CN208257918U - Merge the mobile Internet voice platform system of public switched telephone network PSTN - Google Patents

Abstract

The utility model discloses a kind of mobile Internet voice platform systems of combining PSTN.Above system includes: mobile Internet voice platform, and at least one PSTN access end system being connected respectively with the mobile Internet voice platform；Wherein, each PSTN access end system includes: the channel access server CES for being responsible for the PSTN access upper audio processing of end system, is connected respectively with the channel server CS of the mobile Internet voice platform and channel management server CMS；It is responsible for the Computer &Telephony Integration Server CTIS of the PSTN access end system bottom audio processing, is connected with the CES.By adopting the above technical scheme, under the unavailable scene of Internet network, user to use mobile Internet real-time voice call product have strong demand again in the case where, realize the purpose of real time full duplex voice communication requirements, provide PSTN network access mobile Internet voice platform channel.

Description

Merge the mobile Internet voice platform system of public switched telephone network PSTN

Technical field

The utility model relates to the communications fields, in particular to a kind of fusion public switched telephone network (Public Switched Telephone Network, referred to as PSTN) mobile Internet voice platform system.

Background technique

With the development of information technology, the arrival of mobile internet era, mobile social activity product gradually rises, at present market Upper common such as microblogging, QQ, wechat, footpath between fields footpath between fields, YY product are also all being provided from media information publication, instant messaging, stranger While the business scenarios services such as friend-making, audio-video live streaming, social element is incorporated to diversification, so that all kinds of mobile Internets APP can have product function abundant and good user experience.

On social platform product, the exchange of person to person is fundamental, and common communication function mainly has: picture and text are instant Message, asynchronous speech message (half-duplex network phone (Voice Over Internet Protocol, referred to as VOIP) voice Message), real-time voice call (full duplex VOIP voice communication) etc. forms of expression.Wherein " picture and text instant message " be it is most convenient, Most common media of communication, it transmits text in the networks such as WiFi/GSM/CMDA by client application software and server Or multi-medium data, realize picture and text instant messaging；" asynchronous voice " technology is a kind of network-based " push to speak " business, should Business function starts recording module when user presses record button, and acquisition, which is spoken, audio signal and to be encoded, compress, and passes through network Server transport is to other side's mobile phone, and after receiver equipment receives audio data, click play button is listened to by loudspeaker.It is this Scheme carries out social communication by recorded speech short message of speaking, and release both hands input information, reduce and use threshold, and convenience has one It is fixed to be promoted；The VOIP technology that " real-time voice call " is and develops rapidly, present broadband network and 4G mobile network are general In the case where and, real-time voice, which is applied, to be also widely developed and applies in mobile Internet, and continuous speech exchange increases The strong presence in virtual network social environment, has also obtained the favor of user.The internets such as QQ, wechat, YY social category Product all realizes one-to-one, the multi-to-multi real time phone call product between good friend, this VOIP network exchanged close to black phone The speech communication product communication experience more comfortable to people, but it equally also proposes more the data transmission quality of communication network High requirement.

Consider in view of factors such as construction cost and eases of use, mobile Internet real-time voice call product on the market It is substantially and uses the broadband Internet WiFi and mobile 3G/4G network as communication bearer network, while it is soft to pass through server Exchange system carries out audio data jitter buffer, codec compression, audio mixing synthesis processing and distribution and delivers, and realizes multi-party real-time language The transmission switching subsystem of sound call.Broadband and 4G network have been popularized at present, although can achieve in 4G network theory per second The bandwidth speed (12.5MB about per second) of 100Mbps, but transmitted on internet due to IP data packet and rely on multiple network The network problem that the wireless signals power factor such as routing node data exchange and terminal network signal interference, shielding causes, still The network access quality of some mobile device is bad, leads to Internet network interruption or data transfer delay shake occur The scene being not suitable for using VOIP real-time voice such as larger.

