WO2024067103A1

WO2024067103A1 - Method and apparatus for determining call quality of video service

Info

Publication number: WO2024067103A1
Application number: PCT/CN2023/118623
Authority: WO
Inventors: 李腾飞; 王蕾
Original assignee: 中兴通讯股份有限公司
Priority date: 2022-09-29
Filing date: 2023-09-13
Publication date: 2024-04-04
Also published as: CN117835303A

Abstract

Provided in the embodiments of the present application are a method and apparatus for determining the call quality of a video service. The method comprises: acquiring an uplink real-time transport protocol (RTP) packet loss condition of a video service according to an uplink RTP data packet state of the video service, which state is indicated by a core network, and acquiring a downlink packet data convergence protocol (PDCP) packet loss condition of the video service according to a downlink PDCP state report of the video service that a base station instructs a terminal to report; and determining the call quality of the video service on the basis of the uplink RTP packet loss condition and the downlink PDCP packet loss condition.

Description

Method and device for determining call quality of video service

CROSS-REFERENCE TO RELATED APPLICATIONS

This application is based on Chinese patent application CN202211202780.9, filed on September 29, 2022, with the invention name “Method and device for determining call quality for video services”, and claims the priority of the patent application, and all the contents disclosed therein are incorporated into this application by reference.

Technical Field

The embodiments of the present application relate to the field of communications, and in particular, to a method and device for optimizing call quality of a video service.

Background technique

As two important services in the 3rd Generation Partnership Project (3GPP) mobile communication system, voice and video calls are of great concern to operators. In the 2G and 3G eras, and even in the early 4G era, due to limited bandwidth resources and high traffic charges, operators only focused on promoting voice services and ensuring call experience. With the reduction of 4G traffic charges and the further enrichment of bandwidth resources by 5G network deployment, operators have begun to promote video services and pay more and more attention to the call experience of video services.

Since video services contain both audio and video information, problems such as voice articulation, intermittent, single-channel, image freeze, mosaic, black screen, and audio and video asynchrony will affect the call experience of video users. In order to ensure the call experience of video users, operators require the base station side to evaluate the quality of video services and switch users with poor call quality of video services to other cells that can provide better services.

The base station generally indirectly reflects the call experience of the video service by evaluating the channel quality of the service cell, but there is no absolute mapping relationship between the two, and there may be misjudgment problems. In the video service, both sound and picture information are transmitted in the unacknowledged mode (UM). The packet loss assessment can be as close to the call experience index of the video service as possible, but the base station cannot accurately assess the packet loss situation. There is currently no effective solution to this problem.

Summary of the invention

The embodiments of the present application provide a method and device for determining call quality of a video service, so as to at least solve the problem in the related art that the packet loss situation cannot be accurately evaluated on the base station side.

According to one embodiment of the present application, a method for determining the call quality of a video service is provided, comprising: obtaining the uplink Real-time Transport Protocol (RTP) packet loss situation of the video service according to the uplink RTP packet status of the video service indicated by the core network, and obtaining the downlink Packet Data Convergence Protocol (PCDP) packet loss situation of the video service according to the downlink PCDP packet loss situation of the video service reported by the terminal indicated by the base station; determining the call quality of the video service based on the uplink RTP packet loss situation and the downlink PCDP packet loss situation.

According to another embodiment of the present application, a device for determining call quality of a video service is provided, comprising: an acquisition module, configured to acquire the uplink RTP packet loss situation of the video service according to the uplink real-time transport protocol RTP data packet status of the video service indicated by the core network, and acquire the downlink packet data convergence protocol of the video service reported by the terminal according to the instruction of the base station. The PDCP status report is used to obtain the downlink PCDP packet loss situation of the video service; and a determination module is configured to determine the call quality of the video service based on the uplink RTP packet loss situation and the downlink PCDP packet loss situation.

According to another embodiment of the present application, a computer-readable storage medium is provided, in which a computer program is stored, wherein the computer program is configured to execute the steps of any of the above method embodiments when running.

