CN101753577A - VoIP communication system based on SIP protocol and communication method thereof - Google Patents
VoIP communication system based on SIP protocol and communication method thereof Download PDFInfo
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Abstract
The invention provides a VoIP communication system based on an SIP protocol and a communication method thereof. The system comprises a user interface module, a main state control module and a communication management module, the main state control module is respectively in communication linkage with the user interface module and the communication management module, and the communication management module also comprises a network initialization module, a call control module, a network message monitoring module and a cancellation module. The invention further provides a communication method based on the communication system so that the VoIP communication system based on the SIP protocol in the invention is mounted at an intelligent terminal and connected with a Web server, an MySql server and an Asterisk server mutually through a network, so as to realize the organic integration of the services, such as intelligent speech synthesis recognition, multi-protocol instant messages, short messages, Email and the like, in the same network.
Description
Technical field
The present invention relates to of the application of a kind of voip technology, particularly run on the VoIP speech business on the intelligent terminal in the converged communication field.
Background technology
For a long time, how people grow apart from voice communication paying close attention to.In the past more than 100 year, we rely on public telephone network PSTN (Public switched Telephone Network) and carry out voice communication.When converse in two places, the both sides of a fixing lane assignment to this dialogue are just arranged, though may also there be unnecessary frequency range in this circuit, out of Memory can not transmit simultaneously.When data communication service occurring, thisly expose its level of resources utilization and the low defective of reliability gradually based on Circuit-switched communication mode.
Moreover, the appearance of voice and aggregation of data business has higher requirement also for traditional circuit switching.Along with development of internet technology, all in development inevitably, this just requires Web server must have with the user to carry out data, the mutual ability of voice and video image most of application software, and this certainly will require the voice-and-data business to merge mutually.
On price, because the great number charge of long distance communication on telephone network, people wish to utilize the low-cost characteristic of internet to finish identical business.
VoIP (the Voice Over Internet Protocol) technology that transmits speech data with the form of grouping is an important substitute of our the circuit exchanging telephone technology used for many years.Its key property based on Packet Based Network has solved the problem of the utilization of resources, has also brought lower cost to voice communication service simultaneously.Voice-and-data application that voip technology is integrated, the new business that is provided is that operators have brought bigger market efficiency.VoIP is the abbreviation of Voice over InternetProtocol, that is to say the IP phone of normal theory, refer to will the simulation voice signal through overcompression, with the coding after, transmit in IP network with the form of data packet.The basic principle of VoIP is: the compression algorithm by voice is compressed processing to speech data, then these speech datas are packed by the TCP/IP standard, through IP network packet is delivered to reception ground, after passing through decompression processing again, revert to original voice signal, thereby reach the purpose that transmits voice by the Internet.This voice transfer mode has that cost of equipment is low, speech and data integration, bandwidth requirement are low, the characteristics such as extensive use of IP, and market prospects are wide.
The realization of voip technology need rely on a series of technology, and these technology comprise talk various network protocols, promptly ICP/IP protocol, be responsible for signaling control Session Initiation Protocol, with the closely-related Session Description Protocol of SIP (SDP), be responsible for the Real-time Transport Protocol of voice transfer etc.Wherein Session Initiation Protocol is because all advantages of himself become the main flow voip signaling protocol.
Evolution on the voip technology is broadly divided into three developing stage:
Phase I: 1996 to 1999.In this stage, voip technology and application main feature be: on using, mainly be that elementary voice communication is provided on enterprise network, the Internet on a small scale, speech quality can't guarantee.The signaling control protocol of using during this mainly be H.323v1/v2, MGCP etc., do not insert communication network as the user with the telecommunications network intercommunication or by enterprise gateway.
Second stage: 1999 to 2005.In this stage, commercial application model is full-fledged gradually, and having engendered on the market to provide the manufacturer of setting up extensive commercial VoIP long-distance service net, for the public provides the long-distance speech communication service that service quality guarantees.The main agreement that adopts can realize the internetworking intercommunication with the conventional telecommunications net for H.323v4+H.248.Simultaneously, H.323 the problems of agreement begin to display, and can't satisfy the application demand of enterprise market.As complex structure, cost height, to support difference of multiple spot user etc.
