CN109981527A - Method, apparatus, electronic equipment and the storage medium of association process - Google Patents

Abstract

Embodiments of the present invention provide a method, apparatus, electronic device, and storage medium for association processing. The method includes acquiring the protocol IP address/port number of the interconnection between the calling networks of at least one first media stream in the first time period, where the first media stream is the media that cannot be associated with the invitation INVITE message of the first time period flow, and obtain the caller identity ID and the caller IP address/port number of the INVITE message of at least one session initiation protocol SIP call flow in the second time period, the second time period is before the first time period and is the same as the first time period Adjacent; if the calling IP address/port number of the first media stream is consistent with the calling IP address/port number of the INVITE message in the second time period, then backfill the calling ID of the INVITE message in the second time period to first media stream. In the method, after the association between the INVITE message and the media stream fails, the secondary association is performed, and the voice quality evaluation is simply and economically realized without changing the protocol.

Description

Method, apparatus, electronic device and storage medium for association processing

technical field

The embodiments of the present invention relate to the field of communication technologies, and in particular, to a method, an apparatus, an electronic device, and a storage medium for association processing.

Background technique

With the large-scale commercial use of VoLTE (Voice Over LTE, voice service based on LTE network) network, VoLTE high-definition voice calls are more and more popular among users. It is very important to ensure smooth voice conversations and no dropped calls.

In order to further monitor and improve the VoLTE voice service quality, it is necessary to evaluate the voice service quality.

In the process of one session, the SIP (Session Initiation Protocol) process on the signaling plane is performed first, and then the media stream process on the media plane is performed. The time of the signaling plane process includes all media planes. process time.

The SIP call flow includes multiple SIP messages, starting with an INVITE (invite) message, indicating that the calling party invites the called party to make a voice call, and the INVITE message includes identification information: the calling party ID (Identification).

The media stream carries RTP (Real-time Transport Protocol, real-time transport protocol) and RTCP (Real-time Control Protocol, real-time transmission control protocol). The media stream includes multiple data packets. connection protocol) package.

The SIP call flow and the media stream carry different protocols, and through different interfaces, when evaluating the voice quality, it is necessary to first associate the SIP call flow with the media stream, and obtain the signaling and media plane information of a session before voice quality can be performed. Evaluate.

However, since the caller ID is not included in the data packet, it is difficult to associate the SIP call flow with the media flow.

In the prior art, the method of associating the SIP call flow with the media stream is:

When the terminal is in a session, the SIP flag is filled in the data packet on the media plane, so that when performing voice evaluation, the SIP flag in the packet can be quickly matched and associated with the SIP process to which the data packet belongs.

The disadvantages of the prior art are:

It requires a specific terminal to support the completion, the data packet structure needs to be changed, and the currently used protocol standard needs to be changed, which is difficult and costly to implement.

SUMMARY OF THE INVENTION

In view of the defects of the prior art, embodiments of the present invention provide a method, an apparatus, an electronic device, and a storage medium for association processing.

On the one hand, an embodiment of the present invention provides a method for association processing, the method includes:

obtaining the protocol IP address/port number of the interconnection between the calling networks of at least one first media stream in the first time period, where the first media stream is a media stream that cannot be associated with the INVITE message of the first time period, and obtain the calling identity ID and calling IP address/port number of the INVITE message of at least one session initialization protocol SIP call flow in the second time period, the second time period is before the first time period and is the same as the first time period. adjacent;

If the calling IP address/port number of the first media stream is consistent with the calling IP address/port number of the INVITE message of the second time period, the calling ID of the INVITE message of the second time period is backfilled to the first media flow.

On the other hand, an embodiment of the present invention provides an apparatus for association processing, and the apparatus includes:

The obtaining module is used to obtain the IP address/port number of at least one first media stream in the first time period, the first media stream is a media stream that cannot be associated with the INVITE message of the first time period, and obtains the second time The caller identity ID and the caller IP address/port number of the INVITE message of at least one SIP call flow in the segment, the second time period is before the first time period and adjacent to the first time;

The backfilling module is used for, if the calling IP address/port number of the first media stream is consistent with the calling IP address/port number of the INVITE message of the second time period, then the calling ID of the INVITE message of the second time period Backfill to the first media stream.

On the other hand, an embodiment of the present invention also provides an electronic device, including a memory, a processor, a bus, and a computer program stored in the memory and running on the processor, where the processor implements the above method when executing the program A step of.

On the other hand, an embodiment of the present invention further provides a storage medium on which a computer program is stored, and when the program is executed by a processor, the steps of the above method are implemented.

It can be seen from the above technical solutions that the method, apparatus, electronic device and storage medium for association processing provided by the embodiments of the present invention can improve the success of association by performing secondary association after the association between the INVITE message and the media stream fails. In the case of not changing the protocol, the voice quality assessment can be performed accurately, so that the voice quality assessment can be implemented simply and economically.

Description of drawings

1 is a schematic flowchart of a method for association processing provided by an embodiment of the present invention;

2 is a schematic diagram of an RTP database provided by another embodiment of the present invention;

3 is a schematic diagram of a SIP database provided by another embodiment of the present invention;

4 is a schematic flowchart of a method for association processing provided by another embodiment of the present invention;

5 is a schematic diagram of data association provided by another embodiment of the present invention;

6 is a schematic structural diagram of an apparatus for association processing provided by another embodiment of the present invention;

FIG. 7 is a schematic structural diagram of an electronic device according to another embodiment of the present invention.

Detailed ways

In order to make the purposes, technical solutions and advantages of the embodiments of the present invention clearer, the technical solutions in the embodiments of the present invention will be clearly described below with reference to the drawings in the embodiments of the present invention. Obviously, the described embodiments are the Inventive Embodiments are some of the embodiments, but not all of the embodiments.