According to another embodiment of the present application, an electronic device is provided, including a memory and a processor, wherein the memory stores a computer program, and the processor is configured to run the computer program to execute the steps in any one of the above method embodiments.

BRIEF DESCRIPTION OF THE DRAWINGS

1 is a schematic diagram of a network architecture of a method for determining call quality for running a video service according to an embodiment of the present application;

2 is a flow chart of a method for determining call quality of a video service according to an embodiment of the present application;

3 is a structural block diagram of a device for determining call quality of a video service according to an embodiment of the present application;

4 is a structural block diagram of a device for determining call quality of a video service according to another embodiment of the present application;

5 is a structural block diagram of a device for determining call quality of a video service according to another embodiment of the present application;

FIG6 is a flowchart of a method for evaluating a video user's call experience according to an embodiment of the present application.

Detailed ways

The embodiments of the present application will be described in detail below with reference to the accompanying drawings and in combination with the embodiments.

It should be noted that the terms "first", "second", etc. in the specification and claims of this application and the above-mentioned drawings are used to distinguish similar objects, and are not necessarily used to describe a specific order or sequence.

FIG1 is a schematic diagram of a network architecture for a method for optimizing call quality of a video service according to an embodiment of the present application. Each embodiment of the present application can be run on the network architecture shown in FIG1. As shown in FIG1, the network architecture includes: a core network, a base station, and a terminal, wherein the terminal is in a service cell of the base station, and the core network is used to report the uplink RTP status of the video service performed by the terminal to the base station. In the network architecture, when the quality of the video service performed by the terminal in the service cell deteriorates, the call quality of the video service will be optimized to ensure the call experience of the video user.

In this embodiment, a method for optimizing call quality of a video service running on the above network architecture is provided. FIG. 2 is a flow chart of a method for determining call quality of a video service according to an embodiment of the present application. As shown in FIG. 2 , the process includes the following steps:

Step S202, obtaining the uplink RTP packet loss situation of the video service according to the uplink Real-time Transport Protocol RTP data packet status of the video service indicated by the core network, and obtaining the downlink PCDP packet loss situation of the video service according to the downlink Packet Data Convergence Protocol PDCP status report of the video service reported by the terminal indicated by the base station;

Step S204: determining the call quality of the video service based on the uplink RTP packet loss situation and the downlink PCDP packet loss situation.

In an exemplary embodiment, determining the call quality of the video service further includes: determining the call quality of the video service only according to the uplink RTP packet loss situation; or determining the call quality of the video service only according to the downlink PCDP packet loss situation.

In step S202 of this embodiment, it includes: when the core network indicates that the data packet state of the uplink RTP of the video service is in a normal transmission state, obtaining the uplink RTP packet loss of the video service; when the core network indicates that the uplink RTP of the video service is in an abnormal transmission state, stopping obtaining the uplink RTP packet loss of the video service Condition.

In this embodiment, the abnormal transmission state includes: a transmission stop state and an unknown transmission state.

By indicating the status of the uplink RTP data packet of the corresponding service through the core network, the uplink RTP packet loss situation can be obtained more accurately, thereby solving the problem of misjudgment of uplink RTP packet loss in scenarios such as call initiation and three-party calling.

In step S202 of this embodiment, it also includes: obtaining the uplink RTP packet loss situation according to the sequence number of the received uplink RTP data packet.

In step S204 of this embodiment, the content of the downlink PDCP status report includes at least one of the following: the downlink PDCP packet loss rate, the number of packet losses, and the maximum number of consecutive packet losses of the video service.

In an exemplary embodiment, instructing the terminal to report the downlink PDCP status report of the video service includes one of the following: instructing the terminal to report the downlink PDCP status report once within a preset time interval; instructing the terminal to trigger the reporting of the downlink PDCP status report when the downlink PDCP packet loss situation of the video service meets a first preset condition.

