CN102546996A - Method for realizing bit-by-bit receiving and transmitting of called number under SIP (Session Initiation Protocol) - Google Patents
Method for realizing bit-by-bit receiving and transmitting of called number under SIP (Session Initiation Protocol) Download PDFInfo
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- CN102546996A CN102546996A CN2010106102068A CN201010610206A CN102546996A CN 102546996 A CN102546996 A CN 102546996A CN 2010106102068 A CN2010106102068 A CN 2010106102068A CN 201010610206 A CN201010610206 A CN 201010610206A CN 102546996 A CN102546996 A CN 102546996A
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Abstract
The invention provides a method for realizing bit-by-bit receiving and transmitting of a called number under an SIP (Session Initiation Protocol). The method comprises the following steps of: a, using a plurality of INVITE transactions in a conversation; b, establishing the conversation according to the called number and establishing new INVITE transactions through subsequent numbers by a number transmitting party; confirming the current INVITE transactions carrying more complete called numbers by a number receiving party; and releasing other INVITE transactions carrying incomplete called numbers by two parties; and c, establishing the calling by using the INVITE transactions carrying complete called numbers through the two parties. According to the method disclosed by the invention, under the condition that proprietary extension for the SIP is not needed, the transmitting and receiving of the bit-by-bit numbers can be realized.
Description
Technical field
The present invention relates to the SIP communications field, relate in particular to a kind of use Session Initiation Protocol and carry out session foundation, and when call setup initially can not get access to full called number, follow-up transmission is the method for called number by turn.
Background technology
In the existing communication system, often run into the situation that dials by turn in the terminal, the user can't send a definite end and indicate to switching system in dialing procedure.Switching system is analyzed through built-in number analysis mechanism, just can navigate to the type of number and the direction of sending according to the former position of number, even so; Switching system might not be known the actual complete length of called number, and such as certain out number, perhaps concrete length have only landing office to know; In this case; Out again after calling can not wait number to collect, can only initiate outbound call earlier according to former item sign indicating numbers of receiving, and then send the follow-up number that the terminal is dialed; Recipient's (such as landing office) up to called number confirms that number collects, and just no longer needs the dialing of receiving terminal.
The trunk protocol of traditional switched telephone network has generally all been considered above-mentioned situation, such as agreements such as No. 1, No. 7, all has regulation one cover signaling mechanism to realize this demand.
Session Initiation Protocol is widely used in the foundation of various sessions at present, but does not stipulate a kind of method of the follow-up called number of transmission of standard, generally all is to use all number sign indicating number.If must send follow-up number; Everybody uses and more just is to use INFO to carry follow-up number at present; But,, can adopt the mode of TEXT so the form of INFO needs the receiving-transmitting sides regulation of holding consultation because agreement do not have concrete regulation; Also can adopt the mode of DTMF, can certainly adopt other privately owned extended message bodies to carry.
In sum, utilize the existing standard Session Initiation Protocol, neither one standard uniform way realizes the transmission and the reception of called number by turn.
Summary of the invention
The invention provides and realize the method that called number is received and dispatched by turn under a kind of Session Initiation Protocol, it need not carry out privately owned expansion and negotiation to message format, only uses Session Initiation Protocol to support the characteristic of eventful affair to realize.
This method comprises: use a plurality of INVITE affairs in dialogue of a.; B. send the number root and set up dialogue, set up new INVITE affairs through follow-up number according to the part called number; Receiving number side confirms current INVITE affairs of carrying full called number; Both sides discharge other and carry the not INVITE affairs of full called number; C. both sides use the INVITE affairs of carrying full called number to set up calling.
Further, the part called number among the step b is enough to make the direction of sending affirmation receiving number side, number side; Other carries not, and the release of the INVITE affairs of full called number realizes through the mode of using failure response or timer expiry or other sip agreements to allow; Full called number is confirmed by receiving number side among the step c, and to sending the loopback success response of number side.
Preferably, a plurality of INVITE affairs, according to the order of setting up, its called number that carries is based on the called number of its previous INVITE affairs and progressively is tending towards complete; Last INVITE affairs has been carried full called number; Carry the not INVITE affairs of full called number, receive a number after more full INVITE affairs, can discharge in receiving number side; Set up when calling out, use last to carry the INVITE affairs of full called number.
Can find out by technical scheme of the present invention, use the standard Session Initiation Protocol to support the characteristics of a plurality of affairs, can accomplish the transmission and the reception of follow-up called number by turn.The distinguishing feature of this invention is, a plurality of INVITE affairs both were related, and were relatively independent again, because the called number that they carry all is a complete relatively called number, this provides a lot of possibilities for concrete implementation of using.
