP address/source port number of said first sip request message is converted into said the 2nd IP address/port number at said NAT gateway place;
The reception subelement is used to receive said registration and acting server is directed against the SIP response message that said first sip request message returns, and a said SIP response message carries said the 2nd IP address/port number;
Safeguard subelement, be used for extracting said the 2nd IP address/port number, and set up the corresponding relation between a said IP address/port number and said the 2nd IP address/port number from a said SIP response message.
Preferably, above-mentioned corresponding relation is set up the unit and can be comprised:
Acquiring unit, being used for through carrying out alternately with said NAT gateway, obtaining said NAT gateway is the 2nd IP address/port number that a said IP address/port number is shone upon;
Safeguard subelement, be used for setting up the corresponding relation between a said IP address/port number and said the 2nd IP address/port number according to second public network IP address/port numbers of obtaining.
Preferably, said SIP conversation element can also be further used for when sending any sip message, according to said corresponding relation, the IP address/port number in Via header field and the Contact header field in this sip message being replaced with the 2nd IP address/port number.
Preferably, the source IP address/source port number of said media message is the IP address/port number that UA monitors media message, and the purpose IP address/destination slogan of said media message is the IP address/port number that media message is monitored in said session opposite end.Said media message comprises live transmission protocol message and RTCP Real-time Transport Control Protocol message.
In sum, the embodiment of the invention has realized that through to expanding existing Session Initiation Protocol the NAT of sip message and media message passes through.
The above only is an execution mode of the present invention; Should be pointed out that for those skilled in the art, under the prerequisite that does not break away from the principle of the invention; Can also make some improvement and retouching, these improvement and retouching also should be regarded as protection scope of the present invention.
Claims (10)
1. the session method of a dialogue-based initiation protocol SIP, the user agent UA in the private network is connected to public network through network address translation NAT gateway, it is characterized in that, and said session method comprises:
UA sets up the corresponding relation between an IP address/port number and the 2nd IP address/port number; A said IP address/port number is the employed source IP address/source port number of sip message that UA sends; To be the NAT gateway carry out public network IP address/port numbers of obtaining after NAT handles to the source IP address/source port number in the first kind message to said the 2nd IP address/port number, and the purpose IP address/destination slogan of said first kind message is that the registration and IP address/port number, the source IP address/source port number of acting server that are arranged in the public network are a said IP address/port number;
UA utilize sip message to said registration and acting server is registered and is consulted to set up and the session opposite end between session; And the media message that generates in the session process is encapsulated in the sip message of specified type as the sip message body; And the sip message of said specified type being transmitted to said session opposite end through said registration and acting server, the sip message of said specified type is used to supply the session opposite end to resolve the media message wherein to be carried; Wherein, after said corresponding relation was set up, UA earlier according to said corresponding relation, replaced with the 2nd IP address/port number with the IP address/port number in this sip message header field part, and then this sip message is sent when sending any sip message.
2. session method as claimed in claim 1 is characterized in that, sets up said corresponding relation and comprises:
UA is that a said IP address/port number, purpose IP address/destination slogan are first sip request message of the IP address/port number of said registration and acting server to registration and acting server transmission source IP address/source port number, and the source IP address/source port number of said first sip request message is converted into said the 2nd IP address/port number at said NAT gateway place;
UA receives said registration and acting server is directed against the SIP response message that said first sip request message returns, and a said SIP response message carries said the 2nd IP address/port number;
UA extracts said the 2nd IP address/port number from a said SIP response message, set up the corresponding relation between a said IP address/port number and said the 2nd IP address/port number.
3. session method as claimed in claim 1 is characterized in that, sets up said corresponding relation and comprises:
UA is through carrying out alternately with said NAT gateway, obtains said NAT gateway and the source IP address/source port number in the said first kind message carried out the 2nd IP address/port number that obtains after NAT handles;
UA sets up the corresponding relation between a said IP address/port number and said the 2nd IP address/port number according to said the 2nd IP address/port number of obtaining.
4. like claim 2 or 3 described session methods; It is characterized in that; Source IP address/the source port number of said media message is the IP address/port number that UA monitors media message, and the purpose IP address/destination slogan of said media message is the IP address/port number that media message is monitored in said session opposite end.
5. session method as claimed in claim 4 is characterized in that, said media message comprises live transmission protocol message and RTCP Real-time Transport Control Protocol message.
6. a user agent UA is connected to public network through network address translation NAT gateway, it is characterized in that said UA comprises:
Corresponding relation is set up the unit; Be used to set up the corresponding relation between an IP address/port number and the 2nd IP address/port number; A said IP address/port number is the employed source IP address/source port number of sip message that UA sends; To be the NAT gateway carry out public network IP address/port numbers of obtaining after NAT handles to the source IP address/source port number in the first kind message to said the 2nd IP address/port number, and the purpose IP address/destination slogan of said first kind message is that the registration and IP address/port number, the source IP address/source port number of acting server that are arranged in the public network are a said IP address/port number;
The SIP conversation element; Be used to utilize sip message to said registration and acting server is registered and is consulted to set up and the session opposite end between session; And the media message that generates in the session process is encapsulated in the sip message of specified type as the sip message body; And the sip message of said specified type being transmitted to said session opposite end through said registration and acting server, the sip message of said specified type is used to supply the session opposite end to resolve the media message wherein to be carried; Wherein, after said corresponding relation is set up, when sending any sip message, according to said corresponding relation the IP address/port number in this sip message header field part is replaced with the 2nd IP address/port number earlier, and then this sip message is sent.
7. UA as claimed in claim 6 is characterized in that, said corresponding relation is set up the unit and comprised:
Send subelement; Being used for to registration and acting server transmission source IP address/source port number is that a said IP address/port number, purpose IP address/destination slogan are first sip request message of the IP address/port number of said registration and acting server, and the source IP address/source port number of said first sip request message is converted into said the 2nd IP address/port number at said NAT gateway place;
The reception subelement is used to receive said registration and acting server is directed against the SIP response message that said first sip request message returns, and a said SIP response message carries said the 2nd IP address/port number;
Safeguard subelement, be used for extracting said the 2nd IP address/port number, and set up the corresponding relation between a said IP address/port number and said the 2nd IP address/port number from a said SIP response message.
8. UA as claimed in claim 6 is characterized in that, said corresponding relation is set up the unit and comprised:
Acquiring unit is used for through carrying out alternately with said NAT gateway, obtains said NAT gateway and the source IP address/source port number in the said first kind message is carried out the 2nd IP address/port number that obtains after NAT handles;
Safeguard subelement, be used for setting up the corresponding relation between a said IP address/port number and said the 2nd IP address/port number according to said the 2nd IP address/port number of obtaining.
9. like claim 7 or 8 described UA; It is characterized in that; Source IP address/the source port number of said media message is the IP address/port number that UA monitors media message, and the purpose IP address/destination slogan of said media message is the IP address/port number that media message is monitored in said session opposite end.
10. UA as claimed in claim 9 is characterized in that, said media message comprises live transmission protocol message and RTCP Real-time Transport Control Protocol message.
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