CN117176972B - Cloud conference audio and video transmission system and method based on WebRTC technology - Google Patents
Cloud conference audio and video transmission system and method based on WebRTC technology Download PDFInfo
- Publication number
- CN117176972B CN117176972B CN202311024476.4A CN202311024476A CN117176972B CN 117176972 B CN117176972 B CN 117176972B CN 202311024476 A CN202311024476 A CN 202311024476A CN 117176972 B CN117176972 B CN 117176972B
- Authority
- CN
- China
- Prior art keywords
- audio
- video
- conference
- module
- person
- Prior art date
- Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
- Active
Links
- 230000005540 biological transmission Effects 0.000 title claims abstract description 50
- 238000005516 engineering process Methods 0.000 title claims abstract description 34
- 238000000034 method Methods 0.000 title claims abstract description 31
- 238000004891 communication Methods 0.000 claims abstract description 50
- 238000012545 processing Methods 0.000 claims abstract description 6
- 230000002452 interceptive effect Effects 0.000 claims description 3
- 230000008569 process Effects 0.000 claims description 3
- 238000009877 rendering Methods 0.000 claims description 3
- 230000003044 adaptive effect Effects 0.000 description 5
- 230000011664 signaling Effects 0.000 description 5
- 238000010586 diagram Methods 0.000 description 4
- 230000009286 beneficial effect Effects 0.000 description 2
- 230000008859 change Effects 0.000 description 2
- 230000000694 effects Effects 0.000 description 2
- 230000004048 modification Effects 0.000 description 2
- 238000012986 modification Methods 0.000 description 2
- 230000000903 blocking effect Effects 0.000 description 1
- 125000004122 cyclic group Chemical group 0.000 description 1
- 230000003993 interaction Effects 0.000 description 1
- 238000012423 maintenance Methods 0.000 description 1
- 238000005457 optimization Methods 0.000 description 1
- 238000012795 verification Methods 0.000 description 1
Landscapes
- Two-Way Televisions, Distribution Of Moving Picture Or The Like (AREA)
- Telephonic Communication Services (AREA)
Abstract
The application discloses a cloud conference audio and video transmission system and a method based on a WebRTC technology, wherein the system comprises: the device comprises a communication module, a media negotiation module and a network transmission module. The communication module is responsible for carrying out audio and video communication with other participants, and can be subdivided into a point-to-point audio and video communication sub-module and a multi-person conference sub-module, which are respectively used for establishing one-to-one and multi-person audio and video connection relations based on the WebRTC technology; the media negotiation module is used for negotiating media parameters between two communication parties to ensure consistency; the network transmission module is responsible for processing the transmission of audio and video data. The system can also comprise a self-adaptive adjustment module, which can dynamically adjust parameters such as a coder and decoder, resolution, frame rate and the like according to network bandwidth and delay conditions, and ensures the quality and fluency of audio and video. The application solves the problem of comprehensive functional coverage and meets the communication and cooperation requirements of different scenes of users.
Description
Technical Field
The application relates to a video conference system, belongs to the field of audio and video transmission, and particularly relates to a cloud conference audio and video transmission system and method based on a WebRTC technology.
Background
WebRTC has been widely used in the field of real-time communications, including video conferencing applications. There are many other communication platforms or software available on the market, such as Zoom, microsoft Teams, telecommunications conferences, etc.
Although WebRTC has made a significant progress in the field of real-time communications as a prior art solution, there are drawbacks in terms of network compatibility, signaling server dependence, network transmission performance, conference size limitations, and lack of customized functionality.
In terms of network compatibility, webRTC implementations between different browsers and devices may present compatibility issues because WebRTC relies on browser and device support. This may result in a failure to achieve fully compatible audio-video communications on certain browsers or devices, limiting the user's choice and experience.
In terms of signaling server dependency, webRTC needs to use a signaling server to establish a connection between two parties to a communication. The signaling server can act to coordinate and exchange media related information. However, the deployment and maintenance of signaling servers may require additional cost and complexity.
