CN104506523A - Call forwarding method under VoIP (voice over Internet protocol) of intelligent terminal - Google Patents
Call forwarding method under VoIP (voice over Internet protocol) of intelligent terminal Download PDFInfo
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- CN104506523A CN104506523A CN201410802281.2A CN201410802281A CN104506523A CN 104506523 A CN104506523 A CN 104506523A CN 201410802281 A CN201410802281 A CN 201410802281A CN 104506523 A CN104506523 A CN 104506523A
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/1066—Session management
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/1066—Session management
- H04L65/1069—Session establishment or de-establishment
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04W—WIRELESS COMMUNICATION NETWORKS
- H04W4/00—Services specially adapted for wireless communication networks; Facilities therefor
- H04W4/16—Communication-related supplementary services, e.g. call-transfer or call-hold
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Abstract
The invention relates to a telephone call forwarding technology, and discloses a high-efficiency call forwarding method under the VoIP (voice over Internet protocol) of an intelligent terminal, the problems that basic call flows need to be changed, interaction signaling needs to be added, and call forwarding cannot be successfully performed under the condition that multiple calls exist simultaneously are solved by adopting the call forwarding scheme under the VoIP of the intelligent terminal in the prior art. The call forwarding method comprises the following steps: A, establishing a first call between a first terminal and a second terminal; B, when the second terminal needs to forward the call of the first terminal to a third terminal, firstly, keeping the first call, and then, sending an INVITE message of a second call to a VoIP server, wherein the INVITE message carries the relevant information of the first call; C, after the INVITE message is received by the VoIP server, establishing the second call between the second terminal and the third terminal, and associating the first call with the second call; D, after the second terminal is hung up, establishing a third call between the first terminal and the third terminal by the VoIP server. The call forwarding method disclosed by the invention is suitable for single-terminal and multi-call telephone call forwarding.
Description
Technical field
The present invention relates to call switching technique, be specifically related to the efficient call forwarding method of a kind of intelligent terminal VoIP.
Background technology
Current network develop rapidly, the network bandwidth is more and more higher, and network data transmission cost is more and more lower.Add that WLAN's in recent years is universal, 3G/4G bandwidth is more and more higher, and the super-speed development of intelligent terminal.Compared to the high charge of Incumbent, the VoIP call of " freely " has very huge advantage in competition.People are more and more higher to the development of the demand of VoIP business also with network.VoIP (Voice over IP) is the switching network based on IP grouping, and by carrying out digitlization, compression, packing to traditional analog voice signal, a series of process such as encapsulation framing, make speech business can be carried by IP network.In a broad sense, VoIP refers to multimedia services such as carrying data, voice, fax and image over data networks, and in a narrow sense namely refers to and upload sending voice business at IP.RTP (Real-time Transfer Protocol, RTP) is the main protocol of media information in VoIP, comprises voice, image etc.
SIP (Session Initiation Protocol, session initiation protocol) be one for setting up, changing and stop the application layer control protocol of Multimedia session, session wherein can be IP phone, Multimedia session or multimedia conferencing.Session Initiation Protocol is as a kind of text based application layer protocol, and the advantage of the aspects such as simple with it, easy-to-use, easy debugging, independence, high expansion, becomes the most important control signal bearing protocol in VoIP field.SDP (Session Description Protocol, Session Description Protocol) be a kind of application layer control protocol being used for describing Multimedia session, it is a text based agreement, for the negotiation etc. of the medium type in session establishment process and encoding scheme.SDP is generally embedded in SIP message body and transmits.
Call diversion is one of function the most basic in traditional telephony call, and especially in the internal telephone system of enterprise, group, its application is very general.Call diversion under VoIP, the control signal of standard relies on SIP REFER to realize.Traditional analog station, due to the restriction of number of keys, the extensibility of its function is very poor, there is contradiction between call diversion and calling keep in application.SIP REFER is when design, and consider compatibility issue when connecing IAD (Intergrated Access Device, integrated access equipment) to analog station, the realization of switching is very complicated.IAD, as the access device between analog station and VoIP system, is responsible for the conversion between circuit signal to IP message.