The call establishment process of a session is as follows:

Process on the signaling plane: VoLTE uses SIP to establish, modify and delete sessions, the calling party sends a SIP message INVITE to the network side, and the SIP call signaling process begins, and the calling party and the called party complete the negotiation.

Media plane process: After the calling party and the called party complete the negotiation, the calling party and the called party establish a media stream and conduct communication exchange. The communication exchange method is to exchange RTP and RTCP data packets.

The call release process is:

Process on the signaling plane: one party sends a BYE (on-hook) message.

Media plane process: no longer exchange packets.

The SIP call process includes multiple SIP messages. It starts from the INVITE message and stops after 100trying (response), 180ring (ringing information), 200ok (success indication) and ACK (determination message). The INVITE message includes identification information. It is: caller ID (Identification, identification), caller IP address/port number.

The media stream includes multiple data packets, the data packets belong to IP packets, and the data packets include identification information, specifically: the calling IP address/port number, and the identification information of the data packets in the media streams of one session is the same.

Media streams include real-time transport protocol RTP streams and real-time transport control protocol RTCP streams.

Correspondingly, the data packets include RTP packets and RTCP packets. RTCP is a protocol used together with RTP. When the media plane of a session is started, two ports will be occupied at the same time, which are used by RTP and RTCP respectively.

The calling party and the called party periodically exchange RTP packets and RTCP packets. The size of the resources occupied by the RTP packets and the RTCP packets and the sending period may be different. Among them, the RTP packets encapsulate the voice data, that is, the calling voice of the calling party and the called party, and the RTCP packets encapsulate the voice data. Network status data, providing information about session quality or performance quality, such as the total number of RTP packets, the number of lost RTP packets, and the jitter of RTP packets.

Since the data packet does not include the calling ID, in the embodiment of the present invention, the calling IP address/port number is used as a common point for association.

If the INVITE message and the media stream belong to the same session, the calling IP address in the INVITE message is the same as the calling IP address in the data packet, and the calling port number in the INVITE message is the same as the calling port in the data packet. The numbers are also the same.

Optionally, the calling ID is the unique identifier of the terminal when the terminal subscribes to the network, and will not change over time. Optionally, the calling IP address and port number are the IP address assigned by the IMS network to the terminal when the terminal is attached to the IMS network IP address and port number. If disconnection occurs, the calling IP address and port number will be reassigned.

The port number is a unique identifier of a logical port used to distinguish services, and is in one-to-one correspondence with an IP address. Either the calling IP address or the port number can identify the identity of the current calling party.

That is to say, if the calling IP address/port number in the INVITE message is the same as the calling IP address/port number in the data packet, the calling ID in the INVITE message can be backfilled into the data packet, so that the The data packet is associated with the calling party and the called party, and the session is evaluated.

Voice quality assessment can be performed on either the RTP stream or the RTCP stream. Of course, voice evaluation can also be performed in combination with RTP streams or RTCP streams.

In practical applications, in the step of collecting the SIP call flow, the INVITE message may be lost, and the IP address of the data packet cannot be associated with the INVITE message, so the data packet cannot be backfilled, so the data packet is discarded, which is equivalent to the media. The voice service quality is not evaluated for some data packets on the face, which affects the accuracy of the voice service quality evaluation as a whole. This problem can be avoided by applying the method of the embodiment of the present invention.

FIG. 1 shows a schematic flowchart of a method for association processing provided by an embodiment of the present invention.

As shown in Figure 1, the method provided by the embodiment of the present invention specifically includes the following steps:

Step 11. Obtain the protocol IP address/port number of the interconnection between the calling networks of at least one first media stream in the first time period. The first media stream cannot be associated with the INVITE message of the first time period. media stream, and obtain the caller identity ID and the caller IP address/port number of the INVITE message of at least one session initiation protocol SIP call flow in the second time period, the second time period is before the first time period and is the same as first time adjacent;

Before this step, collect the SIP call flow in the second time period and the first time period on the signaling plane, and cache the collected INVITE message of the SIP call flow for the second time period and the media streams of the first time period are associated.

The media stream in each time period is associated with the SIP call flow. In the embodiment of the present invention, the association between the media stream in the first time period and the SIP call flow is taken as an example for description.

Optionally, the INVITE message is associated with the data packet, thereby associating the SIP call flow with the media stream.

Optionally, the association refers to finding out whether the calling IP address of the media stream within the first time period is consistent with the calling IP address of the INVITE message within the first time period, or finding the calling party of the media stream within the first time period. Whether the port number is the same as the port number of the INVITE message in the first time period.

If any one of the calling IP address and the port number is found to be consistent, it can be considered that the association is successful. The embodiment of the present invention takes the calling IP address as an example for description.

If the calling IP address of the media stream in the first time period is consistent with the calling IP address of the INVITE message, it is considered that the association is successful, and the calling ID of the INVITE message is backfilled into each data packet of the media stream.

If they are inconsistent, it is considered that no consistent INVITE message is collected within the first time period, so the INVITE message cannot be associated with the media stream to obtain the first media stream.

Optionally, the first time period and the second time period can be set according to the actual situation. For example, the first time period is (9:00-9:30) for 30 minutes, and the second time period is (8:30-9:30). 9:00] These 30 minutes.

Optionally, each first media stream in the first time period is associated with the SIP call flow in the second time period, respectively.

In the case that the association between the media stream in the first time period and the SIP call flow fails, an attempt is also made to associate with the SIP call flow in the second time period at another time.

Optionally, searching for the calling IP address of the INVITE message in the second time period according to the calling IP address of each first media stream.

