CN101631174B - Network telephone real-time identification and filtering method based on session initiation protocol - Google Patents

Network telephone real-time identification and filtering method based on session initiation protocol Download PDF

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Publication number
CN101631174B
CN101631174B CN2009101159866A CN200910115986A CN101631174B CN 101631174 B CN101631174 B CN 101631174B CN 2009101159866 A CN2009101159866 A CN 2009101159866A CN 200910115986 A CN200910115986 A CN 200910115986A CN 101631174 B CN101631174 B CN 101631174B
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layer
sip
flow
information
session
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CN101631174A (en
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王攀
张顺颐
许娇
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Nanjing Youshutong Information Technology Co ltd
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RUNTREND TECHNOLOGY Inc
Suzhou RunTrend Technology Inc
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Abstract

The invention discloses a network telephone real-time identification and filtering method based on a session initiation protocol. The invention comprises a data acquisition layer, a protocol analysis layer, a flow identification layer, a flow control layer and a presentation layer in sequence according to the data flow direction, wherein the flow identification layer works in an unidirectional flow collection mode, the payload depth detection is performed to received grouping information according to the payload characteristic of an SIP, the SIP session information is transmitted to the flow control layer, and the flow control layer works in a parallel mode and performs filtering to the illegal SIP session by using a signaling masquerading technology. The invention can rapidly and accurately identify the network telephone information stream based on the SIP signaling and also enables the illegal SIP session to be effectively blocked or filtered.

