CN1416249A - Distributed calling center based on Internet protocol - Google Patents
Distributed calling center based on Internet protocol Download PDFInfo
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- CN1416249A CN1416249A CN 02137269 CN02137269A CN1416249A CN 1416249 A CN1416249 A CN 1416249A CN 02137269 CN02137269 CN 02137269 CN 02137269 A CN02137269 A CN 02137269A CN 1416249 A CN1416249 A CN 1416249A
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- 238000004088 simulation Methods 0.000 description 7
- 238000004891 communication Methods 0.000 description 3
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Abstract
The central system consists of several calling centers. Each calling center includes the telephone switching system, the computer telephone integrated service and the customer data center. Further, the voice processing unit and the gateway is included. The voice-processing unit compresses or decompresses the voice signal. The gateway takes the charge for building the calling connection to other calling center as well as controlling the calling procedure. In the calling period, the data of the calling party through the gateways in the calling and called parties are sent to the computer telephone integrated server at the called party. The voice signal send and received in the packet switching network is completely by the gateways in the calling and called parties.
Description
Technical field
The present invention relates to the interconnection technique of call center, strange land, particularly a kind of distributed call center based on Internet Protocol (IP).
Background technology
As shown in Figure 1, the call center generally comprises telephone switching system, cti server and customer data center etc., and wherein, telephone switching system mainly is responsible for according to certain allocation algorithm, and the phone that the user is squeezed into is reasonably distributed to the seat treatment people on backstage; Computer telephone integration (CTI) server then provides the contact interface of switch and computer intercommunication, make the information such as calling number that computer can provide according to switch, user's data in closing of the circuit, is presented at before seat personnel's the computer screen; User data that the customer data central store is relevant and Business Information.
Telephone switching system generally can adopt the computer card mode to realize, the function of conventional switch promptly is provided by the computer speech integrated circuit board of special use, makes up call center system.These special-purpose computer voice plate cards adopt special-purpose digital signal processor (DSP) chip to improve the speech processes ability, and analog of telephone line and digital telephone line interface can be provided respectively.Simultaneously, also user's the phone and the seat personnel on backstage can be connected, carry out communication on telephone by special voice seat integrated circuit board.
The call center generally is widely used in the service trade such as bank, telecommunications and consulting and the after-sale service department of large enterprises.Expansion along with scale of operation, the corresponding growth of service object's quantity, service area also no longer is confined to a ground and may be extended to the strange land, and a tame enterprise may set up several call centers to improve efficiency of service and to save operation cost in different geographical position for this reason.But the call center that these strange lands distribute is separate often, can cause the idle and waste of resource thus.For example, when the call center on a certain ground is in the overload operation state and during the call center on another ground professional not enough, if UNICOM not mutually between the telephone switching system of two call centers, then customer data can't be shared, traffic carrying capacity also can't be diverted to idle call center from the call center of excess load, thereby cause the waste of resource on the one hand, cause the decline of service quality on the other hand again.
Summary of the invention
The purpose of this invention is to provide a kind of distributed call center system, interconnected between each call center by the packet switching network, therefore each call center can realize the unified distribution of strange land call processing resource on the basis of data sharing, improves the utilance of resource and reduces the operation cost of call center.
Be made up of some call centers according to distributed call center of the present invention system, each described call center comprises telephone switching system, computer telephone integration server and customer data center, further comprises Audio Processing Unit and gateway, wherein,
Audio Processing Unit links to each other and links to each other with gateway through bus with telephone switching system, be used for telephone switching system is offered gateway behind the audio digital signals stream boil down to voice signal bag that bus provides, and the voice signal bag decompress(ion) that gateway provides is condensed to audio digital signals stream after bus offers telephone switching system; And
Gateway links to each other with Audio Processing Unit with telephone switching system, be used for setting up to call out being connected and controlling calling procedure between the calling and called side according to the calling control command of telephone switching system and address, calling and called side and the gateway of other call center, during calling out, the calling and called side gateway voice signal bag that provides of Audio Processing Unit separately is encapsulated as grouped data after the packet switching network is sent to recipient's gateway, and will be descapsulated into the Audio Processing Unit that offers behind the voice signal bag separately from the grouped data that the packet switching network receives.
In above-mentioned distributed call center system, gateway also links to each other with the customer data center with the computer telephone integration server, during call setup, calling party's gateway offers callee's computer telephone integration server with the customer data of calling party's customer data central store through callee's gateway, and offers the callee by this computer telephone integration server.
In above-mentioned distributed call center system, reasonable is that described bus meets the integrated agreement of many equipment suppliers (MVIP), H.100 agreement or an agreement in the agreement H.110.
In above-mentioned distributed call center system, reasonable to be the described packet switching network be the Internet, call setup between the call center gateway and control and gateway based on Internet Protocol (IP) to the encapsulation of grouped data and decapsulation based on an agreement in H.323 agreement, MGCP or the conversation initialized protocol.