By instructing the terminal to report the downlink PDCP status report of the corresponding service through the base station, the problem of misjudgment of downlink PDCP packet loss in scenarios such as uplink sudden interference and insufficient uplink power can be solved.

After step S204 in this embodiment, the method further includes: optimizing the call quality of the video service when the call quality of the video service deteriorates to reach a second preset condition.

In an exemplary embodiment, the second preset condition includes one of the following: the uplink RTP packet loss rate of the video service meets the first threshold value and the number of packet losses meets the second threshold value; the number of continuous uplink RTP packet losses of the video service meets the third threshold value; the duration of the uplink RTP packet not received by the video service meets the fourth threshold value; the downlink PDCP packet loss rate of the video service meets the fifth threshold value and the number of packet losses meets the sixth threshold value; the number of continuous downlink PDCP packet losses of the video service meets the seventh threshold value.

In an exemplary embodiment, optimizing the call quality of the video service includes: switching users whose call quality of the video service has deteriorated from the current serving cell to a target cell whose call quality meets the conditions; or, scheduling and optimizing the call quality of the video service.

In an exemplary embodiment, before switching a user whose call quality of the video service has deteriorated from the current serving cell to a target cell whose call quality meets the conditions, it also includes: determining whether the target cell meets the call quality conditions by evaluating the uplink and downlink channel conditions of the target cell or the uplink and downlink packet loss of historical users in the target cell.

Through the above steps, the uplink RTP packet loss of the video service is obtained according to the uplink Real-time Transport Protocol RTP data packet status of the video service indicated by the core network, and the downlink PCDP packet loss of the video service is obtained according to the downlink Packet Data Convergence Protocol PDCP status report of the video service reported by the terminal indicated by the base station. The packet loss of the video service can be obtained more accurately on the base station side, which is conducive to the evaluation of the call quality of the video service. For users with poor call quality, the video call quality of the user can be guaranteed by triggering switching to other cells with better call quality, scheduling optimization and other strategies. It solves the problem that the packet loss situation cannot be accurately evaluated on the base station side when evaluating the quality of the video service in the related technology, and improves the call quality of the video service.

Through the description of the above implementation methods, those skilled in the art can clearly understand that the method according to the above embodiment can be implemented by means of software plus a necessary general hardware platform, or by hardware, but in many cases the former is a better implementation method. Based on this understanding, the technical solution of the embodiment of the present application, or the part that contributes to the prior art, can be embodied in the form of a software product, which is stored in a storage medium (such as a read-only memory/random access memory (ROM/RAM), magnetic A disk or optical disk) includes several instructions for enabling a terminal device (which may be a mobile phone, a computer, a server, or a network device, etc.) to execute the methods described in the various embodiments of the present application.

In this embodiment, a device for determining call quality of a video service is also provided, and the device is used to implement the above-mentioned embodiments and preferred implementation modes, and the descriptions that have been made are not repeated here. As used below, the term "module" can be a combination of software and/or hardware that implements a predetermined function. Although the devices described in the following embodiments are preferably implemented in software, the implementation of hardware, or a combination of software and hardware, is also possible and conceivable.

FIG3 is a structural block diagram of a device for determining call quality of a video service according to an embodiment of the present application. As shown in FIG3 , the device includes: an acquisition module 10 and a determination module 20 .

The acquisition module 10 is configured to acquire the uplink RTP packet loss of the video service according to the uplink real-time transport protocol RTP data packet status of the video service indicated by the core network, and acquire the downlink PCDP packet loss of the video service according to the downlink packet data convergence protocol PDCP status report of the video service reported by the terminal indicated by the base station;

The determination module 20 is configured to evaluate the call quality of the video service according to at least one of the following: the RTP packet loss situation, the downlink PCDP packet loss situation.

