CN102572146B - 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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- CN102572146B CN102572146B CN 201110452093 CN201110452093A CN102572146B CN 102572146 B CN102572146 B CN 102572146B CN 201110452093 CN201110452093 CN 201110452093 CN 201110452093 A CN201110452093 A CN 201110452093A CN 102572146 B CN102572146 B CN 102572146B
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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 telephone communication method in a kind of VoIP system.
Background technology
Conversation initialized protocol (Session Initiation Protocol, be called for short SIP) is widely used in the networking telephone (Voice over Internet Protocol is called for short VoIP) system.Yet, existing most VoIP systems are only supported the communication between the SIP client, also there are the situations such as poor expandability, node load be overweight mostly in 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 telephone communication method in a kind of VoIP system, the method can realize the communication between standard SIP client and regular handset, simultaneously can support the large-scale consumer concurrent call, have good stability, extensibility is strong and the characteristics such as load balancing.
SIP client and Mobile telephone communication method in a kind of VoIP system provided by the invention is characterized in that, the method comprises the following steps:
(1) the voice gateways server is regularly reported the load state of self to the gateway management server;
(2) described gateway management server according to load state, is stored the relative address information of each described voice gateways server in the mode of priority query;
(3) the SIP client sends registration message to sip server, and described sip server returns to 200OK message, shows and succeeds in registration;
(4) described sip server is searched the own nearest described gateway management server of distance by IP address proximity principle;
(5) the described gateway management server of described sip server request is that described SIP client is distributed the voice gateways server that load is lighter;
(6) described gateway management server takes out the lighter described voice gateways server of load and returns to described sip server from self priority query;
(7) described sip server is the relative address information of the described voice gateways server of described SIP client maintenance;
(8) described SIP client is come calling handset by the mode that sends invitation message to described sip server, and described mobile phone is accepted the invitation, and sets up the session success between described SIP client and described mobile phone;
(9) described SIP client and described mobile phone communicate;
(10) judge whether to finish communication, if finish, enter step (11), proceed communication otherwise enter step (9);
(11) described SIP client by the mode that sends byebye message to described sip server finish with mobile phone between communicate by letter, conversation end between described SIP client and described 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 in the situation that do not change original voice gateways server by the mode of newly-increased voice gateways server, have good extensibility.
(3) load balancing: in the present invention, all voice gateways servers are all managed by the gateway management server, the gateway management server is according to the load state (CPU of each voice gateways server, internal memory and the network bandwidth etc.) come to be they distributing user, the voice gateways server lighter to those loads, the user who distributes is more, and the voice gateways server heavier to those loads, the user who distributes is also corresponding less, so just be unlikely to occur part voice gateways server load overloading, and the situation that another part voice gateways server " be 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 that the SIP signaling is to the flow path switch figure of pstn signaling.
Embodiment
As shown in Figure 1, SIP client of the present invention and Mobile telephone communication method mainly comprise the following steps:
(1) the voice gateways server is regularly reported the load state of self to the gateway management server;
Described voice gateways server refers to specifically provide the server of public switched telephone network (PSTN) (Public SwitchedTelephone Network, be called for short PSTN) access service, is mainly used in carrying out the format conversion between signaling protocol and media stream data.The management server of described gateway management server finger speech sound gateway server mainly is responsible for safeguarding the load information of each voice gateways server of self managing.
Described load state comprises CPU, internal memory and the network bandwidth even load information of described voice gateways server.
(2) described gateway management server according to load state, is stored the relative address information of each described voice gateways server in the mode of priority query;
After the described voice gateways server of described gateway management server reception sends to the packet of self, these packets are resolved, obtain the load information of described voice gateways server, and finally store described voice gateways server relative address information according to these load informations in the mode of priority query.
(3) the SIP client sends registration (REGISTER) message to sip server, and described sip server returns to 200OK message, shows and succeeds in registration;
(4) described sip server is searched the own nearest described gateway management server of distance by IP address proximity principle;
At first described sip server can read the address information of described 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 described gateway management server.
