CN101022471A

CN101022471A - Method and system for realizing public telephone exchange network simulation service

Info

Publication number: CN101022471A
Application number: CN200610007656.1A
Authority: CN
Inventors: 黄世碧; 毛凌志; 王鹏
Original assignee: Huawei Technologies Co Ltd
Current assignee: Huawei Technologies Co Ltd
Priority date: 2006-02-15
Filing date: 2006-02-15
Publication date: 2007-08-22
Anticipated expiration: 2026-02-15
Also published as: WO2007093116A1; CN100571299C

Abstract

A method for realizing simulation service of public telephone exchange network includes carrying out conversion between simulation user line signaling and SIP signaling in calling-conversation process by access signaling adaptive entity, realizing simulation user line signaling information transmission between SIP application server and said entity through SIP protocol and simultaneously carrying out relevant treatment by said server according to analysis of simulation user line signaling information for realizing PSTN simulation service.

Description

A kind of method and system that realize public telephone exchange network simulation service

Technical field

The present invention relates to public switched telephone network (PSTN) (PSTN, Public Switched TelephoneNetwork) artificial service technology, refer to a kind of method and system of carrying out the public telephone exchange network simulation service of analog line Signalling exchange in the session that are implemented in especially.

Background technology

The analog line signaling is to upload the analog telephone terminal of passing and the signaling between switching equipment at the analog subscriber circuit, the analog line signaling mainly comprises following four types signaling: the User Status signaling of 1) describing User Status, the dislodging machine state of User Status signaling reflection analog telephone terminal, switching equipment can detect user's machine of whether seting out by detecting having or not of electric current on the subscribers feeder; 2) digital signalling is the number information that the user pulled out that the analog telephone terminal sends to switching equipment; 3) ringing-current and signal tone are switching equipment notifies the result's of user call proceeding signaling to analog telephone terminal being used to of sending, as tone signal and bell signals such as dialing sound, busy tone, ring-back tones; 4) hooking signaling is the signaling that the analog telephone terminal is sent to switching equipment, and it is mutual generally to be used for triggering the analog line signaling information that the call session process carries out between user and the switching equipment.

In general, analog line Signalling exchange between analog telephone terminal and the switching equipment occurs in call session and sets up or the call session period for removal, but, part PSTN supplementary service need realize in the call session process, just carries out the analog line Signalling exchange in user's the communication process.With the call waiting is example, analog telephone user A and telephone subscriber B are in communication process, if there is third party telephone subscriber C to call out analog telephone user A, switching equipment will send the Call Waiting tone signal has third party's calling to analog telephone user A with notice, telephone subscriber C hears ring-back tone, and at this moment analog telephone user A can have following three kinds of selections:

1) if refuse the incoming call of telephone subscriber C, analog telephone user A does not do any operation, and switching equipment will stop to send the stand by tone signal after a period of time;

2) if keep telephone subscriber B, and change the conversation with telephone subscriber C into, then analog telephone user A presses R key or hooking on phone, and after hearing dialing tone again by numerical key 2;

3) if finish conversation with telephone subscriber B, change the conversation with telephone subscriber C into, then analog telephone user A presses R key or hooking on phone, and after hearing dialing tone again by numerical key 1.

From above-mentioned example as seen, in the call session process of analog telephone user A and telephone subscriber B, for realizing call waiting, analog telephone terminal and switching equipment need carry out the analog line Signalling exchange, send stand by tone signal, analog telephone terminal as switching equipment to the analog telephone terminal and send digital signallings such as R key, numerical key 2, numerical key 1 to switching equipment.

Business such as other business of carrying out the analog line Signalling exchange in the call session process also has calling transfer, Three-Way Calling, searches malicious call, calling back when busy, storing call when busy, the definition of relevant these business, information such as manipulate, can repeat no more here with reference to national communication industry relevant criterion.

At present, no matter be the narrow band switching machine equipment of carrying control unification, also be based on the Softswitch that carrying control is separated, the switching equipment service processing software of its core all can be regarded a finite-state automata as, needs the driving of the input and output incident of dependence analog line signaling and network side signaling to finish various PSTN business.For the Business Processing that needs to carry out the analog line Signalling exchange in the call session process, the event-driven that also needs to rely on the analog line signaling in the call session could realize corresponding business.

Continuous maturation along with group technology, develop based on Circuit-switched conventional telecommunication network to broadband telecommunication net based on packet switching, the grouping telecommunications network is prepared or set up in a lot of operations commercial city, though it is developing tendency in future that the user uses packet communicate terminal to insert the grouping telecommunications network, but operator is in the process of construction of grouping telecommunications network, need progressively to existing PSTN/ integrated services digital network (ISDN, Integrate Service digital Network) network is transformed, realize that existing PSTN/ISDN network is to next generation network (NGN, Next Generation Network) smooth evolution, after requiring existing P STN/ISDN core network to be grouped telecommunications core network to replace it, existing PSTN/ISDN terminal use perception can keep the terminal of existing network less than network change, User Network Interface, professional experiences etc. are constant.In the research of each present relevant criterion tissue, above-mentioned packet telecommunications network is applied to the transformation of PSTN/ISDN core net and the application of replacement, also can be described as PSTN/ISDN emulation (PSTN/ISDN Emulation).

IP Multimedia System (IMS, IP Multimedia Subsystem) is the subsystem of third generation partner program (3GPP) at the support IP multimedia service of Rel5 version proposition as a kind of network based on packet switching.The core characteristics of IMS are to adopt session initiation protocol (SIP, SessionInitiation Protocol) and with the independence that inserts, IMS is a multimedia control/calling controlling platform on packet domain, support conversation class and non-conversation class multimedia service, for the multimedia application in future provides a general business platform.Under network integration development trend, many international and domestic organizing all in the network integration scheme of research based on IMS, purpose is to make IMS become general-purpose platform based on the SIP session, supports simultaneously to move and fixing plurality of access modes.Because the research of the network integration just begins, that is that all right is ripe for technology, and its standardization effort also correspondingly becomes the emphasis of present research.In the framework of NGN, terminal and access network are various, and have only an IMS network based on the core network of SIP session, and IMS provides service for fixing and portable terminal simultaneously.The IMS network as a kind of be the network of core with the IP group technology, be a kind of integration program of network evolution, also be a main trend of future development simultaneously.

