CN101437032B - System for monitoring VOIP voice quality based on SIP protocol and detection method thereof - Google Patents
System for monitoring VOIP voice quality based on SIP protocol and detection method thereof Download PDFInfo
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Abstract
The invention discloses an SIP protocol-based VOIP voice quality monitoring system and a detection method thereof and relates to a computer network. For overcoming the drawbacks of high cost and inflexibility of voice quality monitoring equipment in the prior art, a system and a method for monitoring the VOIP voice quality are designed. The method comprises the following steps: a voice data acquisition module extracts an SIP data packet and transmits the SIP data packet to a session tracking module for call state tracking; a voice data processing module carries out monitoring and maintenance analysis on a call established successfully; and finally, a voice quality estimation module calls an e-Model to calculate a voice information quality index MOS according to indexes such as network data delay, packet loss rate and jitter. The system can carry out accurate analysis of voice session and provide effective quality index and is low in cost. The system and the method can be widely used for monitoring, analysis and control of network flow, QoS guarantee and in other fields related to flow engineering.
Description
Technical field
The present invention relates to computer networking technology, especially computer network flow monitoring, network service quality and network performance analysis technical field.
Background technology
VoIP can provide value added service easily as CRBT, voice mail etc., and further popularizing of IP network is rapidly developed VOIP in the world.Along with China's IP telephony network scale and flow rapidly increase, the major issue that is faced is exactly, and: VoIP is based on that IP network realizes, yet IP network " do one's best " special, brought a series of problems such as time delay, shake, packet loss, make that QoS can not get guaranteeing at all.
Voice quality is most important concerning Network, and inferior voice quality may cause customer churn, has influence on the popularization of VoIP, thereby restricts its development.Guarantee the voice quality of VoIP, just must monitor it.The validity of voice quality monitoring depends on the analysis and assessment to session data, and some renowned companies have been devoted to the research of VOIP performance at present.
With the immediate existing monitoring product of the present invention NetTool that is FLUKE networks
TMSeries IIInline Network Tester (brand-new hand-hold type VoIP tester has quickened IP telephony deployment and failure diagnosis speed communication news [R] .2005 June) can online connection at this product aspect the VoIP, deeply see clearly voip call, the quick diagnosis IP phone starts, conversation control problem, and measures critical call quality index.But it uses hardware device to realize the test of voice quality, can bring two problems based on hardware: cost and flexibility; The flexibility of hardware device simultaneously is far away not as good as software, and in the face of special assessment models under special detection index under the varying environment and the various environment, hardware device is difficult to adjust timely.
Summary of the invention
Technical problem to be solved by this invention is: design a kind of VOIP based on SIP (Session initiation Protocol) (networking telephone) voice quality monitoring system, and detection method, solve in the existing hardware equipment, cost height and flexibility are low, in the face of special assessment models under special detection index under the varying environment and the various environment, be difficult to adjust timely, at the problems referred to above, we propose corresponding solution.
The VOIP voice quality monitoring system based on SIP of the present invention's design mainly comprises following four modules: speech data collection module, session tracking module, language data process module, speech quality evaluation module.Speech data collection module: connect network interface card based on INET socket (the address family socket that ICP/IP protocol is supported) layer building, main be responsible for gathering by network interface card catch the filtration that is operated the data that the system kernel space obtains, filter out other incoherent packets, therefrom extract SIP (Session initiation Protocol) bag and RTP (real time transport protocol) bag; The session tracking module: this module is responsible for the tracking of process is set up in session, and catches and with the media information of the voice communication of setting up, the SIP packet is carried out the session coupling, and whether the state exchange mechanism identification session that utilizes the SIP session to set up is set up or discharged; The language data process module: this module is responsible for the voice flow control head safeguarding each session information and belong to same session, when conversation end, the voice flow control head is submitted to the speech quality evaluation module, resolve media negotiation information, determine the characteristic value of this session, timestamp is stamped in rtp packet header formed the voice flow control head, the voice flow control head of same session is connected into chained list according to characteristic value; The speech quality evaluation module: thus this module is responsible for the voice flow control head of each session is analyzed the various voice quality indexs that calculate conversation, analyze the rtp voice packet header of each session, therefrom obtain the information of voice flow, the e-Model model that calls this module calculates the voice messaging quality, finally obtain MOS (average suggestion value), MOS value Chang Zuowei weighs the important indicator of communication system voice quality.
