CN102548025B

CN102548025B - Method for reducing mobile voice over internet protocol (VoIP) call setup delay

Info

Publication number: CN102548025B
Application number: CN201210071573.4A
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: 2015-04-29
Anticipated expiration: 2032-03-16
Also published as: CN102548025A

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

A kind of mobile voip call setup delay method of reduction

Technical field

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

Background technology

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

Current, rapidly, voip technology also expands to voice, video and data etc. at interior multimedia service by pure speech transmission service to the development of IP based network high-quality Real-time multimedia.The key realizing VoIP is signaling technology, current extensive use have two kinds of Signalings: H.323 with SIP (Session Initiation Protocol, Session initiation Protocol).

H.323 powerful but very complicated, it specify complete multimedia application architecture, require higher, implement more difficult.Session Initiation Protocol is simple, easily expansion and a realization, and text based telephone signaling control protocol, it is one and accesses irrelevant application layer signaling protocol with bottom.Based on the next generation network of SIP, seamless fusion can comprise various types of network such as 3G, WLAN (WLAN), PSTN (public telephone DIALOGUES), Internet, can voice-bearer, video, data sharing is in interior media business.

Typical Session Initiation Protocol realizes system architecture and comprises sip server and user agent, and user agent is divided into User Agent Client (caller) and subscriber proxy server (called).Sip server is divided into according to function: proxy server, Redirect Server and location server.Wherein, proxy server is transmitted to other sip servers or user agent the sip message received; Redirect Server is redirected the sip message received; Location server management position information.In mobile VoIP system, each user needs the position in position server end registration oneself, namely first user need configure an IP address (or Care-of Address), and notifies location server in join domain by sending REGISTER information.

When setting up a calling, caller connect with location server and determine called after, send INVITE request message, request message sent to called after multiple sip server route.If successfully process INVITE request message called, will send a state value is that the response message of 200OK is to caller.After receiving 200OK information, caller transmission ACK message indicates and proper reception of 200OK information, and whole flow process as shown in Figure 1.

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

For the VoIP delay problem based on SIP, document [1] analyzes wireless fading channel and sets up the time delay that SIP session produces, and document [2] establishes SIP session establishment time delay analytical model in multi-speed wireless network.But these study the calling establishment time delay not considering to produce due to mobile management in mobile VoIP system, and only provide some notional results, fail to provide the effective ways reducing 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 mobile voip call setup delay method of reduction.

Thought of the present invention is: for the voip call setup delay problem based on SIP, in the position updating process of caller, preset the called threshold value being connected the remaining time of carrying out when position is registered with 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, the time delay spent by caller access location server can be saved, and directly set up session with called.

Implementation step of the present invention is as follows:

Step (1), caller sends REGISTER log-on message and carries out caller location renewal to location server; Owing to comprising a large amount of positional informations in location server, therefore need a criterion to determine candidate called subscriber, the called threshold value being connected the residue hour of log-on carried out when position is registered with SIP location server of caller setting, using this threshold value as user's Criterion of Selecting;

Step (2), location server receives and after processing this log-on message, send a 200OK response message to caller, this 200OK response message comprises all called positional information that can be arrived by calling;

Step (3), the called and SIP location server that determining step (2) obtains is connected the residue hour of log-on carried out when position is registered and whether is greater than threshold value, if be less than, then caller conventionally connects with called; Otherwise, continue next step;

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

When called mobility is very high, namely called connection with location server carries out to change positional information frequently when position is registered, then the residue registration time of staying is shorter, and the positional information extracted so in advance can not be used again.Therefore need to choose the called subscriber having enough remaining times.By the threshold tau of pre-defined residue hour of log-on _th, only have when called remaining time higher than this value time, just can extract this called positional information in advance.Usually, this threshold value is set to the half etc. of called position hour of log-on.

Step (5), caller directly sends INVITE request message to called; If successfully process INVITE request message called, then sending state value is that 200OK response message is to caller;

Step (6), caller sends ACK confirmation message to called after receiving 200OK response message, connect.

Beneficial effect

If caller is not by judging whether the called residue hour of log-on carried out when position is registered that is connected with SIP location server is greater than default threshold value, thus extract called positional information in advance, then need first to send INVITE to location server, and check called current location, this will produce the calling establishment time delay of long period.Adopt operation of the present invention, the operation of this step can be omitted, greatly reduce the session establishment time.

Accompanying drawing explanation

Fig. 1 general SIP voip call Establishing process

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

Detailed description of the invention

If in SIP voip call process of establishing, general calling Time Created is 100ms, and remains hour of log-on threshold tau _thusually the half of call set-up time is taken as, i.e. τ _th50ms can be set to.Suppose in network, have 2 called A, B, wherein the residue hour of log-on T of called A _a(40ms) 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) higher than threshold tau _th, the session establishing method that caller adopts the present invention to propose is connected with called B.

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

Step (1), caller sends REGISTER log-on message and carries out caller location renewal to location server, and presets the called threshold value being connected the residue hour of log-on carried out when position is registered with 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 of call set-up time.In this example, caller can calculate divided by 2 the threshold value remaining hour of log-on call set-up time 100ms is 50ms, if this value is kept at memory cell MEM_0 by software programming.

Step (2), location server receives and after processing this registration message, sends a 200OK response message to caller, and this 200OK packets of information, containing all called positional information that can be arrived by calling, comprises called A and called B in the present embodiment;

Step (3), whether the caller residue hour of log-on that more called A is connected with called B and SIP location server when carrying out position registration is respectively greater than default threshold tau _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 caller takes out MEM_A, MEM_B and MEM_0 from memory cell, and compares the size of these numerical value.

Due to the residue hour of log-on T of called A _a(40ms) lower than threshold tau _th(50ms), caller adopts general session establishing method to be connected with called A, and concrete steps refer to step (4)-(6):

Step (4) caller sends INVITE request message, after multiple sip server route, send this message to called A;

Step (5) is if called A successfully processes INVITE request message, then sending state value is that the response message of 200OK is to caller;

After step (6) caller receives 200OK information, send ACK message and indicate correct reception to called A.

Due to the residue residence time T of called B _b(70ms) higher than threshold value (50ms), then the method that calling and called B connects is shown in step (7)-(8).

Step (7), caller directly sends INVITE request message to called; If successfully process INVITE request message called, then sending state value is that the message of 200OK is to caller;

Step (8), caller sends ACK confirmation message to called after receiving 200OK information, connect.

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

1) caller according to 200OK information extract in advance residue residence time be greater than the called of threshold value, determine called without the need to connecting with SIP location server again.

2) caller directly sends INVITE request message to called, without the need to sending to called by this message after multiple sip server route.

The operation of these two aspects can reduce SIP session connection time delay greatly.

Claims

1. reduce a mobile voip call setup delay method, it is characterized in that, comprise the following steps:

Step (1), caller sends REGISTER log-on message and carries out caller location renewal to location server, and the called threshold value being connected the residue hour of log-on carried out when position is registered with SIP location server of caller setting;

Step (3), called and the SIP location server that determining step (2) obtains is connected the residue hour of log-on carried out when position is registered and whether is greater than default threshold value, if be less than this threshold value, then caller conventionally connects with called; Otherwise, continue to perform next step;