If the caller does not turn off the network, leave the home, or turn on the airplane mode, etc., between the first time period and the second time period, the calling IP address/port number is usually unchanged, and it can be considered that the calling IP If the address/port number is the same, it is indeed the same caller.

Since there is no INVITE message associated with the media stream in the first time period, it can continue to find out whether the same caller initiates a session in the second time period.

Step 12: If the calling IP address/port number of the first media stream is consistent with the calling IP address/port number of the INVITE message in the second time period, backfill the calling ID of the INVITE message in the second time period to first media stream.

If there is an INVITE message in the second time period that matches the IP address of the first media stream in the first time period, it means that before the first time period, the same caller has initiated another session in the second time period. The INVITE message of the session corresponding to the second time period has been collected, and the collection of the INVITE message of the session corresponding to the first time period has been lost.

Optionally, the voice quality has a lot to do with time and location. Since the time is adjacent, even if the two sessions are different, it can be considered that the caller has not moved. The voice quality of the two sessions is similar. The caller ID of the INVITE message is backfilled into the data packet of the first media stream.

The voice quality evaluation is performed according to the SIP call flow corresponding to the INVITE message and the first media stream.

The SIP call flow corresponding to the INVITE message and the first media stream are correlated to correspond to a complete session, and voice quality evaluation is performed for the complete session.

If the IP addresses of the first media stream in the first time period and the INVITE message in the second time period do not match the IP addresses, it is considered that the first media stream cannot be associated, and the process ends.

In this embodiment of the present invention, when the association fails for the first time (the first media stream and the INVITE message of the first time period), a second association (the first media stream and the INVITE message of the second time period) is also performed. Improve the success rate of associations.

If the secondary association is successful, the caller ID of the INVITE message is backfilled to the first media stream, and then the SIP call process corresponding to the INVITE message and the first media stream are evaluated for the quality of the voice service to reduce the situation of discarding the first media stream. , to improve the overall accuracy of speech quality assessment.

The method for association processing provided in this embodiment can improve the success rate of association by performing a secondary association after the association between the INVITE message and the media stream fails, and can accurately evaluate the voice quality without changing the protocol. Therefore, voice quality evaluation can be implemented simply and economically.

On the basis of the above embodiment, in the method for association processing provided by another embodiment of the present invention, before the step of acquiring the IP address of at least one first media stream in the first time period, the method includes:

collecting multiple second media streams within the first time period;

Obtain the calling IP address/port number of each second media stream;

According to the calling IP address/port number of each second media stream, find out whether there is a consistent INVITE message within the first time period;

If it does not exist, the second media stream is recorded as the first media stream, and the calling IP address/port number of the first media stream is stored in the RTP database;

Correspondingly, the step of acquiring the calling IP address/port number of at least one first media stream in the first time period is as follows:

The calling IP address/port number of at least one first media stream is extracted from the RTP database.

Optionally, the second media stream is collected from the Mb interface, that is, the media plane information during the actual session.

Optionally, the Mb interface is located between the IMS (IP Multimedia Subsystem, IP Multimedia Subsystem) network to the IPv6 (Internet Protocol Version 6, the sixth edition of IP) network, and IPv6 is designed to replace IPv4 (Internet Protocol Version 6). 4, the fourth edition of IP, the next generation IP protocol that forms the basis of today's Internet technology).

Optionally, first associate the second media stream collected in each time period, determine whether it is associated with the INVITE message, and obtain the first media stream that cannot be associated.

FIG. 2 is a schematic diagram of an RTP database provided by another embodiment of the present invention.

As shown in FIG. 2 , the first media stream is parsed to generate an RTP database, where the RTP database includes information of multiple first media streams, including the calling IP address, port number, and SSRC (Synchronization source, defining a synchronization source) , time, total RTP packets, RTP packet loss and other information.

Optionally, when performing secondary association, the calling IP address/port number of at least one first media stream is extracted from the RTP database.

Other steps in this embodiment are similar to those in the foregoing embodiment, and are not repeated in this embodiment.

The method for association processing provided in this embodiment is easy to read by independently forming a database for media streams that fail to be associated.

On the basis of the above embodiment, in the method for association processing provided by another embodiment of the present invention, before the step of acquiring the caller ID and IP address of the INVITE message of at least one SIP call flow in the second time period, the method includes: :

collecting multiple SIP call flows in the first time period and the second time period;

Store the calling ID and calling IP address/port number of the INVITE message of each SIP call process into the SIP database;

The steps of acquiring the caller ID and IP address of the INVITE message of at least one SIP call flow in the second time period are as follows:

The caller ID and the caller IP address/port number of the INVITE message of at least one SIP call flow of the second time period are extracted from the SIP database.

Optionally, put the INVITE messages of the SIP call flow collected in each time period into the SIP database, so that the media streams of each time period can be associated, especially when the first media stream cannot be associated in the first time period. Next, extract the INVITE message of the second time period from the SIP database for association.

If there is no INVITE message, other SIP messages of the SIP call flow are put into the SIP database.

Optionally, the SIP call flow is collected from the Mw interface, which is an IMS domain signaling interface, specifically the SBC (Session Border Control, session border controller) and CSCF (Call Session Control Function) network elements of VoLTE. between interface.

FIG. 3 is a schematic diagram of a SIP database provided by another embodiment of the present invention.

As shown in Figure 3, the SIP database can store a variety of information, including calling ID, called ID, calling IP address, port number, CALL ID (SIP-level calling ID), time and other information.

Other steps in this embodiment are similar to those in the foregoing embodiment, and are not repeated in this embodiment.

In the method for association processing provided by this embodiment, the SIP call flow of each time period is put into the SIP database, so that the second time can be extracted from the SIP database when the first media stream cannot be associated in the first time period. The segment's INVITE message is associated.