The present invention also provides a kind of VOIP speech quality detection method, and this method comprises the steps:
B1, network interface card is set to promiscuous mode, and the configuration-system initial parameter is set up the data buffer zone, catches network packet;
B2, speech data collection module judges whether the packet catch comprises the Session Initiation Protocol content, if execution in step B3 then; Otherwise execution in step B6;
B3, the session tracking module is analyzed the SIP bag, and follows the tracks of talking state, judges whether a session sets up success; If the success execution in step B5; Otherwise execution in step B4;
B4, the session tracking module deposits SIP packet data session information in the conversation message chained list, handles next SIP bag;
B5, the language data process module is extracted the media communication information of session negotiation in the SIP packet, port numbers and coded system, execution in step B7;
B6, whether speech data collection module judgment data bag is the RTP data flow, if execution in step B7 then; Abandon this packet otherwise carry out;
B7, the language data process module is carried out the maintenance of session voice data flow to the RTP packet, if this session discharges then abandons this bag; Otherwise execution in step B8;
B8, the speech quality evaluation module is analyzed the session voice data, passes through e-Model Model Calculation MOS value according to parameters such as data delay, packet loss.
Adopt VOIP voice quality monitoring system of the present invention and detection method, carry out multianalysis from application layer to network layer to the packet Business Stream, can reflect the state of both call sides in real time, have higher certainty of measurement, adopt modular construction, strengthened the extensibility of system.The present invention is applicable to network traffics, network service quality monitoring.
Description of drawings
Fig. 1 VOIP voice quality monitoring system architecture
Fig. 2 session tracking module schematic diagram
Fig. 3 language data process module diagram
Fig. 4 speech quality evaluation module workflow diagram
Fig. 5 speech quality detection method flow process of the present invention
Fig. 6 dialog-ID list item form schematic diagram
Embodiment
Below at accompanying drawing and instantiation enforcement of the present invention is elaborated.
Be illustrated in figure 1 as the VOIP voice quality monitoring system architectural schematic based on Session Initiation Protocol, this monitoring system mainly is made up of following four modules:
1. speech data collection module, the mainly responsible collection of this module is caught by network interface card and is operated the data message copy that the system kernel space obtains, at first with packet that captures and the keyword of judging SIP " sip " compare, if coupling thinks that then the packet that captures is the SIP bag, otherwise abandon, import the SIP bag into the session tracking module simultaneously.
2. session tracking module, this module are responsible for tracking that process is set up in session, catch the media information that is about to the voice communication of setting up, and the sip bag of receiving is carried out the session coupling.At first the sip bag of receiving is carried out lexical analysis, analysis wherein comprises the message of which kind of type, if set up the sip message of session, just searches the sip conversation recording according to the diagnostic criterium of same session.Whether the state exchange mechanism identification session that utilizes the sip session to set up is set up or is discharged.Set up state for session, parse media negotiation information such as port numbers, speech coding etc. and import the speech quality evaluation module into.If the conversation end state then notifies the language data process module to obtain the voice flow control head.
3. language data process module, each session information of this resume module and the voice flow control head (having beaten the rtp packet header of timestamp) that belongs to same session.When receiving that message is set up in session that terminal is sent, at first resolve media negotiation information, determine characteristic value such as communication port, the media transmission protocol etc. of this session.From the network intercepting submodule of speech data collection module, obtain rtp (RTP) packet header then, stamp timestamp simultaneously, the afterbody that according to characteristic value the voice flow control head of same session is added chained list afterwards, this chained list is the doubly linked list structure, when system initialization, distribute, safeguard by this module; Linked list head with the voice flow control head when conversation end imports the speech quality evaluation module into.
4. speech quality evaluation module, thereby the voice flow control head analysis that this module is finished each session calculates the call voice quality: at first obtain voice flow control head linked list head from the language data process module, travel through each voice control head of voice flow control head chain table analysis (having beaten the rtp packet header of timestamp) afterwards, therefrom obtain the information of voice flow: network delay, packet loss, type of coding; The formula 1 of then each impairment value being brought into the e-Model model of simplification obtains the R value; Calculate the Mos value by formula 2 at last.
R=R0-Icodec-Idelay-Ipacketloss (formula 1)
Wherein, R0: basic signal to noise ratio, take ITU-T Recommendation G.113 in the general value Icodec of R0 of record: the encoding and decoding distortion, take the G.113 code impairment value Idelay in the document of ITU-T Recommendation: the average delay damage, by formula 3 calculating and get; Ipacketloss: certain section interior packet loss of time.
