CN101150763B - A terminal and method for testing real time service transmission performance of WiMAX network - Google Patents
A terminal and method for testing real time service transmission performance of WiMAX network Download PDFInfo
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Abstract
The invention discloses a method for testing realtime service transmission performance of the WiMAX network, belongs to the wireless broadband access field. In the method, a mobile terminal establishes RTSP media session connection with a streaming media server; the streaming media server receives a session request according to the mobile terminal, determins corresponding coding speed, encapsulates realtime UDP messages, then sends to the mobile terminal through the WiMAX network, the mobile terminal analyzes received UDP messages; when the network connection is time-out, the mobile terminal stops the RTSP media session connection with the streaming media server, calculates performance indexes of the transmission rate, the packet loss rate, time delay and the time delay jitter according to the analysis result of the UDP messages. The invention also discoses a terminal for testing realtime service transmission performance of the WiMAX network. The invention acquires performance indexes such as the transmission rate, the packet loss rate, time delay and the time delay jitter, etc., under different Qos allocation, improves testability and analyzability.
Description
Technical field
The present invention relates to WiMAX and insert field, the terminal and the method for particularly a kind of test WiMAX (WorldInteroperability for Microwave Access, global interoperability inserting of microwave) network real-time transport service performance.
Background technology
WiMAX is a kind of WiMAX access technology that occurs in recent years, and its full name is global interoperability inserting of microwave standard.Characteristics such as WiMAX uses multi-carrier modulation technology, data service at a high speed can be provided, and have the frequency spectrum resource utilization rate height, and coverage is big.The cost of WiMAX is relatively low, can facilitate for individual, family, enterprise, good mobile multimedia broadband service and wireless data transmission at a high speed.
As shown in Figure 1; As a kind of reference model end to end, the network configuration of WiMAX system comprises customer mobile terminal MSS (Mobile Service Subcrible), access service network ASN (Access Service Network), connectivity serving network CSN (Connection Service Network) and application service provider ASP (Application Service Provider) network composition at least.Wherein, ASN comprises the base station BS (Base Station) of handling IEEE 802.16d/e protocol air interface message and realizes the access gateway AG W (Access Gateway) of ASN to the CSN interface function, can be connected on a plurality of CSN; CSN is made up of router, authentication, mandate, charging AAA agency or server, dynamic host allocation protocol Dynamic Host Configuration Protocol server, customer data base, internet gateway device, connects for WiMAX user provides IP, and is connected on the ASP through Ethernet; ASP provides various upper layer application data services for WiMAX user, like streaming media service, FTP (FTP) service, HTTP (Hypertext Transmission Protocol) service and Email (Email) service etc.With regard to network interface, the R1 interface is the interoperability interface between MSS and the BS, and R3 is the logic interfacing between AGW and the CSN, and R6 is the interoperability interface between BS and the AGW.
Compare with existing mobile communication technology, the WiMAX technology can provide higher data rate, stronger data service disposal ability, and carrying the upper layer data business through the WiMAX technology becomes a kind of more economy means flexibly.In order to support the various application data services of MSS visit ASP side; The IEEE802.16d/e protocol definition 4 kinds of dispatching services types; Be divided into unsolicited grant service UGS, real-time inquiry business RTPS, non real-time inquiry business nRTPS and Best-Effort service BE by dispatching priority order from high to low, every kind of type of service has different bandwidth application ways, bandwidth scheduling characteristics and Qos (Quality of Service) parameter set.Wherein, RTPS is used to support the variable bit rate real time business, is for the business demand that satisfies dynamic change designs, like streaming media service etc.BS is that RTPS provides periodic unicast inquiry request chance, and bans use of other contention requests chances and incidentally request.RTPS requires to have minimum assurance speed MRR, and to maximum delay and maximum delay jitter-sensitive.
Along with popularizing and multimedia technology application on the internet of the Internet, it is one of professional that streaming media service becomes main RTPS that ASP provides.Streaming Media is meant the continuous time-base media that in Internet, uses the stream transmission technology, like audio frequency, video or multimedia file.
As shown in Figure 2, the protocol stack of Streaming Media mainly comprises realtime transmission protocol RTP, RTCP Real-time Transport Control Protocol RTCP and real-time Transmission session protocol RTSP, and they all are positioned on transmission control protocol/User Datagram Protocol TCP/UDP.
Wherein RTP is used for Internet to go up a kind of host-host protocol to multimedia data stream.RTP be applicable to one to one or the transmission situation of one-to-many under work, its objective is provides temporal information and realizes that stream is synchronously.RTP uses UDP to transmit data usually, when application program begins a RTP session, will use two ports: give RTP for one, give RTCP for one.RTP itself is each parameter of Measurement Network not, and flow control or congested control are not provided yet, and it relies on RTCP that these services are provided.Usually the RTP algorithm not as one independently network layer realize, but as the part of application code.