In terms of network transmission performance, webRTC generally uses a UDP protocol when transmitting audio and video data, and UDP is fast, but has a high requirement on network stability. In an unstable or high packet loss rate network environment, audio and video quality may be affected, resulting in problems such as delay, jitter, and frame loss.
In terms of meeting size limitations, in the native implementation of WebRTC, the size of multi-person video conferences is limited by the devices and networks. As the number of people involved in a conference increases, more bandwidth and processing power is required, which may lead to reduced performance or failure to meet the needs of a large-scale conference.
Existing WebRTC videoconferencing applications typically lack flexible customization functionality in terms of customization functionality. The user may not be able to adjust the bandwidth, resolution, codec, etc. parameters according to his own needs to optimize the conference experience.
Disclosure of Invention
According to one aspect of the application, in order to provide better audio and video conference experience, a cloud conference audio and video transmission system based on the WebRTC technology is provided, and the system comprises a communication module, a media negotiation module and a network transmission module; wherein,
The communication module is used for processing the audio and video stream by adopting the WebRTC technology according to the instruction input by the user on the terminal interactive interface, and establishing one-to-one or multi-person audio and video communication connection between the current user and other participants;
The media negotiation module is used for negotiating media parameters among communication parties and keeping consistency;
The network transmission module is used for transmitting the audio and video data.
Optionally, the terminal may be a smart phone or a tablet computer.
Optionally, the communication module includes:
And the point-to-point audio and video call sub-module is used for establishing point-to-point audio and video call connection according to a gateway or SIP mode selected by a user, and capturing, encoding and decoding, transmitting, playing and rendering the audio and video.
Optionally, the webRTC library employed by the communication module adds support for two codec capabilities omx.img and omx.hisi.
Optionally, the communication module further includes: and the multi-person conference sub-module is used for providing multi-person video conference functions, including creating a conference room, inviting other participants and managing the conference.
Optionally, the multi-person conference module uses Videoroom plugins or MCU conferences to enable multi-person video communications.
Optionally, the system further comprises: and the self-adaptive adjustment module is used for dynamically adjusting relevant parameters of the audio and video data according to network conditions.
Optionally, the network conditions include, but are not limited to: network bandwidth, delay conditions.
Optionally, the relevant parameters of the audio-video data include, but are not limited to: the codec, resolution, frame rate are dynamically adjusted.
Optionally, the adaptive adjustment module circularly calls a peerconnection.getstats () method provided by WebRTC to monitor googRtt parameter changes in audioTrack in real time;
if googRtt > =300±5 and more than 3 times in succession, prompting to close the current camera;
If googRtt is more than 100+/-5 and is more than 3 times continuously, changing the resolution of the camera to the lowest resolution supported by the current equipment, and changing the frame rate to 15;
If googRtt is less than 80 plus or minus 5 and continuously exceeds 10 times, prompting to open the camera;
if googRtt < 50+ -5 and continuously exceeds 10 times, restoring the resolution set by the user;
The above determination is performed in the order from top to bottom using the if else method, and if any one of the conditions is satisfied, the other is not performed.
Optionally, the system further comprises: and the live broadcast and recorded broadcast module is used for carrying out live broadcast and recorded broadcast on the conference content generated by the multi-person conference submodule, and comprises connection and pushing of a streaming media server, generation and sharing of a watching link and storage and playback of recorded content.
Optionally, the live broadcast and the recorded broadcast are respectively realized through a Streaming plug-in and a Recordplay plug-in.
Optionally, the media parameters include, but are not limited to: decoder, resolution, frame rate.
Optionally, the network transmission module discovers the best network path using ICE protocol and performs candidate selection.
According to another aspect of the application, a cloud conference audio/video transmission method based on WebRTC technology is provided, including:
after receiving an instruction initiated by a user from a user interface, the communication module establishes one-to-one audio and video call or establishes a multi-person video conference;
the media negotiation module negotiates media parameters between the two communication parties and keeps consistent;
And the network transmission module transmits the audio and video data.