Description of drawings
The sketch map that Fig. 1 uses Session Initiation Protocol to carry out network interconnection for a kind of soft switchcall server;
The use Session Initiation Protocol that Fig. 2 provides for the embodiment of the invention carries out the concrete realization handling process sketch map of follow-up by turn number transmitting-receiving.
Embodiment
In Fig. 1, provided a sketch map that soft switchcall server uses Session Initiation Protocol to carry out network interconnection.Use the Session Initiation Protocol of IP based network transmission to carry out intercommunication between soft switchcall server, UAC here and UAS represent originating end and the receiving terminal that SIP calls out respectively.
The embodiment of the invention, can but be not limited to the network architecture shown in Fig. 1.Fig. 1 has been merely the explanation of Fig. 2 and a simple signal is provided.
A kind of execution mode of the method that the present invention carried in the SIP session is following:
Session is set up: UAC carries out number analysis according to the part called number of receiving, analyzes in the time of enough can confirming the UAS direction, just initiates an INVITE according to current number and calls out, and continues to prepare to receive follow-up called number then; UAS extracts called number after receiving that first initial INVITE calls out, and carries out number analysis, and analysis result does not collect, and just continues to wait for that UAC sends follow-up number.
Follow-up number sends and receives: after UAC gets access to one or the follow-up called number of multidigit; Number be connected to before received the last period called number the back; Just form a most complete present called number, initiate new INVITE affairs according to this number then; UAS finds that number upgrades to some extent after receiving new INVITE request, just proceeds number analysis, to previous INVITE request echo failure response (like 484 responses) early, finishes this INVITE affairs simultaneously.Said process is along with the transmission and the reception of follow-up number move in circles.
Call setup: in the process that above-mentioned follow-up number receives, UAS constantly uses new number to carry out number analysis, and after confirming that number has collected, UAC and UAS just set up calling according to the INVITE affairs that last has carried the called number that collects.
Eventful affair is safeguarded: carry not a plurality of INVITE affairs of all number sign indicating number; All in the described process of above-mentioned execution mode; Use failure response to be released successively, therefore both sides can safeguard a plurality of INVITE affairs simultaneously, and this is an optimal way of the present invention.During practical implementation, can be according to own characteristic, in the scope that the sip agreement allows; Simultaneously a plurality of affairs are managed; Realize the final release of the infull INVITE affairs of number through some other modes, such as discharging through the affairs timer expiry, unified discharge etc. behind the call setup or when finishing.
For ease of above-mentioned execution mode is understood, will do further explanation to concrete implementation of the present invention with the concrete example that is embodied as below.
As shown in Figure 2, the called number of assumption of complete consists of: number 0+ number 1+ number 2.
Wherein: number 0: the several leading position (1 or multidigit) of called number, it is enough to confirm to call out is to send out to which UAS.Number 1: following the follow-up number of number 0 in the called number, can be 1, also can be multidigit.Number 2: following the follow-up number of number 1 in the called number, can be 1, also can be multidigit.
The processing of UAC:
Step 1 receives that the user dials the called number that cries by turn, when receiving number 0, through the number analysis of inside, can judge the UAS that which will send to confirm.
Step 2 is used number 0 to send invite as called number and is given UAS, is called for short INVITE0 here.At this moment, write down the transaction information of the INVITE0 request of oneself sending, write down the number of having received 0 simultaneously.
Step 3 before the final response of receiving the UAS loopback, continues to dial by turn the called number 1 that cries if receive the user once more, just combines number 0 and number 1, and be used as called number and send INVITE1 to UAS, is new affairs here certainly.In addition, equally to write down the transaction information of the INVITE1 request of oneself sending, write down the number 0+ number of having received 1 simultaneously.Same, receive that once more the user dials called number 2,3, the 4...... that cries by turn, processing is not always the case.
Step 4 if receive the final response of the failure of UAS loopback, will be compared the failure response of the invite affairs of whether current record so earlier, if not, then only need to finish its pairing those invite affairs, otherwise, whole session finished.
Step 5 if receive the final response of the success of UAS loopback, is then upgraded the current sessions information of local record immediately according to the session information of success response, so just can guarantee the consistent affairs with the final activity of UAS.So far, the number process of transmitting has just been ended, and new invite affairs are not sent to UAS again in the back.
The processing of UAS:
Whether step 1 receives the INVITE0 that UAC sends, collected by the number analysis program decision called number (number 0) of inside, if do not collect or when temporarily can't judge, then continue to wait for.
Step 2 behind the INVITE1 that reception UAC sends, returns 484 for earlier previous INVITE0.To judge whether called number " number 0+ number 1 " collects by the number analysis program of inside equally, if also do not collect or when temporarily can't judge, then continue to wait for.Make mistakes if handle, send 484 then for earlier previous INVITE, finish current affairs then, and call release.Same, receive INVITE2,3,4...... that UAC sends once more, operation is not always the case.