For R＜0:MOS=1
For 0≤R≤100:MOS=1+0.035 * R+R (R-60) (100-R) * 7 * 10
-6(formula 2)
For R＞100:MOS=4.5;
Idelay=0.02 * d when 0≤delay＜100
Idelay=0.06 * d-4 when 100≤delay＜200
Idelay=0.12 * d-16 when 200≤delay＜300
Idelay=0.16 * d-28 when 300≤delay＜400
Idelay=0.06 * d+12 (formula 3) when 400≤delay＜500
Be illustrated in figure 2 as session tracking module schematic diagram, it is by two submodules: sip message resolution module and session matching module are finished the parsing of the sip message of receiving and session coupling.
The sip message analyzing sub-module is responsible for the SIP bag that receives is handled, obtain the pipeline fd that communicates by letter with speech data collection module read hold descriptor fdr and network intercepting module and voice conversation module communication pipeline fd2 read to hold descriptor fdr2, and the new pipeline fd3 that sets up with voice communication.Open up subprocess, opening voice conversation module in subprocess, and with fdr2 read hold descriptor and pipeline fd3 read hold descriptor fdr3 to pass to the voice conversation module; Read the sip data flow by descriptor fdr in parent process, resolve the sip character stream afterwards, and the content of resolving is inserted in the Record structure, this content is a media negotiation information, comprising: content behind message kind, message header position, the header field.
The session matched sub-block is responsible for the sip message in the Record structure is mated, if session success or end then are filled into session information sesssion info structure, the url of url of session originating end (network address) and called end is filled in tport field and the fport field respectively and transmits by writing of pipeline fd3 and hold fd3[0] give the language data process module, if session is not finished as yet and is continued to read duct size information.
The Record data structure:
typedef?struct
{
int?type；
int?contentlenth；
Cseqtype cseq; The Cseq territory * of/* message/
Fromtype from; The from territory * of/* message/
Totype to; The to territory * of/* message/
Callidtype call_id; The Call_ID header field * of/* message/
Viatype via; The via territory * of/* message/
Methodtype?method；
Sdptype?sdp；
}Record；
The Sesssion_info data structure:
typedef?struct
{
Int i; // conversation end or beginning 1-begin 0-and finish
int?tport；
int?fport；
Int tlen; The length of //to.url
Int flen; The length of //from.url
Unsigned int code; // coded message
}session_info；
Be illustrated in figure 3 as the language data process module diagram, it is by the I/O module, and three submodules of link module and packet handing module are formed.The I/O module uses the I/O multiplexing to be responsible for the reception of coordination network packet and SIP session information; Packet handing module is to be responsible for communicating by letter and data flow that format is received of this module and SIP conversation module and speech data collection module; Link module maintain sessions information chained list, the rtp bag that the transmission of processed voice data acquisition module comes is if conversation end then calls the speech quality evaluation module and analyzes, if new session is then created a new record in the session information chained list.
Control I/O module is used whether the I/O multiplexing is checked has the transmission of session tracking module among pipeline fd3 and the pipeline fd2 input: if pipeline fd3 is readable, call packet handing module and read pipeline data stream fd3 simultaneously according to data structure voice_info format, the data structure that will format the back acquisition then is sent in the link module respectively; If pipeline fd2 is readable, at first read pipeline fd2 data flow and storage, then read pipeline fd2 data flow again, and will pass to the speech quality evaluation module after the character stream format.
The voice_info structure:
Struct voice_info // voice conversation information nodes
{
Session_info session; // the session={i that receives, tport, fport, tlen, flen}
Char*turl; The url of // originating end
Char*furl; The url of // receiving terminal
Struct voice_info*next; The next voice channel information of // sensing
Struct rtp_packet*tfirst; // point to same session originating end rtp voice packet header
Struct rtp_packet*tlast; // point to same session originating end rtp voice packet tail
Struct rtp_packet*ffirst; // point to same session called end rtp voice packet header
Struct rtp_packet*flast; // point to same session called end rtp voice packet tail
}；
Be illustrated in figure 4 as the speech quality evaluation module and use network delay, packet loss, parameters such as type of coding are finished the calculating to voice quality parameters MOS value, its workflow diagram:
1. when module starts, call ping () function earlier and send ICMP (internet control message protocol) bag, calculate the network delay of both call sides by transmitting time and time of reception, and write down the network delay time t0;
2. the time field of module from the structure rtp_packet of first next node of language data process module transmission read the zero hour of the time of reception ts of first bag as conversation; Module is read the speech coding type from the srtp.pt field of rtp_packet structure, obtain the maximum R0 of this kind coding, and code impairment Icodec is with coding output speed V and become frame delay t1; Read the speech data length L then from the dalalen field of rtp_packet structure, obtain wrapping conversion delays t2=L/V, the sequence number that obtains first bag at last from the srtp.sq field of rtp_packet structure is as minmal sequence Smin;
3. module further travels through the number bag of whole session information chained list as the speech data number N that receives, and the srtp.sq field of noting in the structure when traversing last struct rtp_packet structure makes maximum sequence number Smax, uses Smin and Smax to calculate packet loss Ipacketloss.In the end obtain the moment tb of end of conversation in the time field among struct rtp_packet, the time span by tb-ts obtains whole conversation obtains average received and postpones t3.
