CN102572146A - Method for communication between session initiation protocol (SIP) client and mobile phone in voice over Internet protocol (VoIP) system - Google Patents
Method for communication between session initiation protocol (SIP) client and mobile phone in voice over Internet protocol (VoIP) system Download PDFInfo
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- CN102572146A CN102572146A CN2011104520938A CN201110452093A CN102572146A CN 102572146 A CN102572146 A CN 102572146A CN 2011104520938 A CN2011104520938 A CN 2011104520938A CN 201110452093 A CN201110452093 A CN 201110452093A CN 102572146 A CN102572146 A CN 102572146A
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Abstract
The invention discloses a method for communication between a session initiation protocol (SIP) client and a mobile phone in a voice over Internet protocol (VoIP) system. The method comprises the following steps that: (1), voice gateway servers regularly report own load conditions to gateway management servers; (2) the gateway management servers store the related information of each voice gateway server in a priority queuing way according to the load conditions; (3) the SIP client transmits a registration message to an SIP server; (4) the SIP server searches for a gateway management server which is closest to the SIP server; (5) the gateway management server allocates a voice gateway server with a light load to the SIP client; (6) the gateway management server extracts the voice gateway server with the light load, and returns the voice gateway server with the light load to the SIP server; (7) the SIP server maintains the related address information of the voice gateway server; (8) the SIP client calls the mobile phone, and successfully establishes a session with the mobile phone; and (9) the SIP client communicates with the mobile phone, and continues the communication or finishes the session. By the method, the communication between a standard SIP client and an ordinary mobile phone can be realized; and the method has the advantages of high stability, high expansibility, load balance and the like.
Description
Technical field
The invention belongs to applications of computer network field, be specifically related to SIP client and mobile communication method in a kind of VoIP system.
Background technology
Conversation initialized protocol (Session Initiation Protocol is called for short SIP) is widely used in the networking telephone (Voice over Internet Protocol the is called for short VoIP) system.Yet; Existing most VoIP systems are only supported the communication between the SIP client; Also there are situation such as poor expandability, node load be overweight mostly in the support SIP client few in number and the VoIP system of mobile communication, can not satisfy the demand of large-scale consumer concurrent call.
Summary of the invention
The object of the present invention is to provide SIP client and mobile communication method in a kind of VoIP system; This method can realize the communication between standard SIP client and the regular handset; Simultaneously can support the large-scale consumer concurrent call, have good stability, extensibility is strong and characteristics such as load balancing.
SIP client in a kind of VoIP system provided by the invention and mobile communication method is characterized in that, this method may further comprise the steps:
(1) the voice gateways server is regularly reported the load state of self to the gateway management server;
(2) said gateway management server is stored the relative address information of each said voice gateways server according to load state with the mode of priority query;
(3) the SIP client is sent registration message to sip server, and said sip server returns 200OK message, shows and succeeds in registration;
(4) said sip server is searched the own nearest said gateway management server of distance through IP address proximity principle;
(5) the said gateway management server of said sip server request is that said SIP client is distributed the voice gateways server that load is lighter;
(6) said gateway management server takes out the lighter said voice gateways server of load and returns to said sip server from self priority query;
(7) said sip server is the relative address information of the said voice gateways server of said SIP client maintenance;
(8) said SIP client is come calling handset through the mode of sending invitation message to said sip server, and said mobile phone is accepted the invitation, and sets up the session success between said SIP client and the said mobile phone;
(9) said SIP client and said mobile phone communicate;
(10) judge whether to finish communication,, then get into step (11), proceed communication otherwise get into step (9) if finish;
(11) said SIP client through the mode of sending byebye message to said sip server finish with mobile phone between communicate by letter conversation end between said SIP client and the said mobile phone.
Compared with prior art, the present invention has the following advantages:
(1) stability: the present invention is based on distributed computing technology and realize, can effectively solve the Single Point of Faliure problem that exists in traditional integrated system, thereby guarantee the stability of whole VoIP system.
(2) extensibility: the present invention can realize the lifting of whole system service performance simply through the mode of newly-increased voice gateways server under the situation that does not change original voice gateways server, have good expandability.
(3) load balancing: in the present invention; All voice gateways servers are all managed through the gateway management server, and the gateway management server comes to be they distributing user, the voice gateways server lighter to those loads according to the load state (CPU, internal memory and the network bandwidth etc.) of each voice gateways server; The user who distributes is more; And the voice gateways server heavier to those loads, the user of distribution is also corresponding less, so just is unlikely to occur part voice gateways server load overloading; And the situation that another part voice gateways server " nobody shows any interest in ", thereby guaranteed load balancing.
Description of drawings
Fig. 1 is SIP client and mobile communication overall flow figure;
Fig. 2 is the flow path switch figure of SIP signaling to pstn signaling.
Embodiment
As shown in Figure 1, SIP client of the present invention and mobile communication method mainly may further comprise the steps:
(1) the voice gateways server is regularly reported the load state of self to the gateway management server;
Said voice gateways server refers to specifically provide the server of PSTN (Public SwitchedTelephone Network is called for short PSTN) access service, is mainly used in the format conversion of carrying out between signaling protocol and the media stream data.The management server of said gateway management server finger speech sound gateway server mainly is responsible for safeguarding the load information of each voice gateways server of self managing.