Rtcp protocol also is positioned on the udp protocol, with RTP flow control and congested control service is provided, and periodically transmits RTCP and wraps to streaming media server.Owing to contain quantity of data packets, the quantity of data packets of losing, out of order quantity of data packets and the time information that is used to estimate propagation delay time that receives to some extent in the RTCP bag; Make streaming media server can utilize these information dynamically to change transmission rate, even change PT Payload Type;
The RTSP agreement is the multimedia programming agreement of a client/server, and it is positioned on the Transmission Control Protocol, can the transmission of control flows media data on IP network, the remote control function of stream medium data is provided simultaneously.In addition, the RTSP agreement is again an application layer protocol, and an a whole set of fluidisation service based on Internet is provided.
The method of the existing service transmission performance of testing stream media in the WiMAX network normally adopts a client end of playing back request streaming media playing, by streaming media server the form of Media Stream with the RTP bag is sent to MSS through the WiMAX network.MSS at first carries out the buffering of a period of time, begins streaming media playing then, is judged the real time service transmission performance of the WiMAX network between end-to-end according to the streaming media playing effect of MSS by the user.But it is poor that above-mentioned method of testing can be analyzed, and can't accurately add up the indexs such as packet loss, transmission rate, time delay and delay variation of MSS, can't quantize comparative analysis to the test case under the different Q os configuration.
Summary of the invention
Technical problem to be solved by this invention is, a kind of terminal and method of testing real time service transmission performance of WiMAX network is provided, and obtains the test index of MSS simplely, improved the accuracy of test greatly.
In order to address the above problem, the invention discloses, a kind of method of testing WiMAX network real-time service transmission performance, this method may further comprise the steps:
A: the network of portable terminal through the base station connects, obtain medium methods availalbe and initialization descriptor after, dispose corresponding qos parameter, foundation is connected with the real-time Transmission session protocol RTSP media session of streaming media server;
B: said streaming media server distributes corresponding resource according to the session information from the conversation request that said portable terminal is received; Confirm respective coding speed; Use user datagram protocol UDP to encapsulate real-time UDP message; Send to said portable terminal through the WiMAX network then, said portable terminal is resolved according to udp protocol the UDP message of receiving;
C: after the network connection is overtime; Said terminating mobile terminal is connected with the real-time Transmission session protocol RTSP media session of said streaming media server; According to analysis result, calculate transmission rate, packet loss, time delay and delay variation performance index then to said UDP message.
Said method, among the said step B, the real-time UDP message that said streaming media server sends is realtime transmission protocol RTP message or RTCP Real-time Transport Control Protocol RTCP message;
It is RTP message or RTCP message that said portable terminal is resolved the real-time UDP message of receiving according to different port numbers, and then according to stream media protocol RTP message or RTCP message is done further parsing.
If said portable terminal parses the stem and the data division of RTP message from the data division of said UDP message, wherein the stem of RTP message comprises version number, fills number, flag, loadtype, serial number, timestamp and the synchronisation source identifier information in flag, extension bit, contribution source.
If said portable terminal parses sender's message of stem, sender and one or more Receiver Report piece of RTCP message from the data division of said UDP message, wherein:
The stem of RTCP message comprises version number, fills flag, accepts report blocks counting and contribution source information;
Sender information comprises the message number of NTP timestamp, RTP timestamp, transmission and the byte number of transmission.
In the method for above-mentioned test WiMAX network real-time service transmission performance, said portable terminal calculates the size of transmission rate to the numerical value of the Length field in the UDP message of the receiving statistics that superposes;
If the UDP message received of said portable terminal is the RTP message, then the number to this message adds 1 cumulative statistics in real time, and the moment of receiving this message is carried out real time record, calculates the size of packet loss, time delay and delay variation performance index.
The invention also discloses; A kind of terminal of testing real time service transmission performance of WiMAX network; This terminal comprises media session module, UDP message receiver module, UDP message process unit; Wherein UDP message receiver module links to each other with the UDP message process unit, and the media session module is set up network with streaming media server and is connected;
The media session module is used for the terminal use and sends the media session request to streaming media server, and the answer of receiving stream media server and network connect regularly, after the acquisition medium SDP description document, disposes corresponding qos parameter;
UDP message receiver module is used to receive the UDP message, and according to the form of udp protocol the UDP message is resolved;
The UDP message process unit; Be used to receive the UDP message data part that UDP message receiver module is transmitted; Resolve according to RTP message format that defines in the stream media protocol or RTCP message format; And provide interface function to supply each module to call, be used for calculating respectively transmission rate, packet loss, time delay and delay variation performance index.