FIG. 4 is a structural block diagram of a device for determining call quality of a video service according to an embodiment of the present application. As shown in FIG. 4 , in addition to all the modules shown in FIG. 3 , the acquisition module 10 further includes:

The acquisition unit 11 is configured to acquire the uplink RTP packet loss status of the video service when the core network indicates that the data packet status of the uplink RTP of the video service is in a normal transmission state;

The stopping unit 12 is configured to stop obtaining the uplink RTP packet loss situation of the video service when the core network indicates that the uplink RTP of the video service is in an abnormal transmission state, wherein the abnormal transmission state includes: a stopped transmission state and an unknown transmission state.

FIG. 5 is a structural block diagram of a device for determining call quality of a video service according to an embodiment of the present application. As shown in FIG. 5 , the device includes, in addition to all the modules shown in FIG. 4 , further includes:

The optimization module 30 is configured to optimize the call quality of the video service when the call quality of the video service deteriorates to reach a second preset condition.

It should be noted that the above modules can be implemented by software or hardware. For the latter, it can be implemented in the following ways, but not limited to: the above modules are all located in the same processor; or the above modules are located in different processors in any combination.

To facilitate the understanding of the technical solutions provided by the embodiments of the present application, the embodiments will be described in detail below in combination with specific scenarios.

In view of the problems existing in the related technologies: Although the call quality of the video service is evaluated by packet loss, which can be as close to the call experience index of the video service as possible, the base station side cannot accurately evaluate the packet loss. The embodiment of the present application provides a method for evaluating the call experience of video users. First, the core network indicates the status of the uplink RTP data packet of the corresponding video service to solve the problem of misjudgment of uplink RTP packet loss in scenarios such as call initiation and three-party call. Secondly, the base station instructs the terminal to report the downlink PDCP status report of the corresponding service to solve the problem of misjudgment of downlink PDCP packet loss in scenarios such as uplink sudden interference and insufficient uplink power. Finally, the call experience of video users is evaluated based on the actual packet loss of uplink RTP and downlink PDCP. For users with poor call quality, strategies such as switching and scheduling optimization can be triggered to other cells with better call quality to ensure the call experience of video users.

FIG. 6 is a flow chart of a method for evaluating a video user's call experience according to an embodiment of the present application. As shown in FIG. 6 , the method The following steps are involved:

Step S602: The base station reflects the call quality of the video service according to the uplink RTP packet loss and the downlink PCDP packet loss.

In one embodiment, for uplink quality detection, RTP packet loss can reflect both packet abandonment and air interface packet loss. However, considering that some terminals may abandon RTP packets first and then assemble PDCP packets, RTP packet loss is more suitable for reflecting the uplink quality status than PDCP packet loss.

For downlink quality detection, RTP packet loss can reflect transmission packet loss, base station packet abandonment and air interface packet loss at the same time. However, transmission packet loss is not a problem of the current service cell. Even if other target neighboring cells are selected to trigger handover, the transmission packet loss problem may still exist. PDCP packet loss can only reflect base station packet abandonment and air interface packet loss. Therefore, PDCP packet loss is more suitable for reflecting the downlink quality status than RTP packet loss.

In this embodiment, considering that problems such as voice articulation, intermittent, single-channel, screen freeze, mosaic, black screen, and sound and picture asynchrony will affect the call experience of video users, and the degree of impact of different problems on the call experience may vary, it can be considered to count the uplink and downlink packet loss of sound and picture separately. When any of the following conditions is met, it is considered that the quality of video call experience has deteriorated, and it is necessary to select a target neighboring area to trigger switching.