Said load state comprises CPU, internal memory and the network bandwidth even load information of said voice gateways server.
(2) said gateway management server is stored the relative address information of each said voice gateways server according to load state with the mode of priority query;
After said gateway management server receives said voice gateways server and sends to the packet of self; These packets are resolved; Obtain the load information of said voice gateways server, and finally store said voice gateways server relative address information with the mode of priority query according to these load informations.
(3) the SIP client is sent registration (REGISTER) message to sip server, and said sip server returns 200OK message, shows and succeeds in registration;
(4) said sip server is searched the own nearest said gateway management server of distance through IP address proximity principle;
Said sip server can at first read the address information of said gateway management server from the configuration file of self, and the principle of finally closing on according to the IP address is selected one apart from self nearest said gateway management server.
(5) the said gateway management server of said sip server request is that said SIP client is distributed the voice gateways server that load is lighter;
(6) said gateway management server takes out the lighter said voice gateways server of load and returns to said sip server from self priority query;
(7) said sip server is the relative address information of the said voice gateways server of said SIP client maintenance;
(8) said SIP client invites the mode of (INVITE) message to come calling handset through sending to said sip server, and said mobile phone is accepted the invitation, and sets up the session success between said SIP client and the said mobile phone;
(9) said SIP client and said mobile phone communicate;
(10) judge whether to finish communication,, then get into step (11), proceed communication otherwise get into step (9) if finish;
(11) said SIP client through the mode of sending goodbye (BYE) message to said sip server finish with mobile phone between communicate by letter conversation end between said SIP client and the said mobile phone;
Shown in Fig. 2 the first half, step (8) comprising in SIP client in the VoIP system of the present invention and the mobile communication method:
(8-1) said SIP client is sent INVITE and is given said sip server;
(8-2) said sip server is transmitted to said voice gateways server with this INVITE;
(8-3) said voice gateways server is resolved the INVITE of receiving, and it is packaged into initial address (IAM) message sends to said mobile phone;
(8-4) said mobile phone can be at first to a said voice gateways server response address complete (ACM) message after receiving this IAM message;
(8-5) said voice gateways server is resolved the ACM message of receiving, and it is packaged into 180 jingle bell forwards to said sip server;
(8-6) said sip server returns to said SIP client with this 180 jingle bell message, shows that said mobile phone is just in ring;
(8-7) said mobile phone can be replied (ANN) message to one of said voice gateways server response once more after the request of accepting session;
(8-8) said voice gateways server is resolved the ANN message of receiving, and it is packaged into the 200OK forwards to said sip server;
(8-9) said sip server returns to said SIP client with this 200OK message, shows that said mobile phone accepted conversation request this time;
After (8-10) said SIP client receives this 200OK message, send an affirmation (ACK) message can for said sip server;
(8-11) said sip server is given said voice gateways server with this ACK forwards, has just set up a session between said SIP client like this and the said mobile phone;
Shown in Fig. 2 the latter half, step (11) comprising in SIP client in the VoIP system of the present invention and the mobile communication method:
(11-1) said SIP client is sent goodbye (BYE) message to said sip server;
(11-2) said sip server is given said voice gateways server with this BYE forwards;
(11-3) said voice gateways server is resolved receiving BYE message, and it is packaged into circuit discharges (REL) message and send to said mobile phone;
(11-4) said mobile phone can respond circuit of said voice gateways server and discharge completion (RLC) message after receiving this REL message;
(11-5) said voice gateways server is resolved the RLC message of receiving, and it is packaged into the 200OK forwards to said sip server;
(11-6) said sip server returns to said SIP client with this 200OK message, shows this conversation end.
Instance:
Utilize SIP client and mobile communication method in the VoIP system that the present invention sets forth, the laboratory provides 10 station server nodes, and 1 Daepori leads to personal computer (PC, Personal Computer) and 1 smart mobile phone, and related hardware disposes as follows:
| Machine name | CPU | Internal memory | Hard disk | The network bandwidth |
| Server A-J | (2.83GHz four nuclear) | 4G | 500G | 100M |
| Common PC | (2.00GHz double-core) | 2G | 250G | 100M |
The hardware of table 1 super node and ordinary PC and network configuration
| Machine name | Model | Operating system | Internal memory | Communication type |
| Mobile phone terminal | HTC?WildFire | Peace tall and erect 2.2 | 256M | Move GSM |
The hardware of table 2 smart mobile phone and network configuration
Server A, B, C, D are as sip server, and server E, F are as the gateway management server, and server G, H, I, J are as the voice gateways server.Common PC goes up installation code SIP client.
Under the experimental situation of putting up, be no less than 10 times speaking test with standard SIP client and mobile phone.
Through above-mentioned repeatedly test; Adopt SIP client and mobile communication method in the VoIP system that the present invention designed can realize communicating by letter between standard SIP client and the regular handset well; And, have good real-time performance and robustness navigating to corresponding cellphone subscriber being no more than in time of 0.1 second under the situation of 10 station servers.