In the above-mentioned terminal, said UDP message process unit comprises RTP message processing module (MPM), RTCP message processing module (MPM) and record statistical module, wherein
The RTP message processing module (MPM) is used to receive the UDP message data part that UDP message receiver module is transmitted, and resolves according to the RTP message format that defines in the stream media protocol;
The RTCP message processing module (MPM) is used to receive the UDP message data part that UDP message receiver module is transmitted, and resolves according to the RTCP message format that defines in the stream media protocol;
The record statistical module is used to provide interface function to supply UDP message receiver module, RTP message processing module (MPM), RTCP message processing module (MPM) to call, and is used for calculating respectively transmission rate, packet loss, time delay and delay variation performance index.
The stem of said UDP message comprises source port, target port, length and verification and information, and said UDP message receiver module is according to wherein target port information, and according to the form of udp protocol the UDP message is resolved.
The stem of the RTP message that said RTP message processing module (MPM) is received comprises version number, fills number, flag, loadtype, serial number, timestamp and the synchronisation source identifier information in flag, extension bit, contribution source.
The RTCP message that said RTCP message processing module (MPM) receives comprises sender's message of stem, sender and one or more Receiver Report piece, wherein
The stem of RTCP message comprises version number, fills flag, accepts report blocks counting and contribution source information;
Sender information comprises the message number of NTP timestamp, RTP timestamp, transmission and the byte number of transmission.
In the above-mentioned terminal, said record statistical module also is used to add up the numerical value of each statistical variable, comprises sum and total size of the paid-in RTP message of terminal use of the paid-in RTP message of sum, terminal use of the RTP message that streaming media server has sent;
This module also is used to write down the numerical value of each record variable simultaneously; The time of reception that comprises delivery time that the terminal use sets up session connection and keep the RTP message that duration, terminal use receive, RTP message that the terminal use receives, and they are kept in the global structure variable.
To above-mentioned terminal, network connect overtime after, said record statistical module is to the numerical value of the Length field in the UDP message of the receiving statistics that superposes, the size of calculating transmission rate;
If the UDP message received is the RTP message, then the number to this message adds 1 cumulative statistics in real time, and the moment of receiving this message is carried out real time record, calculates the size of packet loss, time delay and delay variation performance index.
The method of the present invention through the Streaming Media message of receiving is resolved obtains performance index such as transmission rate, packet loss, time delay and delay variation under the test case under the different Q os configuration, improved testability and analyticity greatly.
Description of drawings
Fig. 1 is a WiMAX system configuration sketch map;
Fig. 2 is the sketch map of the protocol stack of streaming media service;
Fig. 3 is the modular structure sketch map of the mobile terminal MS S of employing technical scheme of the present invention;
Fig. 4 is for adopting the flow chart of technical scheme testing real time service transmission performance of WiMAX network of the present invention.
Embodiment
Below in conjunction with accompanying drawing the technical scheme that the present invention adopted is done further explain.
Multithreading has been used at a kind of terminal of testing real time service transmission performance of WiMAX network, can create a TCP process and a UDP process, is respectively applied for to set up flow media session connection and receiving stream media message; Wherein the TCP process comprises a RTSP thread, and the UDP process then comprises a UDP thread, real time record statistics thread, a RTCP thread and an a plurality of RTP thread.
The concrete modular structure at this terminal is as shown in Figure 3; Comprise media session module, UDP message receiver module, UDP message process unit; Wherein the UDP message process unit comprises RTP message processing module (MPM), RTCP message processing module (MPM) and record statistical module again; UDP message receiver module links to each other with RTP message processing module (MPM), RTCP message processing module (MPM) and the record statistical module of UDP message process unit respectively; The record statistical module also links to each other with RTP message processing module (MPM), RTCP message processing module (MPM) respectively, and the media session module is carried out network with streaming media server and is connected, and wherein each functions of modules is following:
The media session module: be used for the terminal use and send the media session request to streaming media server, the answer of receiving stream media server and network connect regularly.This module is set up in the session connection process through WiMAX network and streaming media server, observes RTSP agreement and Transmission Control Protocol.
UDP message receiver module: this module comprises a UDP thread; Be used for the UDP message that the receiving stream media server sends through the WiMAX network; And the UDP message is resolved according to the form of udp protocol; Wherein resolving is according to the destination port number in the UDP message, judges that just the data division in the UDP message belongs to RTP message or RTCP message, is transmitted to the RTP message processing module (MPM) respectively then and the RTCP message processing module (MPM) is handled.
UDP message process unit: comprise RTP message processing module (MPM), RTCP message processing module (MPM) and record statistical module.
