CN101741829A - Domain control server for integrated control of sound images - Google Patents
Domain control server for integrated control of sound images Download PDFInfo
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- CN101741829A CN101741829A CN200910044047A CN200910044047A CN101741829A CN 101741829 A CN101741829 A CN 101741829A CN 200910044047 A CN200910044047 A CN 200910044047A CN 200910044047 A CN200910044047 A CN 200910044047A CN 101741829 A CN101741829 A CN 101741829A
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Abstract
The invention relates to a domain control server for integrated control of sound image information, which comprises a service interface layer based on the standard RFC3261 SIP stack, an authentication service module, an SIP outbound proxy service module, an SIP registration service module, an SIP redirection service module, a media routing control logic module, an SIP terminal emulation logic module, a domain configuration management module and an non-standard SIP sound image system interface. The domain control server provides access interfaces meeting the SIP standards for various actual or abstract SIP terminals in a platform, simultaneously provides the functions of equipment management, geographic position locating, routing information service and the like for the entire network, provides a strict security mechanism and a strict authentication mechanism, and realizes centralized management and communication with standard digitized terminals.
Description
Technical field
The present invention relates to a kind of audio and video information transmission control equipment, particularly a kind of domain control server that is used for the audio and video information Comprehensive Control.
Background technology
Perfect day by day communication network provides carrier for the transmission of remote image and sound, the maturation of the communication technology and development have solved the problems such as distance, intercommunication and bandwidth bottleneck in the audio and video information transmission course, make the real-time remote transmission of image and acoustic information become possibility.The transmission of audio and video information is by the initial transmission developing into of broadcast type signal current twocouese, networking, interactive mode, multi-functional, comprehensive acoustic image transmission system.Based on the acoustic image transmission system of synthesization, we can carry out the acoustic image business of numerous forms such as video conference, video monitoring, visible dispatching, visual commander, remote teaching, for our work and life brings more convenient and efficient.Though comparatively extensive use is used in the transmission of video image, yet restricted by the historical conditions of network technical development, the real-time Transmission of video/audio on network lacks ripe and complete condition all the time, thereby cause the realization mechanism confusion of these systems, not having unified standard can comply with, various informative and incompatible.So, can't be interconnected between the different system, image can't intercommunication, and the image between the departments at different levels can't call mutually, and equipment can't be controlled mutually, finally forms " isolated island " of audio and video information one by one.Therefore, it is current that what press for most is that unified transmission standard and interconnected mechanism are set up in the networking of video image and audio frequency transmission, and effectively solve the transmission control of automation and the problem of network management, network environment that is similar to the acoustic image resource interconnectivity and sharing of Internet of real realization accomplishes to obtain in real time all kinds of audio and video informations in the network whenever and wherever possible, " unblocked ".
Summary of the invention
The purpose of this invention is to provide a kind of domain control server that is used for the audio and video information Comprehensive Control, it is with the defined outgoing call acting server of standard Session Initiation Protocol (Outbound Proxy), registrar (Registrar Server), Redirect Server (Redirection Server), the access interface that meets the SIP standard is provided for all kinds of reality in the platform or abstract sip terminal, simultaneously, software platform also provides the equipment control to whole network, the location, geographical position, functions such as routing iinformation service, and strict security mechanisms and Authentication mechanism is provided.Domain control server comprises the business interface layer 1 based on standard RFC3261SIP protocol stack, authentication service module 2, SIP outgoing call proxy service module 3, SIP registration service module 4, SIP redirect services module 5, medium route control logic module 6, sip terminal analog logic module 7, territory Configuration Manager 8 and non-standard SIP acoustic image system interface 9, wherein standard RFC3261 Session Initiation Protocol stack receives the standard RFC3261 Session Initiation Protocol data of automatic network, business interface layer 1 is finished the analysis of data and is converted to described authentication service module 2, SIP outgoing call proxy service module 3, SIP registration service module 4, SIP redirect services module 5, medium route control logic module 6, sip terminal analog logic module 7, the concrete call instruction of non-standard SIP acoustic image system interface 9, territory Configuration Manager 8 is finished SIP redirect services module 5, medium route control logic module 6, the support of sip terminal analog logic module 7, SIP outgoing call proxy service module 3 is carried out authentication based on authentication service module 2, and sip terminal analog logic module 7 is controlled each terminal according to the interface that non-standard SIP acoustic image system interface 9 provides.
Described business interface layer 1 also will receive the control signal from figure control interface.