The present invention not only is confined to above-mentioned embodiment; Persons skilled in the art are according to content disclosed by the invention; Can adopt other multiple embodiment embodiment of the present invention, therefore, every employing project organization of the present invention and thinking; Do some simple designs that change or change, all fall into the scope of the present invention's protection.
Claims (3)
1. SIP client and the mobile communication method in the VoIP system is characterized in that this method may further comprise the steps:
(1) the voice gateways server is regularly reported the load state of self to the gateway management server;
(2) said gateway management server is stored the relative address information of each said voice gateways server according to load state with the mode of priority query;
(3) the SIP client is sent registration message to sip server, and said sip server returns 200OK message, shows and succeeds in registration;
(4) said sip server is searched the own nearest said gateway management server of distance through IP address proximity principle;
(5) the said gateway management server of said sip server request is that said SIP client is distributed the voice gateways server that load is lighter;
(6) said gateway management server takes out the lighter said voice gateways server of load and returns to said sip server from self priority query;
(7) said sip server is the relative address information of the said voice gateways server of said SIP client maintenance;
(8) said SIP client is come calling handset through the mode of sending invitation message to said sip server, and said mobile phone is accepted the invitation, and sets up the session success between said SIP client and the said mobile phone;
(9) said SIP client and said mobile phone communicate;
(10) judge whether to finish communication,, then get into step (11), proceed communication otherwise get into step (9) if finish;
(11) said SIP client through the mode of sending byebye message to said sip server finish with mobile phone between communicate by letter conversation end between said SIP client and the said mobile phone.
2. SIP client in the VoIP system according to claim 1 and mobile communication method is characterized in that, step (8) comprising:
(8-1) said SIP client is sent invitation message and is given said sip server;
(8-2) said sip server is transmitted to said voice gateways server with this invitation message;
(8-3) said voice gateways server is resolved the invitation message of receiving, and it is packaged into initial address message sends to said mobile phone;
(8-4) said mobile phone can be at first to address complete message of said voice gateways server response after receiving this address message;
(8-5) said voice gateways server is resolved the address complete message of receiving, and it is packaged into 180 jingle bell forwards to said sip server;
(8-6) said sip server returns to said SIP client with this 180 jingle bell message, shows that said mobile phone is just in ring;
(8-7) said mobile phone can be once more to response message of said voice gateways server response after the request of accepting session;
(8-8) said voice gateways server is resolved the response message of receiving, and it is packaged into the 200OK forwards to said sip server;
(8-9) said sip server returns to said SIP client with this 200OK message, shows that said mobile phone accepted conversation request this time;
After (8-10) said SIP client receives this 200OK message, send an acknowledge message can for said sip server;
(8-11) said sip server is transmitted to said voice gateways server with this acknowledge message, has just set up a session between said SIP client like this and the said mobile phone.
3. SIP client in the VoIP system according to claim 1 and 2 and mobile communication method is characterized in that,
Step (11) comprising:
(11-1) said SIP client is sent byebye message to said sip server;
(11-2) said sip server is transmitted to said voice gateways server with this byebye message;
(11-3) said voice gateways server is resolved receiving BYE message, and it is packaged into the circuit release message sends to said mobile phone;
(11-4) said mobile phone can respond circuit Release complete of said voice gateways server after receiving this release message;
(11-5) said voice gateways server is resolved the circuit Release complete of receiving, and it is packaged into the 200OK forwards to said sip server;
(11-6) said sip server returns to said SIP client with this 200OK message, shows this conversation end.
Priority Applications (1)
| Application Number | Priority Date | Filing Date | Title |
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| CN 201110452093 CN102572146B (en) | 2011-12-30 | 2011-12-30 | Method for communication between session initiation protocol (SIP) client and mobile phone in voice over Internet protocol (VoIP) system |
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| CN 201110452093 CN102572146B (en) | 2011-12-30 | 2011-12-30 | Method for communication between session initiation protocol (SIP) client and mobile phone in voice over Internet protocol (VoIP) system |
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Cited By (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN103401882A (en) * | 2013-08-16 | 2013-11-20 | 深圳市宏电技术股份有限公司 | VOIP gateway voice link backup method and system |
| CN103581005A (en) * | 2013-11-04 | 2014-02-12 | 惠州Tcl移动通信有限公司 | Voice gateway system and method for achieving internet telephone paging |
| CN104683253A (en) * | 2013-11-27 | 2015-06-03 | 北京大唐高鸿数据网络技术有限公司 | Dynamic load balancing method for unified communication system |
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| CN103401882B (en) * | 2013-08-16 | 2016-12-28 | 深圳市宏电技术股份有限公司 | VoIP gateway voice link backup method and system |
| CN103581005A (en) * | 2013-11-04 | 2014-02-12 | 惠州Tcl移动通信有限公司 | Voice gateway system and method for achieving internet telephone paging |
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| CN104683253A (en) * | 2013-11-27 | 2015-06-03 | 北京大唐高鸿数据网络技术有限公司 | Dynamic load balancing method for unified communication system |
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