In above-mentioned distributed call center system, reasonable is that the compressed format of described audio digital signals stream is observed agreement G.711, G.723 agreement, G.728 agreement or an agreement in the agreement G.729.
Therefore, intrasystem each call center of distributed call center of the present invention all comprises gateway, therefore can realize calling out by the packet switching network and connect and exchange customer data and voice signal, realize the unified distribution of data sharing and call processing resource thus, improved the utilance of resource and reduced the operation cost of call center.
Description of drawings
Fig. 1 is the composition frame chart of prior art call center.
Fig. 2 is two schematic diagrames that the call center interconnects by IP network in the distributed call center of the present invention system.
Fig. 3 is the functional stratification figure of the preferred embodiment in call center in the distributed call center of the present invention system.
Embodiment
In distributed call center of the present invention system, can unified allocation of resources for the call resources that makes a plurality of call centers, customer data can be shared between each call center, should make them by network interconnection.Obviously, the network between the computer system connects can adopt multiple mode and medium, but considers factors such as realizing cost, applied environment and technology trends, is a kind of preferable mode by packet switching network interconnection.Be most widely used and still have bright development prospect based on the packet switching network of Internet Protocol (IP), therefore in the present invention, reasonable is by IP network each call center to be linked together.
Fig. 2 is two schematic diagrames that the call center interconnects by IP network in the distributed call center of the present invention system.Each call center comprises telephone switching system, computer telephone integration server, customer data center, Audio Processing Unit (said units is not all drawn) and gateway.The call proceeding that will insert its extension set A2 when call center A desire is during to the extension set B1 of call center B, in A inside, call center, under the control of telephone switching system, calling party's's (be call center A in extension set A2) address information (for example the extension number of calling party's telephone number and access etc.), callee's's (being extension set B1 in the call center B) address information (comprises gateway G under the callee
BNumber for access and callee's extension number) and the control command of set up calling out be provided for gateway G
AGateway G
AThen determine gateway G according to Number for access
BThe IP address and according to the call setup control command and the gateway G of telephone switching system
BSet up the TCP session through IP network communication, and send the address information of calling and called side to gateway G
BSelect as another kind, also can in telephone switching system, comprise the table of comparisons of Number for access and gateway ip address, so gateway G
BThe IP address also can directly offer gateway G by telephone switching system
AAs also having a kind of selection, callee's Number for access and extension number can be a number according to certain principle combinations, gateway G
ACan from this number, extract the Number for access and the extension number of the IP address that is used for definite callee's gateway according to this rule respectively.Gateway G
BAccording to callee's extension number with call proceeding to extension set B1.If extension set B1 off-hook, i.e. call through between extension set A2 and the B1, then calling and called side's gateway G
AWith G
BBetween the parameter in audio compression and the packet processing method is held consultation.After negotiation is finished at calling and called side gateway G
AWith G
BBetween set up the full duplex logical channel, so can converse between the calling and called side.
During call setup, calling party's gateway G
AAlso with the customer data of calling party's customer data central store through callee's gateway G
BOffer the computer telephone integration server in the callee call center, therefore when the beginning Speech Communication, the seat personnel of extension set B1 can read customer data from call center A by the display that links to each other with the computer telephone integration server, for example customer name, buy information such as product, service log and enjoyment service range.
Reasonablely being, the calling between the gateway controls/and the process of foundation follows H.225/Q.931 agreement, and H.245 agreement is followed in negotiations process, and these agreements all belong to H.323 protocol stack.
When client (extension set A2) converses with seat personnel (extension set B1), telephone switching system in call center A and the B at first will be audio digital signals stream from the simulated voice digital coding of extension set A2 or B1, be sent to the Audio Processing Unit that links to each other separately through bus then and compress processing.Voice signal bag after the compression offers the gateway G that links to each other separately
AOr G
B, then at gateway G
AOr G
BCarry out packet transaction, that is, continuous voice signal bag is encapsulated as the packet switched data-IP frame that is suitable in IP network, transmitting.These IP frames are by calling and called side gateway G
AOr G
BSend to IP network, the router in the network is sent to recipient's gateway according to the address that frame head comprises with them.Recipient's gateway G
BOr G
ATo receive IP frame removal frame head and rearrangement processing (decapsulation) and be continuous voice signal bag, then the voice signal bag is delivered to the Audio Processing Unit that links to each other separately and made decompression, continuous number voice signal after the processing bus of flowing through offers telephone switching system, offers recipient B1 or A2 by it.
Reasonablely be that above-mentioned voice signal wraps in transmission between the gateway and follows RTP/RTCP module in the protocol stack H.323.Reasonablely be that the compressed format of above-mentioned audio digital signals stream is observed agreement G.711, G.723 agreement, G.728 agreement or an agreement in the agreement G.729.