(5) the described gateway management server of described sip server request is that described SIP client is distributed the voice gateways server that load is lighter;
(6) described gateway management server takes out the lighter described voice gateways server of load and returns to described sip server from self priority query;
(7) described sip server is the relative address information of the described voice gateways server of described SIP client maintenance;
(8) described SIP client invites the mode of (INVITE) message to come calling handset by sending to described sip server, and described mobile phone is accepted the invitation, and sets up the session success between described SIP client and described mobile phone;
(9) described SIP client and described mobile phone communicate;
(10) judge whether to finish communication, if finish, enter step (11), proceed communication otherwise enter step (9);
(11) described SIP client by the mode that sends goodbye (BYE) message to described sip server finish with mobile phone between communicate by letter, conversation end between described SIP client and described mobile phone;
As shown in Fig. 2 the first half, in the SIP client in VoIP system of the present invention and Mobile telephone communication method, step (8) comprising:
(8-1) described SIP client sends INVITE to described sip server;
(8-2) described sip server is transmitted to described voice gateways server with this INVITE;
(8-3) described voice gateways server is resolved the INVITE of receiving, and it is packaged into initial address (IAM) message sends to described mobile phone;
(8-4) described mobile phone, can be at first to a described voice gateways server response address complete (ACM) message after receiving this IAM message;
(8-5) described voice gateways server is resolved the ACM message of receiving, and it is packaged into 180 jingle bell message is transmitted to described sip server;
(8-6) described sip server returns to described SIP client with this 180 jingle bell message, shows that described mobile phone is just in ring;
(8-7) described mobile phone after the request of accepting session, can be replied (ANN) message to one of described voice gateways server response again;
(8-8) described voice gateways server is resolved the ANN message of receiving, and it is packaged into 200OK message is transmitted to described sip server;
(8-9) described sip server returns to described SIP client with this 200OK message, shows that described mobile phone accepted this time session request;
(8-10) described SIP client after this 200OK message, sends a confirmation (ACK) message can for described sip server;
(8-11) described sip server is transmitted to described voice gateways server with this ACK message, has just set up a session between said SIP client and described mobile phone;
As shown in Fig. 2 the latter half, in the SIP client in VoIP system of the present invention and Mobile telephone communication method, step (11) comprising:
(11-1) described SIP client sends goodbye (BYE) message to described sip server;
(11-2) described sip server is transmitted to described voice gateways server with this BYE message;
(11-3) described voice gateways server is resolved receiving BYE message, and it is packaged into circuit discharges (REL) message and send to described mobile phone;
(11-4) described mobile phone after receiving this REL message, can respond circuit release of described voice gateways server and complete (RLC) message;
(11-5) described voice gateways server is resolved the RLC message of receiving, and it is packaged into 200OK message is transmitted to described sip server;
(11-6) described sip server returns to described SIP client with this 200OK message, shows this conversation end.
Example:
Utilize SIP client and Mobile telephone communication method in VoIP system set forth in the present invention, the laboratory provides 10 station server nodes, and 1 Daepori leads to personal computer (PC, Personal Computer) and 1 smart mobile phone, and the related hardware configuration is as follows:
| Machine name | CPU | Internal memory | Hard disk | The network bandwidth |
| Server A-J | (2.83GHz four core) | 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 | Android 2.2 | 256M | Mobile 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.The upper installation code SIP client of common PC.
Under the experimental situation of putting up, be no less than the speaking test of 10 times with standard SIP client and mobile phone.
Through above-mentioned repeatedly test, adopt SIP client and Mobile telephone communication method in the designed VoIP system of the present invention can realize well communicating by letter between standard SIP client and regular handset, and in the situation that 10 station servers can navigate to corresponding cellphone subscriber being no more than in time of 0.1 second, have real-time and robustness preferably.
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 to implement 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 protection of the invention.
Claims (3)
1. SIP client and the Mobile telephone communication method in a VoIP system, is characterized in that, the method comprises the following steps:
(1) the voice gateways server is regularly reported the load state of self to the gateway management server;
(2) described gateway management server according to load state, is stored the relative address information of each described voice gateways server in the mode of priority query;
(3) the SIP client sends registration message to sip server, and described sip server returns to 200OK message, shows and succeeds in registration;
(4) described sip server is searched the own nearest described gateway management server of distance by IP address proximity principle;
(5) the described gateway management server of described sip server request is that described SIP client is distributed the voice gateways server that load is lighter;
(6) described gateway management server takes out the lighter described voice gateways server of load and returns to described sip server from self priority query;
(7) described sip server is the relative address information of the described voice gateways server of described SIP client maintenance;
(8) described SIP client is come calling handset by the mode that sends invitation message to described sip server, and described mobile phone is accepted the invitation, and sets up the session success between described SIP client and described mobile phone;
(9) described SIP client and described mobile phone communicate;
(10) judge whether to finish communication, if finish, enter step (11), proceed communication otherwise enter step (9);
(11) described SIP client by the mode that sends byebye message to described sip server finish with mobile phone between communicate by letter, conversation end between described SIP client and described mobile phone.