Thus, under the unavailable scene of Internet network, user to use mobile Internet real-time voice converse product In the case where having strong demand again, using which kind of platform architecture, can be realized real time full duplex voice communication requirements, be at present urgently It solves the problems, such as.

Utility model content

The main purpose of the utility model is that a kind of mobile Internet voice platform system of combining PSTN is disclosed, with It at least solves in the related technology under the unavailable scene of Internet network, user converses to using mobile Internet real-time voice Product has strong demand again in the case that, the problem of also lacking the platform architecture that can be realized real time full duplex voice communication requirements.

According to the utility model, a kind of public switched telephone network (Public Switched Telephone is provided Network, referred to as PSTN) mobile Internet voice platform system characterized by comprising mobile Internet voice is flat Platform, and at least one PSTN access end system being connected respectively with the mobile Internet voice platform；Wherein, Ge Gesuo Stating PSTN access end system includes: the channel access server for being responsible for the PSTN access upper audio processing of end system CES is connected with the channel server CS of the mobile Internet voice platform and channel management server CMS respectively；It is responsible for Computer &Telephony Integration Server (the Computer Telephony of the PSTN access end system bottom audio processing Integration, referred to as CTIS), it is connected with the CES.

Compared with prior art, the utility model embodiment has at least the following advantages: by public switched telephone network PSTN Mobile Internet voice platform is accessed, the mobile Internet voice platform system of combining PSTN is formed, using the system, Under the unavailable scene of Internet network, user to use mobile Internet real-time voice call product have the case where strong demand again Under, the purpose of real time full duplex voice communication requirements is realized, PSTN network access mobile Internet voice platform is provided Channel.

Detailed description of the invention

Fig. 1 is the mobile Internet voice platform according to the fusion public switched telephone network PSTN of the utility model embodiment The structural block diagram of system；

Fig. 2 is the structural block diagram that end system is accessed according to the PSTN of the preferred embodiment in the utility model；

Fig. 3 is the structural schematic diagram according to the CES and CTIS of the preferred embodiment in the utility model；

Fig. 4 is the structural block diagram according to the mobile Internet voice platform of the preferred embodiment in the utility model；

Fig. 5 is the mobile Internet voice according to the fusion public switched telephone network PSTN of the preferred embodiment in the utility model The structural block diagram of plateform system.

Specific embodiment

The specific implementation of the utility model is made a detailed description with reference to the accompanying drawings of the specification.

Fig. 1 is the mobile Internet voice platform according to the fusion public switched telephone network PSTN of the utility model embodiment The structural block diagram of system.As shown in Figure 1, the mobile Internet voice platform system includes: mobile Internet voice platform 10, with And at least one PSTN access end system 20 being connected respectively with the mobile Internet voice platform 10；Wherein, Ge Gesuo Stating PSTN access end system 20 includes: the channel access server for being responsible for the PSTN access upper audio processing of end system 20 (CES) 200, respectively with the channel server (CS) 100 and channel management server of the mobile Internet voice platform 10 (CMS) 102 are connected；It is responsible for the Computer &Telephony Integration Server of the next audio processing of the PSTN access end system 20 (CTIS) 202, it is connected with the CES 200.

It is in mobile Internet voice platform system shown in Fig. 1, public switched telephone network (PSTN) access is mobile mutual Network voice platform, form the mobile Internet voice platform system of combining PSTN, using the system, Internet network not Can be under scene, user realizes real-time to using in the case that mobile Internet real-time voice call product has strong demand again The purpose of full-duplex voice communication requirements provides the channel of PSTN network access mobile Internet voice platform.

Preferably, as shown in Fig. 2, PSTN access end system 200 can also include: WEB server 204；It is responsible for tune Room channel information is obtained with the interface of the WEB server, the room channel letter is broadcasted when user accesses corresponding process The interactive voice answering system (IVRS) 206 of breath, is connected with the WEB server and the CTIS respectively.

