CN117835303A - Method and device for determining call quality of video service - Google Patents
Method and device for determining call quality of video service Download PDFInfo
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
- H04N17/00—Diagnosis, testing or measuring for television systems or their details
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Abstract
The embodiment of the invention provides a method and a device for determining the call quality of video service. The method comprises the following steps: acquiring an uplink RTP packet loss condition of a video service according to an uplink real-time transport protocol RTP data packet state of the video service indicated by a core network, and acquiring a downlink PCDP packet loss condition of the video service according to a downlink packet data convergence protocol PDCP state report of the video service reported by a base station indication terminal; and determining the call quality of the video service based on the uplink RTP packet loss condition and the downlink PCDP packet loss condition. Therefore, the problem that the packet loss condition cannot be accurately evaluated at the base station side in the related technology can be solved, and the effect of improving the call quality of video service is achieved.
Description
Technical Field
The embodiment of the invention relates to the field of communication, in particular to a method and a device for determining the call quality of video service.
Background
Voice and video calls are of interest to operators as two important services in the third generation partnership project (3rd Generation Partnership Project,3GPP) mobile communication system. In the 2G and 3G times, even in the 4G early times, due to limited bandwidth resources and high traffic charge, operators only pay attention to popularization of voice service and conversation experience guarantee. With the reduction of 4G traffic charges and the further enrichment of bandwidth resources by 5G network deployment, operators begin to popularize video services, and the conversation experience of the video services is also more and more concerned.
Because the video service contains two kinds of information of sound and picture, the problems of sound word spitting, intermittent and single-pass, picture blocking, mosaic and black screen, and the problem of sound and picture non-synchronization can affect the conversation experience of the video user. In order to guarantee the call experience of the video users, the operators require the base station side to evaluate the video service quality and switch the users with poor call quality of the video service to other cells which can provide better service.
The base station side 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 relation between the two, and a misjudgment problem may exist. The two information of sound and picture in video service are transmitted in non-affirmed transmission mode (Unacknowledged Mode, UM), the communication experience index of video service can be maximally close to the evaluation of packet loss condition, but the base station side can not evaluate the packet loss condition accurately, and no effective solution is available at present for the problem.
Disclosure of Invention
The embodiment of the invention provides a method and a device for determining the call quality of video service, which at least solve the problem that the packet loss condition cannot be accurately evaluated at a base station side in the related technology.
According to one embodiment of the present invention, there is provided a call quality determining method for a video service, including: acquiring an uplink Real-time transport protocol (Real-time Transport Protocol, RTP) packet loss condition of a video service according to an uplink RTP packet status of the video service indicated by a core network, and acquiring a downlink PCDP packet loss condition of the video service according to a downlink packet data convergence protocol (Packet Data Convergence Protocol, PCDP) status report of the video service reported by a base station indication terminal; and determining the call quality of the video service based on the uplink RTP packet loss condition and the downlink PCDP packet loss condition.
According to another embodiment of the present invention, there is provided a call quality determining apparatus for a video service, including: the acquisition module is used for acquiring the uplink RTP packet loss condition of the video service according to the uplink real-time transmission protocol (RTP) data packet state of the video service indicated by the core network and acquiring the downlink PCDP packet loss condition of the video service according to a downlink Packet Data Convergence Protocol (PDCP) state report of the video service reported by a base station indication terminal; and the determining module is used for determining the call quality of the video service based on the uplink RTP packet loss condition and the downlink PCDP packet loss condition.
According to a further embodiment of the invention, there is also provided a computer readable storage medium having stored therein a computer program, wherein the computer program is arranged to perform the steps of any of the method embodiments described above when run.
According to a further embodiment of the invention, there is also provided an electronic device comprising a memory having stored therein a computer program and a processor arranged to run the computer program to perform the steps of any of the method embodiments described above.
According to the invention, the uplink RTP packet loss condition of the video service is obtained according to the uplink RTP data packet state of the video service indicated by the core network, and the downlink PCDP packet loss condition of the video service is obtained according to the downlink packet data convergence protocol PDCP state report of the video service reported by the base station indication terminal, so that the packet loss condition of the video service can be more accurately obtained at the base station side, and the assessment of the call quality of the video service is facilitated. Therefore, the problem that the packet loss condition cannot be accurately evaluated at the base station side in the related technology can be solved, and the effect of improving the call quality of video service is achieved.