Optionally, for the one-to-one audio/video call, the system communicates according to the gateway or SIP mode selected by the user, and the application establishes audio/video connection and realizes audio/video transmission through WebRTC technology.
Optionally, for the multi-person video conference, the system establishes the multi-person video conference according to a Videoroom plug-in or MCU conference mode selected by a user; the Videoroom plug-in establishes a multi-person video room and invites other participants to join in to realize multi-person communication; the MCU conference processes audio and video streams among a plurality of participants through the central control unit, so that the multi-person video conference is realized.
Optionally, the method further comprises: and the self-adaptive adjustment module dynamically adjusts relevant parameters of the audio and video data according to network conditions.
Optionally, the dynamic adjustment includes:
Circularly calling a peerconnection.getstats () method provided by WebRTC to monitor googRtt parameter changes in audioTrack in real time;
If googRtt > =300±5 and the number of times exceeds 3 continuously, controlling to close the current camera, and simultaneously prompting the user;
If googRtt is more than 100+/-5 and is more than 3 times continuously, changing the resolution of the camera to the lowest resolution supported by the current equipment, and changing the frame rate to 15;
If googRtt is less than 80 plus or minus 5 and continuously exceeds 10 times, prompting to open the camera;
if googRtt < 50+ -5 and continuously exceeds 10 times, restoring the resolution set by the user;
The above determination is performed in the order from top to bottom using the if else method, and if any one of the conditions is satisfied, the other is not performed.
Optionally, the method further comprises:
And the live broadcast and recorded broadcast module is used for carrying out live broadcast and recorded broadcast on conference contents of the multi-person video conference.
Optionally, the live broadcast and recording broadcast module carries out real-time live broadcast of conference content through a Streaming plug-in; the Recordplay plug-in is used to record conference content.
The application has the beneficial effects that:
1) Comprehensive functional coverage: the application provides a plurality of functions such as point-to-point audio and video call, multi-person video conference, live broadcast recording and broadcasting and the like, and meets the communication and cooperation requirements of users in different scenes. Compared with other prior art schemes with only partial functions, the comprehensive function coverage of the application is more comprehensive.
2) Based on WebRTC technology: the application adopts the WebRTC technology as the bottom layer for realizing, which ensures that the audio and video communication is more stable and efficient. WebRTC provides lower latency and better network adaptability by using protocols such as UDP and TCP, as well as adaptive tuning and network transport optimization.
3) Mobile application platform: the system of the application can be operated on mobile equipment such as smart phones, tablet computers and the like. This enables users to conduct meetings and communications anywhere and anytime, increasing flexibility and convenience.
4) Adaptive adjustment and personalized settings: the application has the self-adaptive adjusting function, and dynamically adjusts parameters such as a coder and decoder, resolution, frame rate and the like according to network conditions so as to provide optimal audio and video quality and fluency. Meanwhile, the user can adjust parameters such as transmission mode, bandwidth limitation, resolution ratio and the like in application setting according to personalized requirements so as to meet specific requirements of different users.
5) Real-time live broadcast and recorded broadcast functions: the app of the application supports the functions of live broadcast and recorded broadcast, and users can live broadcast conference contents to other people in real time for watching or record the conference for subsequent playback and sharing. This provides the user with more communication and sharing options.
Drawings
Fig. 1 is a block diagram of the internal structural relationship of a WebRTC technology-based audio and video transmission system for a cloud conference according to an embodiment of the present application;
Fig. 2 is a workflow diagram of a WebRTC technology-based cloud conference audio/video transmission system according to an embodiment of the present application;
fig. 3 (a) and fig. 3 (b) are respectively audio/video effect comparison diagrams according to an embodiment of the application.
Detailed Description
The present application is described in detail below with reference to examples, but the present application is not limited to these examples.