On the basis of the above embodiment, in the method for association processing provided by another embodiment of the present invention, after the step of acquiring the caller ID and IP address of the INVITE message of at least one SIP call flow in the second time period, the method includes: :

acquiring the calling ID and IP address of the INVITE message of at least one SIP call flow in a third time period, where the third time period is a time period after the first time period;

Correspondingly, if the calling IP address/port number of the first media stream is consistent with the calling IP address/port number of the INVITE message in the second time period, the calling ID of the INVITE message in the second time period is backfilled to The steps of the first media stream are as follows:

If the calling IP address/port number of the first media stream is consistent with the calling IP address/port number of the INVITE message of the second time period, and the calling IP address/port number of the INVITE message of the second time period is the same as the calling IP address/port number of the INVITE message of the second time period The calling IP address/port number of the INVITE message in the three time periods is the same, and the calling ID of the INVITE message in the second time period is consistent with the calling ID of the INVITE message in the third time period, then the second time period The caller ID of the INVITE message is backfilled to the first media stream.

Suppose that the calling IP address/port number is assigned to the calling party A by the IMS network side in the second time period, and a session of the second time period is initiated. The IP address of caller A is assigned to caller B by the IMS network side, and there is no INVITE message associated with the media stream in the session of caller B in the first time period. Sessions that are not initiated by the same caller.

That is, if the calling IP address is reassigned, and the IP addresses of the second time period and the first time period are the same, but not the same caller, the INVITE message of the second time period should not be the same as the first time period. media stream association, and the caller ID of the INVITE message in the second time period should not be backfilled to the first media stream.

In the embodiment of the present invention, since there is no INVITE message associated with the media stream in the first time period, it is possible to continue to find out whether the same caller initiates a session in the second time period and the third time period.

The media stream does not contain the caller ID. First, the INVITE messages in the second time period and the third time period of the signaling plane are used for association. If two INVITE messages in the second time period and the third time period are the calling IP addresses/ The port number is the same and the calling ID is the same, then the INVITE messages of the second time period and the third time period correspond to the same caller, so it can be considered that the first time period between the second time period and the third time period is the same. The first media stream and the INVITE message in the second time period are the same caller, and the INVITE message in the third time period is the same caller.

In this case, the caller ID of the INVITE message of the second time period may be backfilled to the first media stream.

Other steps in this embodiment are similar to those in the foregoing embodiment, and are not repeated in this embodiment.

In the method for association processing provided in this embodiment, in order to avoid mistakenly assuming that the INVITE message of the second time period should not be associated with the first media stream of the first time period, the second time period and the third time period of the signaling plane are used first. If the association is successful, it is determined that it is the same caller, and then the first media stream of the first time period is associated with the INVITE message of the second time period. If the association is successful, it can be determined that it is the same caller. Call.

On the basis of the above embodiment, in the method for association processing provided by another embodiment of the present invention, the first media stream includes a start time point, and the SIP flow includes multiple SIP messages. The IP address of the stream is consistent with the IP address of the INVITE message in the second time period, then after the step of backfilling the caller ID of the INVITE message in the second time period to the first media stream, the method includes:

Taking the start time point of the first media stream as the start time of the session;

Determine whether there is a BYE message in the SIP process corresponding to the INVITE message;

If there is a BYE message, the time point of the BYE message is used as the end time of the session.

Optionally, after the INVITE message in the second time period is successfully associated with the first media stream, it is considered that the INVITE message corresponds to a session with the first media stream, or the session corresponding to the INVITE message is considered to be a session corresponding to the first media stream. The voice quality of the INVITE message is similar to that of the first media stream.

Optionally, the voice evaluation needs to determine the start time and end time of a session.

The start time of a session is when the calling party initiates an INVITE message, and the time point of the INVITE message is taken as the start time of the session.

The end time of one session is the BYE message of the SIP call process. Before step 11, the entire SIP call process is collected, and the SIP message of the SIP process is obtained.

After the INVITE message is successfully associated, other SIP messages in the SIP process of the INVITE message that have been successfully associated are analyzed according to time to determine whether there is a BYE message in the SIP process corresponding to the INVITE message, and if there is a BYE message, Take the time point of the BYE message as the end time of the session.

The time point of the last data packet of the media stream is the end time of the media plane information. In practical applications, in the step of collecting the SIP call flow, the BYE message may be lost, so the exact end time cannot be determined, but according to historical experience The data can determine that when the media stream is no longer sent, the BYE message will be sent in a very short period of time, so the time point of the last data packet of the media stream can be used as the end time of a session.

If there is no BYE message, the end time of the media stream is taken as the end time of the session.

After determining the start time and end time of a session, you can perform voice quality assessment on the session.

Other steps in this embodiment are similar to those in the foregoing embodiment, and are not repeated in this embodiment.

In the method for association processing provided in this embodiment, the time point of the INVITE message of the secondary association is used as the start time of the session corresponding to the first media stream, so that one session can be retrieved.

In order to more fully understand the technical content of the present invention, on the basis of the foregoing embodiment, the method for association processing provided in this embodiment is described in detail.

The embodiments of the present invention can be used to improve the method for correlating the SIP call signaling process and the RTP/RTCP media stream, increase the correlation of the correlation, and improve the integrity of the VoLTE voice evaluation.

The method specifically includes: analyzing and analyzing the SIP call flow collected on the Mw interface, and generating a SIP protocol message database (such as the aforementioned SIP database) without an INVITE message; Database (such as the aforementioned RTP database); perform the association judgment between the SIP message and the proprietary RTP database, determine whether the proprietary RTP data stream is associated with the SIP protocol message, if the association is successful, improve the SIP call start time, and establish a new SIP Call flow, which evaluates the call.