Drawings
Fig. 1 is a schematic diagram of a network architecture of a call quality determining method of operating a video service according to an embodiment of the present invention;
fig. 2 is a flowchart of a call quality determination method of a video service according to an embodiment of the present invention;
fig. 3 is a block diagram of a call quality determining apparatus of a video service according to an embodiment of the present invention;
fig. 4 is a block diagram showing a construction of a call quality determining apparatus for a video service according to another embodiment of the present invention;
fig. 5 is a block diagram showing a construction of a call quality determining apparatus for a video service according to still another embodiment of the present invention;
fig. 6 is a flowchart of a video user session experience assessment method according to an embodiment of the present invention.
Detailed Description
Embodiments of the present invention will be described in detail below with reference to the accompanying drawings in conjunction with the embodiments.
It should be noted that the terms "first," "second," and the like in the description and the claims of the present invention and the above figures are used for distinguishing between similar objects and not necessarily for describing a particular sequential or chronological order.
Fig. 1 is a schematic diagram of a network architecture of a call quality determining method for running a video service according to an embodiment of the present invention, where embodiments of the present invention may be run on the network architecture shown in fig. 1, as shown in fig. 1, and the network architecture includes: the system comprises a core network, a base station and a terminal, wherein the terminal is positioned in a service cell of the base station, and the core network is used for reporting an uplink RTP state of 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 is poor, the call quality of the video service is optimized to ensure the call experience of the video user.
In this embodiment, a method for determining call quality of a video service running in the above network architecture is provided, and fig. 2 is a flowchart of a method for optimizing call quality of a video service according to an embodiment of the present invention, as shown in fig. 2, where the flowchart includes the following steps:
step S202, acquiring an uplink RTP packet loss condition of a video service according to an uplink real-time transport protocol RTP data packet state of the video service indicated by a core network, and acquiring a downlink PCDP packet loss condition of the video service according to a downlink packet data convergence protocol PDCP state report of the video service reported by a base station indication terminal;
step S204, the conversation quality of the video service is determined based on the uplink RTP packet loss condition and the downlink PCDP packet loss condition.
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 condition; or, the communication quality of the video service is determined only according to the packet loss condition of the downlink PCDP.
In step S202 of the present embodiment, it includes: under the condition that a core network indicates that the data packet state of the uplink RTP of the video service is in a normal transmission state, acquiring the packet loss condition of the uplink RTP of the video service; and under the condition that the core network indicates that the uplink RTP of the video service is in an abnormal transmission state, stopping acquiring the packet loss condition of the uplink RTP of the video service.
In this embodiment, the abnormal transmission state includes: stop transmission state and unknown transmission state.
The core network indicates the state of the corresponding service uplink RTP data packet to more accurately acquire the packet loss condition of the uplink RTP, so that the problem of misjudgment of the packet loss of the uplink RTP in the scenes of calling, three-party communication and the like can be solved.
In step S202 of the present embodiment, further includes: and acquiring the uplink RTP packet loss condition according to the sequence number of the received uplink RTP data packet.
In step S204 of the present embodiment, the content of the downlink PDCP status report includes at least one of the following: and the downlink PDCP packet loss rate, the packet loss number and the maximum continuous packet loss number of the video service.
In an exemplary embodiment, the downlink PDCP status report instructing the terminal to report the video service includes one of: the terminal is instructed to report the downlink PDCP status report once in a preset time interval; and when the downlink PDCP packet loss condition of the video service meets a first preset condition, the terminal is indicated to trigger the reporting of the downlink PDCP status report.
The base station instructs the terminal to report the corresponding service downlink PDCP status report, so that the problem of misjudgment of downlink PDCP packet loss under the scenes of uplink burst interference, insufficient uplink power and the like can be solved.
After step S204 of the present embodiment, further includes: and under the condition that the conversation quality of the video service is deteriorated to reach a second preset condition, optimizing the conversation quality of the video service.