RFC3261 Session Initiation Protocol stack provides the basic communication control ability of standard RFC3261 Session Initiation Protocol, and business interface layer 1 provides the user capture that domain control server is managed, disposes interface, the interface of cooperating between a plurality of territories networking time domains and the territory; Non-standard SIP acoustic image system interface 9 provides the specific access control ability to all kinds of non-standard SIP acoustic image application systems with the form of open plug-in unit; Territory Configuration Manager 8 is used to dispose the actual acoustic image application system of all management of domain control server, and the routing relation in this territory between these terminals and the equipment; Sip terminal analog logic module 7 is used for the non-standard sip terminal managed by abstract this territory of form of software; When medium route control logic module 6 needs this calling is carried out the control of medium route by territory Configuration Manager 8 in the terminal call process; SIP redirect services module 5 is used to cooperate medium route control logic, and the terminal of finishing in the medium routing procedure is redirected; Location registration information when SIP registration service module 4 starts according to any sip terminal is carried out target localization in the SIP calling procedure; SIP outgoing call proxy service module 3 is outgoing call acting servers of sip terminal regulation, it is when the call request that receives sip terminal, find that by the location of " SIP registration service " target terminal, the routing relation that is disposed by the territory Configuration Manager determine routing policy, make up and set up the feasible path of calling out between the different terminals;
Its control method may further comprise the steps:
The first step, the terminal equipment initialization;
In second step, finish the registration of terminal equipment by SIP registration service module;
The 3rd goes on foot, and is finished according to the RFC3261 agreement by SIP outgoing call proxy service module and calls out;
In the 4th step, judge the connection type between terminal equipment;
The 5th step; According to the digital terminal between terminal equipment to digital terminal or pseudo-terminal to pseudo-terminal or digital terminal to pseudo-terminal or pseudo-terminal are interrupted the calling of classifying of four kinds of different connection types to numeral.
In the 5th step, digital terminal to calling (S-S calling) performing step of digital terminal is: A, calling are initiated digital terminal SUA1 and are sent the outgoing call proxy service module of asking domain control server to the SIP of target number terminal SUA2;
B, outgoing call proxy service module are carried out authentication by authentication module to digital terminal SUA1, if authentication is failed then refused call request; If C authentication success is by the position of SIP registration service localizing objects digital terminal SUA2; D, location positioning success back are routed to SUA2 by SIP registration service module with SUA1; 5, SUA2 carries out media negotiation by RFC3261 agreement with SUA1, sets up multimedia transmission channel based on the standard multimedia host-host protocol until end of calling.
In the 5th step, pseudo-terminal to calling (A-A calling) performing step of digital terminal is: A, calling initiation pseudo-terminal SAUA1 send the SIP request to target simulation terminal SAUA2, after the authentication success, and the position of location pseudo-terminal SAUA2; After successfully locate the position of B, SAUA2, find that by the territory Configuration Manager medium route control logic module and sip terminal analog logic module make this transmission channel be in ready state between the two based on the analogue transmission passage; The request that C, SAUA1 send is routed to SAUA2, carry out media negotiation and success by RFC3261 agreement after, medium route control logic module and sip terminal analog logic module physics are set up between the two connecting channel until end of calling.
In the 5th step, digital terminal to the calling between the pseudo-terminal (S-A calling) performing step is: A, calling initiation digital terminal SUA1 send the SIP request to target simulation terminal SAUA1, after the authentication success, and the position of location pseudo-terminal SAUA1; After successfully locate the position of B, SAUA1, find to exist simulation and Digital Transmission passage between the two by the territory Configuration Manager, medium route control logic module and sip terminal analog logic module make this transmission channel be in ready state; The request that C, SUA1 send is routed to SAUA1, carry out media negotiation and success by RFC3261 agreement after, medium route control logic module and sip terminal analog logic module physics are set up between the two connecting channel until end of calling.
In the 5th step, pseudo-terminal to the calling between the digital terminal (A-S calling) performing step is: A, calling initiation pseudo-terminal SAUA1 send the SIP request to target number terminal SUA1, after the authentication success, and the position of location number word terminal SUA1; After successfully locate the position of B, SUA1, find to exist simulation and Digital Transmission passage between the two by the territory Configuration Manager, medium route control logic module and sip terminal analog logic module make this transmission channel be in ready state; The request that C, SAUA1 send is routed to SUA1, carry out media negotiation and success by RFC3261 agreement after, medium route control logic module and sip terminal analog logic module physics are set up between the two connecting channel until end of calling.
Using technical scheme of the present invention can reach existing and newly-built all kinds of acoustic image application systems interconnect, the acoustic image resource is shared fully, transmission channel is fully multiplexing, the audio and video information visit is easy in real time, the acoustic image transmission security reliable, control and management automation and intellectuality, platform are followed unified standard, are easy to later stage expansion and integrated purpose.Particularly, because the non-standard sip terminal of being managed in this territory of sip terminal analog logic module abstracts, domain control server can will originally not support the simulation audio-visual equipment of Digital Control ability to adopt abstract technology, utilizing computer software is the sip terminal of a symbol RFC3261 standard with such simulated sound as device abstract, so can realize and the standard digital terminal between unified management and communication.
Description of drawings
Fig. 1 is a principle of the invention block diagram;
Fig. 2 is the terminal call overview flow chart;
Fig. 3 is calling (S-S calling) flow chart between the digital terminal;
Fig. 4 is calling (A-A calling) flow chart between the pseudo-terminal;
Fig. 5 is that digital terminal is to the calling between the pseudo-terminal (S-A calling) flow chart;
Fig. 6 is that pseudo-terminal is to the calling between the digital terminal (A-S calling) flow chart;
Fig. 7 is the instance model schematic diagram that call out in a territory.
Wherein: 1, business interface layer 2, authentication service module 3, SIP outgoing call proxy service module 4, SIP registration service module 5, SIP redirect services module 6, medium route control logic module 7, sip terminal analog logic module 8, territory Configuration Manager 9, non-standard SIP acoustic image system interface
Embodiment
In the network interconnection environment of a complexity, we can be divided into this network some territories (domain) usually.The basic conception in territory is: by the network element that locus, compass of competency or function logic is identical and close, from logic they being organized together, form a territory; By certain territory administrative mechanism they are carried out logic manage, form a relatively independent network component units.