Phase III: the phase III that is the VoIP development from now on.In order to obtain more universal application, this stage will solve problem and the further service quality of improving that second stage runs on the one hand, will introduce more new function, new business on the other hand.The improvement of problem mainly is the work of the following aspects:
Service quality: this is to carry out the transition to the most basic requirement that VoIP, IPPBX replace PBX by PSIN.The service quality here mainly is requirement system lowest latency time (40 milliseconds) that can guarantee to reach voice transfer and QoS etc.
High availability: though VoIP has become inexorable trend, compare with traditional PSIN, in maturity, stability, availability, manageability, and even aspect such as extensibility still need be strengthened.
Open and compatible: in the face of the enterprise market, one of maximum problem that VoIP is current is exactly how under open architecture, and the intercommunication that can reach each tame VoIP product or solution is with compatible, and the interoperability of IP phone and black phone.
Present stage, voip signaling protocol can be divided into three kinds substantially, and promptly (1) gatekeeper (CK) H.323 follows LAN and goes up multimedia conferencing communication protocol, provides and calls out control, call manager and conferencing function etc.; (2) MGCP MGCP, control media gateway state also indicates their transfer mediums to assigned address; (3) SIP is with control of client/server distributed call and capability negotiation.SIP is the same with SMTP with HTTP, is a kind of text based agreement.Many programmers understand this agreement very much.They find that Session Initiation Protocol is very simple and are easy to fix a breakdown.H.323 agreement is write with binary code, does not have the programmer of rich experiences and developing instrument all to be unfamiliar with this agreement.SIP is the agreement of an application layer, with its characteristics small and exquisite and easy to use, is obtaining use more and more widely.
Summary of the invention
Goal of the invention: the present invention proposes a kind of converged communication solution of new generation, relates generally to a kind of VoIP communication system and communication means thereof based on Session Initiation Protocol.
Technical scheme:
The present invention adopts following technical scheme for achieving the above object:
A kind of VoIP communication system based on Session Initiation Protocol comprises Subscriber Interface Module SIM, major state control module, communication management module; The major state control module communicates to connect with Subscriber Interface Module SIM and communication management module respectively; Wherein:
Subscriber Interface Module SIM is responsible for the demonstration at interface, to the response of user's input and the realization of telephone directory function;
Communication management module is responsible for the mutual of signaling based on the Session Initiation Protocol stack, sets up and stop session;
The major state control module is used to receive the message that Subscriber Interface Module SIM and communication management module are sent, and message is resolved, and carry out the switching of major state according to the result who resolves; Send message informing Subscriber Interface Module SIM and communication management module simultaneously, make them make corresponding operation.
The communication management module of VoIP communication system of the present invention comprises netinit module, Call Control Block, internet message monitoring modular and nullifies module; Wherein:
Netinit module: finish the initialization of protocol stack, and after initialization is finished, register to server; Call Control Block: be responsible for making a call, incoming call answering, on-hook;
Internet message is monitored module: the sip message of circular wait network side; When the sip message of receiving network side transmitted, message is replied and decoded, by message dilivery notice major state control module, make it the switching of completion status simultaneously;
Nullify module: be responsible for before application program withdraws from, to the server requests logging off users, and all resources of release busy.
A kind of based on communication means of the present invention, comprise the steps:
1) start the Subscriber Interface Module SIM process, the initialization global variable is created and is used the householder interface, shows dialog box;
2) create pipeline between major state control module and Subscriber Interface Module SIM and the communication management module respectively, be used for Inter-Process Communication.