IVRS 40 encapsulates HTTP access interface, and 30 interface of called Web service device obtains room channel information, visits in user Room information (type, number etc.) is broadcasted when asking corresponding process, is entered for telephone subscriber by IVR process selecting.Telephone subscriber When selection enters channel, room id information is passed to CTIS by the interface with CTIS by IVRS.Finally called by CTIS LibCES the corresponding interface enters channel.

IVRS audits voice social platform room sound function, voice social platform (CMS) realize one it is virtual " thoroughly It is bright " corresponding room audio is delivered to CTIS gateway by user.One channel User Manager of design, Ke Yiji are needed inside CTIS It records current gateway node phone and participates in number information, while realizing that audio stream subscribes to function by the channel User Manager.

CTIS is the main body of voice social platform channel audio data docking PSTN network management, it is responsible for calling insertion The libCES of CTIS shares the request interface into voice social platform channel that library module provides, and voice frequency is added in user Road.CTIS realizes channel and the design of channel Member Manager, for managing corresponding internal storage data, is calling insertion CTIS's Data when the shared library module of libCES enters (leaving) room channel, in maintenance manager.To save CTIS computer room flow band Width, the downlink voice of a room channel only passes through a Virtual User channel, and (corresponding Virtual User administrative mechanism is by CMS/ CES/libCES cooperation is completed) issue CTIS.CTIS calls the corresponding interface of libCES to complete access voice social platform room The function of channel, meanwhile, libCES can adjust back output audio data after being successfully accessed room.At this point, CTIS is managed according to oneself The room channel and information about firms of reason, by certain the room voice bag data (G.711a-law) constantly received by driving API to pass It is sent to the corresponding bottom hardware channel of board, realizes that the audio on telephone line plays.

To realize telephone subscriber and voice social platform user live chat communication function, at this moment libCES will be speech User creates full-duplex voice row of channels up and down, for sending the uplink and downlink audio data of user's speech.

Preferably, the CTIS includes: downlink reception port, uplink sending port, respectively with the downlink reception port The computer telephone integration CTI controller connected with the uplink sending port, the CTI relaying being connect with the CTI controller Switching equipment.

CTIS is generally deployed on Industrial Control Computer, can be inserted or connect common number in the industrial personal computer PCI slot Relaying switching equipment (such as: three remittance SHD-120D-CT digital junction sound cards, eastern into Keygoe3003 multimedia switchboard), number Word relays switching equipment backboard and accesses digital junction coaxial cable.The hardware driving of sound card is installed on industrial personal computer simultaneously, CTIS, which is started by software program and operates sound card driving, to be realized to the access control of trunk (such as: telephone connection, audio The basic operations such as recording, audio broadcasting).

Need to connect board hardware in view of CTIS, and hardware driving version is limited by various types of hardware product producer, often There is a problem of supporting quality inconsistent operating system.Therefore, CTIS is also likely to be present a possibility that multiple systems version, often It is windows platform and Linux platform.Therefore libCES needs to provide the version of a variety of operating systems, corresponding Windows system, it should it is libCES.dll, corresponding linux system, it should be libCES.so.

LibCES is run in the runtime by integrated be loaded into the memory address space that CTIS process is distributed of operating system.It is The application layer of CES business integrates SDK, it externally provides service and realizes that we design it and provide function grade interface tune by API With convenient for integrated access side's fast integrations such as CTIS.

LibCES and CES is communicated, and is transmitted signaling by TCP channel, is transmitted voice by UDP channel.By all kinds of instructions Agreement and audio stream data are packaged processing, and the SDK development kit for providing api class is used for application layer is integrated.

For libCES by C language function pointer conceptual design readjustment notification interface, the corresponding application layer program of CTIS will Oneself be used to receive instruction and the function address of audio data and be transmitted in libCES module memory, libCES receive it is not of the same race After class data, unpacks and CTIS application layer is passed data to by call back function pointer after converting.