In our audio and video information Comprehensive Control platform network system, the set that a department, mechanism or acoustic image application system are managed, related audio/video devices, transmission channel and central apparatus formed is represented in territory usually.For example, command centre of Department of Public Security of Shanxi Province has managed the video conferencing system of two cover IP based networks, an one cover simulation video monitoring system and a cover are based on the image delivering system of matrix, so, if we these systems from being organized as the network element of an audio and video information Comprehensive Control platform in logic, a Here it is territory.Be distributed on the structure principle in territory, can constitute incidence relation between any two territories and the territory.But too Fu Za territory incidence relation can cause managing confusion, is unfavorable for Network Management.Therefore the architecture in territory is organized into level shape structure (tree), and this can not only satisfy managerial needs, also meets organization's structure of public security system reality simultaneously.
Sip terminal refers to abide by video, the voice frequency terminal equipment of RFC3261 consensus standard design, for example video conference terminal, video server equipment, character matrix, video telephone etc.Sip terminal should be stipulated by the RFC3261 agreement, the action treatment mechanism of SIP INVITE, the ACK of support standard, OPTIONS, REGISTER, BYE and CANCEL request and agreement, support SDP media description and negotiation, for this platform the ability of special support, the sip terminal that needs to use these special abilities should possess the support that SIP expands, but this is optional rather than necessary.Actual this platform of media delivery mechanism is not arranged by force, but from the consideration of interconnecting property, media delivery should be selected following agreement as far as possible:
RTP/RTCP: be used for real-time media transmission (as application such as video conference, monitoring in real time)
RTSP: be used for transmission of multimedia streams and location (application such as checking) as video request program, video recording
Analog image virtual transmission agreement (AGVP): the transmission of analog image and digital network are irrelevant, and this agreement has only defined the exchange requirement of analog image transmission, and concrete transmission is finished by control analog image switching equipment.As Extended Protocol, the standard sip terminal is optional to the support of this agreement.
The data format of medium is the MPEG-4/H.264 of choice criteria as far as possible also.
Owing to must consider full compatibility to different acoustic image application systems, require transmission control protocol, one side is the current system environments of compatibility effectively, also has good autgmentability on the other hand, and this is the whole platform construction technical problem of core the most.(Session Initialization Protocol, conversation initialized protocol RFC3261) are a kind of agreement that can satisfy this application need to SIP.Current, Session Initiation Protocol has become the de facto standard of realizing multimedia transmission control based on the IPv4/IPv6 network, and products such as a large amount of IP phone, video conference, video server are all supported this agreement; A new generation the 3G mobile communications network, also with Session Initiation Protocol as the core technology that multimedia service is provided in the IP Multimedia System; Simultaneously, Session Initiation Protocol also is the accepted standard communication protocol of recommending in Ministry of Public Security's " requirement of supervision of the cities alarm network system current techique ".This agreement well-known architecture and outstanding extended capability make its standard that can be consistent in field of multimedia communication, can carry out special case processing and expansion/expansion at different situations again.
The conversion of non-standard → standard Session Initiation Protocol and level of abstraction are realized: in a large number based on the acoustic image application systems of digitlization transmission, its transmission control protocol is not based on the standard Session Initiation Protocol for existing.Based on early stage multimedia communication international standard agreement H.323 these systems have, and more have in a large number based on the self-defined host-host protocol of manufacturer.Solve the compatibility of nonstandard protocol application system to standard Session Initiation Protocol audio and video information Comprehensive Control platform, we can introduce an intermediate layer (mainly realizing with software or dedicated gateway equipment).This intermediate layer is abstracted into the standard sip terminal with each Terminal Type of these application systems, provides standardized transmission control interface to platform; The inner conversion that realizes controlling to the specific transmission control mode relevant with system by the SIP transmission.This mechanism can make the user of platform need not to be concerned about the specific implementation form of particular system, without exception with the unified sip terminal that is considered as standard of the audio and video information terminal in the platform (video source, audio-source, video object, audio frequency target etc.), the control that conducts interviews of the Session Initiation Protocol by standard.
The conversion of simulation and other form acoustic image host-host protocol → standard Session Initiation Protocol and level of abstraction are realized: the realization principle of this layer is with " conversion of non-standard → standard Session Initiation Protocol and level of abstraction are realized ", by introducing an intermediate layer, with the audio and video information terminal of simulation abstract be the sip terminal of standard, and then can adopt the control that conducts interviews of the Session Initiation Protocol of standard.Be that with the application system difference of digitlization transmission means simulated sound can't directly be transmitted to digital transport network as the video/audio information of application system, and the SIP application system is in principle all at digitlization video/audio media format.
Addressed this problem dual mode, and dual mode can be used simultaneously:
The first, partly to expand/redefine by media description Session Initiation Protocol, as the transmitting SIP terminal, its media delivery form is specific, special-purpose form to agreement for the simulated sound that takes out, and does not support the compatibility with the digitlization transmission.Like this, the transmission of simulated sound picture only is supported in simulated sound as carrying out in the transmission network.In fact, this application is very general.For example, based on the supervision of the cities video of analogue transmission, our typical demand is that the video image with the different location switches on certain monitor by the analogue transmission network and checks.