And for intelligent terminal, based on its high intelligent and programmability, by expansion Session Initiation Protocol, the flow process of switching can be simplified completely significantly.Shorten transit time, reduce taking of Internet resources.
For analog station, traditional call diversion flow process is as follows:
1. caller A and called B is formed and converses;
Either party hooking of 2.A or B, the calling of the opposing party kept, the opposing party now listens holding tone.For B hooking, the now calling of A is kept (flow process of A hooking is the same, repeats no more);
3.B dials, and number is third party C;
Talkthrough between 4.B and C; Or C ring, B listens ring-back tone;
5.B on-hook.A and C forms call; Or A listens ring-back tone, formed with A after C off-hook and converse;
In the process that server (telephone exchange) is held, flow process is as follows:
Call between 1.A and B is once independently call out concerning server;
After A keeps by 2.B, B calls out C again.Concerning server, the action that B calls out C is also once independently call out;
Call between 3.A and B is now hold mode.Call between B and C is now call or ringing condition;
4. because analog station does not support multi-way call, two callings can only be had at most simultaneously, and only have one to converse as active state; At server end, just can set up the association between two independent call by the number of B, after B on-hook, A and C can form call;
In the present invention, be unified address, be called " caller " by the A in above-mentioned flow process, B is called " called ", and because B is the executor of switching, therefore B also claims " mediator ", and C is called " by mediator ".Calling between calling and called is called " first calling ", mediator and being called " second call out " by the calling between mediator, caller and be called " the 3rd call out " (together lower) between mediator.
In the calling of traditional analog station, only allow a phone can only have two callings, one of them is hold mode, and another is talking state, and two calls switch by hooking action at most simultaneously.
And the VoIP application program on intelligent terminal, particularly in the industry that some are special (as LTE-R), a terminal allows there is multiple calling, one of them is talking state, and other all-calls are all hold mode.Switching cannot be realized according to the mode of analog station because server cannot know switching action for be multiple be in the calling of hold mode which.
For intelligent terminal voip call switching, the implementation of current main flow adopts SIP REFER to realize, and also has part manufacturer to use SIP INFO to realize.
1) SIP INFO implementation is adopted:
1. mediator carries by the number of mediator in INFO, by VoIP server by number notifying caller.
2. the serviced device release of the calling of mediator.
3. caller initiates new calling again from backstage, and calling is arrived by mediator.
If 4. call out unsuccessful, server incites somebody to action call diversion side again, and caller re-ringing, to recover original call.
Single SIP INFO, autgmentability is very poor, and shortcoming is obvious.Manufacturer is had to use in early days, now substantially superseded.
2) SIP REFER implementation is adopted:
In SIP REFER standard, in message body, carry refer-to and referred-by field, represent respectively by mediator's number and mediator's number.New calling is all that server is initiated by resolving REFER message.
There is the time of (this action can be arranged, as pressed a string special number or special button) according to switching action, being divided into again three kinds of modes:
1, N/A REFER, namely transfers after mediator's dialing.REFER message sends immediately after dialing.Transfer after successfully (caller and conversed between mediator set up), server notifies mediator by NOTIFY.
2, have the REFER of early notification, mediator after by mediator's ring, transfers.REFER message sends after mediator hears ring-back tone.Same, after transfer successfully (caller and conversed between mediator set up), server notifies mediator by NOTIFY.
3, have the REFER of notice, mediator after by mediator's off-hook, transfers.REFER message mediator and by mediator between converse set up after send.Same, after transfer successfully (caller and conversed between mediator set up), server notifies mediator by NOTIFY.
SIP REFER implementation is adopted to there is following defect for intelligent terminal voip call switching in conventional art:
1. flow process is complicated, and switching flow process disunity, causes logic complicated, underaction.