As shown in fig. 1, in one embodiment of the present application, an internal structural relationship block diagram of a WebRTC technology-based cloud conference audio/video transmission system includes: the device comprises a communication module, a media negotiation module and a network transmission module. The communication module is used for processing the audio and video stream by adopting the WebRTC technology according to the instruction input by the user on the terminal interactive interface, and establishing one-to-one or multi-person audio and video communication connection between the current user and other participants;
The media negotiation module is used for negotiating media parameters between two communication parties and keeping consistency;
The network transmission module is used for transmitting the audio and video data.
In one embodiment, the communication module further comprises: and the point-to-point audio and video call sub-module is used for establishing point-to-point audio and video call connection according to a gateway or SIP mode selected by a user, and capturing, encoding and decoding, transmitting, playing and rendering the audio and video.
In one embodiment, the library file of WebRTC employed by the communication module is optimized compared to the prior art: modifying webtc source codes, and increasing judgment processing on OMX.IMG and OMX.hisi; modifying the sdp negotiation content. Since WebRTC defaults to soft solution, a stuck and stained screen situation occurs.
In one embodiment, the WebRTC source code modification is as follows:
The application can effectively solve the problem of audio/video jamming or screen display, and particularly aims at the problem of the Hua-Cheng decoding H264 format jamming.
In one embodiment, the communication module further comprises: and the multi-person conference sub-module is used for providing multi-person video conference functions, including creating a conference room, inviting other participants and managing the conference.
In one embodiment, the multi-person conference module may use Videoroom plugins or MCU conferences to enable multi-person video communications.
In one embodiment, the system further comprises an adaptive adjustment module, which is used for dynamically adjusting parameters such as a codec, resolution, frame rate and the like according to network bandwidth and delay conditions so as to ensure optimal audio/video quality and smoothness.
In order to ensure audio quality, especially for the problem of poor audio quality in weak network, the self-adaptive adjustment module of the application adopts a cyclic calling mode, such as calling the peerconnection.getstats () method provided by WebRTC once every second to monitor googRtt parameter change in audioTrack in real time, and makes a judgment according to googRtt parameter change:
if googRtt > =300±5 and more than 3 times in succession, prompting to close the current camera;
If googRtt is more than 100+/-5 and is more than 3 times continuously, changing the resolution of the camera to the lowest resolution supported by the current equipment, and changing the frame rate to 15;
If googRtt is less than 80 plus or minus 5 and continuously exceeds 10 times, prompting to open the camera;
if googRtt < 50+ -5 and continuously exceeds 10 times, restoring the resolution set by the user;
The above determination is performed in the order from top to bottom using the if else method, and if any one of the conditions is satisfied, the other is not performed.
In one embodiment, the system further comprises a live broadcast and recorded broadcast module for supporting live broadcast and recorded broadcast of the conference content, wherein the live broadcast and recorded broadcast module comprises connection and pushing of a streaming media server, generation and sharing of a watching link, and storage and playback of recorded content. The support is realized by live broadcasting and recorded broadcasting of the conference content through a Streaming plug-in and a Recordplay plug-in respectively.
In one embodiment, the network transmission module uses the ICE protocol to find the best network path and make candidate selections.
In one embodiment, the media negotiation module is configured to negotiate media parameters between two parties of communication to keep them consistent. The parameters include, but are not limited to: codec, resolution, frame rate.
Referring to fig. 2, a flow chart of the operation of the system described above is shown in one embodiment.
(1) User interaction interface
The user launches the app application at the terminal and logs in. The user opens the app and inputs login credentials for identity verification; the application verifies the user identity and loads the main interface.
(2) Point-to-point audio-video call
The user selects a point-to-point audio and video call function; the system invokes audio and video capturing functions of the device, such as a microphone and a camera, to acquire audio and video data of the user; the system establishes audio and video connection with another user through a WebRTC technology; the system encodes and decodes the audio and video data through a codec (such as VP8, VP9, H.264, opus, PCMU, etc.); and playing the decoded data at a receiving end to realize real-time audio and video call between the two parties.