The second, to whole platform definition audio and video information routing architecture, with the route access point of " analog to digital " conversion equipment (as Video Codec, video server etc.) as analog to digital, and with this conversion equipment same abstract be sip terminal.Digitized sip terminal such as need are directly visited pseudo-terminal, by the route converting system of platform, access process can be divided into two big links: the first, and the exchange of analog video network portion makes the audio and video information of pseudo-terminal accurately exchange to conversion equipment; The second, control transformation equipment is based on the digital audio and video information of SIP host-host protocol after digitalized S IP terminal provides conversion.
Extremely shown in Figure 7 as Fig. 1, domain control server comprises the business interface layer 1 based on standard RFC3261 Session Initiation Protocol stack, authentication service module 2, SIP outgoing call proxy service module 3, SIP registration service module 4, SIP redirect services module 5, medium route control logic module 6, sip terminal analog logic module 7, territory Configuration Manager 8 and non-standard SIP acoustic image system interface 9, wherein standard RFC3261 Session Initiation Protocol stack receives the standard RFC3261 Session Initiation Protocol data of automatic network, business interface layer 1 is finished the analysis of data and is converted to described authentication service module 2, SIP outgoing call proxy service module 3, SIP registration service module 4, SIP redirect services module 5, medium route control logic module 6, sip terminal analog logic module 7, the concrete call instruction of non-standard SIP acoustic image system interface 9, territory Configuration Manager 8 is finished SIP redirect services module 5, medium route control logic module 6, the support of sip terminal analog logic module 7, SIP outgoing call proxy service module 3 is carried out authentication based on authentication service module 2, and sip terminal analog logic module 7 is controlled each terminal according to the interface that non-standard SIP acoustic image system interface 9 provides.
Described business interface layer 1 also will receive the control signal from figure control interface.
Its control method may further comprise the steps:
The first step, the terminal equipment initialization;
In second step, finish the registration of terminal equipment by SIP registration service module;
The 3rd goes on foot, and is finished according to the RFC3261 agreement by SIP outgoing call proxy service module and calls out;
In the 4th step, judge the connection type between terminal equipment;
The 5th step; According to the digital terminal between terminal equipment to digital terminal or pseudo-terminal to pseudo-terminal or digital terminal to pseudo-terminal or pseudo-terminal are interrupted the calling of classifying of four kinds of different connection types to numeral.
Digital terminal to calling (S-S calling) performing step of digital terminal is: A, calling are initiated digital terminal SUA1 and are sent the outgoing call proxy service module of asking domain control server to the SIP of target number terminal SUA2; B, outgoing call proxy service module are carried out authentication by authentication module to digital terminal SUA1, if authentication is failed then refused call request; If C authentication success is by the position of SIP registration service localizing objects digital terminal SUA2; D, location positioning success back are routed to SUA2 by SIP registration service module with SUA1; 5, SUA2 carries out media negotiation by RFC3261 agreement with SUA1, sets up multimedia transmission channel based on the standard multimedia host-host protocol until end of calling.
Pseudo-terminal to calling (A-A calling) performing step of digital terminal is: A, calling initiation pseudo-terminal SAUA1 send the SIP request to target simulation terminal SAUA2, after the authentication success, and the position of location pseudo-terminal SAUA2; After successfully locate the position of B, SAUA2, find that by the territory Configuration Manager medium route control logic module and sip terminal analog logic module make this transmission channel be in ready state between the two based on the analogue transmission passage; The request that C, SAUA1 send is routed to SAUA2, carry out media negotiation and success by RFC3261 agreement after, medium route control logic module and sip terminal analog logic module physics are set up between the two connecting channel until end of calling.
Digital terminal to the calling between the pseudo-terminal (S-A calling) performing step is: A, calling initiation digital terminal SUA1 send the SIP request to target simulation terminal SAUA1, after the authentication success, and the position of location pseudo-terminal SAUA1; After successfully locate the position of B, SAUA1, find to exist simulation and Digital Transmission passage between the two by the territory Configuration Manager, medium route control logic module and sip terminal analog logic module make this transmission channel be in ready state; The request that C, SUA1 send is routed to SAUA1, carry out media negotiation and success by RFC3261 agreement after, medium route control logic module and sip terminal analog logic module physics are set up between the two connecting channel until end of calling.
Pseudo-terminal to the calling between the digital terminal (A-S calling) performing step is: A, calling initiation pseudo-terminal SAUA1 send the SIP request to target number terminal SUA1, after the authentication success, and the position of location number word terminal SUA1; After successfully locate the position of B, SUA1, find to exist simulation and Digital Transmission passage between the two by the territory Configuration Manager, medium route control logic module and sip terminal analog logic module make this transmission channel be in ready state; The request that C, SAUA1 send is routed to SUA1, carry out media negotiation and success by RFC3261 agreement after, medium route control logic module and sip terminal analog logic module physics are set up between the two connecting channel until end of calling.