Step 3 if receive certain invite that UAC sends, is found the enough words of its entrained called number (hypothesis is INVITE2) here, just need carry out a series of built-in functions, determines whether this session goes on (promptly accepting still refusal).If accepted this calling, will give response of successful such as UAC loopback 18* or 2** according to situation so, the process that collects the digits finishes, and continues follow-up normal call flow then.Otherwise, with regard to the corresponding failure response of loopback, end session.Identical with UAC, so far, the number receiving course has just been ended, and the invite affairs (directly loopback failure response) that UAC sends are no longer handled in the back.
In the present embodiment, a plurality of INVITE affairs both were related, and were relatively independent again, because the called number that they carry all is a complete relatively called number, this provides a lot of possibilities for the concrete implementation of using.
In sum, in the whole SIP call establishment that the embodiment of the invention provides, need not carry out privately owned expansion and negotiation, utilize the foundation and the release of a plurality of INVITE affairs, realize that called number receives and dispatches by turn message format.
The above; Be merely the preferable embodiment of the present invention, but protection scope of the present invention is not limited thereto, any technical staff who is familiar with the present technique field is in the technical scope that the present invention discloses; The variation that can expect easily or replacement all should be encompassed within protection scope of the present invention.Therefore, protection scope of the present invention should be as the criterion with the protection range of claim.
Claims (8)
1. a Session Initiation Protocol is realized the method that called number is received and dispatched by turn down, it is characterized in that this method comprises:
A. use a plurality of INVITE affairs in a dialogue;
B. send the number root and set up dialogue, set up new INVITE affairs through follow-up number according to the part called number; Receiving number side confirms current INVITE affairs of carrying full called number; Both sides discharge other and carry the not INVITE affairs of full called number;
C. both sides use the INVITE affairs of carrying full called number to set up calling.
2. method according to claim 1 is characterized in that step b, and described part called number is enough to make the direction of sending affirmation receiving number side, number side.
3. method according to claim 1 is characterized in that step b, and described release realizes through failure response.
4. method according to claim 1 is characterized in that step b, and described release realizes through timer expiry.
5. method according to claim 1 is characterized in that step c, and described full called number confirmed by receiving number side, and to sending the loopback success response of number side.
6. method according to claim 1 is characterized in that step a, described a plurality of INVITE affairs, and according to the order of setting up, its called number that carries is based on the called number of its previous INVITE affairs and progressively is tending towards complete; Last INVITE affairs has been carried full called number.
7. method according to claim 6 is characterized in that step b, and the described not INVITE affairs of full called number of carrying receive a number after more full INVITE affairs in receiving number side, are released.
8. method according to claim 7 is characterized in that step c, uses last to carry the INVITE affairs of full called number in the described calling.
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| Application Number | Priority Date | Filing Date | Title |
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| CN2010106102068A CN102546996A (en) | 2010-12-28 | 2010-12-28 | Method for realizing bit-by-bit receiving and transmitting of called number under SIP (Session Initiation Protocol) |
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| CN2010106102068A CN102546996A (en) | 2010-12-28 | 2010-12-28 | Method for realizing bit-by-bit receiving and transmitting of called number under SIP (Session Initiation Protocol) |
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Cited By (1)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN108809911A (en) * | 2017-05-04 | 2018-11-13 | 中兴通讯股份有限公司 | The method, apparatus and storage medium of two-stage dialing are realized in VoLTE networks |
Citations (2)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN1878162A (en) * | 2005-06-07 | 2006-12-13 | 中兴通讯股份有限公司 | Mapping method for protocol interconnection entity to superposition transmitting-receiving code signalling |
| CN1984199A (en) * | 2005-12-14 | 2007-06-20 | 华为技术有限公司 | Method for overlapped transmitting number by session initial protocol |
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- 2010-12-28 CN CN2010106102068A patent/CN102546996A/en active Pending
Patent Citations (2)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN1878162A (en) * | 2005-06-07 | 2006-12-13 | 中兴通讯股份有限公司 | Mapping method for protocol interconnection entity to superposition transmitting-receiving code signalling |
| CN1984199A (en) * | 2005-12-14 | 2007-06-20 | 华为技术有限公司 | Method for overlapped transmitting number by session initial protocol |
Cited By (2)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN108809911A (en) * | 2017-05-04 | 2018-11-13 | 中兴通讯股份有限公司 | The method, apparatus and storage medium of two-stage dialing are realized in VoLTE networks |
| CN108809911B (en) * | 2017-05-04 | 2021-11-12 | 中兴通讯股份有限公司 | Method, device and storage medium for realizing secondary dialing in VoLTE network |
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Application publication date: 20120704 |