3) start major state control module process, connect database, take out the user profile that is stored in the database, user profile is presented on the interface;
4) enter the message circulation state, wait for user's operation or pipeline message;
5) when the user operates or pipeline message is arranged, response user's operation or response pipeline message, and message or pipeline message that the user is operated are sent to communication management module;
6) start the communication management module process;
7) enter message circulation, wait for pipeline message;
8) when pipeline message, communication management module is sent to the major state control module with pipeline message, and the major state control module is carried out the transition of system's major state according to pipeline message;
9) the major state control module sends pipeline message to Subscriber Interface Module SIM and communication management module;
10) behind the communication management module process initiation, at first create sub-thread;
11) network side message is monitored and handled to the sub-thread of communication management module;
12) the communication management module main thread is to the Asterisk server registration;
13) the communication management module main thread enters message circulation, waits for pipeline message;
14) the communication management module main thread is done operations such as dialing, on-hook according to receiving pipeline message;
15) communication management module sends pipeline message to the major state control module.
Beneficial effect:
Among the present invention based on the VoIP communication system, be installed in intelligent terminal, and by network and Web server, the MySql server, the Asterisk server is interconnected, thereby the intelligent sound in realization and the same network synthesizes organically blending of services such as identification, multi-protocols instant message, note, Email.
Description of drawings
Fig. 1 is the VoIP system based on Session Initiation Protocol among the present invention.
Fig. 2 is the host state machine state transition diagram.
Fig. 3 is that the local user removes electric flow chart.
Fig. 4 is local user's refusing incoming call flow chart.
Fig. 5 is the startup flow process of VoIP Phone application program.
Fig. 6 is the system architecture of VoIP Phone application program.
Embodiment
Below in conjunction with accompanying drawing technical scheme of the present invention is elaborated:
1) structure of Asterisk sip server:
The present invention uses Asterisk software to realize sip server.Asterisk is an open source software that runs on the (SuSE) Linux OS, and it can realize communicating by letter between the data networking equipment of Circuit-switched Telecommunication network equipment and packet switch.Analog telephone is connected on the analog interface plate card of sip server by telephone wire, and this sip server is connected in the local area network (LAN).Like this, the network user just can pass through local area network (LAN), is registered on the sip server, thereby communicates with the user of other network users or analog telephone.At first, Asterisk software is installed on (SuSE) Linux OS.Install after the Asterisk, it is configured, just revise the content of these two configuration files of sip.conf and zaptel.conf, for the network user and analog telephone user set up number of the account, the design dialing rule.Only had the user of Asterisk number of the account, just can be registered on the sip server.After the registration, the user need make a call according to dialing rule.Secondly, make up Asterisk server and unified address list scheduling mechanism.
2) structure of the webserver:
At first, under windows, use VC6.0 to create MFC AppWizard application program.This application program is finished the function of VoIP Phone, comprises registration, calls out, and answers.
Then, use Inno Setup software, the MFC application program of making in the previous step is made into the setup.exe installation file.
Once more, in the jsp webpage, insert hyperlink, be linked to the setup.exe installation file that previous step is made.
At last, Tomcat software is installed on the WEB server host, the jsp file of making in the previous step is put into the acquiescence web path of Tomcat.Tomcat is exactly the web server software of this problem VoIP system.
So far, the user can pass through the IE browser, and visit is positioned over the jsp webpage on the Tomcat webserver, and downloads and installs VoIP Phone application program from this webpage.
3) software system framework of VoIP Phone application program:
The major function of VoIP Phone is to realize conversation, and offers good operation interface of user.For this reason, this VoIP Phone application system is divided into three big modules, is respectively Subscriber Interface Module SIM, major state control module, communication management module.Subscriber Interface Module SIM is responsible for the demonstration at interface, to the response of user's input and the realization of telephone directory function.The major state control module is the central control module of whole software system, and the message that its reception Subscriber Interface Module SIM and communication management module are sent is resolved message, and carried out the switching of major state with this.The major state control module also needs to send message, and notice Subscriber Interface Module SIM and communication management module make them make corresponding operation.Communication management module is the communication infrastructure of whole telephone set system, it mainly is responsible for the mutual of signaling, set up and stop session, it is based on the Session Initiation Protocol stack, and comprises netinit module, Call Control Block, internet message monitoring modular and the several submodules of cancellation module.The whole software system configuration as shown in Figure 6.