Description

The networking telephone Real time identification and the filter method of session initiation protocol based
Technical field
The invention belongs to network communications technology field, relate in particular to a kind of networking telephone Real time identification of session initiation protocol based and the method for filtration.
Background technology
The networking telephone (VoIP; Voice over Internet Protocol) be a kind of telephone system of Vo IP transmission technology; Can adopt internet and the interconnected environment of global ip widely;, better service more than traditional business are provided, and it relies on cheap telephone expenses price and voice quality preferably to obtain masses' extensive use.VoIP has two kinds of application layers commonly used control (signaling) agreements, a kind ofly is agreement H.323, and another kind is Session initiation Protocol (SIP, Session Initiation Protocol).Session Initiation Protocol is an application layer control signaling protocol of using based on realization real time communication in the IP network of HTTP.
Under the attraction of huge interests, a large amount of unlawful VoIP operations are flooded with regular telecommunication market, not only cause legal operator telephone traffic to run off, and have more broken the competition situation of original telecommunication market, have brought huge impact for traditional voice service.Simultaneously, because unlawful VoIP lacks necessary service quality guarantee, only depend on simple low price to attract users, this is in the interests of also having damaged the user to a certain extent.
Yet the method that traditional networking telephone to session initiation protocol based (SIP) filters can not be filtered the information flow that blocks SIP phone effectively.
Such as traditional I P address filter method, in case filtered user's IP address, then other all business that comprise normal online of user all are under an embargo, and this method is infeasible.
And traditional port detection method, though agreement has been stipulated the default port of SIP signaling, therefore most ports that adopt dynamic assignment of using can't discern the VoIP information flow based on SIP from port simply.
Traditional for another example forthright series connection flow control methods; Need checkout equipment be deployed on the network traffics true path, also might form and handle bottleneck and Single Point of Faliure, can influence whole network topology structure; Increased the propagation delay time of whole network simultaneously; Legacy network is brought certain adverse influence, but parallel way can't actual contact arrive real data flow again, can't directly operate and control data stream.
Summary of the invention
The objective of the invention is: in order to solve above-mentioned deficiency of the prior art; A kind of method of the networking telephone being carried out Real time identification and filtration is provided; Especially be directed against the networking telephone of session initiation protocol based SIP; On the basis that one-way flow is gathered, accurately discern real-time networking telephone information flow, and under paralleling model, it is carried out filtration rapidly and efficiently based on the SIP signaling.
Technical scheme of the present invention is:
A kind of networking telephone Real time identification and filter method of session initiation protocol based; Comprise data collection layer 1, protocal analysis layer 2 and presentation layer 3 also comprise the flow identification layer 4 that is operated under the one-way flow acquisition mode; And be operated in the flow control layer 5 under the paralleling model; Data flow through successively from the bottom up data collection layer 1, protocal analysis layer 2, flow identification layer 4, flow control layer 5 and presentation layer 3, wherein, flow identification layer 4 comprises following steps:
(4.1) receive the grouping information that the protocal analysis layer provides;
(4.2) carry out pay load deep according to the payload characteristic of SIP and detect,, continue execution in step (4.3), otherwise abandon grouping, do not do operation, go to step (4.1) then if mate successfully;
(4.3) preserve the SIP session information, form the call detail record of SIP;
Flow control layer 5 comprises following steps:
(5.1) from the call detail record of SIP, read the details of this session;
(5.2) forge the on-hook signaling of a SIP, and this session information is write in the signaling;
(5.3) on-hook signaling of forging is sent to the callee;
(5.4) callee receives the on-hook signaling of forgery, callee's on-hook, and calling is filtered.
The more detailed technical scheme of the present invention is: wherein, the grouping information that said protocal analysis layer 2 provides comprises the header information and the payload information thereof of IP packet header and transmission control protocol/UDP.The SIP session information that said flow identification layer 4 is preserved comprises the payload information that caller IP address, called IP address, source port, destination interface and IP divide into groups.In the said flow identification layer 4, when the grouped data of other sessions arrives, upgrade the SIP session information, and form the call detail record of new SIP.
The invention has the beneficial effects as follows:
1. can fast, accurately discern networking telephone information flow, professional and not influence of non-SIP session to other based on the SIP signaling;
2. can make operator to implementing effectively to block to filter, to ensure the interests of traditional voice transmission and legitimate network phone operation based on the real-time network telephone service of SIP signaling;
3. can carry out statistical analysis to real-time network phone information flow, be convenient to control SIP and call out influence traditional voice service based on the SIP signaling;
4. the real-time network telephone service based on the SIP signaling is carried out performance evaluation, flow control, call follow and Traffic Anomaly detection etc.;
5. the real-time network phone information based on the SIP signaling is carried out security monitoring.
Description of drawings
Below in conjunction with accompanying drawing and embodiment the present invention is further described:
Fig. 1 is the overall system flow chart of the preferred embodiments of the present invention;
Fig. 2 is the flow chart of protocal analysis layer in the preferred embodiments of the present invention;
Fig. 3 is the flow chart of flow identification layer in the preferred embodiments of the present invention;
Fig. 4 is the flow chart of flow control layer in the preferred embodiments of the present invention.
Wherein: 1 data collection layer; 2 protocal analysis layers; 3 presentation layers; 4 flow identification layers; 5 flow control layers.
Embodiment
Embodiment: as shown in Figure 1, system is divided into 5 aspects, is respectively data collection layer 1, protocal analysis layer 2, flow identification layer 4, flow control layer 5 and presentation layer 3 from the bottom up by data flow.When the networking telephone is communicated by letter, at first pass through data collection layer 1, the message stream data of the networking telephone is carried out data acquisition and duplicated, to last layer protocal analysis layer various grouping informations are provided, ensure data integrity, reliable.This layer is bitstream data with the interface of last layer protocal analysis layer.