In one embodiment, for the uplink and downlink packet loss of voice, the 5G Quality of Service Identifier 1 (5QI1) service of the 5G system is used as an example to illustrate:

The uplink RTP packet loss rate of 5QI1 service meets the threshold and the number of packet losses meets the threshold;

Considering that when the local user is talking, the 5QI1 service sends an activation packet uplink, and when the local user is not talking, the 5QI1 service sends a silent packet uplink, and the loss of silent packets has little impact on the call experience. Therefore, the packet loss rate can be calculated according to a certain conversion factor;

The number of continuous uplink RTP packet losses of 5QI1 service meets the threshold;

5QI1 service does not receive the uplink RTP packet for a period that meets the threshold;

The downlink PDCP packet loss rate of 5QI1 service meets the threshold and the number of packet losses meets the threshold;

Considering that the 5QI1 service sends activation packets downlink when the other end user is talking, and sends silent packets downlink when the other end user is not talking, the loss of silent packets has little impact on the call experience, and a certain conversion factor can be used in the calculation of packet loss rate;

The number of consecutive downlink PDCP packet losses for 5QI1 service meets the threshold;

Regarding the uplink and downlink packet loss of the picture, this embodiment takes the 5QI2 service of the 5G system as an example to illustrate:

The uplink RTP packet loss rate of the 5QI2 service meets the threshold and the number of packet losses meets the threshold;

The number of consecutive uplink RTP packet losses of 5QI2 service meets the threshold;

The duration of the uplink RTP packet received by the 5QI2 service does not meet the threshold.

The downlink PDCP packet loss rate of 5QI2 service meets the threshold and the number of packet losses meets the threshold;

The number of consecutive downlink PDCP packet losses for 5QI2 services meets the threshold.

Reflecting the call quality of the video service according to the uplink RTP packet loss, including: indicating the uplink RTP data packet status of the corresponding service through the core network;

In one embodiment, under normal circumstances, the base station side calculates the RTP packet loss situation based on the SN number of the received RTP packet, and for the 5QI1 service, the activation packet and silent packet loss situation is calculated based on the sending rules of the activation packet and silent packet. For example: the SN number of the uplink RTP packet received by the 5QI1 service last time is 502, and the SN number of the RTP packet received this time is 509. The time interval between the two uplink RTP packets is 400ms. With the activation packet sending period of 20ms and the silent packet sending period of 160ms, it is estimated that the 5QI1 service uplink loses 4 RTP activation packets and 2 RTP silent packets.

If the base station detects that no uplink RTP packet has been received for a long time, it is generally caused by the poor uplink channel conditions of the serving cell. The base station needs to select a target neighboring cell to trigger switching, except for some special scenarios. For example, in the call initiation scenario, the uplink RTP packet cannot be generated because the other end is not connected; in the three-party call scenario, the user call is held and the uplink RTP packet cannot be generated; in the camera-off scenario, the uplink RTP packet does not need to be generated. For these special scenarios, the core network indicates the status of the corresponding service uplink RTP data packet, where the status attributes include but are not limited to: normal transmission, stop transmission, unknown, etc., which can solve the problem of misjudgment of uplink RTP packet loss in scenarios such as call initiation and three-party call. Taking the 5G system as an example, for new service scenarios, the core network can indicate in the PDU session resource establishment request (PDU SESSION RESOURCE SETUP REQUEST) and PDU session resource modification request (PDU SESSION RESOURCE MODIFY REQUEST) messages; for service status attribute change scenarios, the core network can indicate in the PDU SESSION RESOURCE MODIFY REQUEST message, and the base station only performs uplink RTP packet loss assessment for services in normal transmission status. In particular, for inter-station handover scenarios, since the service status attributes have not changed, the base station on the handover target side may not be able to obtain the latest status of the corresponding service from the core network. The base station on the handover source side needs to indicate the latest status of the corresponding service to the base station on the target side in the handover request message.

In one embodiment, reflecting the call quality of the video service according to the downlink PCDP packet loss situation includes: instructing the terminal to report a downlink PDCP status report of the corresponding service through the base station;

In one embodiment, under normal circumstances, the base station side estimates the PDCP packet loss situation based on the feedback results of the received downlink PDCP packets. For example, if 20 downlink PDCP packets sent by the 5QI1 service within a period of time have 18 successful feedback results and 2 failed feedback results, it is considered that the downlink PDCP packet loss rate of the 5QI1 service within this period of time is 10%.