In an exemplary embodiment, the second preset condition includes one of: the uplink RTP packet loss rate of the video service meets a first threshold value and the packet loss number meets a second threshold value; the number of continuous packet loss of the uplink RTP of the video service meets a third threshold value; the duration that the video service does not receive the uplink RTP packet meets a fourth threshold value; the downlink PDCP packet loss rate of the video service meets a fifth threshold value and the number of lost packets meets a sixth threshold value; and the number of continuous packet loss of the downlink PDCP of the video service meets a seventh threshold value.
In an exemplary embodiment, optimizing the call quality of the video service includes: switching the user with deteriorated call quality of the video service from the current service cell to a target cell with call quality meeting the condition; or, the call quality of the video service is scheduled and optimized.
In an exemplary embodiment, before the user with poor call quality of the video service is handed over from the current serving cell to the target cell with call quality meeting the condition, the method further includes: and determining whether the target cell meets the call quality meeting condition by evaluating the uplink and downlink channel conditions of the target cell or the uplink and downlink packet loss conditions of the historical users in the target cell.
Through the steps, the uplink RTP packet loss condition of the video service is obtained according to the uplink real-time transmission protocol RTP data packet state of the video service indicated by the core network, and the downlink PCDP packet loss condition of the video service is obtained according to the downlink packet data convergence protocol PDCP state report of the video service reported by the base station indication terminal, so that the packet loss condition of the video service can be more accurately obtained at the base station side, the assessment of the call quality of the video service is facilitated, and the video call quality of a user with poor call quality can be ensured by triggering strategies such as switching, scheduling optimization and the like to other better cells. The problem that the packet loss condition cannot be accurately evaluated at the base station side when the video service quality is evaluated in the related technology is solved, and the effect of the conversation quality of the video service is improved.
From the description of the above embodiments, it will be clear to a person skilled in the art that the method according to the above embodiments may be implemented by means of software plus the necessary general hardware platform, but of course also by means of hardware, but in many cases the former is a preferred embodiment. Based on such understanding, the technical solution of the present invention may be embodied essentially or in a part contributing to the prior art in the form of a software product stored in a storage medium (e.g. Read-Only Memory/Random Access Memory, ROM/RAM), magnetic disk, optical disk) comprising instructions for causing a terminal device (which may be a mobile phone, a computer, a server, or a network device, etc.) to perform the method according to the embodiments of the present invention.
The embodiment also provides a device for determining call quality of video service, which is used for implementing the foregoing embodiments and preferred embodiments, and is not described in detail. As used below, the term "module" may be a combination of software and/or hardware that implements a predetermined function. While the means described in the following embodiments are preferably implemented in software, implementation in hardware, or a combination of software and hardware, is also possible and contemplated.
Fig. 3 is a block diagram of a call quality determining apparatus for a video service according to an embodiment of the present invention, as shown in fig. 3, the apparatus comprising: an acquisition module 10 and a determination module 20.
The acquiring module 10 is configured to acquire an uplink RTP packet loss condition of a video service according to an uplink RTP packet status of the video service indicated by a core network, and acquire a downlink PCDP packet loss condition of the video service according to a downlink packet data convergence protocol PDCP status report of the video service reported by a base station indication terminal;
a determining module 20, configured to evaluate the call quality of the video service according to at least one of the following: and the RTP packet loss condition and the downlink PCDP packet loss condition.
Fig. 4 is a block diagram of a call quality determining apparatus for a video service according to an embodiment of the present invention, and as shown in fig. 4, the apparatus includes, in addition to all the modules shown in fig. 3, an acquisition module 10 including:
an obtaining unit 11, configured to obtain an uplink RTP packet loss condition of the video service when the core network indicates that a data packet state of the uplink RTP of the video service is in a normal transmission state;
a stopping unit 12, configured to stop acquiring an uplink RTP packet loss condition of the video service when the core network indicates that the uplink RTP of the video service is in an abnormal transmission state, where the abnormal transmission state includes: stop transmission state and unknown transmission state.
Fig. 5 is a block diagram of a call quality determining apparatus for a video service according to an embodiment of the present invention, as shown in fig. 5, which includes, in addition to all the modules shown in fig. 4:
and an optimizing module 30, configured to optimize the call quality of the video service when the call quality degradation of the video service reaches a second preset condition.