This plug-in component operation is on the Windows operating system platform, and the API that using system provides develops.Whole system is made of three threads, and Subscriber Interface Module SIM, major state control module, communication management module are respectively used a thread.
Communication management module comprises two threads again, and main thread is responsible for mutual with the major state control module, and sub-thread is responsible for monitoring the sip message of network.Communicate by message dilivery between three threads.
4) the startup flow process of VoIP Phone application program
Start process description:
1) start the Subscriber Interface Module SIM process, the initialization global variable is created and is used the householder interface, shows dialog box;
2) create pipeline between major state control module and Subscriber Interface Module SIM and the communication management module respectively, be used for Inter-Process Communication.
3) start major state control module process, connect database, take out the user profile that is stored in the database, user profile is presented on the interface;
4) enter the message circulation state, wait for user's operation or pipeline message;
5) when the user operates or pipeline message is arranged, response user's operation or response pipeline message, and message or pipeline message that the user is operated are sent to communication management module;
6) start the communication management module process;
7) enter message circulation, wait for pipeline message;
8) when pipeline message, communication management module is sent to the major state control module with pipeline message, and the major state control module is carried out the transition of system's major state according to pipeline message;
9) the major state control module sends pipeline message to Subscriber Interface Module SIM and communication management module;
10) behind the communication management module process initiation, at first create sub-thread;
11) network side message is monitored and handled to the sub-thread of communication management module;
12) the communication management module main thread is to the Asterisk server registration;
13) the communication management module main thread enters message circulation, waits for pipeline message;
14) the communication management module main thread is done operations such as dialing, on-hook according to receiving pipeline message;
15) communication management module sends pipeline message to the major state control module.
5) research of communication management module in the VoIP Phone application program
Communication management module is the communication infrastructure of whole telephone set system, and it mainly is responsible for the mutual of signaling, sets up and stop session.It comprises netinit module, Call Control Block, internet message monitoring module and nullifies the several submodules of module based on OISP, EXOSIP, ORTP, Mediastream protocol stack.It comprises two threads, and main thread is responsible for mutual with the major state control module, and sub-thread is responsible for monitoring the sip message of network.
Introduce the function of each module below:
The netinit module is at first finished the initialization of three protocol stacks, and after initialization is finished, registers to server.After registration was finished, this soft phone just entered the idle condition of waiting for subscriber dialing or waiting for incoming call.
Call Control Block is responsible for making a call, incoming call answering, on-hook.
Internet message is monitored module, is an independently thread, the sip message of circular wait network side.In case receive the sip message of network side transmitted, just message is replied and decode, by message dilivery notice major state control module, make it the switching of completion status simultaneously.
Nullify module, be responsible for before application withdraws from, to the server requests logging off users, and all resources of release busy.
Introduce the workflow of communication management module below.
Communication management module is an independently thread, is called sip process.It is set up by the major state control module, and communicates by message dilivery and major state control module.Sip process at first creates a sub-thread, is used for internet message and monitors submodule.In case sub-thread creation is finished, the main thread of sip process just calls the netinit module, finishes the initialization of protocol stack, and registers to server.In case the message that the main thread of sip process is sent with regard to circular wait major state control module is finished in registration.Now remove electricity with the local user, after the conversation, local user's on-hook is an example, the workflow of sip process is described, as shown in Figure 3.
In Fig. 3, when the local user dialled, sip process received Message_Dial message, called sip_invite () and made a call.Server echo message EXOSIP_CALL_PROCEEDING and EXOSIP_CALL_RINGING, expression is called out and is transferred.The called subscriber answers, and server is sent message EXOSIP_CALL_ANSWERED here, and the expression conversation is set up.After the conversation, local user's on-hook, sip process receives Message_HangUp message, calls sip_call_terminate (), finishes conversation.Server loopback EXOSIP_CALL_RELEASED, expression conversation resource discharges.