In scenarios such as uplink burst interference and insufficient uplink power, the feedback results of some downlink PDCP packets may not be parsed normally, resulting in misjudgment of packet loss. By instructing the terminal to report the status of the downlink PDCP data packet of the corresponding service on the base station side, the misjudgment of downlink PDCP packet loss in scenarios such as uplink burst interference and insufficient uplink power can be solved; among them, the status report includes two modes: periodic type and event type. For example: the periodic type can consider reporting a status report every 2s, and the report content includes but is not limited to the corresponding service downlink PDCP packet loss rate, number of packet losses, maximum number of continuous packet losses, etc.; the event type can consider requiring the corresponding service downlink DPCP packet loss rate, number of packet losses, maximum number of continuous packet losses, etc. to trigger the status report when one or more conditions are met. The periodic and event-type status reports can be used separately to save signaling overhead, or they can be used at the same time to improve detection efficiency. The specific selection can be made according to the actual network situation.

Step S604: Switch users with poor video service call quality to other cells with good video service quality, or perform scheduling optimization strategies to ensure the call experience of video users.

In one embodiment, according to the relevant strategies of step S604 and step S606, the base station only performs uplink RTP packet loss evaluation on the services in the normal transmission state, and performs downlink PDCP packet loss evaluation according to the PDCP status report reported by the terminal, so as to obtain the actual packet loss situation to evaluate the call experience of the video user. For users with poor call quality of the video service, the call experience of the video user can be guaranteed by triggering switching to other better cells, scheduling optimization and other strategies. In particular, when selecting the switching target cell, since it is impossible to predict the uplink/downlink packet loss situation of the user after the switching, it can be determined by judging the uplink/downlink channel of the target cell. The system can infer the uplink/downlink packet loss of historical users based on the conditions, thereby reducing the probability that the call experience of video users will not improve or even deteriorate after switching.

The above method involved in the embodiment of the present application is applicable to the security strategies of Long Term Evolution (LTE), New Radio (NR) and subsequent 3GPP other mobile systems.

In the above-mentioned embodiments of the present application, in order to address the problem that the base station side cannot accurately evaluate the call experience of video users, first, the core network indicates the status of the uplink RTP data packet of the corresponding service, so as to solve the problem of misjudgment of uplink RTP packet loss in scenarios such as voice call initiation and three-party call; secondly, the base station instructs the terminal to report the downlink PDCP status report of the corresponding service, so as to solve the problem of misjudgment of downlink PDCP packet loss in scenarios such as uplink sudden interference and insufficient uplink power; finally, the call experience of video users is evaluated based on the actual packet loss of uplink RTP and downlink PDCP. For users with poor call quality of video services, strategies such as switching to other better cells and scheduling optimization can be triggered to ensure the call experience of video users.

The following is an implementation of the above-mentioned video user call experience evaluation method in a specific scenario:

Assuming that a user is conducting a video call service on a 5G base station and is initially in the call initiation stage, the other user does not answer the call. The core network indicates through the PDU SESSION RESOURCE MODIFY REQUEST message that the uplink RTP of the corresponding service is in a stopped transmission state. The base station side does not perform uplink RTP packet loss assessment on the relevant service to prevent the condition of not receiving the uplink RTP packet of the corresponding service for a long time from triggering switching.

After the user at the other end answers the call, the core network indicates that the uplink RTP of the corresponding service is in normal transmission status through the PDU SESSION RESOURCE MODIFY REQUEST message, and the base station side starts to evaluate the uplink RTP packet loss of the relevant service. If the uplink RTP packet loss rate of the corresponding service within the evaluation period of 2s is greater than 6% and the number of packet losses is greater than 10, the video call experience is poor, and the target cell is selected to trigger the handover, otherwise the packet loss evaluation continues.