RFC3261 Session Initiation Protocol stack provides the basic communication control ability of standard RFC3261 Session Initiation Protocol, and business interface layer 1 provides the user capture that domain control server is managed, disposes interface, the interface of cooperating between a plurality of territories networking time domains and the territory; Non-standard SIP acoustic image system interface 9 provides the specific access control ability to all kinds of non-standard SIP acoustic image application systems with the form of open plug-in unit; Territory Configuration Manager 8 is used to dispose the actual acoustic image application system of all management of domain control server, and the routing relation in this territory between these terminals and the equipment; Sip terminal analog logic module 7 is used for the non-standard sip terminal managed by abstract this territory of form of software; When medium route control logic module 6 needs this calling is carried out the control of medium route by territory Configuration Manager 8 in the terminal call process; SIP redirect services module 5 is used to cooperate medium route control logic, and the terminal of finishing in the medium routing procedure is redirected; Location registration information when SIP registration service module 4 starts according to any sip terminal is carried out target localization in the SIP calling procedure; SIP outgoing call proxy service module 3 is outgoing call acting servers of sip terminal regulation, it is when the call request that receives sip terminal, find that by the location of " SIP registration service " target terminal, the routing relation that is disposed by the territory Configuration Manager determine routing policy, make up and set up the feasible path of calling out between the different terminals;
Acoustic image terminal in the existing system abstract (virtual) is become sip terminal.Abstract sip terminal provides the interface that meets above-mentioned sip terminal requirement by the simulation of computer software to platform, and the inner control visit of finishing concrete application system.
Each sip terminal all should be under the jurisdiction of certain territory, and this is consistent with the agreement of RFC3261.Sip terminal at first should be registered to the registrar (Registrar Server) in affiliated territory after starting, and exists to register it, writes down its network site and other relevant information.After registration, sip terminal has following name form:
[friendly name]＜sip/sips: domain name under the Termination ID @ 〉
Wherein, friendly name is that optionally friendly name is used for understanding intuitively the purposes implication of each terminal when platform management.This terminal of Sip/sips identifier declaration is a sip user, and wherein sips represents the transmission start Transport Layer Security mechanism (TLS) of Session Initiation Protocol.Termination ID must keep the overall situation unique under terminal in the territory, ad hoc proposal adopts the digital coding mechanism of class telephone number, so that provide visit to the terminal that is similar to video telephone.Some typical sip terminals names as:
" meeting room 1 camera "＜sip:07315061@hnga.gov.cn 〉
sips:07332000@zhuzhou.hnga.gov.cn
Medium route: after in a single day the SIP multimedia transmission mechanism of standard determine the clear and definite position of communicating pair, directly between two sip terminals, set up point-to-point connection, carry out multimedia transmission by the media formats of consulting.For the SIP acoustic image terminal of standard, this yes best the most direct mode.Yet for the terminal of the non-standard application system of a large amount of existence, the medium that we can't directly set up between the sip terminal connect, and this wherein must be through certain path gating (route) and media format conversion.Though virtual sip terminal will be simulated camera and has been abstracted into a terminal equipment that meets the SIP standard, yet the image of simulation camera can't be directly be transferred to the standard sip terminal with the form of digital picture.This wherein must deliver to the SIP encoder with the exchange through two analog matrixs of the image of camera, and the standard sip terminal obtains this image by the visit to the SIP encoder then.This process promptly is referred to as the medium routing procedure.
In this platform, finally the visit of whole acoustic image network signal terminal is all called out this standardisation process by SIP and finish.But, because network is a distributed frame, there is the relational structure in a plurality of territories, and aforementioned medium routing procedure.Therefore, be a quite complicated process to the addressing of whole network and medium route, this complex process should not waited by user, terminal and bear, and the foundational system of reason platform is finished automatically.Therefore,, in this platform, utilize SIP outgoing call agency, make whole addressing and medium routing procedure transparence as cutting point according to the agreement of RFC3261.
SIP outgoing call agency is a kind of SIP Call Agent mechanism that RFC3261 arranges, its basic principle is: sip terminal is sent to nearest (normally under territory) SIP outgoing call agency with call request, as for how seeking target, intermediate demand which kind of carries out handle operation, all act on behalf of inside by the SIP outgoing call and finish, no matter terminal is this details.By SIP outgoing call agency mechanism, platform software can be finished the terminal addressing and the work of medium route of this platform pellucidly.
SIP registrar server (Registrar Server): SIP registrar server is another important Session Initiation Protocol functional unit of RFC3261 agreement, and it mainly acts on is registration and the sip terminal position enquiring of finishing sip terminal.
SIP is redirected (Redirection): in the process of sip terminal (UA1) calling SIP terminal (UA2), through the location, find that in fact call targets UA2 should transfer to another terminal UA3, this process is referred to as to be redirected.
Domain control server has been born the role of core in this system.Domain control server has been managed all acoustic image application systems that this territory is had jurisdiction over, and this comprises: the SIP application system of standard (as the SIP conference terminal among the figure, SIP Conference server, SIP acoustic image terminal), off-gauge digital sound are the analog audio-video application system of exchcange core and the acoustic image application system of other form (as wireless telecommunications commander, third party's central control system etc.) as application system (as the non-standard acoustic image terminal among the figure, off-gauge acoustic image Control Server), City Surveillance System, with the matrix.Domain control server is finished centralized management and control to all systems by integrated access control interface to these system equipments, and for example image obtains, exchanges, the configuration of equipment, operation and status poll.Prior, domain control server is by in house software, institute managed the non-standard SIP audio-visual equipment of controlling carry out virtually and abstract, makes it to become a standard sip terminal in logic.