2. need to increase repeatedly Signalling exchange, interacting message is too much, can produce the obvious time delay of call setup, also can take network
Resource is more obvious under 3g/4g condition.
3. change original call flow completely, in order to support SIP REFER, the change of VoIP server end can be very large.
4. need terminal and server to support SIP NOTIFY simultaneously, whether transfer successfully for notice.
5. there is the situation of multiple calling for intelligent terminal simultaneously, SIP REFER cannot know switching action for be the call which is kept, therefore successfully cannot carry out call diversion.
Summary of the invention
Technical problem to be solved by this invention is: the efficient call forwarding method proposing a kind of intelligent terminal VoIP, what the intelligent terminal voip call switching scheme in conventional art that solves existed need to change basic call flow process, need to increase mutual signaling, there is multiple call case at the same time under, successfully cannot carry out the problem of call diversion.
The technical solution adopted for the present invention to solve the technical problems is: the call forwarding method of a kind of intelligent terminal VoIP, comprises the following steps:
A. the first calling is set up between first terminal and the second terminal;
B. when the second terminal needs the call with first terminal to transfer to third terminal, first the first calling is kept, then send the INVITE of the second calling to VoIP server, in this INVITE, carry the first call-related information;
After C.VoIP server receives described INVITE, set up second between the second terminal and third terminal and call out, and the first calling and second are called out and associated;
D., after the second hanging up terminal, VoIP server sets up the 3rd calling between first terminal and third terminal.
Concrete, in steps A, the method setting up the first calling between first terminal and the second terminal comprises:
First terminal is as caller, and after dialing, send INVITE to VoIP server, described INVITE comprises the relevant information of the first calling: call-id, from, to; This INVITE sends to as the second called terminal by VoIP server, now the second terminal ring, and first terminal listens ring-back tone, after the second terminal off-hook, sets up the call between first terminal.
Or the second terminal is as caller, and send INVITE to VoIP server after dialing, described INVITE comprises the relevant information of the first calling: call-id, from, to; This INVITE sends to as called first terminal by VoIP server, now first terminal ring, and the second terminal listens ring-back tone, after first terminal off-hook, sets up the call between the second terminal.
Concrete, in step B, described second terminal is called out the method for carrying out keeping by first and is comprised:
Second terminal sends the REINVITE Messages Message of SDP=0.0.0.0 to VoIP server, and this REINVITE message is sent to first terminal by VoIP server, and now first terminal listens holding tone.
Concrete, in step B, the INVITE that described second terminal sends the second calling to VoIP server comprises described second call-id, from, to information of calling out, and also comprises call-id, from, to information of the first calling.
Concrete, in step C, the first calling comprises with the method for the second correlation of call by described VoIP server:
VoIP server parses call-id, from, to information of the first calling from described INVITE, and according to this information searching parsed to the first calling, then the first calling and the second correlation of call are got up.
Concrete, in step C, the second method of calling out that described VoIP server is set up between the second terminal and third terminal comprises:
VoIP server sends call-id, from, to information of the second calling to third terminal, and now third terminal ring, the second terminal listens ring-back tone.
Particularly, in step D, after the second hanging up terminal, the method for the 3rd calling that VoIP server is set up between first terminal and third terminal comprises:
Second terminal is setting up with third terminal rear on-hook of conversing, second terminal sends BYE to VoIP server, VoIP server SIP terminated between the second terminal after receiving BYE is connected, discharge the first calling and the second calling simultaneously, VoIP server sets up the 3rd calling, send REINVITE (SDP) respectively to first terminal and third terminal and carry out media re-negotiation, between first terminal and third terminal, set up call.
Or in step D, after the second hanging up terminal, the method for the 3rd calling that VoIP server is set up between first terminal and third terminal comprises:
Second terminal with third terminal set up converse before on-hook, second terminal sends BYE to VoIP server, VoIP server discharges the first calling and the second calling immediately after receiving BYE, and send REINVITE (SDP) to first terminal, media are redirected to the source of sound of ring-back tone, now first terminal listens ring-back tone, after third terminal off-hook, VoIP server sends 200OK (SDP) to first terminal, and send REINVITE (SDP) to third terminal, carry out media re-negotiation, between first terminal and third terminal, set up call.