The workflow of protocal analysis layer 2 as shown in Figure 2; The data that transmit from data collection layer are at first carried out the protocal analysis of link layer; (the TCP to TCP/IP is provided then; Transmission Control Protocol/Internet Protocol) agreement of data is resolved; Obtain header information and the necessary packet payload information of enough IP packet header information and TCP/UDP (UDP User Datagram Protocol),, satisfy identification and the perception demand of flow identification layer business in order to offer last layer flow identification layer.The protocal analysis degree of depth of this layer should be analyzed to the 4th layer of the ICP/IP protocol stack, i.e. transport layer.The interface of this layer and flow identification layer is stream (Flow), and the information of stream comprises caller IP address, called IP address, source port, destination interface and protocol type, and wherein protocol type has two kinds of TCP or UDP, storage part payload also in this stream.How much can disposing as required of this analysis layer institute stored packet payload information.
Flow identification layer 4 is core layers of whole framework; Characteristics such as the IP packets headers information that it provides according to the lower-layer protocols analysis layer and the header information of TCP/UDP and payload information effectively identify the network telephone service based on the SIP signaling, and the grouping information that will mate failure abandons and do not handle.This layer is gathered one-way flow; And adopt based on pay load deep and detect (DPI; Deep Paeket Inspection) recognition methods; The information that identification obtains offers flow control layer, and the interface data that provides comprises caller IP address, called IP address, source port, destination interface and application details.Wherein, use details and comprise, be i.e. the payload information of IP grouping except the IP head that IP divides into groups, the content information the TCP/UDP head.
The workflow of flow identification layer 4 is as shown in Figure 3, may further comprise the steps:
(1) receives the grouping information that the protocal analysis layer provides;
(2) carry out pay load deep according to the characteristic of SIP payload and detect,, continue execution in step 3, otherwise abandon grouping, do not do operation, go to step 1 then if mate successfully;
(3) preserve the SIP session information, the key message of SIP session is preserved, the SIP session information comprises caller IP address, called IP address, port, encoding and decoding speech type, calls out initiation time, end of calling time.When the grouping information of other sessions arrives, upgrade corresponding information, form the call detail record (CDR, Call Detail Record) of SIP.
With a concrete signaling packet is example, and the embodiment of " carrying out pay load deep according to the characteristic of SIP payload detects " was described for the 2nd step, and typically the signaling packet payload of SIP INVITE is expressly as follows:
..INVITEsip:88218293181.26.66.5:5060SIP/2.0..Via:SIP/2.0/UDP181.26.66.5:5060;branch=z9hGrbK60058..From:<sip:88218293181.26.66.5:5060>;tag=60056;Wenqent=Wenqentsipstack;index=0;ver=1..To:<sip:88218293181.26.66.5:5060>..Call-ID:60057181.26.66.5..CSeq:20..INVITE..Cont?act:<sip:88218293181.26.66.5:5060>..max-forwards:50..user-agent:tenqe?nt-VQQ..subject:Hello...Allow:INVITE,ACK,CANCEL,BYE...
The flow identification layer receives the data message from the protocal analysis layer, if only can collect forward, such as the data that arrive inside the province outside the province, then detect the feature string that whether contains SIP INVITE calling in the data, i.e. " INVITE SIP: "; If only can collect reverse, such as the data that arrive outside the province inside the province, detect the feature string that whether contains SIP 100 TRYING calling in all packets so, i.e. " 100TRYING ".If do not have, then abandon this grouping information and do not process; If have, explain that promptly this calling is the SIP session.Extract source sip address SIP FROM URI, target sip address SIP TO URI, SIP caller gateway address and the called gateway address of SIP and the general header field of CSeq (Command sequence) in this grouping.All groupings that possesses identical SIP FROM URI, SIP TO URI, SIP caller gateway address and called gateway address of SIP and CSeq territory all belong to same SIP session.So far, carry out pay load deep based on the characteristic of SIP payload and detect completion.
Illegally whether defining of SIP session by the provider customer, flow control layer carries out real time filtering to the illegal SIP session that the flow identification layer identifies, and under parallelly connected listen mode, adopts the signaling camouflage to realize; The mode that promptly adopts bypass to disturb; With third-party mode, the calling subscriber that disguises oneself as sends on-hook signaling to the called subscriber; Making the called subscriber take for is the on-hook signal that the calling subscriber sends, thus hanging wire.
The step that flow control layer 5 is implemented to filter is:
(1) from the call detail record of SIP, reads the details of this session;
(2) forge the on-hook signaling of a SIP, and this session information is write in the signaling;
(3) on-hook signaling of forging is sent to the callee;
(4) callee receives the on-hook signaling of forgery, callee's on-hook, and calling is filtered.
The concrete steps that present embodiment is implemented to filter are:
From the call detail record CDR of SIP, read the details of this session, obtain SIP FROM URI, SIP TO URI, SIP caller gateway address and called gateway address of SIP and CSeq territory in this SIP calling.Forge a SIP BYE order then, pseudo-making method is: according to the data of SIP INVITE signaling packet, replace three parts; The firstth, the method name; Be about to INVITE and replace with BYE, the secondth, the sequence number numerical value of change CSeq adds 1 on original value basis; The 3rd is the method name that changes CSeq, changes BYE into; Then the BYE grouping information after this change is sent to terminal called according to called IP address.Terminal called is received BYE information, and takeing for is the on-hook signal that former calling subscriber sends, thus on-hook.
Be in the presentation layer 3 of the superiors; Be prior art; Identification for the SIP phone real time business has very wide significance and using value, can be applied to the real-time telephone service traffic statistics of SIP and analyze the real-time telephone service performance evaluation of SIP; Real-time phone traffic control of SIP and call follow, fields such as real-time phone traffic abnormality detection of SIP and the real-time phone information security monitoring of SIP.
Above content is to combine concrete execution mode to the further detailed description that the present invention did, and can not assert that practical implementation of the present invention is confined to these explanations.For the those of ordinary skill of technical field under the present invention, under the prerequisite that does not break away from the present invention's design, can also make some simple deduction or replace, all should be regarded as belonging to protection scope of the present invention.