Because the IMS framework has been adopted in the wideband multimedia territory among the NGN, if PSTN/ISDN Emulation Subsystem (PES, PSTN/ISDN Emulation Subsystem) also adopts the network architecture based on IMS, so a lot of network entity functions will obtain merging and sharing, therefore, in ITU Telecommunication Standardization Sector (ITU-T) and ETSI (ETSI) tissue, all set up the research work that relevant standards project carries out this respect.Provided PSTN/ISDN Emulation Subsystem function structure among etsi standard draft TS 02030 V1.2.7 based on IMS, Fig. 1 is based on the PES function structure schematic diagram of IMS, as shown in Figure 1, this PES framework applications AGCF (AGCF, Access Gateway Control Function) and the media gateway adaptive entities of access signaling such as (MG, Media Gateway) realized that the traditional PSTN terminal is adaptive to the access of IMS network.Stipulate according to draft standard TS02030V1.2.7, adaptive entity of access signaling such as AGCF can finish the conversion between analog line signaling and the SIP signaling, but do not realize the PSTN service logic, the application server (AS) that PSTN service logic control is moved to the IMS network is as among the PES AS, promptly should meet the principle that business realizing and core control are separated based on the PES framework of IMS.In addition, etsi standard draft TS3044 gives some idiographic flow definition that realize PSTN simulation services based on IMS.The concrete achievement in research based on the PSTN simulation subsystem of IMS in the research of relevant etsi standard please refer to above-mentioned two draft standards, and this paper no longer describes in detail.

At present, in ETSITS3044V0.1.0, realize that based on IMS the idiographic flow definition of PSTN simulation services has provided call session and set up or the partial simulation subscriber line signaling of call session period for removal and the transformation rule between the SIP signaling, as user's off-hook with after dialing called number, AGCF judges the number termination according to configuration data, analog line signalings such as user's off-hook and called number are converted to SIP conversation initial request (INVITE) message, and this SIP INVITE will send among the PES AS that carries out the control of PSTN service logic.AGCF is after the conversion that realizes corresponding analog line signaling and SIP signaling, though variation has taken place in the expression way to the analog line signaling in Session Initiation Protocol, but the information of the required transmission of original analog line signaling can obtain conversion in Session Initiation Protocol, therefore the information of the required transmission of analog line signaling is not lost, and PES AS can finish corresponding PSTN service logic control according to the analog line signaling information that carries in the SIP signaling like this.

And in ETSI TS3044V0.1.0, for analog line signaling mutual in the call session process, fail to resemble other occur in that call session is set up or analog line signaling of call session period for removal realize and the SIP signaling between conversion, and between AGCF and PES AS, transmit by Session Initiation Protocol.Be by in AGCF, making an explanation and handling for analog line signaling mutual in the call session process in the existing flow process, be not converted to Session Initiation Protocol and send to PES AS, such as, for hooking class business, be to detect after user's the hooking analog line signaling by AGCF, state according to existing call among the AGCF, explain the expressed implication of service code information of follow-up group of hooking signaling and hooking, finish Three-Way Calling, need to carry out the service logic control and the medium control and treatment of analog line Signalling exchange business in the call session processes such as Call Waiting.In the existing scheme, relate to the business of multi-party conversation for Call Waiting etc., AGCF need safeguard a plurality of SIP session status, need judge the current activation of which session according to the analog line signaling as AGCF, and which session is current to be session state information such as to be held; Relate to the business of a plurality of conversation medias for three party service etc., AGCF also needs to control MG and realizes that tripartite media conference bridging connects control.

To sum up, the processing method of mutual analog line signaling in the call session process that provides among the ETSI TS3044V0.1.0, explain by the adaptive entity of access signaling and handle for analog line signaling mutual in the call session process, be not converted to Session Initiation Protocol and be sent to PES AS, the business that causes relying on these analog line signalings like this can only be placed on and realize on the adaptive entity of access signaling and can't concentrate to move among the PES AS and realize.Can not meet design principle fully, thereby cause following problem based on the PES framework of IMS:

1) needs to realize part PSTN service logic control among the AGCF, run counter among the TS 02030V1.2.7 on the business based on the PES framework of IMS and moved to PES AS, and AGCF does not do professional design principle.

2) relating to the business that needs in the call session process to carry out the analog line Signalling exchange in a large number must realize on AGCF, causes AGCF to go up the Control Software more complicated.

As the hooking analog line signaling in the termination conversation procedure in AGCF, caused the relevant business of all hooking analog line signalings on AGCF, to realize, and ETSI TS3044V0.1.0 has only provided business descriptions such as Call Waiting, Three-Way Calling, other business that relies on hooking analog line signaling also has a lot, as calling transfer, search business such as malicious call, calling back when busy, storing call when busy, because can't obtain hooking analog line signaling information on the PES AS, these business also can only realize on AGCF.Like this, AGCF need carry out a large number of services of analog line Signalling exchange in realizing the call session process; For the business that relates to multi-party conversation, AGCF also needs to safeguard a plurality of SIP session status; AGCF also to control MG finish the medium control that relates to a plurality of conversation media business or the like increase in demand AGCF go up the complexity of Control Software.

3) caused the management maintenance of PES business complicated.

From existing scheme, part PSTN business realizes in PES AS, part PSTN business realizes in the adaptive entity of access signaling, this distributed service execution mode makes that problems such as cooperation between the business and conflict are difficult to solve, and in as the adaptive entity of the access signaling of AGCF, need to carry out the business datum configuration, caused professional management maintenance more complicated.

Summary of the invention

In view of this, main purpose of the present invention is to provide a kind of method that realizes public telephone exchange network simulation service, can meet the principle that is separated based on business realizing and core control in the PES framework of IMS.

Another purpose of the present invention is to provide a kind of system that realizes public telephone exchange network simulation service, can meet the principle that is separated based on business realizing and core control in the PES framework of IMS.

