CN101030973A - Webpage calling system and method - Google Patents
Webpage calling system and method Download PDFInfo
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- CN101030973A CN101030973A CN 200610165353 CN200610165353A CN101030973A CN 101030973 A CN101030973 A CN 101030973A CN 200610165353 CN200610165353 CN 200610165353 CN 200610165353 A CN200610165353 A CN 200610165353A CN 101030973 A CN101030973 A CN 101030973A
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Abstract
The invention comprises: client side; internet; enterprises serving side and webpage calling server. Said client side and enterprises serving side are respectively connected to the webpage calling server through internet. Said client side can establish call connection with the enterprise serving side through webpage calling server. The invention supports multiple access modes and supports SIP protocol.
Description
Technical field
The present invention relates to network communications technology field, relate in particular to a kind of Webpage calling system of switching plane Network Based.
Background technology
According to the investigation of IDC to 1000 tame electronics business web sites, the actual access amount of website and the ratio of actual client's amount are 2582: 1 (per 2582 visitors have only one becomes the client), and the ratio in traditional commerce is 122: 1, cause so big gap reason a lot, wherein can not be linked up timely between client and the businessman is an important reasons.As the network marketing based on the website, volume of business is only absolute principle, and the client is in browsing shopping website, and on the webpage of icy and magnanimity information, client's query can not be answered usually at short notice timely, causes client's loss.
Therefore, press for a kind of system that can make the client just can directly carry out the freephone consulting from the webpage.
Summary of the invention
The present invention proposes in order to address the above problem.
The purpose of this invention is to provide a kind of Webpage calling system, comprise client, Internet and enterprises service end, it is characterized in that, this system also comprises the webpage call server, described client links to each other with the webpage call server by Internet respectively with the enterprises service end, and described client can directly be set up to call out with the enterprises service end and be connected by the webpage call server.
Webpage call server wherein comprises:
SIP signaling Exchange Service device is used to finish SIP standard agency, location, registration and redirected; The media services device is used for back-up system prompt tone, audio announcement, Interaction Voice Response and speech coding; The media relays device is used to support directional relay and load balancing; Gateway is selected service unit, be used to receive sip message, according to the selected intended gateway of gateway conditional parameter, transmit sip message to intended gateway, and gateway support lost efficacy or the automatic switchover of gateway route during full load; Authentication and charging service unit are used to realize authenticating user identification and call charging; Management and control device is used to provide the system monitoring management of Web form, realizes the monitoring and the warning of system mode.
Preferably, the load balancing that provides of described media relays device comprises load balancing, the load balancing of media relays and the load balancing of gateway flow of signaling exchange.
Preferably, described webpage call server also comprises data encryption device, is used to guarantee the fail safe of conversing.
Preferably, described webpage call server also comprises the call queue controller, is used for call queuing, and supports to call out automatically distribution policy.
Preferably, described webpage call server is a two-node cluster hot backup, to guarantee the fail safe of system.
Client is set up to call out by webpage call server and enterprises service end in the mode of browsing enterprise web site, activation webpage calling plug-in unit wherein and is connected.The webpage call server is arranged at the machine room of IDC.The enterprises service end receives from the webpage of client by IP phone, soft phone, IAD equipment or PBX system and calls out.
Another aspect of the present invention provides a kind of webpage method of calling, comprises the steps:
S1: the client lands enterprise web site, calls out plug-in unit by the webpage of enterprise web site and calls out the enterprises service end;
S2: the webpage call server that is positioned at the IDC machine room receives the webpage call information, and the webpage call request is transferred to the enterprises service end;
S3: the enterprises service end receives the customer call of transmitting from the webpage call server by network, sets up to be connected with client's direct communication.
Framework of the present invention is flexible, networking is tight, easy to use, can effectively utilize existing public Internet Internet resources, support the multiple network access way, support standard Session Initiation Protocol, the basic speech business is provided, and the increment multimedia service, be the communication system of a cover low cost, high reliability, high manageability, high flexibility.
Description of drawings
Fig. 1 is a system topology schematic diagram of the present invention;
Fig. 2 is that server redundancy/Hot Spare is disposed schematic diagram;
Fig. 3 is that two signal servers are disposed the load balancing schematic diagram;
Fig. 4 is netted media relays principle schematic;
Fig. 5 is for forcing the application schematic diagram of media relays;
Fig. 6 is a method flow schematic diagram of the present invention.