Assume that during a call, the other end user answers another call and the local user is suspended in the three-party call. The core network indicates through the PDU SESSION RESOURCE MODIFY REQUEST message that the uplink RTP of the corresponding service is in a stopped transmission state. The base station does not perform uplink RTP packet loss assessment for related services to prevent the condition of not receiving the uplink RTP packet of the corresponding service for a long time from triggering switching.

After the user on the other end hangs up the other phone, the three-party call process ends and the normal call is restored with the local user. The core network indicates through the PDU SESSION RESOURCE MODIFY REQUEST message that the uplink RTP of the corresponding service is in normal transmission status, and the base station side starts to evaluate the uplink RTP packet loss of the relevant service; if the number of consecutive uplink RTP packet losses of the corresponding service is greater than 5, the condition of poor video call experience is met, and the target cell is selected to trigger switching, otherwise the packet loss evaluation continues.

When the base station sends a downlink PDCP data packet to the user on the relevant service, the base station requires the UE to select the target cell to trigger the handover or improve the user's call experience through scheduling optimization if the downlink PDCP packet loss rate is greater than 6% and the number of packet losses is greater than 10 within the 2s evaluation period of the corresponding service through RRC reconfiguration message, and the poor video call experience condition is met. Otherwise, the packet loss evaluation continues.

Through the embodiment of the present application, when implementing the video user call experience evaluation method, the video user call experience is evaluated according to the actual packet loss of uplink RTP and downlink PDCP. For users with poor call quality of video services, strategies such as switching to other better cells and scheduling optimization can be triggered to ensure the call experience of video users.

An embodiment of the present application further provides a computer-readable storage medium, in which a computer program is stored, wherein the computer program is configured to execute the steps of any of the above method embodiments when run.

In an exemplary embodiment, the above-mentioned computer-readable storage medium may include, but is not limited to: a USB flash drive, a read-only memory (ROM), a random access memory (RAM), a mobile hard disk, a magnetic disk or an optical disk, and other media that can store computer programs.

An embodiment of the present application further provides an electronic device, including a memory and a processor, wherein a computer program is stored in the memory, and the processor is configured to run the computer program to execute the steps in any one of the above method embodiments.

In an exemplary embodiment, the electronic device may further include a transmission device and an input/output device, wherein the transmission device is connected to the processor, and the input/output device is connected to the processor.

For specific examples in this embodiment, reference may be made to the examples described in the above embodiments and exemplary implementation modes, and this embodiment will not be described in detail herein.

Obviously, those skilled in the art should understand that the modules or steps of the above-mentioned embodiments of the present application can be implemented by a general computing device, they can be concentrated on a single computing device, or distributed on a network composed of multiple computing devices, they can be implemented by a program code executable by a computing device, so that they can be stored in a storage device and executed by the computing device, and in some cases, the steps shown or described can be executed in a different order from that here, or they can be made into individual integrated circuit modules, or multiple modules or steps therein can be made into a single integrated circuit module for implementation. In this way, the present application is not limited to any specific combination of hardware and software.

The above description is only the preferred embodiment of the present application and is not intended to limit the present application. For those skilled in the art, the embodiments of the present application may have various modifications and variations. Any modification, equivalent replacement, improvement, etc. made within the principles of the present application shall be included in the protection scope of the present application.

Claims

A method for determining call quality of a video service, comprising:

Obtain the uplink RTP packet loss situation of the video service according to the uplink Real-time Transport Protocol RTP data packet status of the video service indicated by the core network, and obtain the downlink PCDP packet loss situation of the video service according to the downlink packet data convergence protocol PDCP status report of the video service reported by the terminal indicated by the base station;

The call quality of the video service is determined based on the uplink RTP packet loss situation and the downlink PCDP packet loss situation.
The method according to claim 1, wherein obtaining the uplink RTP packet loss situation of the video service according to the uplink RTP data packet status of the video service indicated by the core network comprises:

When the core network indicates that the data packet status of the uplink RTP of the video service is in a normal transmission state, obtaining the uplink RTP packet loss status of the video service;