For achieving the above object, technical scheme of the present invention specifically is achieved in that

A kind of method that realizes the public switched telephone network (PSTN) PSTN simulation services, this method comprises:

In session initiation protocol SIP packet network, the adaptive entity of access signaling carries out conversion between analog line signaling and SIP signaling to analog line signaling information mutual in the call session process, realizes the transmission of described analog line signaling information between adaptive entity of access signaling and sip application server by Session Initiation Protocol; Sip application server carries out respective handling according to described analog line signaling information simultaneously, thereby realizes PSTN simulation services.

The method that the adaptive entity of described access signaling carries out the conversion between analog line signaling and SIP signaling is: respectively each analog line signaling information is carried in the various SIP message, or an above analog line signaling information is carried at passes to described sip application server in the same sip message.

The method that the adaptive entity of described access signaling carries out the conversion between analog line signaling and SIP signaling is: the adaptive entity of described access signaling is resolved described analog line signaling information functional information corresponding, and the function information that parsing is obtained by Session Initiation Protocol passes to described sip application server again.

The method that the adaptive entity of described access signaling carries out the conversion between analog line signaling and SIP signaling is:

Use subscription and the response mechanism of SIP, and the user key-press digital signalling information that applications keys SGML KPML transmits in the analog line signaling information is given described sip application server.

Described SIP signaling is a SIP Info message, or SIP Message message or SIP Invite message.

Comprise the message body of the MIME medium type that carries described analog line signaling information in the described SIP signaling, described MIME medium type comprises:

Be used to identify the MIME medium type field of MIME medium type classification;

Be used to identify the MIME medium sub-type field of the subtype of MIME medium type;

Be used for identifying the coded system field of the coded system that analog line signaling information that the message body of MIME medium subtype carries adopts.

The value of described MIME medium type field is for using application; The value of described MIME medium sub-type field is analog line signaling analog-subscriber-signal.

Described coded system field value is a text expansion Backus normal form ABNF mode, or extending mark language XML mode.

Described analog line signaling comprises: user's hooking signal signaling, and/or digital signalling, and/or ringing-current, and/or tone information, and/or metering pulse signaling and/or reversed polarity signaling and/or frequency shift keying signaling.

Described SIP packet network is IP Multimedia System IMS;

The adaptive entity of described access signaling is AGCF AGCF;

Described sip application server is PSTN simulation services application server PES AS.

The adaptive entity of described access signaling is the integrated access device that access signaling is adaptive and the access bearer adaption function is unified.

Comprise first user and second user in the described conversation procedure, described PSTN is professional to be the Call Waiting artificial service, and this method specific implementation comprises:

A1. after described PES AS receives request from described first user of the 3rd user's request call, call waiting tone is carried at sends to described AGCF in the SIP signaling, the IAD that described AGCF controls in the described SIP packet network is play stand by tone to described first user;

B1. the hooking signal message from described first user that described AGCF will receive is carried at and sends to described PES AS in the SIP signaling, and described PES AS is carried at the dialing sound and sends to described AGCF in the SIP signaling;

C1. described AGCF controls the medium that IAD in the described SIP packet network disconnects described first user and second user and is connected, and to described first user's Sending dialled number sound;

D1. described ACGF will be carried in the SIP signaling and send to described PES AS from described first user's service code information; Described PES AS resolves the service code information that receives knows that current business is that Call Waiting artificial service and described second user are maintenance side;

E1. described PES AS by the Media Resource Server in the described SIP packet network to the playback of described maintenance side, and and described AGCF upgrade Session Description Protocol SDP between described first user and the 3rd user, described AGCF uses the SDP after upgrading to set up media access between described first user and the 3rd user.

A2. after described PES AS receives request from described first user of the 3rd user's request call, call waiting tone is carried at sends to described AGCF in the SIP signaling, the IAD that described AGCF controls in the described SIP packet network is play stand by tone to first user;

B2. after described AGCF receives hooking signal from described first user, control the medium that IAD in the described SIP packet network disconnects described first user and second user and be connected, and to described first user's Sending dialled number sound;

C2. described ACGF will be carried at from described first user's service code information and described hooking signal message and send to described PES AS in the SIP signaling; Described PES AS resolves the information that receives knows that current business is that Call Waiting artificial service and described second user are maintenance side;

D2. described PES AS by the Media Resource Server in the described SIP packet network to the playback of described maintenance side, and and described AGCF upgrade Session Description Protocol SDP between described first user and the 3rd user, described AGCF uses the SDP after upgrading to set up media access between described first user and the 3rd user.

Comprise first user and second user in the described conversation procedure, described PSTN business is the Three-Way Calling artificial service, and this method specific implementation comprises:

A3. after described AGCF receives hooking signal from described first user, control the medium that IAD in the described SIP packet network disconnects described first user and second user and be connected, and to described first user's Sending dialled number sound; Described first user dials the 3rd user's number;

B3. described ACGF will be carried at from described first user's the 3rd user number information and described hooking signal message and send to described PES AS in the SIP signaling; Described PES AS resolves the information that receives knows that current business is the Three-Way Calling artificial service;

C3. described PES AS and described AGCF upgrade the Session Description Protocol SDP between described first user and the 3rd user, and the SDP after described AGCF use is upgraded sets up the media access between described first user and the 3rd user.

After the described step B3, before the step C3, this method further comprises:

Described PES AS changes described second user by Session Initiation Protocol and does not send Media Stream for receiving only.

Comprise first user and second user in the described conversation procedure, described PSTN business is a calling switching artificial service, and this method specific implementation comprises:

A4. behind the hooking signal that described AGCF receives, control the medium that IAD in the described SIP packet network disconnects described first user and second user and be connected from described first user, and to described first user's Sending dialled number sound;

B4. described ACGF will be carried at from described first user's service code and described hooking signal message and send to described PES AS in the SIP signaling; Described PES AS resolves the information that receives to be known that current business is a calling switching artificial service and determines that switching side is the 3rd user;

C4. described PES AS consults the Session Description Protocol SDP between described second user and the 3rd user, and the SDP after the use renewal sets up the media access between described second user and the 3rd user; Discharge described first user's session simultaneously.

Described first user is an analog subscriber.