It should be noted that each of the above modules may be implemented by software or hardware, and for the latter, it may be implemented by, but not limited to: the modules are all located in the same processor; alternatively, the above modules may be located in different processors in any combination.
In order to facilitate understanding of the technical solutions provided by the present invention, the following details will be described in connection with embodiments of specific scenarios.
Aiming at the problems in the related art: the communication quality of the video service is evaluated through packet loss, and although the communication quality can be maximally close to the communication experience index of the video service, the accurate evaluation of the packet loss cannot be achieved by the base station side. The embodiment of the invention provides a video user conversation experience assessment method. The method comprises the steps of indicating the state of an uplink RTP data packet corresponding to the video service through a core network, and solving the problem of misjudgment of the uplink RTP data packet under the scenes of calling, three-party communication and the like. And secondly, the base station instructs the terminal to report the corresponding service downlink PDCP state report, thereby solving the problem of misjudgment of downlink PDCP packet loss in the scenes of uplink burst interference, insufficient uplink power and the like. And finally, evaluating the conversation experience of the video user according to the real packet loss conditions of the uplink RTP and the downlink PDCP, and ensuring the conversation experience of the video user by triggering strategies such as switching, dispatching optimization and the like to other better cells for poor quality users.
Fig. 6 is a flowchart of a video user session experience assessment method according to an embodiment of the present invention, as shown in fig. 6, the method includes the steps of:
in 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.
Specifically, for uplink quality detection, RTP packet loss may reflect packet dropping and air interface packet dropping at the same time, but it is considered that some terminals may perform RTP packet dropping before PDCP packet grouping, so RTP packet dropping is more suitable for reflecting uplink quality status than PDCP packet dropping.
For downlink quality detection, RTP packet loss can reflect transmission packet loss, base station packet loss and air interface packet loss at the same time. However, the transmission packet loss does not belong to the current serving cell, and even if other target neighbor cells are selected to trigger handover, the transmission packet loss problem may still exist. And PDCP packet loss can reflect only the base station packet loss and air interface packet loss, so PDCP packet loss is more suitable for reflecting the downlink quality state than RTP packet loss.
In this embodiment, considering the problems of voice word-spitting, intermittent and single-pass, picture blocking, mosaic and black screen, and the problem of voice and picture non-synchronization, etc., the call experience of the video user may be affected, and the degree of influence of different problems on the call experience may be different, and statistics may be performed on the uplink and downlink packet loss conditions of the voice and picture, respectively, and when any of the following conditions is satisfied, the quality of the video call experience is considered to be poor, and the target neighbor cell needs to be selected to trigger the switching.
Specifically, for the uplink and downlink packet loss situations of the voice, in this embodiment, a 5G service quality indicator 1 (5G Quality of Service Identifier 1,5QI1) service of the 5G system is taken as an example to describe:
the RTP packet loss rate of the 5QI1 service meets the threshold and the number of lost packets meets the threshold;
considering that the 5QI1 service uplink transmits an activation packet when the local end user speaks and the 5QI1 service uplink transmits a silence packet when the local end user does not speak, the silence packet loss has little influence on the conversation experience, so the packet loss rate can be converted according to a certain conversion factor in calculation;
the number of continuous packet loss of the 5QI1 service uplink RTP meets the threshold;
the time length that the 5QI1 service does not receive the uplink RTP packet meets the threshold;
the downlink PDCP packet loss rate of the 5QI1 service meets the threshold and the number of lost packets meets the threshold;
considering that the 5QI1 service downlink transmits an activation packet when the opposite terminal user speaks and the 5QI1 service downlink transmits a silence packet when the opposite terminal user does not speak, the silence packet loss has little influence on the conversation experience, and the packet loss rate can be calculated according to a certain conversion factor;
the number of continuous packet loss of the downlink PDCP of the 5QI1 service meets the threshold;
for the uplink and downlink packet loss of the picture, in this embodiment, a 5QI2 service of a 5G system is taken as an example to describe:
the 5QI2 service uplink RTP packet loss rate meets the threshold and the packet loss number meets the threshold;
the number of continuous packet loss of the 5QI2 service uplink RTP meets the threshold;
the time length that the 5QI2 service does not receive the uplink RTP packet meets the threshold.