Fig. 3 shows the preferred embodiment of circuit switching system, gateway and Audio Processing Unit in the above-mentioned call center.In Fig. 3, simulation extension set plate and switch processing unit constitute telephone switching system, and simulation extension set plate links to each other with the receipt of call control command with switch processing unit.The ip voice resource board, H.323 protocol process module and network interface unit realize the function of Audio Processing Unit of the present invention and gateway, wherein the ip voice resource board links to each other by bus with simulation extension set plate in the telephone switching system and links to each other with protocol process module H.323 with switch processing unit, and H.323 protocol process module links to each other with switch processing unit, network interface unit, computer telephone integration server (not shown) and customer data center (not shown).In above-mentioned call center, in order to improve compress speech and decompression disposal ability, can comprise polylith ip voice resource board, they all pass through bus interconnection, and compress speech and decompressing function are realized by digital signal processor (DSP).Reasonablely be, between simulation extension set plate and the ip voice resource board and the bus that is connected between the ip voice resource board meet the integrated agreement of many equipment suppliers (MVIP), H.100 agreement or an agreement in the agreement H.110.
Working method below in conjunction with call establishment and above-mentioned each unit of speech processes process prescription.
When calling both sides belonged to two different call centers, intracardiac switch processing unit was indicated a certain ip voice resource board to send to protocol process module H.323 and is simulated H.323 call event in the caller calls.H.323 the IP address (being gateway address) of network interface unit in the callee call center is provided according to the callee's gateway Number for access that provides protocol module, and send voip call to set up a TCP session to the gateway (being the H.323 protocol process module and the network interface unit of callee call center) of this IP address by network interface unit, wherein calling party's gateway sends CALL SETUP message H.225/Q.931, divides machine information and from the customer data at customer data center in session; The customer data that callee's gateway then transmits calling party's gateway is delivered to continuous computer-integrated telephone server and is checked the state of callee's extension set, if can connect, then play to callee's extension subscriber on the one hand and send the bell sound, return Q.931 ALERT message to calling party's gateway on the other hand; Q.931 calling party's gateway receives after the ALERT message to calling party's extension subscriber playing RBT.When callee's off-hook, callee's gateway sends Q.931CONNECT message to calling party's gateway, and Q.931 calling party's gateway receives after the CONNECT message promptly according to agreement H.245 the compression protocol and the packet oriented protocol of VoP are held consultation.After negotiation finishes, promptly set up the full duplex logical channel between calling and called side's gateway and create the RTP session, calling party's gateway stops to calling party's extension set playing RBT, and calling and called side can begin conversation.Meanwhile, the computer-integrated telephone server in the callee call center offers customer data the seat personnel of callee's extension set.
During conversing, as shown in Figure 3, one of them calling party (calling party or callee) at first is encoded to audio digital signals stream through simulation extension set plate to the voice signal that another calling party (callee or calling party) transmits, then under the control of switch processing unit, this audio digital signals bus of flowing through is sent to idle ip voice resource board and carries out compress speech and handle to obtain VoP.As mentioned above, voice signal wraps in the transmission in the IP and receives and need finish by gateway.H.323 protocol process module and network interface unit promptly realize gateway function, wherein VoP is encapsulated as processing that IP frame or IP frame be descapsulated into VoP and is finished according to the Real-time Transport Protocol module by protocol process module H.323, and the transmitting-receiving of IP frame in IP network finished by network interface unit.In encapsulation process, H.323 the protocol process module basis is split as a series of grouped data with the VoP that recipient's gateway negotiation parameter provides the ip voice resource board, encapsulation backs such as IP address with recipient's gateway constitute the IP frame again, preferably send into IP network through network interface unit.H.323 protocol process module comprises the correspondence table of extension number and the affiliated gateway ip address of extension set, so it can determine the IP address of recipient's gateway according to recipient's extension number.
Above-mentioned IP frame is sent to recipient's gateway through IP network, and H.323 the IP frame that this gateway network interface card receives is descapsulated into continuous voice signal bag and offers idle ip voice resource board in the protocol process module at it.Decapsulation process is opposite with above-mentioned encapsulation process, is not described in detail herein.The ip voice resource board is an audio digital signals stream with voice signal bag decompression, and offer simulation extension set plate through bus under the control of switch processing unit, simulation extension set plate then is changed to the audio digital signals circulation analog voice signal and offers corresponding extension set.