4. algoritic module calls formula delay=t0+t1+t2+t3 and tries to achieve total delay delay, calculates Idelay (average delay damage) by formula 3;
5. algoritic module calculates MOS (average suggestion value) value by the above-mentioned e-Model formula of simplifying 2;
As shown in Figure 5, for adopting VOIP voice quality testing process of the present invention, mainly comprise following five steps:
1, network interface card is set to promiscuous mode, configures system's initial parameter, sets up the data buffer zone, catches network packet, log-on data collection, session tracking, language data process and speech quality evaluation module;
2, data acquisition module block analysis packet content is carried out SIP keyword coupling, if contain the SIP keyword, then further analyze SIP signaling content, carry out Session Initiation Protocol identification, extract the media communication information of session negotiation in the SIP packet, comprise port numbers and coded system.The dialog-ID list item is specially shown in Figure 6 in the Session Initiation Protocol recognition methods:
The type territory: the SIP packet that expression is received is request message or response message;
The method territory: if request message, expression requesting method such as INVITE, CANCEL, ACK etc.; Response message is represented method parameter in the Cseq territory in this way;
The from territory: expression session transmitting terminal URL and tag parameter, if no tag parameter acquiescence 0 in the SIP bag;
The to territory: expression session receiving terminal URL and tag parameter, if no tag parameter acquiescence 0 in the SIP bag;
The branch territory: its value is affairs of branch parameter value sign in the via territory in the SIP bag of receiving;
State territory: expression session current state;
Session Initiation Protocol identification comprises: data acquisition module searches in the record with receiving SIP whether identical dialog-ID list item is arranged; If not having explanation is new request, then create new list item recording-related information record SDP, if exist identical dialog-ID list item to do following processing: if it is identical with method field among the SIP that receives to find among the list item dialog-ID method field: be considered as repeating transmission if the method method is ACK or BYE, and the startup timeout mechanism, overtime device adds up.The numerical portion that if the method method is INVITE checks the Cseq field with write down in whether identical, identical expression repeating transmission and previous step are done same treatment; If the numeral of Cseq field is the re-INVITE request with different INVITE requests of then receiving in the record, create a new list item preservation relevant information.If the different different phases that then enter same session of method field in the list item that finds with method field among the SIP that receives, if capture BYE message, cancel conversation recording, and send conversation end message, abandon this message data to speech data collection module.
Session Initiation Protocol identification specifically describes and is, the o attribute among the use SDP (Session Description Protocol) is as search sign.The o property column comprises source and the session id of SDP; Record m attribute and a attribute.Used host-host protocol when the row of m attribute has comprised transmission medium, port and type of service; The row of a attribute has illustrated the coded system of voice; If receive that the SDP of identical o attribute then upgrades m attribute and a attribute record in the session; Set up m attribute and a attribute that compares both sides when finishing in session, if the identical port that the type and the both sides of media bearer agreement are used is delivered to the speech data trapping module.Set up failure otherwise thinking converses.
3, be that signaling is set up in session as the SIP signaling, then in storage area, create a conversation recording, store this SIP packet data session information, handle next packet, whether the judgment data bag is the RTP packet, and the waiting voice data processing module is further analyzed the relevant RTP packet of this session; As be the SIP release signaling, then, the RTP packet is carried out the session voice data flow safeguard, if this session discharges then abandons this RTP packet with corresponding conversation recording deletion in the storage area;
4, the language data process module is analyzed the rtp packet of catching, and extracts the voice flow record that relevant parameters deposits this conversation recording correspondence in, passes to the speech quality evaluation module and further analyzes;
5, the speech quality evaluation module is carried out time delay, throughput, the isoparametric calculating of MOS value to the pairing voice flow record of session, calculates data delay, packet loss, shake and shows its result.
Should can be widely used in the fields relevant such as network measure, network monitor, network performance analysis based on the VOIP voice quality monitoring system and the detection method of Session Initiation Protocol with traffic engineering.