The RTP message processing module (MPM): this module comprises one or more RTP threads, is used to receive the UDP message data part that UDP message receiver module is transmitted, and resolves according to the RTP message format that defines in the stream media protocol, and judges the legitimacy of RTP message format.
The RTCP message processing module (MPM): this module comprises a RTCP thread, is used to receive the UDP message data part that UDP message receiver module is transmitted, and resolves according to the RTCP message format that defines in the stream media protocol, and judges the legitimacy of RTCP message format.Said RTCP message comprises RTCP SR Sender Report (Sender Report) message of stem, sender information and one or more Receiver Report piece (Receiver Report Block).
The record statistical module: this module comprises a real time record statistics thread; Be used to write down the numerical value of each record variable; The time of reception that comprises delivery time that the terminal use sets up session connection and keep the RTP message that duration, terminal use receive, RTP message that the terminal use receives, and they are kept in the global structure variable; This module also is used to add up the numerical value of each statistical variable simultaneously; The sum and the total size of the paid-in RTP message of terminal use that comprise the paid-in RTP message of sum, terminal use of the RTP message that streaming media server has sent, and they are kept in the global structure variable.In addition, this module also provides interface function to supply UDP message receiver module, RTP message processing module (MPM), RTCP message processing module (MPM) to call, and is used for calculating respectively performance index such as transmission rate, packet loss, time delay and delay variation.
Wherein transmission rate is a unit with MBit/s usually, refer in certain time interval, and total size of the RTP message that the terminal use received in the unit interval, transmission rate is big more, shows that the network real-time service transmission performance is good more.
Packet loss is meant in the certain hour interval, the ratio of the RTP message number that the number of the RTP message that the terminal use receives and streaming media server send, and packet loss is low more, shows that the network real-time service transmission performance is good more.
Time delay, the moment that is meant the RTP message that the terminal use receives and streaming media server are sent the moment poor of this RTP message, and time delay is more little, shows that the network real-time service transmission performance is good more.
Delay variation is meant the absolute value of difference of time delay and the average delay of some RTP messages that the terminal use receives.
This module statistics that can superpose to the numerical value of the Length field in the UDP message of receiving is to calculate the size of transmission rate; Can also add 1 cumulative statistics in real time to the number of the RTP message received, the moment of the RTP message received carried out real time record, to calculate the size of packet loss, time delay and delay variation performance index, to satisfy the needs of real-time service transmission performance evaluation.
The process of the testing real time service transmission performance of WiMAX network of present embodiment comprises the steps, and is as shown in Figure 4:
Step 401: behind the professional establishment of connection of the up-downgoing of MSS realization and WiMAX network, its media session module is sent RTSP to streaming media server and is selected to ask (Options Request) message, obtains the methods availalbe of streaming media server;
Step 402: after streaming media server is received above-mentioned request message, select response (Options Response) message, carry all methods availalbe information that streaming media server can provide in this message to the media session module feedback RTSP of MSS;
After the media session module of step 403:MSS is received above-mentioned response message, send RTSP to streaming media server and describe request (Describe Request) message, require to obtain the media initialize descriptor that streaming media server provides;
Step 404: streaming media server feedback RTSP describes the media session module that response (Describe Response) message is given the terminal use; The media initialize descriptor of feedback mainly is meant Session Description Protocol SDP (Session Description Protocol) file, and this document has comprised information such as type and the bandwidth of medium;
The media session module of step 405:MSS can dispose corresponding qos parameter after obtaining medium SDP description document;
In the present embodiment, dispose the qos parameter of real-time inquiry business RTPS;
The media session module of step 406:MSS is sent RTSP conversation request (Setup Request) message to streaming media server; The attribute and the transmission mode of session are set in this message; Remind streaming media server to set up session; Simultaneously, initialization " terminal use sets up session connection and keeps duration " is KEEP_SESSION_PERIOD;
Step 407: after streaming media server is received conversation request message, set up session connection, and reply (Setup Response) message, carry Session ID and session related information in this message to the media session module feedback RTSP of MSS session with MSS;