Embodiment
Following examples are used to illustrate the present invention, but are not used for limiting the scope of the invention.
Webpage calling system of the present invention mainly is made up of three parts, as shown in Figure 1, front end is at enterprise web site deploy Webcall assembly, machine room at IDC is disposed the webpage call server, the webpage call server is designed to two-node cluster hot backup, dispose IP phone, soft phone or IAD equipment in enterprises, simultaneously also can with enterprise original PBX system---call center system dock, realizes perfect fusion.Allow the client listen at any time from online business opportunity, operator also can be by the support of Webpage calling system, for the client provides a kind of brand-new value-added service.
For operator, this Webpage calling system has good system robustness, has powerful billing function, supports the billing model based on the high flexible of charging subregion; Support prepaid, back payment policies.Business that can carrier-supporting-carrier is carried out, the exploitation that Webpage calling system is used for the BOSS class provides standard data interface a: CSGate, CSGate comprises three kinds of interface modes: UDP Socket pattern, volatile data base pattern and WebService pattern, realize the integration with original business platform.
High reliability is the essential characteristic of carrier-class server system.The high reliability of Webpage calling system of the present invention can resolve into two aspects: the high reliability of data processing and the high reliability of application service.
1. the high reliability of data processing
Webpage calling system of the present invention adopts MySQL to support as database, Webpage calling system of the present invention is supported in hour employing database replication technology of system scale, when data scale is big, adopt data-base cluster (Cluster) technology to realize data redundancy and backup, ensure the reliability of data.According to the evaluation of the third-party institution to the MySQL reliability, when adopting the MySQL clustered deploy(ment), reliability can reach 99.999%, can reach the requirement of carrier-class systems reliability fully.
2. the high reliability of application service
Design and the realization of the server technical characteristic high reliability of supporting application service of Webpage calling system of the present invention by architecture.
● support SIP SRV mechanism, back-level server system redundancy/Hot Spare is disposed comprehensively
Externally provide service (directly also can provide service, but can lose a lot of outstanding characteristics) by " domain name " addressing system, and the domain name mapping service of SIP service-domain is provided by HR-EDNS by the IP address mode.System supports SIP SRV mechanism in a plurality of links such as service addressing, media relays decision-making.SIP SRV mechanism also can be tackled " envelope IP and end-blocking mouth " behavior of present domestic part ISP effectively except the effect that can reach system backup and load balancing.
The webpage call server of Webpage calling system of the present invention has perfect multiserver data sync and signalling exchange system, support that wherein the deployment of webpage call server redundancy/Hot Spare as shown in Figure 2 by reliability redundant and that Hot Spare is disposed the raising system.
● complete authentification of user mechanism
Server end is supported the WWW authentication and the Proxy authentication of Digest pattern.Registration link and per call link the terminal use all authenticate.The Digest authentication is based on MD5 algorithm (cryptographic algorithm that a kind of intensity is higher).These mechanism can effectively prevent to authenticate user's deception of link.
● the ability that the embedded reply of server software SIP attacks
Server adopts sip message analytics engine efficiently, and has realized multiple fault-tolerant measure in the message parse link, can tackle multiple illegal SIP and attack.
● adopt the VRRP agreement during back-level server networking, realize the server hardware redundancy
Webpage calling system of the present invention has configurable perfect signaling synchronization mechanism, adopts the VRRP agreement during back-level server networking, realizes the server hardware redundancy, and then the availability of safeguards system.
● the fault tolerant mechanism of PSTN gateway and media relays transponder
The normally distributed deployment of PSTN gateway and media relays transponder.Webpage calling system of the present invention is supported the fault-tolerant processing of PSTN gateway by the gateway selector, when a certain PSTN gateway lost efficacy (unreachable/fault), the gateway selector can automatically judge and mark its be " unavailable temporarily ", and the call request that will arrive PSTN is routed on the lower available PSTN Tandem Gateway of priority; Same mechanism, the media relays distributor is also supported the crash handling to the media relays transponder, the media relays transponder of mark inefficacy is " unavailable temporarily " automatically, only carries out media relays by " available transponder " and handles.