The packet loss rate of the downlink PDCP of the 5QI2 service meets the threshold, and the number of the lost packets meets the threshold;
the number of continuous packet loss of the downlink PDCP of the 5QI2 service meets the threshold.
According to the conversation quality of the video service reflected by the upstream RTP packet loss, the method comprises the following steps: indicating the state of the corresponding service uplink RTP data packet through the core network;
specifically, under normal conditions, the base station side calculates the RTP packet loss condition according to the SN number of the received RTP packet, and calculates the activation packet and silence packet loss condition according to the sending rule of the activation packet and silence packet aiming at the 5QI1 service. For example: the SN number of the uplink RTP packet received by the 5QI1 service at the previous time is 502, the SN number of the RTP packet received at the current time is 509, the time interval of the uplink RTP packet received at the second time is 400ms, the transmission period of the active packet is 20ms, the transmission period of the silent packet is 160ms, 4 RTP active packets are calculated to be lost from the uplink of the 5QI1 service, and 2 RTP active packets are lost.
If the base station detects that the uplink RTP packet is not received for a long time, the base station needs to select a target neighbor cell to trigger handover, which is generally caused by too poor uplink channel conditions of the serving cell, except for some special situations, such as: the originating scene cannot generate an uplink RTP packet because the opposite terminal is not connected, the user call in the three-party call scene is kept from generating an uplink RTP packet, the camera closing scene does not need to generate an uplink RTP packet, and the like, and for these special scenes, the state of the corresponding service uplink RTP data packet is indicated through the core network, wherein the state attributes include but are not limited to: normal transmission, stop transmission, unknown and the like, and can solve the problem of misjudgment of uplink RTP packet loss in the scenes of calling, three-way communication and the like. Taking a 5G system as an example, for a newly-built service scenario, the core network may indicate in a PDU session resource establishment request (PDU SESSION RESOURCE SETUP REQUEST) and a PDU session resource modification request (PDU SESSION RESOURCE MODIFY REQUEST) message; for the service state attribute change scenario, the core network may indicate in the PDU SESSION RESOURCE MODIFY REQUEST message, and the base station only performs uplink RTP packet loss evaluation on the service in the normal transmission state. In particular, for an inter-station handover scenario, since the service state attribute is not changed, the handover target side base station may not be able to obtain the latest state of the corresponding service from the core network, and the handover source side base station needs to indicate the latest state of the corresponding service to the target side base station in a handover request message.
Specifically, reflecting the call quality of the video service according to the packet loss condition of the downlink PCDP includes: a base station instructs a terminal to report a corresponding service downlink PDCP state report;
specifically, under normal conditions, the base station side estimates the PDCP packet loss according to the feedback result of the received downlink PDCP packet. For example: and 20 downlink PDCP packets sent in a period of time of the 5QI1 service are fed back to be 18 successful, and 2 failures are fed back, so that the downlink PDCP packet loss rate of the 5QI1 service in the period of time is considered to be 10%.
Under the scenes of uplink burst interference, insufficient uplink power and the like, the feedback result of the base station on some downlink PDCP packets can not be analyzed normally, and the problem of packet loss misjudgment can occur. The base station side instructs the terminal to report the state of the downlink PDCP data packet of the corresponding service, so that the problem of misjudgment of the downlink PDCP data packet under the scenes of uplink burst interference, insufficient uplink power and the like can be solved; wherein the status report includes two modes of periodic and event type, for example: the periodic report of the status can be reported every 2s, and the report content includes but is not limited to the corresponding downlink PDCP packet loss rate, the packet loss number, the maximum continuous packet loss number and the like; the event type can be considered to trigger the reporting of the status report when the packet loss rate, the packet loss number, the maximum continuous packet loss number and the like of the corresponding service downlink DPCP meet one or more conditions. The periodic and event status reports can be used independently to save signaling overhead, or can be used simultaneously to improve detection efficiency, and can be selected reasonably according to actual network conditions.