The invention has the beneficial effects as follows: 1. depart from SIP REFER, greatly simplify the flow process of switching, logic is simple; 2. compared with ordinary call, in ordinary call flow process originally, just add extended field, do not change original Signalling exchange, amendment is simple, compatible good; 3. can support the switching of unit/terminal many callings simultaneously, facilitate user.
Accompanying drawing explanation
Fig. 1 is call diversion principle schematic of the present invention;
Fig. 2 is the call diversion schematic diagram of mediator after the switch under receiver-on-hook condition in embodiment 1;
Fig. 3 is the call diversion schematic diagram of mediator after ring under receiver-on-hook condition in embodiment 2.
Embodiment
The present invention is intended to the efficient call forwarding method proposing a kind of intelligent terminal VoIP, what the intelligent terminal voip call switching scheme in conventional art that solves existed need to change basic call flow process, need to increase mutual signaling, there is multiple call case at the same time under, successfully cannot carry out the problem of call diversion.
In standard SIP protocol, call-id, from and to tri-fields uniquely can determine a calling.And the key of call diversion is to set up the association between twice calling.Core concept of the present invention is the expansion to call signaling, call-id, from and to tri-fields of the first calling are carried as extended field when the second calling is initiated, thus, VoIPVoIP server just can know that the second calling is the call switching action for the first calling, thus release first calling and the second call setup the 3rd are called out, and realize call diversion.
Can as common user terminal, no matter support multiway calling, but a road calling can only be had to be talking state at synchronization, other calling be all necessary for hold mode.For ease of describing, forming calling to only have two terminals simultaneously and setting forth call diversion principle of the present invention:
As shown in Figure 1, it comprises following performing step:
1, calling is called, the first call setup, and VoIP server will set up the session association between calling and called;
2, called by caller maintenance, the session of the first calling still exists, unaffected;
3, called as mediator's calling by mediator, the second call setup, VoIP server sets up mediator and by the session association between mediator;
If when 4 second call setup, do not carry the extended field about the first calling, VoIP server keeps process by common calling, and after mediator's on-hook, two callings are all released, and the 3rd calling also can not be set up; If when the second call setup, carry about first calling extended field extended field, VoIP server will set up the first calling call out with second between associate.
5, when mediator's on-hook, VoIP server sets up the 3rd calling, the dotted portion namely in Fig. 1, discharges the first calling and the second calling simultaneously.
The core point realizing call diversion due to the present invention is to carry the extended field about the first calling when the second calling is initiated, specific implementation can be done interpolation " switching " option on intelligent terminal APP call interface, and (" maintenance " option is the basic function of terminal, original interface exists, for keeping calling), if press " switching " option, then dialing interface can be ejected, dial after by mediator's number mediator, the call-id of the first calling is obtained from software backstage, from, to tri-fields, when the second calling is initiated, these three fields are put in the INVITE of the second calling as extended field, so when VoIP server receives this INVITE, resolve three extended fields simultaneously, then be judged to be that this time calling is for call diversion, VoIP server finds the first calling according to extended field, again the first calling and the second correlation of call are got up, after mediator's on-hook, VoIP server end makes caller and is formed the 3rd calling by mediator.