IVRS obtains voice social platform chatroom room list information, room ID is provided when communicating with CTIS, by CTIS The corresponding interface of libCES is called to complete the function of access mobile Internet voice social platform chatroom channel.Meanwhile LibCES readjustment output room audio data.

Preferably, above-mentioned CES includes: uplink receiving port, the encoder connecting with the uplink receiving port, and described The uplink sending port of encoder connection；Downlink reception port, the jitter buffer control being connected with the downlink reception port Device, the audio decoder being connected with the jitter buffer controller, the mixer being connected with the audio decoder, with institute State the downlink sending port that mixer is connected.

Preferably, the channel transmission control protocol (TCP) for having transmission signaling, institute are established between the CES and the CMS State User Datagram Protocol (UDP) channel established between CES and the CS and have transmission voice.

Preferably, the TCP channel for having transmission signaling, and the UDP of transmission voice are established between the CES and the CTIS Channel.

Fig. 3 is the structural schematic diagram according to the CES and CTIS of the preferred embodiment in the utility model.

As shown in figure 3, CES is the Core server of virtual mobile Internet voice social activity APP terminal, it is each PSTN Upper (host computer for the being compared to automatic control system) audio treatment unit of access point.

Application integration SDK of the libCES as CES establishes communication bridge between the servers such as CES and CTIS.Fig. 3 will Being logically present for the module is ignored in libCES transparency process, convenient for the membership credentials between expression server.

Between CMS after " channel is added " signaling is docked successfully, CES and CS establish UDP communication channel and (use CES Real-time Transport Protocol communication), and enter specified channel and access data.The UDP channel is full-duplex channel, can carry out downstream tones packet Transmission and upstream tones transmission.The processes such as acquisition, coding compression, transmission about VOIP packets of audio data are not the utility model Emphasis, repeat no more.

When receiving downlink voice, the jitter buffer controller of CES is responsible for the transmission jitter cushioning control of audio data, audio Decoder is decoded audio, and subsequent mixer is by decoded audio data and the room background music Media Stream that receives Data (there may be the movements such as the decoding of music media stream, resampling for music media stream receiving module) carry out audio mixing and audio After sampling width, frequency, coded format conversion process, CTIS is then sent to by downlink sending port, board is operated by CTIS It is output to PSTN route.

When receiving ascending voice, the encoder of CES is responsible for carrying out audio data into coding compression, and uplink sending port is responsible for Package is sent, and audio data is finally sent to CS processing.

Wherein, between this system CES and CS audio coding can use SILK encryption algorithm (code rate 30kbps), CES with Audio coding between CTIS can be using G.711a-law encryption algorithm (code rate 64kbps).All audio coding decoding processes It completes to calculate all in CES.

As shown in figure 3, CTIS is access PSTN network, the processing server of voice trunking board equipment is controlled, carries out sound For frequency according to docking with board link data, it is the bottom (slave computer for being compared to automatic control system) of each PSTN access point Audio treatment unit.

CTIS is by digital junction switching equipment, for example, N number of (for example, 32) channel time slot on E1 will be relayed (Timeslot) it is encapsulated as 30 channels (0/16 time slot is the synchronous and signalling time slot), every relaying E1 provides 30 user concurrents Access ability.

CTIS can be each voice social platform room channel creation Virtual User (create and manage in advance), for uniting One receives each channel downlink voice of voice social platform, only need to be in CTIS if pstn telephone user only audits channel content Subscribe to the channel audio stream in portion.Audio data can be delivered to each audio subscriber by CTIS.

CTIS PSTN user wish the seat of honour make a speech when, CTIS by CES be the user creation exclusively enjoy signaling (signaling with CMS carries out TCP communication) and audio stream (audio stream and CS carry out UDP communication) transmission channel, realize that the audio of full duplex counts in real time According to transmitting-receiving.CTIS is interacted with voice plate card audio data, (G.711a-law encodes) voice compression using PCMA.