In the sip terminal abstract architecture, domain control server provides the sip terminal simulation mechanism, and the non-standard sip terminal that this mechanism can be managed domain control server utilizes the computer software simulation to be one and meets RFC3261 required standard sip terminal.The sip terminal of any standard sip terminal or simulation all can be called out these simulation sip terminals, and vice versa.In the media formats negotiations process that SIP calls out, the simulation sip terminal will be that direct transmission, matrix switch or needs process medium route according to the decision of media negotiation content.Where necessary, these simulation sip terminals finally by domain control server platform, the concrete corresponding acoustic image application system control interface of utilization, are finished the control to equipment in the realization system.
Registration: all sip terminals (comprising SUA1, SUA2, SAUA1, SAUA2) all by the RFC3261 agreement, carry out endpoint registration with REGISTER standard SIP request to " the SIP registration service " of domain control server when starting.In the registration process, may need " authentication service " of domain control server to carry out authentication assistance, guarantee fail safe.
Direct calling between the terminal: if can directly set up transmission channel (for example between SUA1 and the SUA2 between the sip terminal (physical end or pseudo-terminal), if perhaps need not between AUA1 and the AUA2 through multistage matrix switch), then the process of its calling is very simple, abides by RFC3261 SIP call flow fully.
Between SUA1 and SUA2, SUA1 calls out SUA2:
The SIP that SUA1 sends INVITE SUA2 asks the outgoing call agency service of domain control server;
The outgoing call agency service is by the position of registration service location SUA2;
The outgoing call agency service finds that by retrieval configuration and medium routing policy can directly set up medium by IP network between SUA1 and the SUA2 connects, and therefore need not to do any particularization and handles, so this call request is forwarded to SUA2;
SUA2 carries out the SIP call treatment by the RFC3261 agreement, carries out media negotiation with SUA1, determines how to carry out between SUA1 and the SUA2 video/audio transmission;
After media negotiation is finished, set up direct media transmission channel between SUA1 and the SUA2, look audio transmission.
If need not between AUA1 and the AUA2 through multistage matrix switch, then the flow process of SAUA1 calling SAUA2 is also similar:
The SIP that SAUA1 sends INVITE SAUA2 asks the outgoing call agency service of domain control server;
The outgoing call agency service is by the position of registration service location SAUA2;
The outgoing call agency service is by retrieval " configuration and medium routing policy ", find that the transmission between SAUA1 and the SAUA2 is based on non-digitalization media delivery mechanism, therefore the outgoing call agency service need make the interchange channel between AUA1 and the AUA2 be in ready state by the analog matrix system control interface; Simultaneously, call request is forwarded to SAUA2;
SAUA2 carries out the SIP call treatment by the RFC3261 agreement, and (this moment, media negotiation was different from the SIP media negotiation of standard, because the Digital Media that need not to realize transmission to carry out media negotiation with SAUA1; In fact, the media negotiation of this moment is used for the connection between definite AUA1 and the AUA2, promptly causes the actual change action of matrix);
After media negotiation is finished, set up the analogue transmission passage between SAUA1 and the SAUA2, the video/audio transmission is carried out.
Calling between the heterogeneity terminal, call out SAUA1 as SUA1:
The SIP that SUA1 sends INVITE SAUA1 asks the outgoing call agency service of domain control server;
The outgoing call agency service is by the position of registration service location SAUA1;
The outgoing call agency service is found not to be based on identical media delivery mechanism between SUA1 and the SAUA1 by retrieval " configuration and medium routing policy ", and does not dispose the medium routing relation between the SUA1 to SAUA1 this moment, so this calling can not be carried out;
The outgoing call agency service is informed the SUA1 call failure.
Owing to do not have the medium routing relation between the current heterogeneity terminal, so call failure.Therefore, realize interconnecting between SUA1 and the SAUA1 if desired, must on domain control server, dispose the medium routing rule.
Call out to disconnect: the SIP outgoing call agency of this platform has stateful proxy mechanism based on the RFC3261 agreement, it is the call conversation between the sip terminal, in the whole dialogue process, SIP outgoing call agency keeps its state machine, and continues to have an effect by outgoing call agency in the Record-Route mechanism maintenance whole dialogue process of SIP.Therefore, when the sip terminal calling needed to disconnect, any one party was sent SIP BYE request, and the SIP outgoing call is acted on behalf of according to the state machine of being kept, and carries out cleaning work (for example analog matrix disconnection commutative relation); Simultaneously, according to the Session Initiation Protocol standard, remove annexation, end dialog between the terminal.
The complete notion that has proposed to make up audio and video information comprehensive transmission Control Network of this programme comprises communication protocol, addressing system, routing mechanism, interconnected mechanism and open interface.Acoustic image is used unified standard, interconnected active demand problem has initiative value for solving for this.Platform effectively solves the connection problem that non-standard SIP acoustic image is applied to standard SIP Control on Communication platform by proposing the special theory of " simulation sip terminal ".By this function, utilize the card module of computer software, can effectively current all kinds of forms be differed, the acoustic image application system of agreement confusion is integrated into this platform with uniform way, has good expansion, expansion capacity simultaneously.The SIP that abides by the RFC3261 standard calls out Control Network, becomes an architecture that is similar to the call network with whole audio and video information network is abstract.This not only meets the related request of the Ministry of Public Security's " requirement of supervision of the cities alarm network system current techique ", also meets the following international standard of multimedia communication simultaneously.This calling Control Network makes the transmission of audio and video information on network manipulate very simple (class call), fully takes into account compatibility simultaneously, and by the seamlessly transit ability of current environment to following multimedia transmission environment." medium route " mechanism that scheme proposes efficiently solves between the acoustic image application system of different architecture how to carry out the problem that medium interconnect.And this process that interconnects is that transparent, automation is carried out for the user.