After the media session module of step 408:MSS is received response message, start a timer T1, and timer duration is the set KEEP_SESSION_PERIOD of step 406;
In the step 409, whether the media session module cycle criterion T1 timer of MSS is overtime, if then change step 417 over to, otherwise carry out next step;
The media session module of step 410:MSS is sent the RSTP expansion to streaming media server and is play (Play) message; The media resource that request playing stream media server provides; Each record variable of initialization or statistical variable initial value are 0 simultaneously, and wherein each variable name and variable code name are as shown in table 1:
Table 1
| Variable name | The variable code name |
| The delivery time of the RTP message that MSS receives | T_SENT_RTP |
| The time of reception of the RTP message that MSS receives | T_RCV_RTP |
| The sum of the RTP message that streaming media server has sent | TOTAL_SENT_RTP_NUM |
| The sum of the paid-in RTP message of MSS | TOTAL_RCV_RTP_NUM |
| Total size of the paid-in RTP message of MSS | TOTAL_RCV_RTP_SIZE |
Step 411: after streaming media server was received the media play request of MSS, structure sent to the UDP message receiver module of MSS then through the RTP message or the RTCP message of UDP encapsulation through the WiMAX network;
The UDP message receiver module of step 412:MSS is resolved according to table 2 the UDP message of receiving, and the destination slogan in the UDP message is judged, if the port numbers of RTP message then continues next step, otherwise changes step 415 over to;
Table 2
| Source?port | Destination?port |
| Length | Checksum |
| UDP?Data | ? |
Stem in the table 2 comprises source port Source port, target port Destination port, length L ength and verification and four fields of Checksum, and the implication of each field is following:
Source port: be the long field of 16Bit, be used to represent the port numbers of transmit leg;
Destination port: be the long field of 16Bit, be used to represent recipient's port numbers.Because it is different receiving the port numbers of RTP message and RTCP message, is RTP message or RTCP message with this content that can distinguish among the UDP Data;
Length: for the long field of 16Bit, be used to represent header length and data length sum, with Byte, promptly 8Bit is a unit;
Checksum: be the field that 16Bit is long, the verification that is used to identify each field of UDP message with;
In addition, the UDP Data in the table 2 is elongated field, because header length is known as 8Byte according to the agreement regulation, therefore the numerical value with the Length field deducts the length that 8Byte is exactly UDP Data.
The interface function that the UDP message receiver module of step 413:MSS calls in the record statistical module carries out overlap-add operation to Length value in the UDP message; As " total size of the paid-in RTP message of terminal use ", i.e. TOTAL_RCV_RTP_SIZE=TOTAL_RCV_RTP_SIZE+Length; Simultaneously, " sum of the paid-in RTP message of terminal use " added up to add 1 operation, i.e. TOTAL_RCV_RTP_NUM=TOTAL_RCV_RTP_NUM+1; Simultaneously, the record current time is T_RCV_RTP, as " time of reception of the RTP message that the terminal use receives ", it is kept in the global structure variable, then the data division in the UDP message is forwarded to the RTP message processing module (MPM) and does further processing;
The RTP message processing module (MPM) of step 414:MSS is done further parsing to UDP message data part according to the structure in the table 3; The interface function that calls the record statistical module is T_SENT_RTP with the records of values of Time-stamp field; And be kept in the global structure variable, change step 409 then over to;
Table 3
| V | P | X | CC | ?M | ?PT | Sequence?Number |
| Timestamp | ? | ? | ? | ? | ? | ? |
| SSRC?Identifier | ? | ? | ? | ? | ? | ? |
| RTP?Data | ? | ? | ? | ? | ? | ? |
Number CC, flag M, loadtype PT, serial number Sequence Number, time stamp T imestamp and the synchronisation source identifier SSRC Identifier that stem in the table 3 comprises the V of version number, fill flag P, extension bit X, CSRC be totally 9 fields, and each field implication is following:
V: be the long field of 2Bit, be used to identify the version number of RTP, default value is 2;
P: be the field that 1Bit is long, whether fill character at the end that is used to identify the RTP message;
X: whether be the long field of 1Bit, being used to be identified at the header back has extended head;
CC: be the long field of 4Bit, be used to identify the number of contribution source CSRC;
M: be the field that 1Bit is long, it is explained by concrete agreement regulation, is used for allowing the important incident of mark in bit stream, like the frame scope etc.;
PT: for the long field of 7Bit, be used to identify the loadtype that is delivered, PT sign indicating number and coded system corresponding relation are defined by additional specifications, and the recipient decodes to the data of accepting to come with this field;
Sequence Number: be the long field of 16Bit, transmit leg whenever sends out a RTP message and just order is added 1, and this field side of being received is used for the loss situation of detection messages;
Timestamp: be the long field of 32Bit, the sampling time of this this RTP message of field reflection;
SSRC Identifier: be the long field of 32Bit, be used to identify RTP message promoter's the source identifier that comes.