Step S604, switching the user with poor call quality of the video service to the cell with good call quality of other video service, or performing a scheduling optimization strategy to ensure the call experience of the video user.
Specifically, according to the related strategies of step S604 and step S606, the base station only performs uplink RTP packet loss evaluation on the service in the normal transmission state, and simultaneously performs downlink PDCP packet loss evaluation according to the PDCP status report reported by the terminal, so as to obtain the real packet loss condition evaluation video user session experience. The user with poor call quality to the video service can ensure the call experience of the video user by triggering strategies such as switching, scheduling optimization and the like to other better cells. Particularly, when the switching target cell is selected, the uplink/downlink packet loss situation after the user switching cannot be predicted in advance, so that the estimation can be performed by judging the uplink/downlink channel condition of the target cell or the uplink/downlink packet loss situation of the historical user, and the problem probability that the video user communication experience after the switching cannot be improved or even deteriorated is reduced.
The method provided by the embodiment of the invention is suitable for long term evolution (Long Term Evolution, LTE), new Radio (NR) and guarantee strategies of other mobile systems of the subsequent 3 GPP.
In the above embodiment of the present invention, aiming at the problem that the base station side cannot accurately evaluate the video user call experience, the core network is used to indicate the state of the corresponding service uplink RTP data packet, so as to solve the problem of uplink RTP packet loss misjudgment in the scenes of voice call initiation, three-way call, etc.; secondly, a base station instructs a terminal to report a corresponding service downlink PDCP state report, so that the problem of misjudgment of downlink PDCP packet loss in the scenes of uplink burst interference, insufficient uplink power and the like is solved; and finally, evaluating the conversation experience of the video user according to the real packet loss conditions of the uplink RTP and the downlink PDCP, wherein the user with poor conversation quality of the video service can ensure the conversation experience of the video user by triggering strategies such as switching, scheduling optimization and the like to other better cells.
The following is a method for implementing the video user session experience assessment in a specific scene:
assuming that the user performs video call service at the 5G base station, the user is initially in a call starting stage, the opposite terminal user does not make a call, the core network indicates that the uplink RTP of the corresponding service is in a transmission stopping state through PDU SESSION RESOURCE MODIFY REQUEST information, and the base station side does not perform uplink RTP packet loss evaluation on the related service, so that the condition that the uplink RTP packet of the corresponding service is not received for a long time is prevented from triggering switching.
After the opposite terminal user makes a call, the core network indicates that the corresponding service uplink RTP is in a normal transmission state through PDU SESSION RESOURCE MODIFY REQUEST information, and the base station side starts to carry out uplink RTP packet loss evaluation on the related service. If the evaluation corresponds to that the uplink RTP packet loss rate in the evaluation period of the service 2s is more than 6% and the packet loss number is more than 10, the condition of poor video call experience is met, the target cell is selected to trigger switching, otherwise, the packet loss evaluation is continued.
Assuming that the user answers other calls by the opposite-end user in the call process, the local-end user is suspended by the three-party call process, the core network indicates that the corresponding service uplink RTP is in a transmission stop state through PDU SESSION RESOURCE MODIFY REQUEST information, and the base station side does not carry out uplink RTP packet loss evaluation on related services, so that the condition that the corresponding service uplink RTP packet is not received for a long time is prevented from triggering switching.
After the opposite terminal user hangs up other telephones, the three-way call process is ended, normal call with the local terminal user is resumed, the core network indicates that the corresponding service uplink RTP is in a normal transmission state through PDU SESSION RESOURCE MODIFY REQUEST information, and the base station side starts to carry out uplink RTP packet loss evaluation on the related service; if the number of the continuous packet loss of the corresponding service uplink RTP is evaluated to be larger than 5, and the condition of poor video call experience is met, selecting a target cell to trigger switching, otherwise, continuing to evaluate the packet loss.
When the base station transmits downlink PDCP data packets to the user on related services, the base station requests the UE to have the downlink PDCP packet loss rate of more than 6 percent and the packet loss number of more than 10 in the corresponding service 2s evaluation period through RRC reconfiguration information, and the condition of poor video call experience is met, a target cell is selected to trigger switching, or the call experience of the user is improved through scheduling optimization. Otherwise, continuing to carry out packet loss evaluation.