In systematic realizing program of the present invention, relate to many-sided load balancing, the technical characteristic of holding load equilibrium mainly contains aspect following three in this embodiment:
1. the load balancing of signaling exchange
In the system that SIP signal server more than is disposed, mainly be the mode that in the domain name mapping of system service, adopts " SRV " record and " the SIP signal server of directed ISP " or " the sip server wheel changes " by HR-EDNS, realize the load balancing of a plurality of SIP signal servers.Fig. 3 is the signal of a kind of situation of signaling exchange load balancing.
In many grouping large scale systems, an important function assembly being arranged in the Webpage calling system server platform of the present invention---SIP calls out distributor (SIPDispatcher).SIP calls out distributor and is responsible for the SIP signal server that the forwarded call request is handled to status of support, and supports SIP SRV addressing, calls out distributor mechanism by SIP, also can realize the load balancing of a plurality of SIP signaling exchange processing servers.
2. the load balancing of media relays
Media relays in the Webpage calling system of the present invention is actually to be finished by RRC, RRS-Dispatcher and three assembly synergistics of RRS-Mediator.RRC is responsible for to RRS-Dispatcher request transponder address, and revises the relevant signaling of SIPNAT according to return information; RRS-Dispatcher is according to the selected actual transponder that carries out media relays---the RRS-Mediator of certain rule, and return information is given RRC; The RRS-Mediator deployment that can distribute, and be controlled by RRS-Dispatcher, the actual media relays task of finishing.The load balancing of media relays mainly realizes by media relays distributor (RRS-Dispatcher).
3.PSTN the load balancing of gateway flow
Webpage calling system support of the present invention realizes and being connected of PSTN by the multichannel gateway device.On the routing decision and load allocating of multichannel gateway device, one independently functional unit---gateway selector (GWSelector) is played the part of crucial role.The system manager can derive priority and weight parameter according to factors such as the price of gateway route, quality, capacity, " priority " decision routing policy, " weight " decision load allocating.
Webpage calling system of the present invention is realized the load balancing of PSTN gateway flow by the gateway selector.
In the Webpage calling system database design, calling record list (CDR table) is the bottleneck of data scale expansion, and according to the data model of CDR table of 1,000,000 grades of records every day, the desirable userbase of single grouping Webpage calling system should be 100,000 grades.
When system scale is big, should adopt " based on the strategy of sign " to carry out user grouping, consider the performance of database performance, every group of user capacity is designed to 100,000 relatively rationally.In each packets inner, system is a complete SIP signaling switching system, and has grouping and independently keep accounts/charge processing system and corresponding database; But it is shared that the media relays server can be all grouping (the whole network).
Adopt and call out distributor processing rule of classification, but user grouping infinite extension in theory.Thus, the level and smooth expansion of Webpage calling system support terminal number of users from 1000 to 1,000 ten thousand.Aspect service quality (Qos) improved mechanism, Webpage calling system of the present invention taked following strategy to make great efforts to improve the service quality of VoIP from architecture Design, reduction Qos problem.
1. directed media transmission trunking
The present invention supports directed media relaying strategy, and cardinal principle is as follows:
● with the grouping of the transponder in the media relays service, each transponder all attach a mark, and which grouping mark it can be and provide medium to transmit to serve;
● the overall situation is safeguarded a directional relay Policy Table, and the list item of this table is made up of " called mark-caller mark-directional relay group number ";
● when the directed media transmission trunking enabled, each needed the calling of relaying all to go to mate the directional relay Policy Table, according to directional relay group number selected transponder in corresponding scope of coupling, finished media relays by this transponder.The directed media transmission trunking can be set called and the caller combination provides service by the medium transponder of determining, can improve the service quality of VoIP under certain conditions greatly.
2. " netted medium " relaying
" netted medium " are a kind of mechanism especially that is used to improve media relays service quality.Under present Internet present situation, between each ISP (ISP), sometimes interconnecting, (more typical situation is non-constant, smaller bandwidth between China Netcom and the China Telecom the Internet, block up and packet loss serious), when the user between the poor ISP of interconnecting carried out IP communication, service quality was difficult to accept usually.The basic principle of " netted media relays " by with in the media delivery route between the ISP " default route " be redirected and be " inner route " between the media relays server, " inner route " is fully controlled by the reasonable deployment of server, can be internal lan, also can be the VPN that has bandwidth to guarantee." bottleneck " link in the media delivery route has replaced uncontrollable factor by controllable factors, can improve the service quality of IP communication greatly.The operation principle of netted media relays as shown in Figure 4.