In addition, the RTP Data in the table 3 is elongated field, and its concrete length is decided according to PT sign indicating number corresponding load type and corresponding code encoding/decoding mode by transmit leg;
The UDP message receiver module of step 415:MSS is forwarded to the RTCP message processing module (MPM) with the data division in the UDP message and does further processing;
The RTCP message processing module (MPM) of step 416:MSS is done further parsing to UDP message data part according to the structure in the table 4; The interface function that calls the record statistical module is TOTAL_SENT_RTP_NUM with " Sender ' s packetcount " field statistics; As " sum of the RTP message that streaming media server has sent ", change step 409 then over to;
Table 4
| ?V | ?P | ?RC | ?PT | Length |
| ?SSRC | ? | ? | ? | ? |
| ?NTP?Timestamp，most?significant?word | ? | ? | ? | ? |
| ?NTP?Timestamp，least?significant?word | ? | ? | ? | ? |
| ?RTP?Timestamp | ? | ? | ? | ? |
| ?Sender′s?packet?count | ? | ? | ? | ? |
| ?Sender′s?octet?count | ? | ? | ? | ? |
| ?Receiver?Report?Block | ? | ? | ? | ? |
Heading in the table 4 comprises the V of version number, fills flag P, receives report blocks counting RC and SSRC, and each field implication is following:
V: be the long field of 2Bit, the same meaning of RTP bag in meaning in the RTCP bag and the table 2;
P: be the field that 1Bit is long, whether fill character at the end that is used to identify the RTCP message;
RC: be the long field of 5Bit, be used for identifying the number of the contained reception report blocks of RTCP message;
PT: being the field of 8 bit long, is constant 200 here, is the message of RTCP SR in order to the sign type;
Length: being the long field of 16Bit, being used to identify the total length of other field except that V, P, RC, PT and Length, is unit with 32Bit;
SSRC: be the long field of 32Bit, be used to identify RTCP SR message promoter's the source identifier that comes;
Sender information in the table 4 comprises that the message of NTP timestamp NTP Timestamp, RTP timestamp RTP Timestamp, transmission counts the byte number Sender ' s octet count of Sender ' s packet count, transmission, and each field implication is following:
NTP Timestamp: be the field that 64Bit is long, the accurate clock when being used to identify RTCP SR and sending constantly, for example quartz clock and global position system GPS etc., if can't obtain accurate clock constantly, it is zero that this field then is set;
RTP Timestamp: for the long field of 32Bit, with the above corresponding synchronization of NTP Timestamp, still with table 2 in the tool Timestamp of RTP message identical unit and side-play amount are arranged;
Sender ' s packet count: be the field that 32Bit is long, be used for sign from beginning to transmit the RTP message when this RTCP SR message produces, transmit leg refers to streaming media server here, the RTP message total of transmission;
Sender ' soctet count: be the field that 32Bit is long, be used for sign from beginning to transmit the RTP message when this RTCP SR message produces, transmit leg refers to streaming media server here, and the total amount of byte of the RTP message of transmission does not wherein comprise stem and filling;
The media session module of step 417:MSS is sent RSTP to streaming media server and is stopped request (Teardown Request) message, and session connection is closed in request;
Step 418: after streaming media server is received and stopped request message, stop replying (Teardown Response) message to the media session module feedback RSTP of MSS, and close session connection:
Each record variable and the statistical variable of record statistical module reading and saving in the global structure variable of step 419:MSS uses the formula in the table 5 to calculate transmission rate, packet loss, time delay and delay variation.
Table 5
| Statistical item | Computing formula |
| Transmission rate | (8*TOTAL_RCV_RTP_SIZE/(KEEP_SESSION_PERIOD/1000) |
| Packet loss | (TOTAL_SENT_RTP_NUM-TOTAL_SENT_RCV_NUM) |
| ? | /TOTAL_SENT_RTP_NUM |
| Time delay | T_RCV_?RTP-T_SENT_RTP |
| Delay variation | After calculating the mean value of all time delays, deduct certain time delay again, ask absolute value. |
Above-mentioned steps 501 to 504 adopts existing techniques in realizing, and wherein, if MSS and streaming media server as offered are good, then the option response of the option of step 501 request and step 502 can be ignored; If the terminal use can pass through other approach, for example HTTP request or the like, then the describe response of the describe of step 503 request and step 504 also can be ignored.
The above only is a most preferred embodiment of the present invention, is not to be used for protection scope of the present invention.All within spirit of the present invention and principle, any modification of being made all should be included within protection scope of the present invention.
Claims (11)
1. a method of testing WiMAX network real-time service transmission performance is characterized in that, this method may further comprise the steps:
A: the network of portable terminal through the base station connects, obtain medium methods availalbe and initialization descriptor after, dispose corresponding qos parameter, foundation is connected with the real-time Transmission session protocol RTSP media session of streaming media server;
B: said streaming media server distributes corresponding resource according to the session information from the conversation request that said portable terminal is received; Confirm respective coding speed; Use user datagram protocol UDP to encapsulate real-time UDP message; Send to said portable terminal through the WiMAX network then, said portable terminal is resolved according to udp protocol the UDP message of receiving;
C: after the network connection is overtime; Said terminating mobile terminal is connected with the real-time Transmission session protocol RTSP media session of said streaming media server; According to analysis result, calculate transmission rate, packet loss, time delay and delay variation performance index then to said UDP message;
Wherein, among the said step B, it is RTP message or RTCP message that said portable terminal is resolved the real-time UDP message of receiving according to different port numbers, and then according to stream media protocol RTP message or RTCP message is done further parsing.