Claims (8)
1. distributed call center system that forms by some call centers, each described call center comprises telephone switching system, computer telephone integration server and customer data center, it is characterized in that, further comprises Audio Processing Unit and gateway, wherein,
Audio Processing Unit links to each other and links to each other with gateway through bus with telephone switching system, be used for telephone switching system is offered gateway behind the audio digital signals stream boil down to voice signal bag that bus provides, and the voice signal bag decompress(ion) that gateway provides is condensed to audio digital signals stream after bus offers telephone switching system; And
Gateway links to each other with Audio Processing Unit with telephone switching system, be used for setting up to call out being connected and controlling calling procedure between the calling and called side according to the calling control command of telephone switching system and address, calling and called side and the gateway of other call center, during calling out, the calling and called side gateway voice signal bag that provides of Audio Processing Unit separately is encapsulated as grouped data after the packet switching network is sent to recipient's gateway, and will be descapsulated into the Audio Processing Unit that offers behind the voice signal bag separately from the grouped data that the packet switching network receives.
2. distributed call center as claimed in claim 1 system, it is characterized in that, described gateway also links to each other with the customer data center with the computer telephone integration server, during call setup, calling party's gateway offers callee's computer telephone integration server with the customer data of calling party's customer data central store through callee's gateway, and offers the callee by this computer telephone integration server.
3. distributed call center as claimed in claim 1 system is characterized in that described called party address comprises callee's gateway Number for access and callee's extension number, and described gateway determines to set up callee's gateway of calling procedure according to described Number for access.
4. distributed call center as claimed in claim 3 system, it is characterized in that, described callee's gateway Number for access and callee's extension number are a number according to certain principle combinations, and described gateway is according to this Rule Extraction callee gateway Number for access and callee's extension number.
5. distributed call center as claimed in claim 1 system is characterized in that, described Audio Processing Unit comprises the digital signal processor of realizing compress speech and decompressing function.
6. as any described distributed call center system among the claim 1-5, it is characterized in that described bus meets the integrated agreement of many equipment suppliers (MVIP), H.100 agreement or an agreement in the agreement H.110.
7. distributed call center as claimed in claim 6 system, it is characterized in that, the described packet switching network is the Internet based on Internet Protocol (IP), call setup between the call center gateway and control and gateway to the encapsulation of grouped data and decapsulation based on an agreement in H.323 agreement, MGCP or the conversation initialized protocol.
8. distributed call center as claimed in claim 7 system is characterized in that, the compressed format of described audio digital signals stream is observed agreement G.711, G.723 agreement, G.728 agreement or an agreement in the agreement G.729.
Priority Applications (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CNB021372691A CN1172496C (en) | 2002-09-29 | 2002-09-29 | Distributed calling center based on Internet protocol |
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| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CNB021372691A CN1172496C (en) | 2002-09-29 | 2002-09-29 | Distributed calling center based on Internet protocol |
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| Publication Number | Publication Date |
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| CN1416249A true CN1416249A (en) | 2003-05-07 |
| CN1172496C CN1172496C (en) | 2004-10-20 |
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| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| CNB021372691A Expired - Lifetime CN1172496C (en) | 2002-09-29 | 2002-09-29 | Distributed calling center based on Internet protocol |
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Cited By (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN100388665C (en) * | 2004-07-22 | 2008-05-14 | 中兴通讯股份有限公司 | Telecommunication network call center system and its realization method |
| CN100428757C (en) * | 2003-09-28 | 2008-10-22 | 维音数码（上海）有限公司 | Telephone auto externally dialing method for client service and its intelligent analysis and management |
| US8443370B2 (en) | 2008-08-26 | 2013-05-14 | Microsoft Corporation | Method of assigning resources to fulfill a service request by a programming model abstraction layer at a data center based at least in part on a reference of the requested resource class indicative of an abstract amount of resources |
-
2002
- 2002-09-29 CN CNB021372691A patent/CN1172496C/en not_active Expired - Lifetime
Cited By (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN100428757C (en) * | 2003-09-28 | 2008-10-22 | 维音数码（上海）有限公司 | Telephone auto externally dialing method for client service and its intelligent analysis and management |
| CN100388665C (en) * | 2004-07-22 | 2008-05-14 | 中兴通讯股份有限公司 | Telecommunication network call center system and its realization method |
| US8443370B2 (en) | 2008-08-26 | 2013-05-14 | Microsoft Corporation | Method of assigning resources to fulfill a service request by a programming model abstraction layer at a data center based at least in part on a reference of the requested resource class indicative of an abstract amount of resources |
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| Publication number | Publication date |
|---|---|
| CN1172496C (en) | 2004-10-20 |
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Effective date of registration: 20170731 Address after: 200233 Shanghai City, Xuhui District Yishan Road No. 810 Shanghai Beiling building 14D Patentee after: Shanghai industry Austrian Communication System Co.,Ltd. Address before: 200233 Shanghai City, Zhongshan Road No. 1800 Mega global building 13 Building E1 room Patentee before: AODIJIAN COMM SYSTEM SHANGHAI |
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Granted publication date: 20041020 |