3. directed signaling process
As shown in Figure 3, dns server can be supported to carry out directed IP address resolution according to the source of DNS request, based on this, the webpage call server is by the deployment of many signal servers at different I SP, can realize the SIP client-requested from different I SP is carried out directional process by corresponding server.Directed signaling process is the optimization to the signaling process path.
4. force the media delivery relaying
The idealized model of media delivery is commonly considered as " peer-to-peer communications (Peer-to-Peer) " in the IP communication, but under particular network condition, by forcing the media delivery relaying, can realize the path optimization of media delivery, thereby help to improve relevant service quality, Fig. 5 example a kind of application scenarios of forcing the media delivery relaying.The setting and the processing of " forcing the media delivery relaying " carried out in support of the present invention to the user.
Aspect communication supervision support scheme, Webpage calling system of the present invention has calling record list, has wherein write down each session initiator's IP address, and this helps to grasp and estimate the positional information of associated call, thereby supports the supervision to IP communication better.In addition, media relays and directed media relaying are forced in the native system support, if certain class session is monitored, only need this class session is forced to be relayed on certain specific " media relays transponder ", " media relays transponder " can intercept and capture the Media Stream of session accordingly, record or be transmitted to specific application in real time, application-specific can be monitored session by media decodes.
The terminal equipment that Webpage calling system of the present invention relates to comprises IP phone (hardware, software), media gateway and the Tandem Gateway of complete series, satisfy the commercial needs in all kinds of mechanism/places, installing terminal equipment is simple, moves, increases, to change extension set simple and convenient.Phone can be distributed in network and can reach Anywhere, just can seamlessly insert corporate HQ, the telephone system of regional general headquarters and branch.
In the present embodiment, concrete terminal configuration is as follows:
The networking telephone (hardware, software)
The network phone provides a preferred plan of saving cost for client's communication requirement.This phone adopts dsp chip to constitute, and is widely used in the broadband IP network environment that meets ICP/IP protocol and carries out voice communication.As be used for the user (LAN, Cable, Modem and XDSL etc.) that the internal lan of enterprises and institutions and wide area network, telecommunications IP phone operator and broadband INTERNET insert.
Principal character:
1. support SIP RFC2543 ﹠amp; RFC3261
2.IEEE 802.3/802.3u 10Base T/100Base TX
3. main G.711 A-/U-law, G.723.1, G.729A/B, G.728, G.726 with the GSM610 audio coder ﹠ decoder (codec)
4.VAD/CNG can save bandwidth, RTP: RTP, RTCP: real time control protocol
5.DHCP: DHCP
6.PPPoE: the point-to-point dialing protocol of Ethernet
7.DNS: domain service protocol
Hardware specification:
| Size | 215mm×190mm×70mm |
| Weight | 1.5 |
| Display screen | |
| 2 row * 16 characters (numeral and literal) show | |
| The grid line interface | 2X 100Base-TLAN mouth |
| Power supply adaptor | 110V or 220V input, 9V DC/500mA output |
Soft phone
The Session Initiation Protocol client software of standard has abundant and advanced functional characteristic, and friendly interface, the Session Initiation Protocol relevant criterion that configuration is succinct, ease for use is strong, compatible up-to-date have good interoperability with other SIP products.
Principal character:
1. strict compatible SIP (RFC3261), RTP (RFC1889) and SDP consensus standards such as (RFC2327)
2. abundant Codec supports; Speex, iLBC, GSM, G.711U, G.711A, G.729A, G.723.1
3. support ICE, UPnP and STUN client NAT crossing technology
4. support multiple calling control; Caller identification, calling maintenance, calling transfer, calling switching etc.