The system specifically includes: a data analysis module, which is used to analyze and analyze the SIP call flow and RTP data flow, and generate a SIP protocol message database without INVITE messages and a proprietary RTP database that cannot be associated with INVITE messages; The SIP protocol message of the message is compared with the RTP data stream that cannot be associated with the INVITE message; the correction module is used to correct the SIP call process for the RTP data stream that is associated successfully, and evaluate the VoLTE voice quality again.

FIG. 4 is a schematic flowchart of a method for association processing provided by another embodiment of the present invention.

As shown in Figure 4, the method of the embodiment of the present invention includes the following steps:

1. Data analysis, parsing into a library

SIP protocol message database: Analyze and analyze the SIP call flow collected on the Mw interface to determine whether there is an INVITE message. If not, analyze other SIP messages. If there is, analyze all SIP messages. SIP protocol Message database, the database includes calling number, called number, IP address, port number, CALL ID, time and other information, as shown in Figure 3.

RTP proprietary database: correlate the RTP data stream with the SIP call signaling process, and determine whether it is associated with the INVITE message. Address, port number, sSRC, time, total RTP packets, RTP packet loss and other information, as shown in Figure 2.

2. Data association

FIG. 5 is a schematic diagram of data association provided by another embodiment of the present invention.

As shown in Figure 5, the RTP data flow in the RTP proprietary database and the SIP messages in the SIP protocol message database are multi-dimensionally correlated to find out the relevant SIP messages:

Step 1, the RTP data flow in the RTP proprietary database and the SIP message in the SIP protocol message database are input into the data association module;

Step 2, the RTP data stream is searched according to the IP address, port number and the IP address and port number carried in all the INVITE messages of the previous time in the SIP protocol message database. If the search results are consistent, go to step 3. If the comparison results If they are inconsistent, the process will end directly;

Step 3, confirm whether the successful SIP message belongs to this RTP data flow, if the SIP message time is before the RTP data flow start time, then consider that the SIP message belongs to the RTP data flow;

Step 4: Correct the start and end time of the SIP session for the successfully associated SIP messages (analyzed in chronological order).

3. Data correction, re-evaluation

Correction of the start time of the SIP call signaling process: the start time point of the RTP packet is used.

SIP call signaling process end time correction: analyze the SIP messages (analyzed by time) output by the data association module to determine whether there is a BYE message in the same SIP session, if there is a BYE message, the time point of the BYE message is used as the SIP call signaling The end time of the process; if there is no BYE message, the end time of the RTP data stream (same IP address, port number, SSRC) is used as the end time of the SIP call signaling process. The formula is as follows:

IF (BYE message exists) then SIP session end time = BYE message time

IF (BYE message does not exist) then SIP session end time = RTP/RTCP stream end time

The revised SIP call signaling process is regarded as a completed call, and the voice quality evaluation module evaluates the call again and outputs the evaluation result.

In the collection step, the INVITE message may be lost, and about 1.3% of the RTP packets cannot be associated with the INVITE message and are discarded, which affects the accuracy of the IMS network voice quality assessment.

The method for association processing provided by this embodiment, by forming a database for the RTP/RTCP that fails to be associated, and forming a database for the SIP process, when the association fails for the first time (the media stream and the INVITE message of the first unit time collected) , and also perform secondary association (association of the media stream with the collected INVITE messages of the first time period and the second time period), which is equivalent to associating RTP/RTCP with all the collected INVITE messages, which improves the success of the association Rate.

If the secondary association is successful, backfill the caller ID of the INVITE message to the media stream, and then evaluate the voice service quality of the media stream. Without changing the protocol, the voice quality evaluation can be accurately performed, which can be simple and economical. Implement voice quality assessment.

FIG. 6 shows a schematic structural diagram of an apparatus for association processing provided by another embodiment of the present invention.

Referring to FIG. 6 , on the basis of the foregoing embodiment, the apparatus for association processing provided in this embodiment includes an acquisition module 61 and a backfilling module 62, wherein:

The obtaining module 61 is configured to obtain the IP address/port number of at least one first media stream in the first time period, the first media stream is a media stream that cannot be associated with the INVITE message of the first time period, and obtain the second time The caller identity ID and the caller IP address/port number of the INVITE message of at least one SIP call flow in the segment, the second time period is before the first time period and adjacent to the first time; the backfill module 62 is used for If the calling IP address/port number of the first media stream is consistent with the calling IP address/port number of the INVITE message of the second time period, the calling ID of the INVITE message of the second time period is backfilled to the first media flow.

The acquisition module 61 collects the SIP call flow in the second time period and the first time period on the signaling plane, and buffers the collected INVITE message of the SIP call flow for the second time period and the first time period. The media streams of the first time period are associated.

The media stream in each time period is associated with the SIP call flow. In the embodiment of the present invention, the association between the media stream in the first time period and the SIP call flow is taken as an example for description.

Optionally, the INVITE message is associated with the data packet, thereby associating the SIP call flow with the media stream.

Optionally, the association refers to finding out whether the calling IP address of the media stream within the first time period is consistent with the calling IP address of the INVITE message within the first time period, or finding the calling party of the media stream within the first time period. Whether the port number is the same as the port number of the INVITE message in the first time period.

If any one of the calling IP address and the port number is found to be consistent, it can be considered that the association is successful. The embodiment of the present invention takes the calling IP address as an example for description.

If the calling IP address of the media stream in the first time period is consistent with the calling IP address of the INVITE message, it is considered that the association is successful, and the calling ID of the INVITE message is backfilled into each data packet of the media stream.

If they are inconsistent, it is considered that no consistent INVITE message is collected within the first time period, so the INVITE message cannot be associated with the media stream to obtain the first media stream.