(3) Multi-person video conferencing.
The user selects a multi-person video conference function; the system creates a conference room and generates a unique room ID; the system joins other participants in the conference room by inviting them to join them; the audio and video data of each participant are transmitted through the WebRTC technology; the system combines and renders the audio and video streams of different participants at the receiving end, so as to realize the multi-user video call.
(4) Live and recorded broadcast.
The user selects a live broadcast or recorded broadcast function; if the live broadcast is selected, the system pushes the conference content to a Streaming media server in real time through a Streaming plug-in; other users may view the meeting content by viewing the live links; if the recording and broadcasting are selected, the system records the conference content through Recordplay plug-ins; the recorded content may be subsequently played back, shared, or archived.
(5) Adaptive adjustment and personalized settings.
The system performs self-adaptive adjustment according to network conditions, such as dynamically adjusting parameters of a coder-decoder, resolution, frame rate and the like according to bandwidth and delay, so as to optimize audio and video quality and fluency. The user may personalize the transmission mode (UDP/TCP), bandwidth limitations, resolution, etc. in the system settings to meet specific needs and optimize the user experience.
Fig. 3 (a) is a screenshot of a meeting using an existing system, and shows that a screen blocking situation occurs, there is a delay, and video smoothness is poor. In the same scene, the cloud conference audio/video transmission system based on the WebRTC technology is used for conference, and the video screenshot state in the conference is shown in fig. 3 (b), so that better video quality can be easily seen.
The application also provides a cloud conference audio/video transmission method based on the WebRTC technology, which comprises the following steps:
after receiving an instruction initiated by a user from a user interface, the communication module establishes one-to-one audio and video call or establishes a multi-person video conference;
the media negotiation module negotiates media parameters between the two communication parties and keeps consistent;
And the network transmission module transmits the audio and video data.
In one embodiment, for the one-to-one audio/video call, the system communicates according to a gateway or SIP mode selected by a user, and the application establishes an audio/video connection and realizes audio/video transmission through WebRTC technology.
In one implementation mode, for the multi-person video conference, the system establishes the multi-person video conference according to a using Videoroom plug-in or MCU conference mode selected by a user; the Videoroom plug-in establishes a multi-person video room and invites other participants to join in to realize multi-person communication; the MCU conference processes audio and video streams among a plurality of participants through the central control unit, so that the multi-person video conference is realized.
In one embodiment, the method further comprises: and the self-adaptive adjustment module dynamically adjusts relevant parameters of the audio and video data according to network conditions.
In one embodiment, the dynamic adjustment includes:
Circularly calling a peerconnection.getstats () method provided by WebRTC to monitor googRtt parameter changes in audioTrack in real time;
if googRtt > =300±5 and more than 3 times in succession, prompting to close the current camera;
If googRtt is more than 100+/-5 and is more than 3 times continuously, changing the resolution of the camera to the lowest resolution supported by the current equipment, and changing the frame rate to 15;
If googRtt is less than 80 plus or minus 5 and continuously exceeds 10 times, prompting to open the camera;
if googRtt < 50+ -5 and continuously exceeds 10 times, restoring the resolution set by the user;
The above determination is performed in the order from top to bottom using the if else method, and if any one of the conditions is satisfied, the other is not performed.
In one embodiment, the method further comprises:
And the live broadcast and recorded broadcast module is used for carrying out live broadcast and recorded broadcast on conference contents of the multi-person video conference.
In one embodiment, the live broadcast and recording broadcast module performs live broadcast of conference content through a Streaming plug-in; the Recordplay plug-in is used to record conference content.
Compared with the prior art, the technical scheme of the application has the advantages and beneficial effects of more comprehensive functional coverage, webRTC technology-based mobile application platform, self-adaptive adjustment and personalized setting, real-time live broadcast and recorded broadcast functions and the like. These advantages and effects enable the application to provide better audio and video communication experience and wider application scenarios.