Embodiment 1:
This example is the situation for mediator and on-hook after being conversed by mediator, and (in order to stress the process of switching, the first call establishment and maintenance being simplified, not illustrating completely) as shown in Figure 2, call diversion flow process is as follows:
1) caller dials called number, the INVITE comprising this calling (the first calling) is sent to VoIP server, contain the relevant information of this calling in this message: call-id1, from1, to1, VoIP server sends to called after receiving this INVITE; Now called terminal ringing, ring-back tone is listened in caller, sets up the call between caller afterwards at called off-hook (being press Answer Key for mobile phone terminal, lower same); In figure, RTP1 represents and sets up call between calling and called, sends RTP Media Stream each other;
2) wish the first call diversion to during by mediator as mediator when called, first need to keep the first calling, the action that REINVITE (SDP=0.0.0.0) in figure is called maintenance caller, the effect of REINVITE (SDP) is that notice caller changes media connection.Because now caller should listen holding tone;
3) after the first calling is kept, calledly initiate new calling (the second calling) as mediator, equally, calledly dial by the number of mediator, new INVITE is sent to VoIP server, the relevant information of the second calling is not only included: call-id2, from2, to2, also carry the first call-related information: call-id1, from1, to1 in this new INVITE;
4) after VoIP server receives new INVITE, find that there is extended field, know that this time calling is for switching, so set up contact by between the first calling with the second calling, preserve the SDP of tripartite's (caller, called, by mediator);
5) VoIP server is to the INVITE being sent normal call by mediator, now by mediator's ring, mediator listens ring-back tone, after by mediator's off-hook, set up the call between mediator, in figure, RTP2 is mediator and by the call Media Stream between mediator;
6), during mediator's on-hook, BYE information is sent to VoIP server;
7) VoIP server receives BYE, and end is connected with the SIP between called;
8) VoIP server sends REINVITE respectively to caller with by mediator, carries the SDP after negotiation and connects for setting up Media Stream;
9) caller and set up the 3rd calling between mediator, in figure, RTP3 is caller and by the call Media Stream between mediator.
Embodiment 2:
This example is for the situation of mediator with (mediator hear the ring-back tone by mediator after) on-hook of not undertaken conversing by mediator, as shown in Figure 3, its with call after on-hook difference in flow process:
1, after mediator's on-hook, VoIP server sends out 180RING to caller, and caller can listen ring-back tone.
2, after mediator's on-hook, VoIP server discharges the first calling and the second calling immediately.
3, by after mediator's off-hook, in the 200OK that VoIP server sends to caller, carry SDP, again consult so do not need to send out a REINVITE.
4 and for by mediator, what preserve due to it is the media information of mediator, so will send REINVITE message to by mediator.
The solution of the present invention is not only supported to the switching of unit/terminal unitary call, also supports the switching of the many callings of unit/terminal,
Such as: user A converses with user D, be in hold mode with the call of user B and C simultaneously, (2 is hold mode server will to record 3 independently calls, 1 is talking state), if now user A wants the call with user C to transfer to user E, the call interface that then user A is first manually switched to C (just illustrates for changing interface herein, tube terminal does not adopt any mode to manage multi-way call, the inevitable mode that each road of operation separately all can be had to converse) press " switching " button, call with D is kept, now three calls of A and BCD are all in hold mode,
Then user A dials, when calling party E, three fields such as call-id, from, to (because " switching " button presses under the interface of conversing with C) of calling out with C are carried, as the extended field of invite message in SIP INVITE.That is, user directly can specify in terminal and want for which call to transfer.Server when parsing SIP INVITE, according to 3 extended fields, just can know this call diversion for be calling between former A and C.When A hangs up relayed call time, the call that server will be connected between CE, call diversion completes.
It should be noted that, existing " maintenance " button separately of terminal can only realize the signaling transfer point of unit/terminal unitary call, because the object when transferring only has one, judges call diversion to which object without the need to server.In order to realize the signaling transfer point of the many callings of unit/terminal in the present invention, end application call interface adding " switching " button, retaining " maintenance " button simultaneously.These two buttons are distinguishing in the software operation that backstage is corresponding: the INVITE after pressing " switching " carries extended field, do not carry extended field by the INVITE after " maintenance ".
Claims (9)
1. a call forwarding method of intelligent terminal VoIP, is characterized in that, comprises the following steps:
A. the first calling is set up between first terminal and the second terminal;
B. when the second terminal needs the call with first terminal to transfer to third terminal, first the first calling is kept, then send the INVITE of the second calling to VoIP server, in this INVITE, carry the first call-related information;
After C.VoIP server receives described INVITE, set up second between the second terminal and third terminal and call out, and the first calling and second are called out and associated;
D., after the second hanging up terminal, VoIP server sets up the 3rd calling between first terminal and third terminal.