A kind of system that realizes the public switched telephone network (PSTN) PSTN simulation services, in session initiation protocol SIP packet network, this system comprises at least: the adaptive entity of access signaling, sip application server;

Also comprise in the adaptive entity of described access signaling: the converting unit of analog line signaling information mutual in the call session process being carried out the conversion between analog line signaling and SIP signaling;

Between described converting unit and described sip application server, realize the transmission of described analog line signaling information by Session Initiation Protocol; Described sip application server carries out respective handling according to described analog line signaling information, realizes PSTN simulation services.

Each analog line signaling information that described converting unit will receive respectively is carried at and passes to described sip application server in the various SIP message, or an above analog line signaling information is carried at passes to described sip application server in the same sip message.

Described converting unit is resolved described analog line signaling information functional information corresponding, and the function information that parsing is obtained by Session Initiation Protocol passes to described sip application server again.

Described converting unit is used subscription and the response mechanism of SIP, and the user key-press digital signalling information that applications keys SGML KPML transmits in the analog line signaling information is given described sip application server.

Described SIP packet network is IP Multimedia System IMS;

The adaptive entity of described access signaling is AGCF AGCF;

As seen from the above technical solution, the present invention has provided a kind of on sip application server such as PES AS, to need in the call session process to realize carrying out the method for the PSTN simulation services of analog line Signalling exchange, the inventive method has met the principle that business realizing and core control are separated in the IMS framework; Can use based on the framework of IMS in conjunction with prior art by the present invention program and to realize 100% of PSTN business is inherited, use unified IMS core network to provide service as PSTN emulation user and IP media user, reduce the network construction cost and the management O﹠M cost of operator, had far-reaching social and economic significance.

Simultaneously, compare with the pattern that has now at the local realization of AGCF PSTN simulation services, the present invention program is provided by the AGCF of access carrier and inserts but the method and system of PSTN simulation services are provided by the PES AS of service provider, guaranteed to realize PSTN simulation services by the PES AS of home domain, supported service mobility, concentrate on the PES AS that has realized at home domain, realize PSTN simulation services uniformly, solved the equal access of PSTN simulation services easily, charge, service conflict, problems such as service dispense have been simplified management and the O﹠M of operator to PSTN simulation services.

Description of drawings

Fig. 1 is based on the PES function structure schematic diagram of IMS;

Fig. 2 is that the analog telephone terminal is used the related entities connection diagram of the adaptive entity access of access signaling based on the packet network of SIP;

Fig. 3 is the flow chart of Call Waiting artificial service embodiment of the present invention;

Fig. 4 is the flow chart of Call Waiting artificial service optimization process embodiment of the present invention;

Fig. 5 is the flow chart of Three-Way Calling artificial service embodiment of the present invention;

Fig. 6 is the flow chart of calling switching artificial service embodiment of the present invention.

Embodiment

Core concept of the present invention is: in packet-based SIP packet network, for the PSTN business of carrying out the analog line Signalling exchange in the call session, the adaptive entity of access signaling carries out conversion between analog line signaling and SIP signaling to analog line signaling information mutual in the call session process, realize the transmission of described analog line signaling information between adaptive entity of access signaling and sip application server by Session Initiation Protocol, sip application server carries out respective handling according to described analog line signaling information simultaneously, thereby realizes PSTN simulation services.

For making purpose of the present invention, technical scheme and advantage clearer, below with reference to the accompanying drawing preferred embodiment that develops simultaneously, the present invention is described in more detail.

The present invention is applicable to that the adaptive entity of analog telephone terminal use access signaling inserts the packet network based on SIP, realize carrying out in the session application scenarios of the PSTN simulation services of user and internetwork analog subscriber Signalling exchange, such as based on the PSTN simulation services of IMS with use the integrated access device of SIP to insert the analog telephone terminal and the application scenarios of realization PSTN simulation services etc. on sip application server.For convenience of description, Fig. 1 is reduced to shown in Figure 2, Fig. 2 is that the analog telephone terminal uses the adaptive entity of access signaling to insert related entities connection diagram based on the packet network of SIP.

As shown in Figure 2, wherein, the analog telephone terminal is to insert the telephone terminal based on the packet network of SIP based on the PES of IMS etc.; Need to prove that having a side in the said call session process of the present invention at least is the analog telephone terminal, and in the call session process, carry out the PSTN business of analog line Signalling exchange.

The adaptive entity of access signaling is a network side entity of finishing the functions such as conversion between analog line signaling and the SIP signaling; In the practical application, as in the PES based on IMS, the adaptive entity function of access signaling can be based on the AGCF in the PES function structure of IMS; It also can be the integrated access device etc. of the adaptive and access bearer adaption function of access signaling unification;

PES AS carries out the control of PSTN service logic, realizes the functional entity of PSTN supplementary service.

L interface between the adaptive entity of analog telephone terminal and access signaling is the interface that transmits the analog line signaling, such as in PES based on IMS, can be that intermediary transmits the analog line signaling by the MG shown in Fig. 1 between the adaptive entity of analog telephone terminal and access signaling, can use between the adaptive entity of MG and access signaling and H.248 wait MGCP; I interface between adaptive entity of access signaling and the PES AS can adopt the SIP signaling protocol, such as in PES, can transmit the SIP signaling by other network entity such as the network entities such as S-CSCF, I-CSCF in the IMS core network between adaptive entity of access signaling and the PES AS based on IMS.

Among the present invention, analog line signaling mutual in the call session process is converted to the SIP signaling, and between adaptive entity of access signaling and PES AS, transmit by the I interface, PES AS has obtained enough analog line signaling informations and has carried out corresponding service processing like this, has met the principle that business realizing and core control based on the PES framework of IMS are separated.Specifically, also comprise in the adaptive entity of access signaling of the present invention: the converting unit of analog line signaling information mutual in the call session process being carried out the conversion between analog line signaling and SIP signaling; Between described converting unit and PES AS, realize the transmission of described analog line signaling information by Session Initiation Protocol; Described PES AS carries out respective handling according to described analog line signaling information, realizes PSTN simulation services.

Described converting unit realizes being converted to:

Each analog line signaling information that converting unit will receive respectively is carried at and passes to described sip application server in the various SIP message, or an above analog line signaling information is carried at passes to PEA AS in the same sip message;

Perhaps, converting unit is resolved the analog line signaling information functional information corresponding that receives, and the function information that parsing is obtained by Session Initiation Protocol passes to PES AS again;

Perhaps, converting unit is used subscription and the response mechanism of SIP, and the user key-press digital signalling information that applications keys SGML KPML transmits in the analog line signaling information is given PES AS.