Claims (4)

1. the networking telephone Real time identification and the filter method of a session initiation protocol based; Comprise data collection layer (1); Protocal analysis layer (2) and presentation layer (3); It is characterized in that: also comprise the flow identification layer (4) that is operated under the one-way flow acquisition mode, and be operated in the flow control layer (5) under the paralleling model, data flow through successively from the bottom up data collection layer (1), protocal analysis layer (2), flow identification layer (4), flow control layer (5) and presentation layer (3); Wherein, flow identification layer (4) comprises following steps:
(4.1) receive the grouping information that the protocal analysis layer provides;
(4.2) carry out pay load deep according to the payload characteristic of SIP and detect,, continue execution in step (4.3), otherwise abandon grouping, do not do operation, go to step (4.1) then if mate successfully;
(4.3) preserve the SIP session information, form the call detail record of SIP;
Flow control layer (5) comprises following steps:
(5.1) from the call detail record of SIP, read the details of this session;
(5.2) forge the on-hook signaling of a SIP, and this session information is write in the signaling;
(5.3) on-hook signaling of forging is sent to the callee;
(5.4) callee receives the on-hook signaling of forgery, callee's on-hook, and calling is filtered.
2. according to the networking telephone Real time identification and the filter method of the session initiation protocol based described in the claim 1, it is characterized in that: the grouping information that said protocal analysis layer (2) provides comprises the header information and the payload information thereof of IP packet header and transmission control protocol/UDP.
3. according to the networking telephone Real time identification and the filter method of the session initiation protocol based described in claim 1 or 2, it is characterized in that: the SIP session information that said flow identification layer (4) is preserved comprises the payload information that caller IP address, called IP address, source port, destination interface and IP divide into groups.
4. according to the networking telephone Real time identification and the filter method of the session initiation protocol based described in claim 1 or 2; It is characterized in that: in the said flow identification layer (4); When the grouped data of other sessions arrives, upgrade the SIP session information, and form the call detail record of new SIP.
CN2009101159866A 2009-08-14 2009-08-14 Network telephone real-time identification and filtering method based on session initiation protocol Expired - Fee Related CN101631174B (en)

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CN101917446A (en) * 2010-08-26 2010-12-15 成都市华为赛门铁克科技有限公司 Multimedia session control method, related device and communication system
CN108123959B (en) * 2017-12-30 2020-11-20 世纪网通成都科技有限公司 Computer readable storage medium for restoring VOIP call ticket and VOIP call ticket restoring system using the same
CN108111530B (en) * 2017-12-30 2020-11-13 世纪网通成都科技有限公司 Computer readable storage medium for detecting VOIP call state and detection system using the same
CN110798461B (en) * 2019-10-23 2022-04-05 国家计算机网络与信息安全管理中心 VoIP (Voice over Internet protocol) association method and device under asymmetric routing network and readable storage medium
CN112929335B (en) * 2021-01-20 2022-09-06 号百信息服务有限公司 Flexible configuration-based baseband signaling filtering system and method

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