The local user has incoming call, and sip process will receive EXOSIP_CALL_INVITE message.At this moment, Message_Ring message takes place in sip process, and whether prompting the local user answer the call.The workflow of local user's refusing incoming call as shown in Figure 4.
6) research of major state module in the VoIP Phone application program
This module is the core of whole VoIP Phone software systems, is controlling the state variation of whole phone and the transmission of messages management of each intermodule.Its main implementation also is based on traditional state machine design thought, be that the major state control module receives the message that other modules are sent, and the driving condition machine produces the transition of state, and generation action in the process of status change, send message to other modules, thereby finish needed function.
The premiere feature of major state control module is to play a part bridge between Subscriber Interface Module SIM and communication management module.It communicates by message dilivery and other two modules.The type of message that transmits is as follows:
typedef?enum
{
Message_Quit，
Message_Dial，
Message_HangUp，
Message_Key，//0-9，key?value?in″key″member
Message_None，//used?only?for?init，means?no?message
Message_Ring，//sip?to?main?to?UI，a?phone?is?coming
Message_Closed，//sip?to?main，the?call?was?hanged?up?by?others
Message_Answer//a?coming?has?came，the?user?wants?to?pick?it?up
}MESSAGE_TYPE；
Message_Quit represents that the user wishes to withdraw from application, and this moment, host state machine should be closed all processes of application, the resource that cleaning takies.
Message_Dial represents that the user presses dial key, wishes to make a phone call, or incoming call answering.
Message_HangUp represents that the user presses on-hook key, may be to hang up the telephone later in conversation, or refuse an incoming call.
Message_Key represents that the user dials, and what another field of message structure body will be stored concrete group of user is which button of 0-9.
Message_None only is used for the initial message structure.
Message_Ring represents that incoming call is arranged.
Message_Closed represents end of conversation.
Message_Answer represents user's incoming call answering.
Another critical function of host state machine according to the pipeline message of receiving, carries out the switching of major state exactly.
Major state defines with an enum type, and is specific as follows:
typedef?enum
{
status_idle，//wait?for?user
status_dial，//user?is?dialing?number
status_busy，//talking?on?phone
status_recv//receive?a?call
}enum_main_status；
extern?enum_main_status?main_status；
The state transition diagram of host state machine as shown in Figure 2.
Claims (3)
1. the VoIP communication system based on Session Initiation Protocol is characterized in that: comprise Subscriber Interface Module SIM, major state control module, communication management module; The major state control module communicates to connect with Subscriber Interface Module SIM and communication management module respectively; Wherein:
Subscriber Interface Module SIM is responsible for the demonstration at interface, to the response of user's input and the realization of telephone directory function;
Communication management module is responsible for the mutual of signaling based on the Session Initiation Protocol stack, sets up and stop session;
The major state control module is used to receive the message that Subscriber Interface Module SIM and communication management module are sent, and message is resolved, and carry out the switching of major state according to the result who resolves; Send message informing Subscriber Interface Module SIM and communication management module simultaneously, make them make corresponding operation.
2. the VoIP communication system based on Session Initiation Protocol according to claim 1 is characterized in that: communication management module comprises netinit module, Call Control Block, internet message monitoring modular and nullifies module; Wherein:
Netinit module: finish the initialization of protocol stack, and after initialization is finished, register to server;
Call Control Block: be responsible for making a call, incoming call answering, on-hook;
Internet message is monitored module: the sip message of circular wait network side; When the sip message of receiving network side transmitted, message is replied and decoded, by message dilivery notice major state control module, make it the switching of completion status simultaneously;
Nullify module: be responsible for before application program withdraws from, to the server requests logging off users, and all resources of release busy.