While the application has been described with reference to the preferred embodiments, it will be understood by those skilled in the art that various changes and modifications can be made without departing from the scope of the application.
Claims (15)
1. The cloud conference audio and video transmission system based on the WebRTC technology is characterized by comprising: the device comprises a communication module, a media negotiation module, a network transmission module and a self-adaptive adjustment module; wherein,
The communication module is used for processing the audio and video stream by adopting the WebRTC technology according to the instruction input by the user on the terminal interactive interface, and establishing one-to-one or multi-person audio and video communication connection between the current user and other participants; the WebRTC library increases the support of two coding and decoding capacities OMX.IMG and OMX.hisi, and modifies the sdp negotiation content;
The media negotiation module is used for negotiating media parameters among communication parties and keeping consistency;
the network transmission module is used for transmitting the audio and video data;
the self-adaptive adjustment module is used for dynamically adjusting relevant parameters of the audio and video data according to network conditions;
the dynamic adjustment specifically comprises:
Circularly calling a peerconnection.getstats () method provided by WebRTC to monitor googRtt parameter changes in audioTrack in the communication module in real time;
if googRtt > =300±5 and more than 3 times in succession, controlling to close the current camera;
If googRtt is more than 100+/-5 and is more than 3 times continuously, changing the resolution of the camera to the lowest resolution supported by the current equipment, and changing the frame rate to 15;
If googRtt is less than 80 plus or minus 5 and continuously exceeds 10 times, prompting to open the camera;
if googRtt < 50+ -5 and continuously exceeds 10 times, restoring the resolution set by the user;
the dynamic adjustment above uses the if else method from top to bottom, and if any condition is satisfied, the other is not performed.
2. The WebRTC technology-based cloud conference audio-video transmission system of claim 1, wherein the communication module includes:
And the point-to-point audio and video call sub-module is used for establishing point-to-point audio and video call connection according to a gateway or SIP mode selected by a user, and capturing, encoding and decoding, transmitting, playing and rendering the audio and video.
3. The WebRTC technology-based cloud conference audio-video transmission system of claim 1, wherein the communication module further includes: and the multi-person conference sub-module is used for providing multi-person video conference functions, including creating a conference room, inviting other participants and managing the conference.
4. The WebRTC technology-based cloud conference audio-video transmission system of claim 3, the multi-person conference sub-module implementing multi-person video communication using Videoroom plug-ins or MCU conferences.
5. The WebRTC technology-based cloud conference audio-video transmission system of claim 1, wherein the network conditions include at least one of: network bandwidth, delay conditions.
6. The WebRTC technology-based cloud conference audio-video transmission system of claim 1, the related parameters of the audio-video data including at least one of: the codec, resolution, frame rate are dynamically adjusted.
7. The WebRTC technology-based cloud conference audio-video transmission system of claim 3, further comprising: and the live broadcast and recorded broadcast module is used for carrying out live broadcast and recorded broadcast on the conference content generated by the multi-person conference submodule, and comprises connection and pushing of a streaming media server, generation and sharing of a watching link and storage and playback of recorded content.
8. The WebRTC technology-based cloud conference audio-video transmission system according to claim 7, wherein the live broadcast and the recorded broadcast are implemented through Streaming plug-in and Recordplay plug-in, respectively.
9. The WebRTC technology-based cloud conference audio-video transmission system of claim 1, wherein the media parameters include at least one of: decoder, resolution, frame rate.
10. The WebRTC technology-based cloud conference audio-video transmission system of claim 1, wherein the network transmission module discovers the best network path using ICE protocol and performs candidate selection.