5. support the multichannel audio mixing,, can realize MPTY by the multichannel audio mixing
6. the support Voice Activity Detection is static/dynamic dithering cache
7. support the band outer (RFC2388) that to consult automatically and be with interior DTMF
8. support audio frequency apparatus to select
9. support volume, microphone to regulate, support silent mode
10. support Presence, support good friend's grouping and state notifying
11. the instant message based on the good friend sends and receives
12. phone directory support
13. cost of the phone call and account balance show (needing the platform support) in real time
14. support Windows98/2000/XP/2003, support Windows Mobile
The Webcall assembly
WebCall is a standard Session Initiation Protocol customer end A ctiveX control, can be deployed in the enterprise web site application such as being used for " click-to-dial ", " Web800 ".
Principal character:
The Codec:G.711U of Zhi Chiing, G.711A, GSM, G.729A, iLBC
2. the DTMF of Zhi Chiing: in band outer (RFC2388) and the band, SDP negotiation is passed through in support
3. control size: 140KB-160KB
4. support automatic MIC to detect (can effectively stop the infecctive call request that does not possess the conversation condition)
5. support volume to regulate
6. anti-harassing and wrecking processing feature (cooperating native system can effectively stop malicious call)
Media gateway (IAD equipment) (single port, Duo Kou)
Connect data network and traditional analog equipment (as: analog telephone, facsimile machine etc.), finish the mutual commentaries on classics of the packet and the signal of telecommunication.
Principal character:
1. support SIP (RFC3261), TCP/UDP/IP, RTP/RTCP, HTTP, ICMPARP/RARP, DNS, DHCP, NTP, PPPoE, STUN, agreements such as TFTP;
2. use powerful Digital Signal Processing (DSP) chip to guarantee that fabulous audio quality is arranged, adopt advanced shake control and hide the message dropping technology;
3. support various speech codings, comprise G.711 (a-law/u-law), G.723.1 (5.3K/6.3K), G.726 (40K/32K/24K/16K), G.729A/B, and iLBC;
4. support route/NAT, Dynamic Host Configuration Protocol server Gateway, DMZ and port mapping function;
5. support bridge mode;
6. support caller identification/restriction, call out and keep, Call Waiting/sudden strain of a muscle is disconnected, calls out switching (change Call Forwarding or calling back), calling transfer (unconditional branch, call forward on busy or transfer on no reply), dial plan, three kinds of DTMF type (In-audio are inside and outside band supported in tripartite talks or the like, RFC2833, SIP INFO)
7. support call waits for that Caller ID shows; Support the polarity inversion charging way;
8. support T.38 and the Pass-Through fax mode; Support DNS SRV to search;
9. support quiet inhibition and Voice Activity Detection, comfort noise produces (CNG) echo and suppresses (G.168) and automatic gain (AGC);
10. support various encryptions and Valuation Standard (BASIC, DIGEST, MD5 and MD5-sess algorithm);
11. support the ISO network configuration second layer and the 3rd layer of QoS (802.1Q VLAN, 802.1p, DiffServ, ToS);
12. support that fire compartment wall/NAT automatic safe penetrates, the user need not change the fire compartment wall setting;
13. support the management of long-range fully-automatic equipment, effectively realize user side " zero " setting, plug and play, private network is transparent penetrates and software automatic updating;
14. can change system configuration easily and download the configuration file of encryption by TFTP/HTTP by plain old telephone voice suggestion, Web interface or relevant intelligent network management system;
15. microminiature lightly designs (size is big as little wallet), is easy to carry;
16. succinct, light and handy global general-use power supply adaptor;
Hardware specification:
| Functional characteristic | Media gateway |
| Ethernet interface | 2RJ 45(WAN/LAN) |
| DHCP/NAT route/bridge joint | Support |
| FXS analog station interface | 1/4/8/16/32 |
| Escaping exit | Support |
| Remote configuration | TFTP/HTTP |
Webpage calling system of the present invention and method are called out soft-switch platform based on webpage, the core of this soft-switch platform except the function of registration with general soft-switch platform, call setup, record maintenance, routing management etc., also have Media Stream and signaling flow according to different network conditions, different needs walk that different paths, NAT instead penetrate, the equilibrium of register command current load, multiple spot heat backup at different sites, Media Stream load balancing, the equilibrium of network routing overhead, IVR, ACD assembly or the like.Based on functions such as the unified management interface of Web, safety management and outstanding network management and early warning, can adapt to domestic and international various complex web border at present.What the operational mode of soft switchcall server of the present invention adopted is the mode of management and flexible operation combination, when originally power system capacity was little, each module was moved on the same system platform, along with the expansion of business, system module can disperse to dispose, and has guaranteed the good autgmentability of system.