Optionally, the first time period and the second time period can be set according to the actual situation. For example, the first time period is (9:00-9:30) for 30 minutes, and the second time period is (8:30-9:30). 9:00] These 30 minutes.

Optionally, each first media stream in the first time period is associated with the SIP call flow in the second time period, respectively.

In the case that the association between the media stream in the first time period and the SIP call flow fails, an attempt is also made to associate with the SIP call flow in the second time period at another time.

Optionally, searching for the calling IP address of the INVITE message in the second time period according to the calling IP address of each first media stream.

If the caller does not turn off the network, leave the home, or turn on the airplane mode, etc., between the first time period and the second time period, the calling IP address/port number is usually unchanged, and it can be considered that the calling IP If the address/port number is the same, it is indeed the same caller.

Since there is no INVITE message associated with the media stream in the first time period, it can continue to find out whether the same caller initiates a session in the second time period.

If there is an INVITE message in the second time period that matches the IP address of the first media stream in the first time period, and before the first time period, the same caller also initiated another session in the second time period, the second The INVITE message of the session corresponding to the time period is collected, but the collection of the INVITE message of the session corresponding to the first time period is lost.

Optionally, the voice quality has a lot to do with time and location. Since the time is adjacent, even if the two sessions are different, it can be considered that the caller has not moved. The voice quality of the two sessions is similar. The caller ID of the INVITE message is backfilled into the data packet of the first media stream.

The voice quality evaluation is performed according to the SIP call flow corresponding to the INVITE message and the first media stream.

The SIP call flow corresponding to the INVITE message and the first media stream are correlated to correspond to a complete session, and voice quality evaluation is performed for the complete session.

If the IP addresses of the first media stream in the first time period and the INVITE message in the second time period do not match the IP addresses, it is considered that the first media stream cannot be associated, and the process ends.

In this embodiment of the present invention, when the association fails for the first time (the first media stream and the INVITE message of the first time period), a second association (the first media stream and the INVITE message of the second time period) is also performed. Improve the success rate of associations.

If the secondary association is successful, the caller ID of the INVITE message is backfilled to the first media stream, and then the SIP call process corresponding to the INVITE message and the first media stream are evaluated for the quality of the voice service to reduce the situation of discarding the first media stream. , to improve the overall accuracy of speech quality assessment.

The apparatus for association processing provided in this embodiment can be used to execute the method in the foregoing method embodiment, and details are not described again in this embodiment.

The apparatus for association processing provided in this embodiment can improve the success rate of association by performing secondary association through the acquisition module and the backfill module after the association between the INVITE message and the media stream fails, and can accurately perform the association without changing the protocol. The voice quality assessment can be carried out simply and economically.

FIG. 7 shows a schematic structural diagram of an electronic device provided by another embodiment of the present invention.

Referring to FIG. 7 , an electronic device provided by an embodiment of the present invention includes a memory (memory) 71, a processor (processor) 72, a bus 73, and a computer program stored in the memory 71 and running on the processor. The memory 71 and the processor 72 communicate with each other through the bus 73 .

The processor 72 is configured to call program instructions in the memory 71 to implement the method as shown in FIG. 1 when executing the program.

In another implementation manner, the processor implements the following method when executing the program:

Before the step of acquiring the IP address of at least one first media stream in the first time period, the method includes:

collecting multiple second media streams within the first time period;

Obtain the calling IP address/port number of each second media stream;

According to the calling IP address/port number of each second media stream, find out whether there is a consistent INVITE message within the first time period;

If it does not exist, the second media stream is recorded as the first media stream, and the calling IP address/port number of the first media stream is stored in the RTP database;

Correspondingly, the step of acquiring the calling IP address/port number of at least one first media stream in the first time period is as follows:

The calling IP address/port number of at least one first media stream is extracted from the RTP database.

In another implementation manner, the processor implements the following method when executing the program:

Before the step of acquiring the caller ID and IP address of the INVITE message of at least one SIP call flow in the second time period, the method includes:

collecting multiple SIP call flows in the first time period and the second time period;

Store the calling ID and calling IP address/port number of the INVITE message of each SIP call process into the SIP database;

The steps of acquiring the caller ID and the caller IP address/port number of the INVITE message of at least one SIP call flow in the second time period are as follows:

The caller ID and the caller IP address/port number of the INVITE message of at least one SIP call flow of the second time period are extracted from the SIP database.

In another implementation manner, the processor implements the following method when executing the program:

After the step of acquiring the caller ID and IP address of the INVITE message of at least one SIP call flow in the second time period, the method includes:

acquiring the calling ID and IP address of the INVITE message of at least one SIP call flow in a third time period, where the third time period is a time period after the first time period;

Correspondingly, if the calling IP address/port number of the first media stream is consistent with the calling IP address/port number of the INVITE message in the second time period, the calling ID of the INVITE message in the second time period is backfilled to The steps of the first media stream are as follows:

If the calling IP address/port number of the first media stream is consistent with the calling IP address/port number of the INVITE message of the second time period, and the calling IP address/port number of the INVITE message of the second time period is the same as the calling IP address/port number of the INVITE message of the second time period The calling IP address/port number of the INVITE message in the three time periods is the same, and the calling ID of the INVITE message in the second time period is consistent with the calling ID of the INVITE message in the third time period, then the second time period The caller ID of the INVITE message is backfilled to the first media stream.