Claims (8)
1. VOIP voice quality monitoring system based on Session Initiation Protocol, comprise: speech data collection module, session tracking module, language data process module, speech quality evaluation module, it is characterized in that, speech data collection module is based on INET socket layer building, connect network interface card, collection is caught by network interface card and is operated the data message copy that the system kernel space obtains, and extracts the SIP packet; The session tracking module carries out the session coupling to the SIP packet, and whether the state exchange mechanism identification session that utilizes the SIP session to set up is set up or discharged; Set up state for session, language data process module parses media negotiation information is determined the characteristic value of this session, timestamp is stamped in rtp voice packet header formed the voice flow control head, according to characteristic value the voice flow control head of same session is connected into chained list; For the conversation end state, then notify the language data process module to obtain the voice flow control head, the linked list head with the voice flow control head when conversation end imports the speech quality evaluation module into; The rtp voice packet header of each session of speech quality evaluation module analysis therefrom obtains the information of voice flow, and the e-Model model that calls in this module calculates the voice messaging quality.
2. VOIP voice quality monitoring system according to claim 1, it is characterized in that: the session tracking module comprises sip message resolution module and session matching module, the sip message resolution module reads the sip data flow by descriptor fdr in parent process, and parsing sip character stream, the content of resolving is inserted in the Record structure sip message in the session matching module coupling Record structure.
3. VOIP voice quality monitoring system according to claim 1, it is characterized in that, the language data process module comprises I/O module, link module and packet handing module, and the I/O module uses the I/O multiplexing to be responsible for the reception of coordination network packet and SIP session information; Packet handing module is responsible for communicating by letter of this module and session tracking module and speech data collection module and data flow that format is received; Link module maintain sessions information chained list, and the rtp bag that will finish when conversing passes to the speech quality evaluation module.
4. VOIP voice quality monitoring system according to claim 1 is characterized in that, described speech quality evaluation module is calculated the delay of measurement voice messaging data Quality, packet loss, shake.
5. a VOIP speech quality detection method is characterized in that, comprises the steps:
B1, network interface card is set to promiscuous mode, and the configuration-system initial parameter is set up the data buffer zone, catches network packet;
B2, speech data collection module judges whether the packet catch comprises the Session Initiation Protocol content, if execution in step B3 then; Otherwise execution in step B6;
B3, the session tracking module is analyzed the SIP packet, and follows the tracks of talking state, judges whether a session sets up success, if successfully then execution in step B5, otherwise execution in step B4;
B4, the session tracking module deposits SIP packet data session information in the conversation message chained list, handles next SIP packet;
B5, the language data process module is extracted the media communication information of session negotiation in the SIP packet: port numbers and coded system, execution in step B7;
B6, whether speech data collection module judgment data bag is the RTP data flow, if execution in step B7 then, otherwise abandon this packet;
B7, the language data process module is carried out the maintenance of session voice data flow to the RTP packet, if this session discharges then abandons this bag, otherwise execution in step B8;
B8, the speech quality evaluation module is analyzed the session voice data, therefrom obtains the information of voice flow: network delay, packet loss, type of coding; The formula R=R0-Icodec-Idelay-Ipacketloss that then each impairment value is brought into the unconcerned type of e-Model of simplification obtains the R value; Determine the Mos value according to the value of R, for R＜0:MOS=1
For 0≤R≤100:MOS=1+0.035 * R+R (R-60) (100-R) * 7 * 10
-6
For R＞100:MOS=4.5, wherein, R0 is basic signal to noise ratio, and Icodec is the encoding and decoding distortion, and Idelay is the average delay damage, and Ipacketloss is the packet loss in certain period.
6. VOIP speech quality detection method according to claim 5, it is characterized in that, described in the step B3 SIP data packet analysis is specifically comprised, analyze SIP signaling content, carry out Session Initiation Protocol identification, extract the media communication information of session negotiation in the SIP packet, comprise port numbers and coded system.
7. VOIP speech quality detection method according to claim 6, it is characterized in that, step B3 further comprises, as the SIP signaling is that signaling is set up in session, then in storage area, create a conversation recording, store this SIP packet data session information, handle next packet, whether the judgment data bag is the RTP packet.
8. VOIP speech quality detection method according to claim 7, it is characterized in that, step B3 further comprises, language data process module analysis RTP packet, as be the SIP release signaling, then, the RTP packet is carried out the session voice data flow safeguard, if this session discharges then abandons this RTP packet with corresponding conversation recording deletion in the storage area.
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| CN101437032A (en) | 2009-05-20 |
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