CTIS carries out memory playing and memory recording by drive control plate card, realizes the real-time audio number with VOIP system It is controlled according to recorded broadcast.Memory PCMA data are delivered to Labcard driver and realize playback by it, and similarly, board can also pass through driving when recording Adjust back the PCMA internal storage data that CTIS program obtains recording.

Preferably, as shown in figure 4, the CMS 102, is connected by full duplex UDP communication channel with the CS 100 It connects；Above-mentioned mobile Internet voice platform 10 can also include: to be responsible for providing the access server of the interface of room channel information ES 104 is connected with the WEB server 204 of the CMS 102 and PSTN access end system 20, wherein institute respectively ES 104 is stated to communicate by hypertext transfer protocol http protocol with the WEB server 204.

Wherein, CMS is the voice-enabled chat channel management server of social interaction server device, it is social platform real-time voice function One of the Core server of energy.In the access scheme, the PSTN accessing user interface logic that chatroom is added, creation one are provided A user and it is added to CS.And the information such as CS distributing user access voice correlation RTP service IP/PORT are returned into client (CES)。

Consider for factors such as management and costs, in the telecom operators such as telecommunications, connection, movement of rental or cooperation number It is typically only capable to receive or breathe out after E1 route to net interior telephone number.Therefore CMS supports more CES node accesses, i.e., many operations Business opportunity room can all dispose IVR system.Therefore, CES also has multiple, and each CES can be added identical according to business demand application Chatroom channel.

CES can virtually a certain number of social platform audio clients, responsible communicate with CMS access Virtual User CS obtains the audios real-time Transmission information such as the port RTP, the SSRC of CS distribution.

It is communicated using the Real-time Transport Protocol and parameter rule and CS to match with CS, receives Chatroom channel voice, shake is slow Punching, audio decoder, and will export audio resampling is sample frequency 8K, the sample bits 8bit that PSTN route is supported G.711a-law PCM data.

(CTI, Computer Telephony Integration, computer telecommunication are integrated by CES and insertion CTIS.CTIS, CTI Server, access PSTN network have the gateway system of IVR function) libCES (Windows system version is LibCES.dll or Linux system version are libCES.so) communication, command information and audio data are output to CTIS, by this Server control is docked with CTI relaying switching equipment API, and media information is output to digital junction chain road.

In view of voice system migrates and develop the factors such as complexity, CES is the application clothes for running on linux system Business program, CTIS are determined according to the demand version of digital junction switching equipment product, it may be possible to Windows server program, It may be Linux server program.And libCES is then the dynamic operation library for being embedded in CTIS.LibCES is only responsible for simple control Instruction and audio data sequenceization encapsulation are packaged transparent transmission work, shake control module without planned network.Therefore, CES and CTIS It need to be deployed in the 1000M local area network of same computer room, it is ensured that possess reliable network communication route between them.LibCES is initial The listening port for actively connecting CES binding when change as client is created that the TCP/UDP communication for stablizing transmission between them Channel.Then, control instruction and packets of audio data are forwarded on a passage.

Telephone subscriber and the APP user's intercommunication of voice social platform carry out the scene of multi-party call, each room channel downlink Voice uses particular virtual subscriber channel, CTIS is sent to by CES, to save flow bandwidth.

Due to linking digital repeatered line, the phone of access PSTN network link by digital speech board after CTIS The line delay and transmission jitter of user's progress audio data transmission are extremely small, and CTIS concentrates on a node and passes through CES accesses CS.Therefore, the user under the same CTIS gateway control listens to the audio-frequency information of same chat but channel, Ke Yitong It crosses unified down going channel and receives data progress internal storage data and subscribe to and distribute.Multiple data channel transmission identical datas are avoided to cause Bandwidth waste.In general the speech user of telephony access, which need to create, exclusively enjoys full-duplex channel, and phone speech user's is upper Downstream tones are transmitted by the designated lane of oneself, do not use Virtual User channel data.