According to the embodiment of the invention, when implementing the video user call experience evaluation party, the video user call experience is evaluated according to the real packet loss conditions of the uplink RTP and the downlink PDCP, and the user with poor call quality of the video service can ensure the call experience of the video user by triggering strategies such as switching, scheduling optimization and the like to other better cells
Embodiments of the present invention also provide a computer readable storage medium having a computer program stored therein, wherein the computer program is arranged to perform the steps of any of the method embodiments described above when run.
In one exemplary embodiment, the computer readable storage medium may include, but is not limited to: a usb disk, a Read-Only Memory (ROM), a random access Memory (Random Access Memory, RAM), a removable hard disk, a magnetic disk, or an optical disk, or other various media capable of storing a computer program.
An embodiment of the invention also provides an electronic device comprising a memory having stored therein a computer program and a processor arranged to run the computer program to perform the steps of any of the method embodiments described above.
In an exemplary embodiment, the electronic apparatus may further include a transmission device connected to the processor, and an input/output device connected to the processor.
Specific examples in this embodiment may refer to the examples described in the foregoing embodiments and the exemplary implementation, and this embodiment is not described herein.
It will be appreciated by those skilled in the art that the modules or steps of the invention described above may be implemented in a general purpose computing device, they may be concentrated on a single computing device, or distributed across a network of computing devices, they may be implemented in program code executable by computing devices, so that they may be stored in a storage device for execution by computing devices, and in some cases, the steps shown or described may be performed in a different order than that shown or described herein, or they may be separately fabricated into individual integrated circuit modules, or multiple modules or steps of them may be fabricated into a single integrated circuit module. Thus, the present invention is not limited to any specific combination of hardware and software.
The above description is only of the preferred embodiments of the present invention and is not intended to limit the present invention, but various modifications and variations can be made to the present invention by those skilled in the art. Any modification, equivalent replacement, improvement, etc. made within the principle of the present invention should be included in the protection scope of the present invention.
Claims (15)
1. A method for determining call quality of a video service, comprising:
acquiring an uplink RTP packet loss condition of a video service according to an uplink real-time transport protocol RTP data packet state of the video service indicated by a core network, and acquiring a downlink PCDP packet loss condition of the video service according to a downlink packet data convergence protocol PDCP state report of the video service reported by a base station indication terminal;
and determining the call quality of the video service based on the uplink RTP packet loss condition and the downlink PCDP packet loss condition.
2. The method according to claim 1, wherein obtaining an uplink RTP packet loss condition of a video service according to an uplink RTP packet state of the video service indicated by a core network, comprises:
under the condition that a core network indicates that the data packet state of the uplink RTP of the video service is in a normal transmission state, acquiring the packet loss condition of the uplink RTP of the video service;
and under the condition that the core network indicates that the uplink RTP of the video service is in an abnormal transmission state, stopping acquiring the packet loss condition of the uplink RTP of the video service.
3. The method of claim 2, wherein the abnormal transmission state comprises: stop transmission state and unknown transmission state.
4. The method according to claim 1, wherein the step of obtaining an uplink RTP packet loss condition of the video service according to an uplink RTP packet status of the video service indicated by the core network, further comprises:
and acquiring the uplink RTP packet loss condition according to the sequence number of the uplink RTP data packet.
5. The method of claim 1, wherein the content of the downlink PDCP status report comprises at least one of: and the downlink PDCP packet loss rate, the packet loss number and the maximum continuous packet loss number of the video service.
6. The method of claim 1, wherein the instructing the terminal to report the downlink PDCP status report for the video service according to the base station comprises one of:
the terminal is instructed to report the downlink PDCP status report once in a preset time interval;
and when the downlink PDCP packet loss condition of the video service meets a first preset condition, the terminal is indicated to trigger the reporting of the downlink PDCP status report.
7. The method of claim 1, wherein after determining the call quality of the video service based on the RTP packet loss situation and the downstream PCDP packet loss situation, further comprising:
and under the condition that the conversation quality of the video service is deteriorated to reach a second preset condition, optimizing the conversation quality of the video service.