Present specification Chinese and English abbreviation explanation:
TU:Transfer Unit, the transmission unit in the network environment (as video exchange matrix, the network switch, protocol conversion equipment etc.).
RFC:Request For Comments, the series standardized file of definition standard the Internet communications agreement.The follow-up different consensus standard of numbering representative.
TU-ADA:Analog to Digital Abstract, the abstract technology of digitlization of transmission unit (TU) in the video/audio analogue transmission network.
AVTCP:Audio﹠amp; Video Transfer Control Protocol, the computer network with standard network protocol of the volume addressing system of acoustic image transmission (video/audio transmission) network, exchange and route control.
AT-RIP:Analog Transfer-Route Information Protocol, the registration of medium route and the location protocol of acoustic image analogue transmission network.
AT-SDP:Analog Transfer-Session Description Protocol, the media description agreement of simulation audio and video information.
V-MGCP:Video-Media Gateway Control Protocol, the video media gateway control protocol.
AVNMP:Audio﹠amp; Video Network Management Protocol, acoustic image transmission network management and control protocol.
Claims (7)
1. domain control server that is used for the audio and video information Comprehensive Control, it is characterized in that it comprises business interface layer (1) based on standard RFC3261SIP protocol stack, SIP outgoing call proxy service module (3), SIP registration service module (4), SIP redirect services module (5), medium route control logic module (6), sip terminal analog logic module (7), territory Configuration Manager (8) and non-standard SIP acoustic image system interface (9), wherein standard RFC3261 Session Initiation Protocol stack receives the standard RFC3261 Session Initiation Protocol data of automatic network, business interface layer (1) is finished the analysis of data and is converted to described SIP outgoing call proxy service module (3), SIP registration service module (4), SIP redirect services module (5), medium route control logic module (6), sip terminal analog logic module (7), the concrete call instruction of non-standard SIP acoustic image system interface (9), territory Configuration Manager (8) is finished SIP redirect services module (5), medium route control logic module (6), the support of sip terminal analog logic module (7), sip terminal analog logic module (7) is controlled each terminal according to the interface that non-standard SIP acoustic image system interface (9) provides;
RFC3261 Session Initiation Protocol stack provides the basic communication control ability of standard RFC3261 Session Initiation Protocol, and described business interface layer (1) provides the user capture that domain control server is managed, disposes interface, the interface of cooperating between a plurality of territories networking time domains and the territory; Non-standard SIP acoustic image system interface (9) provides the specific access control ability to all kinds of non-standard SIP acoustic image application systems with the form of open plug-in unit; Territory Configuration Manager (8) is used to dispose the actual acoustic image application system of all management of domain control server, and the routing relation in this territory between these terminals and the equipment; Sip terminal analog logic module (7) is used for the non-standard sip terminal managed by abstract this territory of form of software; When medium route control logic module (6) needs this calling is carried out the control of medium route by territory Configuration Manager (8) in the terminal call process; SIP redirect services module (5) is used to cooperate medium route control logic, and the terminal of finishing in the medium routing procedure is redirected; Location registration information when SIP registration service module (4) starts according to any sip terminal is carried out target localization in the SIP calling procedure; SIP outgoing call proxy service module (3) is the outgoing call acting server of sip terminal regulation, it is when the call request that receives sip terminal, find that by the location of SIP registration service target terminal, the routing relation that is disposed by the territory Configuration Manager determine routing policy, make up and set up the feasible path of calling out between the different terminals.
2. a kind of domain control server that is used for the audio and video information Comprehensive Control as claimed in claim 1, it is characterized in that it also comprises authentication service module (2), described business interface layer (1) is finished the analysis of data and is converted concrete call instruction to described authentication service module (2) to, described SIP outgoing call proxy service module (3) is carried out authentication based on authentication service module (2), authentication service module (2) provides subscriber authentication mechanism based on the SIP Authentication mechanism that RFC3261 arranged, and guaranteeing can not this platform of access control without the user of authentication.
3. a kind of control method that is used for the domain control server of audio and video information Comprehensive Control as claimed in claim 2 is characterized in that said method comprising the steps of:
The first step, the terminal equipment initialization;
In second step, finish the registration of terminal equipment by SIP registration service module;
The 3rd goes on foot, and is finished according to the RFC3261 agreement by SIP outgoing call proxy service module and calls out;
In the 4th step, judge the connection type between terminal equipment;
The 5th step; According to the digital terminal between terminal equipment to digital terminal or pseudo-terminal to pseudo-terminal or digital terminal to pseudo-terminal or pseudo-terminal are interrupted the calling of classifying of four kinds of different connection types to numeral.