ES can provide the interface of Chatroom channel list, obtain the room list for the opening that can enter.It provides more The room list and room information query service of the features such as more personalized interest matchings, crowd's matching, good friend's matching.ES is mentioned The interface of confession is packaged using http protocol, is integrated HTTP client module for IVRS and is called pulling data.

The cloud portal server that ES is used as mobile Internet voice platform, in addition to providing room for servers such as IVRS Between outside the information such as list, also bear user's access capability service of mobile Internet voice platform.Therefore, ES is stepped on user A whole set of ergasia such as record, authentication, to realize access service function.Meanwhile ES connecing as mobile Internet voice platform Enter server, also achieves the agent functionality of the network communication between mobile client application APP and CMS, i.e., mobile visitor Family end APP interacts (such as: addition channel leaves channel, obtains channel members list, audits, movement of making a speech) with CMS, unlike The communication modes of the direct-connected CMS of CES referred to herein, but TCP long of all communications all by being established between APP and ES The forwarding of connecting communication channel broker is realized.

Mimic of the above-mentioned CES as mobile Internet voice client logic using direct-connected CMS communication plan, and does not have There is the direct-connected ES as APP, and acts on behalf of the communication for realizing APP to CMS by ES.The reason is as follows that:

CES is judged as client when PSTN user needs to enter certain mobile Internet voice APP room channel Whether the communication channel that has initialized establish the channel completed is had.No then default establishes a communication channel, realization pair The subscription listening function of the upper a certain channel of CMS/CS.The function is to export the main operation modes of audio (i.e.: as PSTN system Audit mode), under this mode, listener only receives the audio output of room channel, due to being proprietary between server Design anonymous can be realized, logical design of not commenced business based on user identity.

CES is as client, and PSTN user wishes to participate in channel interaction, and at this time system just needs to establish the language of full duplex Message road.The full duplex voice channels that PSTN speech user is exclusively enjoyed by oneself are realized and other use of mobile Internet voice system Two-way real-time voice exchange between family.And possess the user for exclusively enjoying full-duplex channel, it is inevitable to be needed at other users product end It shows its identity and status information, is at this time still that anonymous is then unable to satisfy demand.So at this moment CES needs first and ES Communication application user identity token (that is: User ID, the ID can be interim distribution and occupy, and be also possible to permanent allocation occupancy), It is later mark by the User ID, room channel is entered as the user that commonly makes a speech and is interacted with other users.

Access server of the ES as mobile client APP, provide TCP long connection service, by with APP client it Between the customization signaling protocol packet arranged carry out service interaction.These signaling protocol packagestructured datas are finally before transmission over networks Binary data stream can be serialized into.In this example, IVRS is due to being external service server, for examining for interface encapsulation Consider, is not easy to be customized TCP long connection data communication with ES, therefore we design the room channel list between IVRS and ES Information data is obtained to be carried out using widely used http protocol in the industry, reduces docking difficulty.Therefore ES can be taken with embedded HTTP Business device function, receives the HTTP request from client and return to room channel information.It can also be by other systems (such as: number According to library) unloading data, and data acquisition clothes are realized by traditional HTTP server (such as: Nginx/Apache/Nodejs) Business.

Fig. 5 is the mobile Internet voice according to the fusion public switched telephone network PSTN of the preferred embodiment in the utility model The structural block diagram of plateform system.As shown in figure 5, the mobile Internet voice platform system includes: mobile Internet voice platform, And at least one PSTN access end system being connected respectively with the mobile Internet voice platform, wherein each PSTN Access end system is deployed in a computer room respectively.