2. the method for claim 1 is characterized in that:
Said portable terminal parses the stem and the data division of RTP message from the data division of said UDP message, and wherein the stem of RTP message comprises version number, fills number, flag, loadtype, serial number, timestamp and the synchronisation source identifier information in flag, extension bit, contribution source.
3. the method for claim 1 is characterized in that:
Said portable terminal parses sender's message of stem, sender and one or more Receiver Report piece of RTCP message from the data division of said UDP message, wherein:
The stem of RTCP message comprises version number, fills flag, accepts report blocks counting and contribution source information;
Sender information comprises the message number of NTP timestamp, RTP timestamp, transmission and the byte number of transmission.
4. like each described method of claim 1 to 3, it is characterized in that:
Said portable terminal calculates the size of transmission rate to the numerical value of the Length field in the UDP message of the receiving statistics that superposes;
If the UDP message received of said portable terminal is the RTP message, then the number to this message adds 1 cumulative statistics in real time, and the moment of receiving this message is carried out real time record, calculates the size of packet loss, time delay and delay variation performance index.
5. the terminal of a testing real time service transmission performance of WiMAX network; It is characterized in that; This terminal comprises media session module, UDP message receiver module, UDP message process unit; Wherein UDP message receiver module links to each other with the UDP message process unit, and the media session module is set up network with streaming media server and is connected;
The media session module is used for the terminal use and sends the media session request to streaming media server, and the answer of receiving stream media server and network connect regularly, after the acquisition medium SDP description document, disposes corresponding qos parameter;
UDP message receiver module is used to receive the UDP message, and according to the form of udp protocol the UDP message is resolved;
The UDP message process unit; Be used to receive the UDP message data part that UDP message receiver module is transmitted; Resolve according to RTP message format that defines in the stream media protocol or RTCP message format; And provide interface function to supply each module to call, be used for calculating respectively transmission rate, packet loss, time delay and delay variation performance index.
6. terminal as claimed in claim 5 is characterized in that:
Said UDP message process unit comprises RTP message processing module (MPM), RTCP message processing module (MPM) and record statistical module, wherein
The RTP message processing module (MPM) is used to receive the UDP message data part that UDP message receiver module is transmitted, and resolves according to the RTP message format that defines in the stream media protocol;
The RTCP message processing module (MPM) is used to receive the UDP message data part that UDP message receiver module is transmitted, and resolves according to the RTCP message format that defines in the stream media protocol;
The record statistical module is used to provide interface function to supply UDP message receiver module, RTP message processing module (MPM), RTCP message processing module (MPM) to call, and is used for calculating respectively transmission rate, packet loss, time delay and delay variation performance index.
7. like claim 5 or 6 described terminals; It is characterized in that; The stem of UDP message comprises source port, target port, length and verification and information, and said UDP message receiver module is according to wherein target port information, and according to the form of udp protocol the UDP message is resolved.
8. terminal as claimed in claim 6; It is characterized in that the stem of the RTP message that said RTP message processing module (MPM) is received comprises version number, fills number, flag, loadtype, serial number, timestamp and the synchronisation source identifier information in flag, extension bit, contribution source.
9. terminal as claimed in claim 6 is characterized in that, the RTCP message that said RTCP message processing module (MPM) receives comprises sender's message of stem, sender and one or more Receiver Report piece, wherein
The stem of RTCP message comprises version number, fills flag, accepts report blocks counting and contribution source information;
Sender information comprises the message number of NTP timestamp, RTP timestamp, transmission and the byte number of transmission.
10. terminal as claimed in claim 6 is characterized in that:
Said record statistical module also is used to add up the numerical value of each statistical variable, comprises sum and total size of the paid-in RTP message of terminal use of the paid-in RTP message of sum, terminal use of the RTP message that streaming media server has sent;
This module also is used to write down the numerical value of each record variable simultaneously; The time of reception that comprises delivery time that the terminal use sets up session connection and keep the RTP message that duration, terminal use receive, RTP message that the terminal use receives, and they are kept in the global structure variable.
11., it is characterized in that like claim 6 or 10 described terminals:
Network connect overtime after, said record statistical module is to the numerical value of the Length field in the UDP message of the receiving statistics that superposes, the size of calculating transmission rate;
If the UDP message received is the RTP message, then the number to this message adds 1 cumulative statistics in real time, and the moment of receiving this message is carried out real time record, calculates the size of packet loss, time delay and delay variation performance index.