11. A transmission method of a WebRTC technology-based cloud conference audio-video transmission system according to any one of claims 1 to 10, comprising the steps of:
after receiving an instruction initiated by a user from a user interface, the communication module establishes one-to-one audio and video call or establishes a multi-person video conference;
the media negotiation module negotiates media parameters between the two communication parties and keeps consistent;
The network transmission module transmits the audio and video data;
The self-adaptive adjustment module dynamically adjusts relevant parameters of the audio and video data according to network conditions;
the dynamic adjustment includes:
Circularly calling a peerconnection.getstats () method provided by WebRTC to monitor googRtt parameter changes in audioTrack in real time;
If googRtt > =300±5 and more than 3 times in succession, prompting to close the current camera;
If googRtt is more than 100+/-5 and is more than 3 times continuously, changing the resolution of the camera to the lowest resolution supported by the current equipment, and changing the frame rate to 15;
If googRtt is less than 80 plus or minus 5 and continuously exceeds 10 times, prompting to open the camera;
if googRtt < 50+ -5 and continuously exceeds 10 times, restoring the resolution set by the user;
the dynamic adjustment above uses the if else method from top to bottom, and if any condition is satisfied, the other is not performed.
12. The transmission method according to claim 11, wherein for the one-to-one audio-video call, the system communicates according to a gateway or SIP mode selected by a user, and the application establishes an audio-video connection and realizes audio-video transmission through WebRTC technology.
13. The transmission method according to claim 11, wherein for the multi-person video conference, the system establishes the multi-person video conference according to a user-selected use Videoroom plug-in or MCU conference mode; the Videoroom plug-in establishes a multi-person video room and invites other participants to join in to realize multi-person communication; the MCU conference processes audio and video streams among a plurality of participants through the central control unit, so that the multi-person video conference is realized.
14. The transmission method according to claim 11, characterized in that the method further comprises:
And the live broadcast and recorded broadcast module is used for carrying out live broadcast and recorded broadcast on conference contents of the multi-person video conference.
15. The transmission method according to claim 14, wherein the live and recorded broadcast module performs live broadcast of conference content through Streaming plug-in; the Recordplay plug-in is used to record conference content.
Priority Applications (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN202311024476.4A CN117176972B (en) | 2023-08-14 | 2023-08-14 | Cloud conference audio and video transmission system and method based on WebRTC technology |
Applications Claiming Priority (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN202311024476.4A CN117176972B (en) | 2023-08-14 | 2023-08-14 | Cloud conference audio and video transmission system and method based on WebRTC technology |
Publications (2)
| Publication Number | Publication Date |
|---|---|
| CN117176972A CN117176972A (en) | 2023-12-05 |
| CN117176972B true CN117176972B (en) | 2024-05-17 |
Family
ID=88932735
Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| CN202311024476.4A Active CN117176972B (en) | 2023-08-14 | 2023-08-14 | Cloud conference audio and video transmission system and method based on WebRTC technology |
Country Status (1)
| Country | Link |
|---|---|
| CN (1) | CN117176972B (en) |
Families Citing this family (1)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN117439976B (en) * | 2023-12-13 | 2024-03-26 | 深圳大数信科技术有限公司 | Audio and video call system based on WebRTC |
Citations (6)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN103650458A (en) * | 2013-08-16 | 2014-03-19 | 华为技术有限公司 | Transmission method, device and system of media streams |
| WO2016046589A1 (en) * | 2014-09-26 | 2016-03-31 | Intel Corporation | Techniques for enhancing user experience in video conferencing |
| CN105721217A (en) * | 2016-03-01 | 2016-06-29 | 中山大学 | Web based audio communication quality improvement method |