The evaluation of VoIP system technical performance does not at present also have ripe standard and system, still uses the evaluation index in the conventional telecommunications technology generally speaking, comprises two aspects:
1. the per second call treatment is counted CPS (Calls Per Second)
2. BHCA disposal ability BHCA (Busy Hour Call Attempts): refer to the number of calls that to handle in the busiest one hour.
It has been generally acknowledged that the emphasis of VoIP server system performance evaluation is the disposal ability or the capacity of following three aspects of assessment:
1. signaling exchanges disposal ability
● the user registration process ability
● call setup and dismounting disposal ability
2. media relays disposal ability
● media relays distribution processor ability
● media relays is transmitted disposal ability
3. media server/value-added service server session concurrent processing ability
Because the difference of VoIP server system deployment conditions (hardware, network), the performance performance of above-mentioned three aspects often is not a balance, just is subjected to the restriction of bandwidth resources such as the media relays disposal ability; The performance performance of authenticating transactions and database server resources such as (relevant) memories is relevant or the like.Generally speaking, it is very difficult the performance of VoIP server being analyzed accurately and estimated.
By calling out instruments such as maker system is carried out simulation test, can obtain understanding substantially to the conversation affair disposal ability of server system, on this basis, in conjunction with concrete business model and conditions such as hardware, network design relation factor being analysed in depth, is rational method and the approach that obtains server system performance index understanding again.
The main relation factor that influences systematic function of the present invention comprises three aspects: the deployment of the quantity of server hardware and configuration, network organizing pattern and band width configuration and each functional unit of webpage call server.
Each functional unit of Webpage calling system of the present invention has multiple combination under different deployment conditions, its performance evaluation is quite complicated.Usually need make a concrete analysis of according to the deployment scenario of project reality.
Here be example only, the performance level of HR-CSS is described briefly with the simple The performance test results of the actual operation system of entry level configuration.
| Project | |
| Hardware server | |
| 2 Dell 2850 are configured to: CPU: two Xeon 3.0GHz, MEM:2GB, HD:SCSI 73GB | |
| Network | Internet (outer net) 100MB; Intranet: the interconnection of 2 station servers, 100MB exchange |
| Server OS | Redhat Enterprise Linux AS4 U2 |
| The webpage call server | The dual system Hot Spare is disposed; Database master-master duplicates; Dispose webpage call server basic functional components/do not dispose expanded function assembly |
According to our test data, under 2 servers that last table exemplifies, dual system deployment conditions, if the network bandwidth is not a system bottleneck, the chances are for the performance index of system:
● the location registration process ability is greater than 10000 register requirement of per second;
● peak value (5 second cycle) call handling capacity is a per second 450-500 call request;
● the BHCA value is approximately 72000;
● according to 60 seconds session model of each session persistence, system can support about 1200 concurrent sessions; If according to 120 seconds session model of each session persistence, the concurrent session number that system can support is greater than 2400.
More than be performance evaluation, can probably understand and estimate the performance level of Webpage calling system of the present invention from related conclusions the system of 2 server configurations of an entry level.
In addition, the webpage method of calling based on above-mentioned calling network system provided by the invention, its method flow is as shown in Figure 6.The step of this method is as follows:
S1: the client lands enterprise web site, calls out plug-in unit by the webpage of enterprise web site and calls out the enterprises service end;
S2: the webpage call server that is positioned at the IDC machine room receives the webpage call information, and the webpage call request is transferred to the enterprises service end;
S3: the enterprises service end receives the customer call of transmitting from the webpage call server by network, sets up to be connected with client's direct communication.
Use server disposition Webpage calling system of the present invention more, better configuration, can provide higher performance index usually.In addition, the deployment scheme of more service device, more network insertion conditions helps configuration combination, the raising system service quality control level of optimizational function assembly, thus the performance of the overall performance of elevator system.Compared with similar products, under same hardware and network configuration, the performance of this webpage calling system performance is more outstanding.