In the prior art, the analog line signaling of calling out initiation or end of calling stage is converted to the sip message that possesses similar implication easily, such as, BYE message in on-hook subscriber signaling and the SIP signaling is all represented the implication of end of calling, so the on-hook analog line signaling in end of calling stage is easy to be converted to the BYE message in the SIP signaling; And mutual analog line signaling is difficult to directly be transformed into the sip message with corresponding meaning in the call session process, needs to adopt suitable sip message to transmit analog line signaling information mutual in the call session process.After call session is set up, according to the Session Initiation Protocol normalized definition, can adopt a message in the Session Initiation Protocol, transmit interactive information in the session as SIP Info message, therefore, the present invention can adopt SIP Info message to transmit analog line signaling information mutual in the call session process; In addition, the present invention also can adopt other sip message, transmit mutual analog line signaling information in the call session process as message such as SIP Message, Invite, just at this moment the SIP session identification of Messgae or the SIP session identification of Invite message and the call session of having set up is different, and the information that needs to carry other in Message or the Invite message is come related these two SIP session identifications inequality.

Transmit mutual analog line signaling information in the call session process in the message such as SIP Info, Message, Invite in order to be implemented in, must be provided with the form of these analog line signaling informations in sip message, so that communicating pair can the corresponding information of correct understanding.Mutual analog line signaling relates to the signalings such as hooking signaling, digital signalling, ringing-current and signal tone in the analog line signaling in the call session process, such as hooking, the hooking user pulled out afterwards service code, dialing tone, busy tone, call waiting tone and bell signal or the like.In order in sip message, to transmit mutual analog line signaling information in the call session process, the present invention is provided with a kind of new multiduty the Internet mail extension (MIME, Multipurpose Internet Mail Extensions) medium type is analog line signaling (analog-subscriber-signal) MIME medium type, and this new analog line signaling MIME medium type comprises following field:

MIME medium type (MIME media type): be used to identify the field of MIME medium type classification, value is for using (application) among the present invention;

MIME medium subtype (MIME subtype): be used to identify the field of the subtype of MIME medium type, use the subtype of analog-subscriber-signal sign MIME medium type to be analog line signaling medium subtype among the present invention;

Mandatory parameter (Required parameters): do not use this territory among the present invention;

Optional parameters (Optional parameters): do not use this territory among the present invention;

Coded system (Encoding scheme): be used for identifying the coded system that the entrained analog line signaling information of message body of MIME medium subtype adopts, can be text expansion Backus normal form (ABNF, Augmented Backus-Naur Form) mode, extending mark language XML mode etc.;

The analog line signaling information that carries described in the message body of MIME medium subtype can include but not limited to following information:

A. user's hooking signaling information;

B. the digital signalling information of analog line signaling: comprise number information that the user pulls out etc.;

C. ringing-current and tone information: be used for ringing-current and tone information that PES AS issues to the adaptive entity of access signaling; Be used to notify the result's of user call proceeding information, as tone signal and bell signals such as dialing sound, busy tone, ring-back tones;

D. metering pulse signaling information;

E. frequency shift keying signaling information;

F. reversed polarity signaling information.

Analog-subscriber-signal MIME medium type according to the invention described above setting, can comprise the message body of analog-subscriber-signal MIME medium type in the sip message, as the sip message body content of carrying hooking subscriber line signaling information can be expressed as follows:

Content-Type：application/analog-subscriber-signal

hook：hook-flash

Wherein, Content-Type:application/analog-subscriber-signal shows that the information of carrying in the message body belongs to the MIME medium type, and the MIME medium subtype of this MIME medium type is analog-subscriber-signal MIME medium type; Hook:hook-flash shows that then the information of carrying in the message body is the hooking signaling information, need to prove, and be example here, also can adopt other character to represent hooking, as long as the user consults.

Be example with Call Waiting, Three-Way Calling, calling switching below, specifically describe the using SIP agreement and between adaptive entity of access signaling and PES AS, transmit the method that mutual analog line signaling information in the call session process is realized PSTN simulation services.

Fig. 3 is the flow chart of Call Waiting artificial service embodiment of the present invention, among Fig. 3, suppose that the adaptive entity of access signaling is AGCF, the analog line signaling adopts Session Initiation Protocol to transmit mutual analog line signaling information in the call session process between AGCF and PES AS, and hypothetical simulation user A is the analog telephone terminal, user B and user C are sip terminal, and the present invention realizes that the method for Call Waiting artificial service may further comprise the steps:

Step 300～step 306: set up basic session between analog subscriber A and user B and call out.

After AGCF receives dialing information from analog subscriber A, send conversation initial request (Invite) sip message of the user B corresponding to PES AS with dialing information; PES AS transmits the InviteB sip message that receives and gives user B, and user B sends to AGCF with 180 Ringing SIP ring signalings via PES AS; After AGCF receives 180 Ringing signalings, the control IAD sends ring-back tone to analog subscriber A, and after receiving the 200OK SIP affirmation signaling of transmitting via PES AS from user B, send the ACK sip response message via PES AS to user B, analog subscriber A and user B communicate.

This step is according to existing PSTN emulation technology based on IMS, and AGCF carries out the conversion between analog line signaling and the SIP signaling, and specific implementation can repeat no more here referring to the relevant criterion draft.

Step 307～step 308: in analog subscriber A and the user B communication process, PES AS receives the conversation initial request (Invite) from the calling analog subscriber A of user C, PES AS judges that the session of analog subscriber A and user B exists, and hypothesis knows the analog subscriber A call waiting of having contracted according to pre-configured business datum, and PES AS transmits call waiting tone analog line signaling information by SIP Info message to AGCF.

In this step, carry the message body of expression call waiting tone analog line signaling information in the SIP Info message, can be expressed as follows:

Content-Type：application/analog-subscriber-signal

tone：call-waiting

Wherein, tone:call-waiting shows that the information of carrying in the message body is call waiting tone information.