In Fig. 5, in mobile Internet voice platform, CMS is connected with ES and CS respectively.PSTN is accessed in end system, CTIS is connected with CES and IVRS respectively, and IVRS is connected with WEB server.WEB server is carried out by http protocol and ES The TCP channel for having transmission signaling is established in communication between CMS and CES, CES and CS, which are established, has the full duplex UDP of transmission voice to believe Road.

In a preferred implementation process, the user of PSTN system accesses under IVRS/CTIS cooperation control, passes through WEB service The interface of device does authentication to ES, then by CES, initiates to request to CMS.CMS establishes Virtual User, asks for point to CS The information such as the address port matched, and reply to CES.Then, the user of PSTN system is accessed by CTIS as communication, IVRS The management for carrying out mathematical logic business, at the end CES, is modeled to a Virtual User as a telephone subscriber, is being chatted by CS It is communicated in its room with other users.

In conclusion public switched telephone network (PSTN) is accessed and is moved by above-described embodiment provided by the utility model Dynamic internet voice platform, forms the mobile Internet voice platform system of combining PSTN, using the system, in Internet Network is unavailable but user is in the case where using product again and having strong demand, can provide the channel of PSTN network access platform, realize The purpose of real time full duplex voice communication requirements.After the system is implemented, more access approach can be effectively provided and produced to terminal Product user.It is also possible to utilize some voices based on pure pstn telephone network construction existing for this technology docking on the market Community enriches the product content of these voice communities, builds closed traditional PSTN speech exchange community and mobile Internet The system platform of user's docking, expands product user face, enriches the communicative channel of people.

Disclosed above is only several specific embodiments of the utility model, and still, the utility model is not limited to this, The changes that any person skilled in the art can think of should all fall into the protection scope of the utility model.

Claims (7)

1. a kind of mobile Internet voice platform system for merging public switched telephone network PSTN characterized by comprising mobile Internet voice platform, and at least one the PSTN incoming end system being connected respectively with the mobile Internet voice platform System；

Wherein, each PSTN access end system includes:

Be responsible for the channel access server CES of the PSTN access upper audio processing of end system, respectively with the mobile interchange The channel server CS of net voice platform is connected with channel management server CMS；

It is responsible for the Computer &Telephony Integration Server CTIS of the PSTN access end system bottom audio processing, with the CES phase Connection.

2. system according to claim 1, which is characterized in that the CES includes:

Uplink receiving port, the audio coder connecting with the uplink receiving port are connect upper with the audio coder Row sending port；

Downlink reception port, the jitter buffer controller being connected with the downlink reception port are controlled with the jitter buffer Audio decoder that device is connected, the mixer being connected with the audio decoder, the downlink being connected with the mixer Sending port.

3. system according to claim 1, which is characterized in that the CTIS includes:

Downlink reception port, uplink sending port are connect with the downlink reception port and the uplink sending port respectively Computer telephone integration CTI controller, the CTI relaying switching equipment being connect with the CTI controller.

4. system according to any one of claim 1 to 3, which is characterized in that established between the CES and the CMS There is the transmission control protocol TCP channel of transmission signaling, the user datagram for having transmission voice is established between the CES and the CS Agreement UDP channel.

5. system according to any one of claim 1 to 3, which is characterized in that established between the CES and the CTIS There are the TCP channel of transmission signaling, and the UDP channel of transmission voice.

6. system according to claim 1, which is characterized in that the PSTN accesses end system further include:

WEB server；

It is responsible for calling the interface of the WEB server to obtain room channel information, when user accesses corresponding process described in casting The interactive voice answering system IVRS of room channel information, is connected with the WEB server and the CTIS respectively.

7. system according to claim 1, which is characterized in that

The CMS is connected by full duplex UDP communication channel with the CS；

The mobile Internet voice platform further include: it is responsible for providing the access server ES of the interface of room channel information, point It is not connected with the WEB server of the CMS and PSTN access end system, wherein the ES passes through Hyper text transfer Agreement http protocol is communicated with the WEB server.