In another implementation manner, the processor implements the following method when executing the program:

The INVITE message of the second time period carries the time point of the INVITE message, and the SIP process includes multiple SIP messages. Correspondingly, if the IP address of the first media stream is consistent with the IP address of the INVITE message of the second time period, Then after the step of backfilling the caller ID of the INVITE message in the second time period to the first media stream, the method includes:

Taking the start time point of the first media stream as the start time of the session;

Determine whether there is a BYE message in the SIP process corresponding to the INVITE message;

If there is a BYE message, the time point of the BYE message is used as the end time of the session.

In another implementation manner, the processor implements the following method when executing the program:

If there is no BYE message, the end time of the media stream is taken as the end time of the session.

In another implementation manner, the processor implements the following method when executing the program:

If the calling IP address/port number of the first media stream is consistent with the calling IP address/port number of the INVITE message of the second time period, the calling ID of the INVITE message of the second time period is backfilled to the first media After the steps of the flow, the method further includes:

The voice quality evaluation is performed according to the SIP call flow corresponding to the INVITE message and the first media stream.

The electronic device provided in this embodiment can be used to execute the program corresponding to the method in the foregoing method embodiment, which is not repeated in this embodiment.

In the electronic device provided in this embodiment, when the processor executes the program, the secondary association is performed after the association between the INVITE message and the media stream fails, so that the success rate of the association can be improved, without changing the protocol. , the voice quality assessment can be performed accurately, so that the voice quality assessment can be realized simply and economically.

Another embodiment of the present invention provides a storage medium, where a computer program is stored on the storage medium, and when the program is executed by a processor, the steps as shown in FIG. 1 are implemented.

In another implementation manner, when the program is executed by the processor, the following method is implemented:

Before the step of acquiring the IP address of at least one first media stream in the first time period, the method includes:

collecting multiple second media streams within the first time period;

Obtain the calling IP address/port number of each second media stream;

According to the calling IP address/port number of each second media stream, find out whether there is a consistent INVITE message within the first time period;

If it does not exist, the second media stream is recorded as the first media stream, and the calling IP address/port number of the first media stream is stored in the RTP database;

Correspondingly, the step of acquiring the calling IP address/port number of at least one first media stream in the first time period is as follows:

The calling IP address/port number of at least one first media stream is extracted from the RTP database.

In another implementation manner, when the program is executed by the processor, the following method is implemented:

Before the step of acquiring the caller ID and IP address of the INVITE message of at least one SIP call flow in the second time period, the method includes:

collecting multiple SIP call flows in the first time period and the second time period;

Store the calling ID and calling IP address/port number of the INVITE message of each SIP call process into the SIP database;

The steps of acquiring the caller ID and the caller IP address/port number of the INVITE message of at least one SIP call flow in the second time period are as follows:

The caller ID and the caller IP address/port number of the INVITE message of at least one SIP call flow of the second time period are extracted from the SIP database.

In another implementation manner, when the program is executed by the processor, the following method is implemented:

After the step of acquiring the caller ID and IP address of the INVITE message of at least one SIP call flow in the second time period, the method includes:

acquiring the calling ID and IP address of the INVITE message of at least one SIP call flow in a third time period, where the third time period is a time period after the first time period;

Correspondingly, if the calling IP address/port number of the first media stream is consistent with the calling IP address/port number of the INVITE message in the second time period, the calling ID of the INVITE message in the second time period is backfilled to The steps of the first media stream are as follows:

If the calling IP address/port number of the first media stream is consistent with the calling IP address/port number of the INVITE message of the second time period, and the calling IP address/port number of the INVITE message of the second time period is the same as the calling IP address/port number of the INVITE message of the second time period The calling IP address/port number of the INVITE message in the three time periods is the same, and the calling ID of the INVITE message in the second time period is consistent with the calling ID of the INVITE message in the third time period, then the second time period The caller ID of the INVITE message is backfilled to the first media stream.

In another implementation manner, when the program is executed by the processor, the following method is implemented:

The INVITE message of the second time period carries the time point of the INVITE message, and the SIP process includes multiple SIP messages. Correspondingly, if the IP address of the first media stream is consistent with the IP address of the INVITE message of the second time period, Then after the step of backfilling the caller ID of the INVITE message in the second time period to the first media stream, the method includes:

Taking the start time point of the first media stream as the start time of the session;

Determine whether there is a BYE message in the SIP process corresponding to the INVITE message;

If there is a BYE message, the time point of the BYE message is used as the end time of the session.

In another implementation manner, when the program is executed by the processor, the following method is implemented:

If there is no BYE message, the end time of the media stream is taken as the end time of the session.

In another implementation manner, when the program is executed by the processor, the following method is implemented:

If the calling IP address/port number of the first media stream is consistent with the calling IP address/port number of the INVITE message of the second time period, the calling ID of the INVITE message of the second time period is backfilled to the first media After the steps of the flow, the method further includes:

The voice quality evaluation is performed according to the SIP call flow corresponding to the INVITE message and the first media stream.

In the storage medium provided in this embodiment, when the program is executed by the processor, the method of the foregoing method embodiment is implemented, which is not repeated in this implementation.

The storage medium provided in this embodiment can improve the success rate of the association by performing a secondary association after the association between the INVITE message and the media stream fails, and can accurately evaluate the voice quality without changing the protocol, so that the Simple and economical implementation of voice quality assessment.

Yet another embodiment of the present invention discloses a computer program product comprising a computer program stored on a non-transitory computer-readable storage medium, the computer program comprising program instructions that, when executed by a computer , the computer can execute the methods provided by the above method embodiments, for example, including:

obtaining the protocol IP address/port number of the interconnection between the calling networks of at least one first media stream in the first time period, where the first media stream is a media stream that cannot be associated with the INVITE message of the first time period, and obtain the calling identity ID and calling IP address/port number of the INVITE message of at least one session initiation protocol SIP call flow in the second time period, the second time period is before the first time period and is the same as the first time period; adjacent;

If the calling IP address/port number of the first media stream is consistent with the calling IP address/port number of the INVITE message of the second time period, the calling ID of the INVITE message of the second time period is backfilled to the first media flow.