8. The method of claim 7, wherein the second preset condition comprises one of:
the uplink RTP packet loss rate of the video service meets a first threshold value and the packet loss number meets a second threshold value;
the number of continuous packet loss of the uplink RTP of the video service meets a third threshold value;
the duration that the video service does not receive the uplink RTP packet meets a fourth threshold value;
the downlink PDCP packet loss rate of the video service meets a fifth threshold value and the number of lost packets meets a sixth threshold value;
and the number of continuous packet loss of the downlink PDCP of the video service meets a seventh threshold value.
9. The method of claim 7, wherein optimizing the call quality of the video service comprises:
switching the user with deteriorated call quality of the video service from the current service cell to a target cell with call quality meeting the condition; or alternatively, the first and second heat exchangers may be,
and under the condition that the current service cell switching is not carried out, the call quality of the video service is scheduled and optimized.
10. The method of claim 9, wherein before switching the user with deteriorated call quality of the video service from the current serving cell to the target cell with call quality meeting the condition, further comprising:
and determining whether the target cell meets the call quality meeting condition by evaluating the uplink and downlink channel conditions of the target cell or the uplink and downlink packet loss conditions of the historical users in the target cell.
11. A call quality determining apparatus for a video service, comprising:
the acquisition module is used for acquiring the uplink RTP packet loss condition of the video service according to the uplink real-time transmission protocol (RTP) data packet state of the video service indicated by the core network and acquiring the downlink PCDP packet loss condition of the video service according to a downlink Packet Data Convergence Protocol (PDCP) state report of the video service reported by a base station indication terminal;
and the determining module is used for determining the call quality of the video service based on the uplink RTP packet loss condition and the downlink PCDP packet loss condition.
12. The apparatus of claim 11, wherein the acquisition module comprises:
an obtaining unit, configured to obtain an uplink RTP packet loss condition of the video service when a core network indicates that a data packet state of the uplink RTP of the video service is in a normal transmission state;
a stopping unit, configured to stop acquiring an uplink RTP packet loss condition of the video service when the core network indicates that the uplink RTP of the video service is in an abnormal transmission state, where the abnormal transmission state includes: stop transmission state and unknown transmission state.
13. The apparatus as recited in claim 11, further comprising:
and the optimizing module is used for optimizing the call quality of the video service under the condition that the call quality of the video service is deteriorated to reach a second preset condition.
14. A computer readable storage medium, characterized in that a computer program is stored in the computer readable storage medium, wherein the computer program, when being executed by a processor, implements the steps of the method according to any of the claims 1 to 10.
15. An electronic device comprising a memory, a processor and a computer program stored on the memory and executable on the processor, characterized in that the processor implements the steps of the method as claimed in any one of claims 1 to 10 when the computer program is executed.
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| Application Number | Priority Date | Filing Date | Title |
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| CN202211202780.9A CN117835303A (en) | 2022-09-29 | 2022-09-29 | Method and device for determining call quality of video service |
| PCT/CN2023/118623 WO2024067103A1 (en) | 2022-09-29 | 2023-09-13 | Method and apparatus for determining call quality of video service |
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| CN202211202780.9A CN117835303A (en) | 2022-09-29 | 2022-09-29 | Method and device for determining call quality of video service |
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| CN105376750B (en) * | 2014-08-29 | 2018-10-30 | 成都鼎桥通信技术有限公司 | Speech quality assessment method and equipment |
| EP3200396B1 (en) * | 2014-09-30 | 2018-11-07 | Huawei Technologies Co. Ltd. | Apparatus, system and method for acquiring quality of service parameter of voice over internet protocol service |
| CN107277499B (en) * | 2016-04-08 | 2019-03-29 | 上海大唐移动通信设备有限公司 | A kind of video quality evaluation method and device |
| CN108401263B (en) * | 2017-02-07 | 2021-09-17 | 大唐移动通信设备有限公司 | Voice quality assessment method and device |
| EP4199526A4 (en) * | 2020-09-14 | 2023-08-23 | Huawei Technologies Co., Ltd. | Communication method and apparatus |
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