Though the present invention specifically illustrates and illustrates in conjunction with a preferred embodiment; but the personnel that are familiar with this technical field are appreciated that; wherein no matter still can make various changes in detail in form, this does not deviate from spirit of the present invention and scope of patent protection.
Claims (10)
1, a kind of Webpage calling system, comprise client, Internet and enterprises service end, it is characterized in that, this system also comprises the webpage call server, described client links to each other with the webpage call server by Internet respectively with the enterprises service end, and described client can directly be set up to call out with the enterprises service end and be connected by the webpage call server.
2, Webpage calling system as claimed in claim 1 is characterized in that described webpage call server comprises:
SIP signaling Exchange Service device is used to finish SIP standard agency, location, registration and redirected;
The media services device is used for back-up system prompt tone, audio announcement, Interaction Voice Response and speech coding;
The media relays device is used to support directional relay and load balancing;
Gateway is selected service unit, be used to receive sip message, according to the selected intended gateway of gateway conditional parameter, transmit sip message to intended gateway, and gateway support lost efficacy or the automatic switchover of gateway route during full load;
Authentication and charging service unit are used to realize authenticating user identification and call charging;
Management and control device is used to provide the system monitoring management of Web form, realizes the monitoring and the warning of system mode.
3, Webpage calling system as claimed in claim 2 is characterized in that the load balancing that described media relays device provides comprises load balancing and the load balancing of media relays and the load balancing of gateway flow that signaling exchanges.
4, Webpage calling system as claimed in claim 2 is characterized in that described webpage call server also comprises data encryption device, is used to guarantee the fail safe of conversing.
5, Webpage calling system as claimed in claim 2 is characterized in that described webpage call server also comprises the call queue controller, is used for call queuing, and supports to call out automatically distribution policy.
6,, it is characterized in that described webpage call server is a two-node cluster hot backup, to guarantee the fail safe of system as the described Webpage calling system of one of claim 1 to 5.
7, Webpage calling system as claimed in claim 1 is characterized in that described client is connected by webpage call server and enterprises service end foundation calling in the mode of browsing enterprise web site, activation webpage calling plug-in unit wherein.
8, Webpage calling system as claimed in claim 1 is characterized in that described webpage call server is arranged at the machine room of IDC.
9, Webpage calling system as claimed in claim 1 is characterized in that described enterprises service end is by IP phone, soft phone, IAD equipment or the reception of the PBX system webpage calling from client.
10, a kind of webpage method of calling is characterized in that this method comprises the steps:
S1: the client lands enterprise web site, calls out plug-in unit by the webpage of enterprise web site and calls out the enterprises service end;
S2: the webpage call server that is positioned at the IDC machine room receives the webpage call information, and the webpage call request is transferred to the enterprises service end;
S3: the enterprises service end receives the customer call of transmitting from the webpage call server by network, sets up to be connected with client's direct communication.
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| CN 200610165353 CN101030973A (en) | 2006-12-18 | 2006-12-18 | Webpage calling system and method |
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|---|---|---|---|
| CN 200610165353 CN101030973A (en) | 2006-12-18 | 2006-12-18 | Webpage calling system and method |
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Cited By (11)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN101959159A (en) * | 2009-07-17 | 2011-01-26 | 浙江省公众信息产业有限公司 | System and method for performing cross-platform on-site disposal by using mobile terminal |
| CN102130971A (en) * | 2011-04-27 | 2011-07-20 | 苏州阔地网络科技有限公司 | Method and system for realizing peer-to-peer (P2P) communication on webpage |
| CN102148775A (en) * | 2010-02-05 | 2011-08-10 | 陈剑峰 | Webpage call service gateway, call service system and method |
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| CN102130971A (en) * | 2011-04-27 | 2011-07-20 | 苏州阔地网络科技有限公司 | Method and system for realizing peer-to-peer (P2P) communication on webpage |
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| CN107295004A (en) * | 2017-07-21 | 2017-10-24 | 881飞号通讯有限公司 | A kind of network voice communication method realized based on web page communication plug-in unit and system |
| CN107295004B (en) * | 2017-07-21 | 2020-09-25 | 881飞号通讯有限公司 | Network voice communication method and system realized based on webpage communication plug-in |
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