It will be understood by those skilled in the art that although some of the embodiments described herein include certain features, but not others, included in other embodiments, that combinations of features of different embodiments are intended to be within the scope of the present invention And form different embodiments.

Those skilled in the art can understand that each step in the embodiments may be implemented by hardware, or by software modules running on one or more processors, or by a combination thereof. Those skilled in the art will understand that a microprocessor or a digital signal processor (DSP) may be used in practice to implement some or all of the functions of some or all of the components according to the embodiments of the present invention. The present invention can also be implemented as apparatus or apparatus programs (eg, computer programs and computer program products) for performing part or all of the methods described herein.

Although the embodiments of the present invention have been described with reference to the accompanying drawings, various modifications and variations can be made by those skilled in the art without departing from the spirit and scope of the present invention, and such modifications and variations all fall within the scope of the appended claims within the limits of the requirements.

Claims (10)

1. A method for association processing, wherein the method comprises: obtaining the protocol IP address/port number of the interconnection between the calling networks of at least one first media stream in the first time period, where the first media stream is a media stream that cannot be associated with the INVITE message of the first time period, and obtain the calling identity ID and calling IP address/port number of the INVITE message of at least one session initialization protocol SIP call flow in the second time period, the second time period is before the first time period and is the same as the first time period. adjacent; If the calling IP address/port number of the first media stream is consistent with the calling IP address/port number of the INVITE message of the second time period, the calling ID of the INVITE message of the second time period is backfilled to the first media flow. 2. The method according to claim 1, wherein before the step of acquiring the IP address of at least one first media stream in the first time period, the method comprises: collecting multiple second media streams within the first time period; Obtain the calling IP address/port number of each second media stream; According to the calling IP address/port number of each second media stream, find out whether there is a consistent INVITE message within the first time period; If it does not exist, the second media stream is recorded as the first media stream, and the calling IP address/port number of the first media stream is stored in the RTP database; Correspondingly, the step of acquiring the calling IP address/port number of at least one first media stream in the first time period is as follows: The calling IP address/port number of at least one first media stream is extracted from the RTP database. 3. The method according to claim 1, wherein before the step of acquiring the caller ID and IP address of the INVITE message of at least one SIP call flow in the second time period, the method comprises: collecting multiple SIP call flows in the first time period and the second time period; Store the calling ID and calling IP address/port number of the INVITE message of each SIP call process into the SIP database; Correspondingly, the steps of acquiring the calling ID and the calling IP address/port number of the INVITE message of at least one SIP call flow in the second time period are as follows: The caller ID and the caller IP address/port number of the INVITE message of at least one SIP call flow of the second time period are extracted from the SIP database. 4. The method according to claim 1, characterized in that: after the step of acquiring the caller ID and IP address of the INVITE message of at least one SIP call flow in the second time period, the method comprises: acquiring the calling ID and IP address of the INVITE message of at least one SIP call flow in a third time period, where the third time period is a time period after the first time period; Correspondingly, if the calling IP address/port number of the first media stream is consistent with the calling IP address/port number of the INVITE message in the second time period, the calling ID of the INVITE message in the second time period is backfilled to The steps of the first media stream are as follows: If the calling IP address/port number of the first media stream is consistent with the calling IP address/port number of the INVITE message of the second time period, and the calling IP address/port number of the INVITE message of the second time period is the same as the calling IP address/port number of the INVITE message of the second time period The calling IP address/port number of the INVITE message in the three time periods is the same, and the calling ID of the INVITE message in the second time period is consistent with the calling ID of the INVITE message in the third time period, then the second time period The caller ID of the INVITE message is backfilled to the first media stream. 5. method according to claim 1, is characterized in that: described first media flow comprises start time point, described SIP flow comprises a plurality of SIP messages, correspondingly, if the IP address of first media flow and the first The IP addresses of the INVITE messages in the two time periods are the same, and after the step of backfilling the caller ID of the INVITE messages in the second time period to the first media stream, the method includes: Taking the start time point of the first media stream as the start time of the session; Determine whether there is a BYE message in the SIP process corresponding to the INVITE message; If there is a BYE message, the time point of the BYE message is taken as the end time of the session. 6. The method according to claim 5, wherein if there is no BYE message, the end time of the media stream is used as the end time of the session. 7. The method according to any one of claims 1-6, wherein: if the calling IP address/port number of the first media stream is associated with the calling IP address/port number of the INVITE message of the second time period Consistent, after the step of backfilling the caller ID of the INVITE message in the second time period to the first media stream, the method further includes: The voice quality evaluation is performed according to the SIP call flow corresponding to the INVITE message and the first media stream. 8. An apparatus for association processing, wherein the apparatus comprises: The obtaining module is used to obtain the IP address/port number of at least one first media stream in the first time period, the first media stream is a media stream that cannot be associated with the INVITE message of the first time period, and obtains the second time The caller identity ID and the caller IP address/port number of the INVITE message of at least one SIP call flow in the segment, the second time period is before the first time period and adjacent to the first time; The backfilling module is used for, if the calling IP address/port number of the first media stream is consistent with the calling IP address/port number of the INVITE message of the second time period, then the calling ID of the INVITE message of the second time period Backfill to the first media stream. 9. An electronic device, characterized in that it comprises a memory, a processor, a bus and a computer program stored in the memory and running on the processor, the processor implementing the program as claimed in claims 1-7 any of the steps. 10. A storage medium on which a computer program is stored, characterized in that: when the program is executed by a processor, the steps of any one of claims 1-7 are implemented.