Step 309～step 310:PES AS returns expression to user C and calls out the 182SIP message of lining up; After AGCF received and carries the SIP Info message of call waiting tone analog line signaling information from PES AS, the control IAD inserted call waiting tone analog line signal to analog subscriber A.

Step 311: analog subscriber A hooking, and send to AGCF by the analog line signaling.

Step 312:AGCF transmits hooking analog line signaling information by SIP Info message to PES AS;

In this step, carry the message body of expression hooking analog line signaling information in the SIP Info message, can be expressed as follows:

Content-Type：application/analog-subscriber-signal

hook：hook-flash

Wherein, hook:hook-flash shows that the information of carrying in the message body is hooking information.

Step 313:PES AS is carried at dialing sound analog line signaling information in the SIP Info message, and sends to AGCF.

After step 314～step 315:AGCF receives and carries the SIP Info message of dialing tone analog line signaling information from PES AS, the medium that the control IAD disconnects analog subscriber A and user B are connected, and to analog subscriber A Sending dialled number sound analog line signaling.

In this step, the specific implementation that the control IAD disconnects the medium connection procedure of analog subscriber A and user B belongs to technology as well known to those skilled in the art, can repeat no more here with reference to relevant criterion.

Step 316: analog subscriber A dials service code such as numerical key 2, and sends to AGCF by the analog line signaling.

Step 317:AGCF is carried at the service code analog line signaling that receives in the SIP Info message, transmits the service code analog line signaling information that analog subscriber A is pulled out to PES AS; After PES AS receives the service code information that analog subscriber A pulled out, know that through resolving analog subscriber A selects to keep user B, connects Call Waiting user C.

Step 318～step 323:PES AS is user B playback to the corresponding voice resource of Media Resource Server application and to maintenance side.

MRS is media resource controlled function (MRFC) among Fig. 1 and a kind of physics realization of two entities of media resource processing capacity (MRFP), is used to network that media resources controls such as playback, meeting bridge are provided.The existing MRS product that media resource is provided in the prior art, among the present invention, MRS can be separated into key-course MRFC entity and Media layer MRFP entity, be a business realizing example in the present embodiment, and expression use media resource is put to calling maintenance side and holded music.In actual the realization, this step is optional, because maintenances side knows will be held as the voice suggestion in conversing etc., so can be without any need for prompt tone.Use MRS and the present invention irrelevant, just a specific embodiment that provides for the operation flow integrality.

Session Description Protocol (SDP) information of the analog subscriber A that step 324～step 325:PES AS consults when using analog subscriber A and user B to set up session is returned 200OK message to user C, with call accepted; After user C receives 200OK message, return and confirm ACK message.

Step 326～step 329:PES AS uses the SDP information of the user C that carries in the Invite A sip message of user C calling analog subscriber A in the step 307 to carry out the SDP renewal to user A initiation change conversation request (Re-Invite), AGCF connects control according to the Media Stream that the SDP after upgrading carries out the IP medium end points of analog subscriber A, between the IP medium end points of the IP of analog subscriber A medium end points and user C, connect, thus the two-way media path of connection analog subscriber A and user C.

In the flow process shown in Figure 3, AGCF converts to each mutual in call session process analog line signaling information and is passed to PES AS after the SIP Info message and handles, in order to reduce the interaction times of transmitting the analog line signaling between AGCF and the PES AS, under the situation of the service logic control and treatment that does not influence PES AS, AGCF can do optimization process to the transmission of analog line signaling information, promptly waits and collects after a plurality of analog line signalings from pseudo subscriber terminal again by the disposable PES of the being sent to AS of SIPInfo message.Fig. 4 is the flow chart of Call Waiting artificial service optimization process embodiment of the present invention, as shown in Figure 4, the analog line signaling informations such as service code that hooking analog line signaling in the step 311 and the analog subscriber A in the step 316 are dialed are carried in the sip message simultaneously, send to PES AS again and handle.

Need to prove, AGCF will realize above-mentioned optimization process, AGCF need possess the ability of judging when service code that the user dialed stops, and the mode of the realization of this ability can be by setting in advance service code on AGCF dialing rule judges when service code stops.

Step 411 among Fig. 4～step 416 has realized the optimization process to step 311～step 317 among Fig. 3, supposes to set in advance in AGCF the dialing rule of service code, specifically describes as follows:

Step 411: analog subscriber A hooking, and send to AGCF by the analog line signaling.

After step 412～step 413:AGCF received hooking analog line signaling, the medium that the control IAD disconnects analog subscriber A and user B were connected, and to analog subscriber A Sending dialled number sound analog line signaling;

Step 414: analog subscriber A dials service code such as numerical key 2, and sends to AGCF by the analog line signaling.

After step 415～step 416:AGCF judges service code dialing termination according to the service code dialing rule that sets in advance, AGCF is carried at the service code analog line signaling information that hooking and analog subscriber A are dialled in the SIP Info message, transmits the hooking of analog subscriber A and the service code analog line signaling information that is pulled out to PES AS; PES AS receive hooking and the service code information dialled after, know that through resolving analog subscriber A selects to keep user B, connects Call Waiting user C.

The realization of the step 300～step 310 among the step 400 among Fig. 4～step 410 and Fig. 3 is in full accord, and subsequent step and the step 318～step 329 among Fig. 3 after the step 416 among Fig. 4 are in full accord, no longer repeat here.

What present embodiment was emphasized is, a plurality of analog line signalings from pseudo subscriber terminal are sent to PES AS by same SIP Info message, has reduced the interaction times of transmitting the analog line signaling between AGCF and the PES AS.

Fig. 5 is the flow chart of Three-Way Calling artificial service embodiment of the present invention, among Fig. 5, suppose that the adaptive entity of access signaling is AGCF, the analog line signaling adopts Session Initiation Protocol to transmit mutual analog line signaling information in the call session process between AGCF and PES AS, and hypothetical simulation user A is the analog telephone terminal, user B and user C are sip terminal, and are in the communication between analog subscriber A and the user B, and the present invention realizes that the method for Three-Way Calling artificial service may further comprise the steps:

Step 500: analog subscriber A hooking, and pass to AGCF by the analog line signaling.