2. SIP client and the Mobile telephone communication method in VoIP system according to claim 1, is characterized in that, step (8) comprising:
(8-1) described SIP client sends invitation message to described sip server;
(8-2) described sip server is transmitted to described voice gateways server with this invitation message;
(8-3) described voice gateways server is resolved the invitation message of receiving, and it is packaged into initial address message sends to described mobile phone;
(8-4) described mobile phone, can be at first to address complete message of described voice gateways server response after receiving this initial address message;
(8-5) described voice gateways server is resolved the address complete message of receiving, and it is packaged into 180 jingle bell message is transmitted to described sip server;
(8-6) described sip server returns to described SIP client with this 180 jingle bell message, shows that described mobile phone is just in ring;
(8-7) described mobile phone, can be again to response message of described voice gateways server response after the request of accepting session;
(8-8) described voice gateways server is resolved the response message of receiving, and it is packaged into 200OK message is transmitted to described sip server;
(8-9) described sip server returns to described SIP client with this 200OK message, shows that described mobile phone accepted this time session request;
(8-10) described SIP client after this 200OK message, sends an acknowledge message can for described sip server;
(8-11) described sip server is transmitted to described voice gateways server with this acknowledge message, has just set up a session between said SIP client and described mobile phone.
3. SIP client and the Mobile telephone communication method in VoIP system according to claim 1 and 2, is characterized in that,
Step (11) comprising:
(11-1) described SIP client sends byebye message to described sip server;
(11-2) described sip server is transmitted to described voice gateways server with this byebye message;
(11-3) described voice gateways server is resolved receiving byebye message, and it is packaged into the circuit release message sends to described mobile phone;
(11-4) described mobile phone after receiving this release message, can respond circuit Release complete of described voice gateways server;
(11-5) described voice gateways server is resolved the circuit Release complete of receiving, and it is packaged into 200OK message is transmitted to described sip server;
(11-6) described sip server returns to described SIP client with this 200OK message, shows this conversation end.
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| 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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| CN102572146A CN102572146A (en) | 2012-07-11 |
| CN102572146B true CN102572146B (en) | 2013-06-26 |
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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 |
| CN103581005B (en) * | 2013-11-04 | 2016-11-02 | 惠州Tcl移动通信有限公司 | A kind of voice gateway system and the method realizing networking 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 |
Citations (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN1783871A (en) * | 2004-12-03 | 2006-06-07 | 上海贝尔阿尔卡特股份有限公司 | Load equalizing system, device and method for SIP telephone service |
| CN101068238A (en) * | 2006-12-18 | 2007-11-07 | 腾讯科技（深圳）有限公司 | Method and system for messaging between public exchange telephone network terminal and immediate communication terminal |
| CN101834877A (en) * | 2010-06-03 | 2010-09-15 | 华中科技大学 | Method and system for balancing dynamic load based on distributed SIP architecture |
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| US8462768B2 (en) * | 2008-06-11 | 2013-06-11 | Verizon Patent And Licensing Inc. | Providing session initiation protocol (SIP) call control functions to public switched telephone network (PSTN)-based call controller |
| US8517849B2 (en) * | 2010-02-19 | 2013-08-27 | Wonderland Nurserygoods Company Limited | Infant swing apparatus |
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| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN1783871A (en) * | 2004-12-03 | 2006-06-07 | 上海贝尔阿尔卡特股份有限公司 | Load equalizing system, device and method for SIP telephone service |
| CN101068238A (en) * | 2006-12-18 | 2007-11-07 | 腾讯科技（深圳）有限公司 | Method and system for messaging between public exchange telephone network terminal and immediate communication terminal |
| CN101834877A (en) * | 2010-06-03 | 2010-09-15 | 华中科技大学 | Method and system for balancing dynamic load based on distributed SIP architecture |
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