CN102548025A

CN102548025A - Method for reducing mobile voice over internet protocol (VoIP) call setup delay

Info

Publication number: CN102548025A
Application number: CN2012100715734A
Authority: CN
Inventors: 黎海涛; 吕海坤
Original assignee: Beijing University of Technology
Current assignee: Beijing University of Technology
Priority date: 2012-03-16
Filing date: 2012-03-16
Publication date: 2012-07-04
Anticipated expiration: 2032-03-16
Also published as: CN102548025B

Abstract

The invention discloses a method for reducing mobile voice over internet protocol (VoIP) call setup delay and belongs to the field of mobile internet protocol (IP) networks. The method comprises the following steps that: a calling party sends REGISTER information to a location server for location updating; after processing the REGISTER information, the location server sends a 200 OK response message to the calling party, wherein 200 OK information contains location information of a user to be called; the calling party extracts a called party whose left retention time is greater than a threshold in advance according to the 200 OK information; the calling party sends acknowledgement (ACK) information to the location server; the calling party directly sends an INVITE request message to the called party; if the called party processes the INVITE request message successfully, the called party sends a message of which a state value is 200 OK to the calling party; and after receiving the 200 OK information, the calling party sends the ACK message to the called party. By the method, the calling party is not required to access the location server and can establish a conversation with the called party directly to ensure that the call setup delay is greatly reduced.

Description

Voip call setup delay method is moved in a kind of reduction

Technical field

The invention belongs to field of wireless, particularly support the professional WLAN of VoIP, broadband wireless access, GSM etc.

Background technology

Along with the fast development of radio network technique, the gradually becoming strong of mobile terminal device function, traditional multimedia service can not satisfy people's demand.How on existing heterogeneous network environment, to develop more mobile multimedia value-added service, be the target that field of telecommunications is pursued, and wherein typical application is VoIP (Voice over Internet Protocol) technology.

Current, IP based network high-quality real-time multimedia application development is rapid, and voip technology also expands to voice, video and data etc. at interior multimedia service by pure speech transmission service.The key that realizes VoIP is a signaling technology, extensive use at present two kinds of signaling systems are arranged: H.323 and SIP (Session Initiation Protocol, Session initiation Protocol).

H.323 powerful but very complicated, it has stipulated complete multimedia application framework, has relatively high expectations, and implements the comparison difficulty.Session Initiation Protocol be one simple, be prone to expansion and realize, text based telephone signaling control protocol, it is one and inserts the application layer signaling protocol that has nothing to do with bottom.Based on the next generation network of SIP, can seamless fusion comprise various types of networks such as 3G, WLAN (WLAN), PSTN (the mutual net of public telephone), Internet, but voice-bearer, video, data sharing are at interior multiple media business.

Typical Session Initiation Protocol realizes that system configuration comprises sip server and user agent, and the user agent is divided into User Agent Client (caller) and subscriber proxy server (called).Sip server is divided into according to function: acting server, Redirect Server and location server.Wherein, acting server is transmitted to other sip servers or user agent to the sip message of receiving; Redirect Server is redirected the sip message of receiving; Location server management position information.In moving VoIP system, each user all need be in the position the own position of server end registration, promptly the user need at first dispose an IP address (or Care-of Address), and notifies location server in join domain through transmission REGISTER information.

Set up one when calling out, caller connect with location server and confirm called after, send the INVITE request message, called through after a plurality of sip server routes request message being sent to.If the called INVITE request message of successfully handling, the response message that will to send a state value be 200 OK is given caller.After receiving 200 OK information, caller is sent ACK message and is come indicating correct to receive 200 OK information, and whole flow process is as shown in Figure 1.

We see from above-mentioned Session Initiation Protocol call establishment, and in the mobile VoIP system based on Session Initiation Protocol, caller and callee need set up session through three-way handshake process.In this process, have the operation of two aspects to increase calling establishment time delay: the INVITE information of 1) shaking hands and sending from caller (MPTY) for the first time needs could arrive called (called party) through the sip server route of some; 2) caller (MPTY) need obtain the current location of called (called party) through connecting with location server.

To the VoIP delay problem based on SIP, document [1] has been analyzed wireless fading channel and has been set up the time delay that the SIP session is produced, and document [2] has been set up SIP session establishment time delay analytical model in the multi-speed wireless network.Yet these researchs do not have to consider to move the calling establishment time delay that produces owing to mobile management in the VoIP system, and only provide some notional results, fail to provide the effective ways that reduce time delay.

[1]H.Fathi，S.Chakraborty，and?R.Prasad，“On?SIP?session?setup?delay?for?VoIP?services?over?correlated?fading?channels，”IEEE?Trans.Veh.Technol.，vol.55，no.1，pp.286-295，Jan.2006.

[2]S.Pack，G.Park，K.Lee，and?W.Lee，“Analysis?of?SIP?transfer?delay?in?multi-rate?wireless?networks，”IEEE?Commun.Lett.，vol.14，no.10，pp.918-920，Oct.2010.

Summary of the invention

The object of the present invention is to provide a kind of reduction to move voip call setup delay method.

Thought of the present invention is: to the voip call setup delay problem based on SIP; In the position updating process of caller; Preestablish the called threshold value that is connected the remaining time when carrying out Location Registration with the SIP location server; If called real surplus hour of log-on is higher than this threshold value, then extracts this called positional information in advance and set up session connection.Like this, can save the spent time delay of caller access location server, and directly set up session with called.