When the core network indicates that the uplink RTP of the video service is in an abnormal transmission state, the acquisition of the uplink RTP packet loss situation of the video service is stopped.
The method according to claim 2, wherein the abnormal transmission state includes: a stopped transmission state and an unknown transmission state.
The method according to claim 1, wherein obtaining the uplink RTP packet loss situation of the video service according to the uplink real-time transport protocol RTP data packet status of the video service indicated by the core network further comprises:

The uplink RTP packet loss situation is obtained according to the sequence number of the uplink RTP data packet.
The method according to claim 1, wherein the content of the downlink PDCP status report includes at least one of the following: the downlink PDCP packet loss rate, the number of packet losses, and the maximum number of consecutive packet losses of the video service.
The method according to claim 1, wherein reporting the downlink PDCP status report of the video service by the terminal according to the instruction of the base station comprises one of the following:

Instructing the terminal to report the downlink PDCP status report once within a preset time interval;

The terminal is instructed to trigger reporting of the downlink PDCP status report when the downlink PDCP packet loss situation of the video service meets a first preset condition.
The method according to claim 1, wherein after determining the call quality of the video service based on the RTP packet loss situation and the downlink PCDP packet loss situation, it also includes:

When the call quality of the video service deteriorates to reach a second preset condition, the call quality of the video service is optimized.
The method according to claim 7, wherein the second preset condition comprises one of the following:

The uplink RTP packet loss rate of the video service meets the first threshold and the number of packet losses meets the second threshold;

The number of continuous uplink RTP packet losses of the video service meets a third threshold;

The duration of the video service not receiving an uplink RTP packet meets a fourth threshold value;

The downlink PDCP packet loss rate of the video service meets the fifth threshold and the number of packet losses meets the sixth threshold;

The number of consecutive downlink PDCP packet losses of the video service meets a seventh threshold.
The method according to claim 7, wherein optimizing the call quality of the video service comprises:

Switching users of the video service whose call quality has deteriorated from the current serving cell to a target cell where the call quality meets the conditions; or,

Without switching the current serving cell, scheduling and optimizing the call quality of the video service.
The method according to claim 9, wherein before switching the user whose call quality of the video service has deteriorated from the current serving cell to the target cell whose call quality meets the condition, it also includes:

Whether the target cell meets the call quality satisfaction condition is determined by evaluating the uplink and downlink channel conditions of the target cell or the uplink and downlink packet loss conditions of historical users in the target cell.
A device for determining call quality of a video service, comprising:

An acquisition module is configured to acquire the uplink RTP packet loss situation of the video service according to the uplink real-time transport protocol RTP data packet status of the video service indicated by the core network, and acquire the downlink PCDP packet loss situation of the video service according to the downlink packet data convergence protocol PDCP status report of the video service reported by the terminal indicated by the base station;

The determination module is configured to determine the call quality of the video service based on the uplink RTP packet loss situation and the downlink PCDP packet loss situation.
The device according to claim 11, wherein the acquisition module comprises:

An acquiring unit, configured to acquire the uplink RTP packet loss status of the video service when the core network indicates that the data packet status of the uplink RTP of the video service is in a normal transmission state;

The stopping unit is configured to stop obtaining the uplink RTP packet loss situation of the video service when the core network indicates that the uplink RTP of the video service is in an abnormal transmission state, wherein the abnormal transmission state includes: a stopped transmission state and an unknown transmission state.
The apparatus according to claim 11, further comprising:

The optimization module is configured to optimize the call quality of the video service when the call quality of the video service deteriorates to reach a second preset condition.
A computer-readable storage medium having a computer program stored therein, wherein the computer program, when executed by a processor, implements the steps of the method described in any one of claims 1 to 10.
An electronic device comprises a memory, a processor and a computer program stored in the memory and executable on the processor, wherein the processor implements the steps of the method described in any one of claims 1 to 10 when executing the computer program.