2. the call forwarding method of a kind of intelligent terminal VoIP as claimed in claim 1, is characterized in that, in steps A, the method setting up the first calling between first terminal and the second terminal comprises:
First terminal is as caller, and after dialing, send INVITE to VoIP server, this INVITE sends to as the second called terminal by VoIP server, now the second terminal ring, first terminal listens ring-back tone, after the second terminal off-hook, sets up the call between first terminal.
3. the call forwarding method of a kind of intelligent terminal VoIP as claimed in claim 1, it is characterized in that, in steps A, second terminal is as caller, and after dialing, send INVITE to VoIP server, this INVITE sends to as called first terminal by VoIP server, now first terminal ring, second terminal listens ring-back tone, after first terminal off-hook, sets up the call between the second terminal.
4. the call forwarding method of a kind of intelligent terminal VoIP as claimed in claim 2 or claim 3, is characterized in that, in step B, described second terminal is called out the method for carrying out keeping by first and comprised:
Second terminal sends the REINVITE message of SDP=0.0.0.0 to VoIP server, and this REINVITE message is sent to first terminal by VoIP server, and now first terminal listens holding tone.
5. the call forwarding method of a kind of intelligent terminal VoIP as claimed in claim 2 or claim 3, it is characterized in that, in step B, the INVITE that described second terminal sends the second calling to VoIP server comprises described second call-id, from, to information of calling out, and also comprises call-id, from, to information of the first calling.
6. the call forwarding method of a kind of intelligent terminal VoIP as claimed in claim 5, is characterized in that, in step C, the first calling comprises with the method for the second correlation of call by described VoIP server:
VoIP server parses call-id, from, to information of the first calling from described INVITE, and according to this information searching parsed to the first calling, then the first calling and the second correlation of call are got up.
7. the call forwarding method of a kind of intelligent terminal VoIP as claimed in claim 6, is characterized in that, in step C, the second method of calling out that described VoIP server is set up between the second terminal and third terminal comprises:
VoIP server sends call-id, from, to information of the second calling to third terminal, and now third terminal ring, the second terminal listens ring-back tone.
8. the call forwarding method of a kind of intelligent terminal VoIP as claimed in claim 7, is characterized in that, in step D, after the second hanging up terminal, the method for the 3rd calling that VoIP server is set up between first terminal and third terminal comprises:
Second terminal is setting up with third terminal rear on-hook of conversing, second terminal sends BYE to VoIP server, VoIP server SIP terminated between the second terminal after receiving BYE is connected, discharge the first calling and the second calling simultaneously, VoIP server sets up the 3rd calling, send REINVITE respectively to first terminal and third terminal and be used for media re-negotiation, between first terminal and third terminal, set up call.
9. the call forwarding method of a kind of intelligent terminal VoIP as claimed in claim 7, is characterized in that, in step D, after the second hanging up terminal, the method for the 3rd calling that VoIP server is set up between first terminal and third terminal comprises:
Second terminal with third terminal set up converse before on-hook, second terminal sends BYE to VoIP server, VoIP server discharges the first calling and the second calling immediately after receiving BYE, and send REINVITE to first terminal, media are redirected to the source of sound of ring-back tone, now first terminal listens ring-back tone, after third terminal off-hook, VoIP server sends 200OK to first terminal, and send REINVITE to third terminal, carry out media re-negotiation, between first terminal and third terminal, set up call.
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| CN114095885A (en) * | 2020-07-31 | 2022-02-25 | 华为技术有限公司 | Call method, system and related device |
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| CN114827099A (en) * | 2022-04-14 | 2022-07-29 | 厦门亿联网络技术股份有限公司 | Interior line communication method and system based on VoIP communication |
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