4. a kind of control method that is used for the domain control server of audio and video information Comprehensive Control as claimed in claim 3, it is characterized in that digital terminal to calling (S-S calling) performing step of digital terminal is in described the 5th step: A, calling are initiated digital terminal SUA1 and are sent the outgoing call proxy service module of asking domain control server to the SIP of target number terminal SUA2; B, outgoing call proxy service module are carried out authentication by authentication module to digital terminal SUA1, if authentication is failed then refused call request; If C authentication success is by the position of SIP registration service localizing objects digital terminal SUA2; D, location positioning success back are routed to SUA2 by SIP registration service module with SUA1; 5, SUA2 carries out media negotiation by RFC3261 agreement with SUA1, sets up multimedia transmission channel based on the standard multimedia host-host protocol until end of calling.
5. a kind of control method that is used for the domain control server of audio and video information Comprehensive Control as claimed in claim 3, it is characterized in that in described the 5th step, pseudo-terminal to calling (A-A calling) performing step of digital terminal is: A, calling initiation pseudo-terminal SAUA1 send the SIP request to target simulation terminal SAUA2, after the authentication success, the position of location pseudo-terminal SAUA2; After successfully locate the position of B, SAUA2, find that by the territory Configuration Manager medium route control logic module and sip terminal analog logic module make this transmission channel be in ready state between the two based on the analogue transmission passage; The request that C, SAUA1 send is routed to SAUA2, carry out media negotiation and success by RFC3261 agreement after, medium route control logic module and sip terminal analog logic module physics are set up between the two connecting channel until end of calling.
6. a kind of control method that is used for the domain control server of audio and video information Comprehensive Control as claimed in claim 3, it is characterized in that in described the 5th step, digital terminal to the calling between the pseudo-terminal (S-A calling) performing step is: A, calling initiation digital terminal SUA1 send the SIP request to target simulation terminal SAUA1, after the authentication success, the position of location pseudo-terminal SAUA1; After successfully locate the position of B, SAUA1, find to exist simulation and Digital Transmission passage between the two by the territory Configuration Manager, medium route control logic module and sip terminal analog logic module make this transmission channel be in ready state; The request that C, SUA1 send is routed to SAUA1, carry out media negotiation and success by RFC3261 agreement after, medium route control logic module and sip terminal analog logic module physics are set up between the two connecting channel until end of calling.
7. a kind of control method that is used for the domain control server of audio and video information Comprehensive Control as claimed in claim 3, it is characterized in that in described the 5th step, pseudo-terminal to the calling between the digital terminal (A-S calling) performing step is: A, calling initiation pseudo-terminal SAUA1 send the SIP request to target number terminal SUA1, after the authentication success, the position of location number word terminal SUA1; After successfully locate the position of B, SUA1, find to exist simulation and Digital Transmission passage between the two by the territory Configuration Manager, medium route control logic module and sip terminal analog logic module make this transmission channel be in ready state; The request that C, SAUA1 send is routed to SUA1, carry out media negotiation and success by RFC3261 agreement after, medium route control logic module and sip terminal analog logic module physics are set up between the two connecting channel until end of calling.
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Cited By (5)
| Publication number | Priority date | Publication date | Assignee | Title |
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| CN102223515A (en) * | 2011-06-21 | 2011-10-19 | 中兴通讯股份有限公司 | Remote presentation meeting system and method for recording and replaying remote presentation meeting |
| CN104956381A (en) * | 2012-11-21 | 2015-09-30 | 格林伊登美国控股有限责任公司 | Graphical user interface for monitoring and visualizing contact center routing strategies |
| US9912813B2 (en) | 2012-11-21 | 2018-03-06 | Genesys Telecommunications Laboratories, Inc. | Graphical user interface with contact center performance visualizer |
| US9912812B2 (en) | 2012-11-21 | 2018-03-06 | Genesys Telecommunications Laboratories, Inc. | Graphical user interface for configuring contact center routing strategies |
| CN114079609A (en) * | 2020-08-03 | 2022-02-22 | 阿里巴巴集团控股有限公司 | Network system control method, device, equipment, medium and network system |
Family Cites Families (1)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN100388733C (en) * | 2002-12-25 | 2008-05-14 | 中兴通讯股份有限公司 | Method for calling route in network of dialog ignition protocol |
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| Publication number | Priority date | Publication date | Assignee | Title |
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| CN102223515A (en) * | 2011-06-21 | 2011-10-19 | 中兴通讯股份有限公司 | Remote presentation meeting system and method for recording and replaying remote presentation meeting |
| CN102223515B (en) * | 2011-06-21 | 2017-12-05 | 中兴通讯股份有限公司 | Remote presentation conference system, the recording of remote presentation conference and back method |
| CN104956381A (en) * | 2012-11-21 | 2015-09-30 | 格林伊登美国控股有限责任公司 | Graphical user interface for monitoring and visualizing contact center routing strategies |
| US9912813B2 (en) | 2012-11-21 | 2018-03-06 | Genesys Telecommunications Laboratories, Inc. | Graphical user interface with contact center performance visualizer |
| US9912812B2 (en) | 2012-11-21 | 2018-03-06 | Genesys Telecommunications Laboratories, Inc. | Graphical user interface for configuring contact center routing strategies |
| US10194028B2 (en) | 2012-11-21 | 2019-01-29 | Genesys Telecommunications Laboratories, Inc. | Graphical user interface for configuring contact center routing strategies |
| CN114079609A (en) * | 2020-08-03 | 2022-02-22 | 阿里巴巴集团控股有限公司 | Network system control method, device, equipment, medium and network system |
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