After step 501～step 502:AGCF received hooking analog line signaling, the medium that the control IAD disconnects A and B were connected, and sent dialing sound analog line signaling to user A.

Step 503: analog subscriber A dials the number that the third party is user C, and by the analog line signaling number information of user C is passed to AGCF.

After step 504～step 505:AGCF judges the subscriber dialing termination according to the dialing rule that sets in advance, AGCF is carried at the analog line signaling information of the number of hooking and the user C that pulled out in the SIP Info message, and send to PES AS, after PES AS receives the number information of hooking and user C, know that through resolving analog subscriber A need keep user B, and connect user C.

Step 506～step 508:PES AS does not send for receiving only by the IP medium end points of Session Initiation Protocol change user B, and specific implementation is a technology as well known to those skilled in the art, can repeat no more here referring to related protocol.

Need to prove that this step is optionally in reality realizes, if omit this step, user B may also send Media Stream, but because user B call out to be kept, analog subscriber A also is what can't receive from the Media Stream of user B, can cause the waste of IP network bandwidth like this.

The SDP that step 509～step 510:PES AS is used analog subscriber A is to the user C conversation request that makes a call; User C returns 180 Ringing SIP ring signalings to PES AS.

Step 511～step 513:PES AS describes according to the SDP of the IP medium end points of the user C that carries in the SIP ring signaling, and the medium connected mode of the IP medium end points of change analog subscriber A is for only receiving the Media Stream from the IP medium end points of user C.

Step 515: analog subscriber A receives the ring-back tone signaling from user C.

Step 516～step 517: user C returns 200OK SIP to PES AS and replys signaling, and with answering call, PES AS returns response ACK message to user C.

The SDP of the IP medium end points of the user C that carries in the SIP ring signaling of step 518～step 521:PES AS according to user C in step 511～step 513 describes, the medium connected mode of the IP medium end points of change analog subscriber A is for sending and receive the Media Stream from the IP medium end points of user C simultaneously, AGCF control IAD realizes that corresponding medium connect control, thereby is communicated with the two-way media path of analog subscriber A to user C.

Fig. 6 is the flow chart of calling switching artificial service embodiment of the present invention, among Fig. 6, suppose that the adaptive entity of access signaling is AGCF, the analog line signaling adopts Session Initiation Protocol to transmit mutual analog line signaling information in the call session process between AGCF and PES AS, and hypothetical simulation user A is the analog telephone terminal, user B and user C are sip terminal, and are in the communication between analog subscriber A and the user B, and the present invention realizes that the method for calling switching artificial service may further comprise the steps:

Step 600: analog subscriber A hooking, and pass to AGCF by the analog line signaling.

After step 601～step 602:AGCF received the hooking subscriber line signaling, the medium that the control IAD disconnects analog subscriber A and user B were connected, and to analog subscriber A Sending dialled number sound analog line signaling.

Step 603: analog subscriber A dials service code, such as ^*12 ^*The number # of user C, and be passed to AGCF by the analog line signaling.

After step 604～step 605:AGCF was known the subscriber dialing termination according to the dialing rule that sets in advance, AGCF was with the hooking that receives and pulled out ^*12 ^*The number # analog line signaling information of user C is carried in the SIP Info message, and send to PES AS, and PES AS receives hooking and pulled out ^*12 ^*After the number # information of user C, know that through resolving it is user C that analog subscriber A selects user B conversation is transferred to transfer side.

This step can also realize like this: AGCF resolves physical simulation subscriber line signaling implication earlier, by agreements such as SIP this function information of switch over operation is passed to PES AS again, wherein the number of user C can be used as a parameter of switch over operation, carry switch over operation as operation field of expansion in Session Initiation Protocol, expand another number field and carry the number of user C, receive the sip message of parameter of the number that carries switch over operation and corresponding forwarded user as PES AS after, implement service logic control according to the implication and the relevant parameter of operation field.

Step 606:PES AS uses the SDP of the IP medium end points of user B to describe the conversation request Invite sip message that makes a call to user C.

Step 607: user C returns the 180Ringing signaling to PES AS, and the SDP that carries the IP medium end points of user C in this ring signaling describes.

Step 608～step 610:PES AS describes according to the SDP of the IP medium end points of the user C that carries among the 180Ringing, and the medium connected mode of the IP medium end points of change user B is for sending and receive the Media Stream from the IP medium end points of user C simultaneously.

Step 611～step 612: user C returns 200 OK responses to PES AS, after PES AS receives 200OK, returns the ACK acknowledge message to user C.

So far under the control of PES AS, user B and user C have set up the two-way media path.

Step 613～step 614:PES AS uses the SIPBye process to discharge the session of analog subscriber A after user B conversation is transferred to user C.

Need to prove, this paper only provides the application flow of the PSTN simulation services that needs to carry out the analog line Signalling exchange in several typical call session processes, do not limit the inventive method and be only applicable to this several business, the inventive method goes for needing in the call session process to carry out other business of analog line Signalling exchange equally; This paper is with based on realizing on AGCF in the PES function structure of IMS that the adaptive entity of access signaling is that example is carried out business description, do not limit the inventive method and be only applicable to this network architecture, the inventive method is equally applicable to meet other network function framework of logical architecture shown in Figure 2, has realized that as use the integrated access device of the SIP of the adaptive entity function of access signaling inserts the analog telephone terminal and realize the application scenarios etc. of PSTN business simulating on sip application server; A complete calling and professional control flow are not represented in the explanation that flow process diagram of being done among the present invention and explanatory note are only done for outstanding key technology of the present invention, do not have all possible branch of limit flow process yet; And incident packet format that the sip message of describing carries and title only for outstanding its analog subscriber signaling information that must carry, do not represent that this is only describing mode.

To sum up, the present invention program has realized on the application server of IMS network realizing needing to carry out the mutual PSTN simulation services of subscriber line signaling in the call session process, has met the principle that business realizing and core control are separated in the IMS framework; Can use based on the framework of IMS in conjunction with prior art by the present invention program and to realize 100% of PSTN business is inherited, use unified IMS core network to provide service as PSTN emulation user and IP media user, reduce the network construction cost and the management O﹠M cost of operator, had far-reaching social and economic significance.