| CN107682657A (en) * | 2017-09-13 | 2018-02-09 | 中山市华南理工大学现代产业技术研究院 | A kind of multi-person speech video call method and system based on WebRTC |
| CN111135569A (en) * | 2019-12-20 | 2020-05-12 | RealMe重庆移动通信有限公司 | Cloud game processing method and device, storage medium and electronic equipment |
| CN115865878A (en) * | 2022-11-25 | 2023-03-28 | 阿里巴巴（中国）有限公司 | Transmission control method and device of media stream, storage medium and electronic equipment |
-
2023
- 2023-08-14 CN CN202311024476.4A patent/CN117176972B/en active Active
Patent Citations (6)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN103650458A (en) * | 2013-08-16 | 2014-03-19 | 华为技术有限公司 | Transmission method, device and system of media streams |
| WO2016046589A1 (en) * | 2014-09-26 | 2016-03-31 | Intel Corporation | Techniques for enhancing user experience in video conferencing |
| CN105721217A (en) * | 2016-03-01 | 2016-06-29 | 中山大学 | Web based audio communication quality improvement method |
| CN107682657A (en) * | 2017-09-13 | 2018-02-09 | 中山市华南理工大学现代产业技术研究院 | A kind of multi-person speech video call method and system based on WebRTC |
| CN111135569A (en) * | 2019-12-20 | 2020-05-12 | RealMe重庆移动通信有限公司 | Cloud game processing method and device, storage medium and electronic equipment |
| CN115865878A (en) * | 2022-11-25 | 2023-03-28 | 阿里巴巴（中国）有限公司 | Transmission control method and device of media stream, storage medium and electronic equipment |
Also Published As
| Publication number | Publication date |
|---|---|
| CN117176972A (en) | 2023-12-05 |
Similar Documents
| Publication | Publication Date | Title |
|---|---|---|
| US10015440B2 (en) | Multiple channel communication using multiple cameras | |
| EP2863632B1 (en) | System and method for real-time adaptation of a conferencing system to current conditions of a conference session | |
| US9781386B2 (en) | Virtual multipoint control unit for unified communications | |
| US8144182B2 (en) | Real time video communications system | |
| EP2569937B1 (en) | Systems and methods for real-time multimedia communication across multiple standards and proprietary devices | |
| EP2625856B1 (en) | Systems and methods for error resilient scheme for low latency h.264 video coding | |
| CN112543297B (en) | Video conference live broadcast method, device and system | |
| RU2398362C2 (en) | Connection of independent multimedia sources into conference communication | |
| US10715764B2 (en) | System and method for scalable media switching conferencing | |
| US20120056971A1 (en) | Virtual Presence Via Mobile | |
| US9398257B2 (en) | Methods and systems for sharing a plurality of encoders between a plurality of endpoints | |
| CN202918417U (en) | Video conversation system based on Android set top box | |
| CN103109528A (en) | System and method for the control and management of multipoint conferences | |
| CN117176972B (en) | Cloud conference audio and video transmission system and method based on WebRTC technology | |
| US7180535B2 (en) | Method, hub system and terminal equipment for videoconferencing | |
| WO2014008506A1 (en) | Systems and methods for ad-hoc integration of tablets and phones in video communication systems | |
| US11882385B2 (en) | System and method for scalable media switching conferencing | |
| CN111385515B (en) | Video conference data transmission method and video conference data transmission system | |
| CN103595948B (en) | The adaptive video call system of resolution ratio and method | |
| CN102438119B (en) | Audio/video communication system of digital television | |
| CN113014950A (en) | Live broadcast synchronization method and system and electronic equipment | |
| Cricri et al. | Mobile and Interactive Social Television—A Virtual TV Room | |
| CN112272281B (en) | Regional distributed video conference system | |
| Andberg | Video conferencing in distance education | |
| US20060156378A1 (en) | Intelligent interactive multimedia system |
Legal Events
| Date | Code | Title | Description |
|---|---|---|---|
| PB01 | Publication | ||
| PB01 | Publication | ||
| SE01 | Entry into force of request for substantive examination | ||
| SE01 | Entry into force of request for substantive examination | ||
| CB03 | Change of inventor or designer information |
Inventor after: Li Jiangang Inventor after: Li Pingshi Inventor after: Cui Yinshi Inventor before: Li Jiangang Inventor before: Cui Yinshi Inventor before: Fu Jinsong |
|
| CB03 | Change of inventor or designer information | ||
| GR01 | Patent grant | ||
| GR01 | Patent grant |