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Families Citing this family (24)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN101488967B (en) * | 2009-01-14 | 2015-05-13 | 深圳市同洲电子股份有限公司 | Video transmission method, embedded monitoring terminal and monitoring platform server |
| WO2011026264A1 (en) * | 2009-09-01 | 2011-03-10 | 华为技术有限公司 | Method and apparatus for detecting multi-service performance in tunnel |
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| CN102075965A (en) * | 2009-11-24 | 2011-05-25 | 中国移动通信集团天津有限公司 | Mobile terminal performance evaluation method and device |
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| CN102571421A (en) * | 2011-12-22 | 2012-07-11 | 中兴通讯股份有限公司 | Method and device for detecting service quality of IP access network |
| CN102932378A (en) * | 2012-11-28 | 2013-02-13 | 河海大学 | RTP (Real-time Transport Protocol)-based stream media transmission method in 3G network |
| CN103167359B (en) * | 2013-03-27 | 2016-03-02 | 华为技术有限公司 | The transmission method of RTP Media Stream and device |
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| US9525629B2 (en) | 2013-12-13 | 2016-12-20 | Tencent Technology (Shenzhen) Company Limited | Method and apparatus for transmitting data packets |
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| CN106972904B (en) * | 2016-11-21 | 2019-04-30 | 深圳信息职业技术学院 | A kind of method and system that clock is synchronous |
| CN109769273A (en) * | 2018-12-07 | 2019-05-17 | 深圳市海派通讯科技有限公司 | Method and system based on the control VoWifi switching of RTP delay variation |
| CN112717380B (en) * | 2019-03-19 | 2023-04-07 | Oppo广东移动通信有限公司 | Network detection method and related device |
| CN110572708A (en) * | 2019-09-16 | 2019-12-13 | 武汉兴图新科电子股份有限公司 | Method for receiving RTP multiplexing stream with high performance |
| CN111263240B (en) * | 2020-02-12 | 2022-07-19 | 青岛海信宽带多媒体技术有限公司 | IPTV4K audio and video playing management method and device and display equipment |
| CN112367271B (en) * | 2020-09-25 | 2023-04-18 | 福建星网智慧科技有限公司 | AI-based congestion control feature extraction method, device, equipment and medium |
| CN113973089B (en) * | 2021-09-30 | 2024-01-26 | 阿里巴巴（中国）有限公司 | Streaming media transmission method, system, equipment and storage medium |
| CN114980300B (en) * | 2022-05-23 | 2024-02-06 | 昆山中科晶上信息技术有限公司 | Method for measuring and calculating industrial grade 5G end-to-end time delay distribution based on UDP protocol and terminal equipment |
| CN115459863A (en) * | 2022-08-23 | 2022-12-09 | 深圳市共进电子股份有限公司 | Network equipment performance test method, device, equipment and storage medium |
| CN115208808B (en) * | 2022-09-14 | 2023-01-24 | 北京智芯微电子科技有限公司 | Service quality testing method and device, chip equipment and storage medium |
Citations (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN1543164A (en) * | 2003-02-18 | 2004-11-03 | ���µ�����ҵ��ʽ���� | Server-based rate control in a multimedia streaming environment |
| CN1728657A (en) * | 2004-07-30 | 2006-02-01 | Ut斯达康通讯有限公司 | Load simulator and simulation system for stream media service |
| CN1852180A (en) * | 2005-12-12 | 2006-10-25 | 华为技术有限公司 | Multi-media service system performance testing method |
-
2007
- 2007-10-18 CN CN2007101514591A patent/CN101150763B/en not_active Expired - Fee Related
Patent Citations (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN1543164A (en) * | 2003-02-18 | 2004-11-03 | ���µ�����ҵ��ʽ���� | Server-based rate control in a multimedia streaming environment |
| CN1728657A (en) * | 2004-07-30 | 2006-02-01 | Ut斯达康通讯有限公司 | Load simulator and simulation system for stream media service |
| CN1852180A (en) * | 2005-12-12 | 2006-10-25 | 华为技术有限公司 | Multi-media service system performance testing method |
Non-Patent Citations (3)
| Title |
|---|
| H. Schulzrinne等.RTP: A Transport Protocol for Real-Time Applications.RFC3550.2003,第4页, 第9页, 第13-16页, 第21-32页, 第35-42页. * |
| 王春波 等.WiMAX与移动流媒体业务.电信科学 12.2006,(12),第78-80页2.1部分、图1-4. |
| 王春波 等.WiMAX与移动流媒体业务.电信科学 12.2006,(12),第78-80页2.1部分、图1-4. * |
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