Simultaneously, the present invention program is provided by the AGCF of access carrier and inserts but the method for PSTN simulation services is provided by the PES AS of service provider, guaranteed to realize PSTN simulation services by the PES AS of home domain, supported service mobility, concentrate, realize uniformly PSTN simulation services on the PES AS that has realized at home domain, solved the problem such as equal access, service conflict, service dispense of PSTN simulation services easily, simplified management and the O﹠M of operator PSTN simulation services.

In addition, between adaptive entity of access signaling and PES AS, also can use subscription and the response mechanism of use SIP given in the mutual sip event bag draft (draft-ietf-sipping-kpml:A Session Initiation Protocol (SIP) Event Package for Key Press Stimulus) of the button of IETF, and applications keys SGML (KPML, Key Press Markup Language) transmits the user key-press digital signalling information in the analog line signaling information.Specific implementation comprises: the dialing rule that PES AS uses subscription (SUBSCRIBE) message of Session Initiation Protocol to carry and adopts KPML to describe is initiated the subscription of analog subscriber key-press event to the adaptive entity of access signaling, when the adaptive entity detection of access signaling arrives user key-press information, if the dialing rule that this user's case information conforms PES AS subscribes to, then the adaptive entity of access signaling sends the number information that SIP notice (Notify) information reporting analog subscriber is dialed to PES AS.

The above is preferred embodiment of the present invention only, is not to be used to limit protection scope of the present invention, all any modifications of being made within the spirit and principles in the present invention, is equal to replacement, improvement etc., all should be included within protection scope of the present invention.

Claims

1. a method that realizes the public switched telephone network (PSTN) PSTN simulation services is characterized in that, this method comprises:

In session initiation protocol SIP packet network, the adaptive entity of access signaling carries out conversion between analog line signaling and SIP signaling to analog line signaling information mutual in the call session process, realizes the transmission of described analog line signaling information between adaptive entity of access signaling and sip application server by Session Initiation Protocol; Sip application server carries out respective handling according to described analog line signaling information simultaneously, realizes PSTN simulation services.

2. method according to claim 1, it is characterized in that, the method that the adaptive entity of described access signaling carries out the conversion between analog line signaling and SIP signaling is: each analog line signaling information is carried at passes to described sip application server in the various SIP message respectively, or an above analog line signaling information is carried at passes to described sip application server in the same sip message.

3. method according to claim 1, it is characterized in that, the method that the adaptive entity of described access signaling carries out the conversion between analog line signaling and SIP signaling is: the adaptive entity of described access signaling is resolved described analog line signaling information functional information corresponding, and the function information that parsing is obtained by Session Initiation Protocol passes to described sip application server again.

4. method according to claim 1 is characterized in that, the method that the adaptive entity of described access signaling carries out the conversion between analog line signaling and SIP signaling is:

5. method according to claim 1 is characterized in that, described SIP signaling is a SIP Info message, or SIP Message message or SIP Invite message.

6. method according to claim 1 is characterized in that, comprises the message body of the MIME medium type that carries described analog line signaling information in the described SIP signaling, and described MIME medium type comprises:

7. method according to claim 6 is characterized in that, the value of described MIME medium type field is for using application; The value of described MIME medium sub-type field is analog line signaling analog-subscriber-signal.

8. method according to claim 6 is characterized in that, described coded system field value is a text expansion Backus normal form ABNF mode, or extending mark language XML mode.

9. method according to claim 1 is characterized in that, described analog line signaling comprises: user's hooking signal signaling, and/or digital signalling, and/or ringing-current, and/or tone information, and/or metering pulse signaling and/or reversed polarity signaling and/or frequency shift keying signaling.

10. according to the described method of claim l, it is characterized in that: described SIP packet network is IP Multimedia System IMS;

The adaptive entity of described access signaling is AGCF AGCF;

11. method according to claim 1 is characterized in that: the adaptive entity of described access signaling is the integrated access device that access signaling is adaptive and the access bearer adaption function is unified.

12. method according to claim 10 is characterized in that, comprises first user and second user in the described conversation procedure, described PSTN is professional to be the Call Waiting artificial service, and this method specific implementation comprises:

B1. the hooking signal message from described first user that described AGCF will receive is carried at and sends to described PES AS in the SIP signaling, and described PES AS is carried at dialing tone and sends to described AGCF in the SIP signaling;

13. method according to claim 10 is characterized in that, comprises first user and second user in the described conversation procedure, described PSTN is professional to be the Call Waiting artificial service, and this method specific implementation comprises:

14. method according to claim 10 is characterized in that, comprises first user and second user in the described conversation procedure, described PSTN business is the Three-Way Calling artificial service, and this method specific implementation comprises:

15. method according to claim 14 is characterized in that, after the described step B3, before the step C3, this method further comprises:

16. method according to claim 10 is characterized in that, comprises first user and second user in the described conversation procedure, described PSTN business is a calling switching artificial service, and this method specific implementation comprises:

B4. described ACGF will be carried at SIP from described first user's service code and described hooking signal message and send to described PES AS in letter the present; Described PES AS resolves the information that receives to be known that current business is a calling switching artificial service and determines that switching side is the 3rd user;

17., it is characterized in that described first user is an analog subscriber according to each described method of claim 12 to 16.

18. a system that realizes the public switched telephone network (PSTN) PSTN simulation services is characterized in that, in session initiation protocol SIP packet network, this system comprises at least: the adaptive entity of access signaling, sip application server;

19. system according to claim 18, it is characterized in that, each analog line signaling information that described converting unit will receive respectively is carried at and passes to described sip application server in the various SIP message, or an above analog line signaling information is carried at passes to described sip application server in the same sip message.

20. system according to claim 18 is characterized in that, described converting unit is resolved described analog line signaling information functional information corresponding, and the function information that parsing is obtained by Session Initiation Protocol passes to described sip application server again.

21. system according to claim 18, it is characterized in that, described converting unit is used subscription and the response mechanism of SIP, and the modern information of user key-press numeral letter that applications keys SGML KPML transmits in the analog line signaling information is given described sip application server.

22. system according to claim 18 is characterized in that, described SIP packet network is IP Multimedia System IMS;

The adaptive entity of described access signaling is AGCF AGCF;