Performing step of the present invention is following:

Step (1), caller are sent the REGISTER log-on message and are carried out the caller location renewal to location server; Owing to comprise a large amount of positional informations in the location server; Event needs a criterion to confirm the candidate called subscriber; The called threshold value that is connected the residue hour of log-on when carrying out Location Registration with the SIP location server is set in caller, and this threshold value is chosen criterion as the user;

After step (2), location server receive and handle this registration message, send one 200 OK response message to caller, this 200 OK information comprises all called positional informations that can be arrived by calling;

Whether the called and SIP location server that step (3), determining step (2) obtain is connected residue hour of log-on when carrying out Location Registration greater than threshold value, if less than, then caller is according to conventional method and called connecting; Otherwise, continue next step;

Step (4), according to 200 OK information, caller is extracted the residue hour of log-on in advance greater than the called positional information of threshold value, send then the ACK confirmation to the position server;

When called mobility very high, promptly called be connected with location server when carrying out Location Registration can be frequent the change positional information, then the residue registration time of staying shorter, the positional information of extracting so in advance can not be used again.Therefore need choose the called subscriber who has enough remaining times.Threshold tau th through the hour of log-on of definition residue in advance has only when be higher than this value called remaining time, just can extract this called positional information in advance.Usually, this threshold value be made as the called Location Registration time half etc.

Step (5), caller are directly sent the INVITE request message to called; If the called INVITE request message of successfully handling, then the transmit status value is that the message of 200 OK is given caller;

Step (6), caller are sent the ACK acknowledge message to called after receiving 200 OK information, connect.

Beneficial effect

If whether caller calledly is not connected residue hour of log-on when carrying out Location Registration greater than preset threshold value through judging with the SIP location server; Thereby extract called positional information in advance; Then need to send earlier INVITE to the position server; And check called current location, this will produce the calling establishment time delay of long period.Adopt operation of the present invention, can omit this operation in step, reduce session settling time greatly.

Description of drawings

Fig. 1 is general, and flow process is set up in the SIP voip call

Fig. 2 reduces the implementation step of voip call setup delay

Embodiment

If the SIP voip call is set up in the process, general calling settling time is 100ms, and residue hour of log-on threshold tau _ThUsually be taken as the half the of call set-up time, i.e. τ _ThCan be made as 50ms.Suppose to have in the network 2 called A, B, wherein the residue hour of log-on T of called A _A(40ms) be lower than threshold tau _Th, caller adopts general session establishing method to be connected with called A; The residue residence time T of called B _B(70ms) be higher than threshold tau _Th, the session establishing method that caller adopts the present invention to propose is connected with called B.

For convenient contrast, list calling and called A below respectively, B sets up the step of session.

Step (1), caller are sent the REGISTER log-on message and are carried out the caller location renewal to location server, and preestablish the called threshold value that is connected the residue hour of log-on when carrying out Location Registration with the SIP location server;

The threshold value of residue hour of log-on is set by caller, and as previously mentioned, this threshold value is generally the half the of call set-up time.In this example caller call set-up time 100ms divided by 2 can calculate the residue hour of log-on threshold value be 50ms, establish this value and be kept at memory cell MEM_0 through software programming.

After step (2), location server receive and handle this registration message, send one 200 OK response message to caller, this 200 OK information comprises all called positional informations that can be arrived by calling, comprises called A and called B in the present embodiment;

Step (3), whether the more called respectively A of caller and called B are connected residue hour of log-on when carrying out Location Registration greater than preset threshold value τ with the SIP location server _Th

If the residue hour of log-on of called A and called B is kept at memory cell MEM_A and MEM_B respectively, then MEM_A, MEM_B and MEM_0 are taken out in caller from memory cell, and compare the size of these numerical value.

Because the residue hour of log-on T of called A _A(40ms) be lower than threshold tau _Th(50ms), caller adopts general session establishing method to be connected with called A, and concrete steps see step (4)-(6) for details:

The INVITE request message is sent in step (4) caller, sends this message to called A through after a plurality of sip server routes;

Step (5) is if called A successfully handles the INVITE request message, and then the transmit status value is that the response message of 200 OK is given caller;

After 200 OK information were received in step (6) caller, transmission ACK message extremely called A came indicating correct to receive.

Because the residue residence time T of called B _B(70ms) be higher than threshold value (50ms), then the method that connects of calling and called B is seen step (7)-(8).

Step (7), caller are directly sent the INVITE request message to called; If the called INVITE request message of successfully handling, then the transmit status value is that the message of 200 OK is given caller;

Step (8), caller are sent the ACK acknowledge message to called after receiving 200 OK information, connect.

Set up process with the general session of caller and called A and compare, the advantage that calling and called B sets up session is:

1) caller is extracted residue residence time called greater than threshold value in advance according to 200 OK information, need not to connect to confirm called with the SIP location server again.

2) caller is directly sent the INVITE request message to called, need not to send to this message called through after a plurality of sip server routes.

The operation of this two aspect can reduce SIP session connection time delay greatly.

Claims

1. one kind is reduced mobile voip call setup delay method, it is characterized in that, may further comprise the steps:

Step (1), caller are sent the REGISTER log-on message and are carried out the caller location renewal to location server, and the called threshold value that is connected the residue hour of log-on when carrying out Location Registration with the SIP location server is set in caller.

Whether the called and SIP location server that step (3), determining step (2) obtain is connected residue hour of log-on when carrying out Location Registration greater than preset threshold value, if less than this threshold value, then caller is according to conventional method and called connecting; Otherwise, continue to carry out next step;