3. the communication means based on the described VoIP communication system of claim 1 is characterized in that: comprise the steps:
1) start the Subscriber Interface Module SIM process, the initialization global variable is created and is used the householder interface, shows dialog box;
2) create pipeline between major state control module and Subscriber Interface Module SIM and the communication management module respectively, be used for Inter-Process Communication;
3) start major state control module process, connect database, take out the user profile that is stored in the database, user profile is presented on the interface;
4) enter the message circulation state, wait for user's operation or pipeline message;
5) when the user operates or pipeline message is arranged, response user's operation or response pipeline message, and message or pipeline message that the user is operated are sent to communication management module;
6) start the communication management module process;
7) enter message circulation, wait for pipeline message;
8) when pipeline message, communication management module is sent to the major state control module with pipeline message, and the major state control module is carried out the transition of system's major state according to pipeline message;
9) the major state control module sends pipeline message to Subscriber Interface Module SIM and communication management module;
10) behind the communication management module process initiation, at first create sub-thread;
11) network side message is monitored and handled to the sub-thread of communication management module;
12) the communication management module main thread is to the Asterisk server registration;
13) the communication management module main thread enters message circulation, waits for pipeline message;
14) the communication management module main thread is done operations such as dialing, on-hook according to receiving pipeline message;
15) communication management module sends pipeline message to the major state control module.
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| CN200910263265XA CN101753577B (en) | 2009-12-17 | 2009-12-17 | VoIP communication system based on SIP protocol and communication method thereof |
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| CN101753577B CN101753577B (en) | 2012-11-14 |
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Cited By (4)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN101895852A (en) * | 2010-07-21 | 2010-11-24 | 中兴通讯股份有限公司 | Method and calling terminal for realizing multiparty call |
| CN102111415A (en) * | 2011-02-28 | 2011-06-29 | 东南大学 | Interactive network voice response system with embedded VoIP and implementation method thereof |
| CN102256020A (en) * | 2011-04-07 | 2011-11-23 | 深圳市共进电子有限公司 | Voice over Internet service control device and method |
| CN108881182A (en) * | 2018-05-30 | 2018-11-23 | 上海携程商务有限公司 | The networking telephone realization method and system of mobile terminal based on IOS |
Family Cites Families (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| GB0707387D0 (en) * | 2007-04-17 | 2007-05-23 | Lucent Technologies Inc | Single radio VCC:LTE-VMSC anchor solution |
| CN101399888B (en) * | 2008-09-26 | 2010-10-06 | 深圳市众方信息科技有限公司 | Network system for processing VoIP service and information synchronization method thereof |
| CN101437032B (en) * | 2008-12-19 | 2011-11-16 | 重庆邮电大学 | System for monitoring VOIP voice quality based on SIP protocol and detection method thereof |
-
2009
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Cited By (7)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN101895852A (en) * | 2010-07-21 | 2010-11-24 | 中兴通讯股份有限公司 | Method and calling terminal for realizing multiparty call |
| WO2012009994A1 (en) * | 2010-07-21 | 2012-01-26 | 中兴通讯股份有限公司 | Method and talking terminal for implementing multiparty call |
| CN101895852B (en) * | 2010-07-21 | 2014-03-12 | 中兴通讯股份有限公司 | Method and calling terminal for realizing multiparty call |
| CN102111415A (en) * | 2011-02-28 | 2011-06-29 | 东南大学 | Interactive network voice response system with embedded VoIP and implementation method thereof |
| CN102256020A (en) * | 2011-04-07 | 2011-11-23 | 深圳市共进电子有限公司 | Voice over Internet service control device and method |
| CN108881182A (en) * | 2018-05-30 | 2018-11-23 | 上海携程商务有限公司 | The networking telephone realization method and system of mobile terminal based on IOS |
| CN108881182B (en) * | 2018-05-30 | 2020-08-25 | 上海华客信息科技有限公司 | IOS-based mobile terminal network telephone realization method and system |
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| CN101753577B (en) | 2012-11-14 |
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| EXPY | Termination of patent right or utility model |