CN116743715A - WebRTC audio and video call function integration method, terminal equipment and storage medium - Google Patents
WebRTC audio and video call function integration method, terminal equipment and storage medium Download PDFInfo
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- 238000004590 computer program Methods 0.000 claims description 19
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/1066—Session management
- H04L65/1101—Session protocols
- H04L65/1108—Web based protocols, e.g. webRTC
Abstract
The invention relates to a WebRTC audio and video call function integration method, terminal equipment and storage medium, comprising the following steps: building sdk based on WebRTC; sdk includes a kernel module sdk-core; sdk-core internally includes a call instance phone; call instance phone includes a call related method and an on method for registering a listening event. The invention establishes connection with the IP-PBX server by receiving and transmitting the SIP packet through the WebSocket technology, realizes the SIP instance registration and the receiving and transmitting of the SIP message between the client and the IP-PBX server, and converts the collected call data into a visual interface, thereby providing interaction convenience for users.
Description
Technical Field
The invention relates to the field of audio and video call, in particular to a WebRTC audio and video call function integration method, terminal equipment and a storage medium.
Background
Currently, in the same industry, web applications implement audio-video calls, call transfer, call recording, DTMF (Dual Tone Multi Frequency, dual tone multi-frequency) and other common call scenarios by using WebRTC technology. The following problems exist in practical Web application development:
(1) The interaction with multiple types of terminals is limited: applications implemented by using only WebRTC can only interact with applications implemented by using WebRTC, and cannot interact with devices such as ip phones and door phones.
(2) Development and study costs are high: the front end develops a call function based on the combination of the SIP protocol and the WebRTC, and needs to be familiar with the audio and video communication principle of the WebRTC technology. At the same time, the relevant standards and specifications of the SIP protocol need to be known in depth, the implementation principle and workflow of the SIP protocol stack are understood, and the protocols associated with SIP, such as SDP, RTP, RTCP, and video coding VP8, H264.
(3) The universality is poor: the method can not be transplanted in other web applications and can not be randomly multiplexed or popularized.
(4) The development period is long: the developer needs to create and complete SIP registration, process various call flows, and implement a call UI, which consumes a long time.
Disclosure of Invention
In order to solve the problems, the invention provides a WebRTC audio and video call function integration method, terminal equipment and a storage medium.
The specific scheme is as follows:
a WebRTC audio and video call function integration method comprises the following steps:
building sdk based on WebRTC; sdk includes a kernel module sdk-core; sdk-core internally includes a call instance phone; the call instance phone comprises a call related method and an on method which is realized based on events.eventemitter in nodejs and is used for registering a monitoring event; sdk-core creates a Session object when creating a calling method in a call related method, and stores call related data and other methods except the calling method in the call related method into the Session;
(1) sdk registration procedure:
s101: after the third party application imports sdk, sending call related parameters to sdk-core;
s102: sdk-core registers to IP-PBX based on the received call related parameters, returns to the call instance phone after registration is successful, and monitors call event based on the on method provided by the call instance phone;
(2) The conversation flow is as follows:
s201: when a call method provided by a call instance phone is called, sdk-core calls a call method in JsSIP based on the call method;
s202: the JsSIP creates an RTCPeerConnection instance of the WebRTC through a call method, and creates an offer through a createOffer method of the RTCPeerConnection instance;
s203: after the JsSIP obtains SDP in the offer, the SDP is encapsulated into a body of an INVITE packet in an SIP protocol format and is sent to the IP-PBX through a websocket;
s204: after the JsSIP receives a 200OK reply SIP packet of the IP-PBX, the call is established successfully, after SDP carried in the SIP packet is analyzed, the SDP is set as an answer to an RTCPeerConnection instance, and the WebRTC carries out ICE connectivity check according to the information carried in the SDP;
s205: after ICE connectivity check is completed, determining the address and port of the receiving and transmitting flows of the two parties of the call, and based on the address and port of the receiving and transmitting flows of the two parties of the call, the WebRTC and the IP-PBX mutually transmit flows;
s206: after the hang-up method in the Session is called, sdk-core sends a BYE packet to the IP-PBX based on the terminate method of the hang-up method calling JsSIP;
s207: the IP-PBX replies 200OK after receiving the BYE packet;
s208: after the JsSIP receives the SIP packet of the IP-PBX reply 200OK, the call ending event is triggered, the flow is removed, the rtcp eerconnection is disconnected, and the call ends.
Further, the method for registering sdk-core with the IP-PBX in step S102 is as follows: sdk-core creates websocket and connects, after connection is successful, the websocket instance is set as a parameter into a constructor for initializing JsSIP and creates the JsSIP instance, and a registration request is initiated to the SIP server by calling a start method of the JsSIP instance.
Further, the sdk further includes an interface module sdk-ui, and in step S101, the third party application needs to send the call related parameters to sdk-ui; sdk-ui sends the call related parameters to sdk-core.
Further, the sdk includes an interface module sdk-ui, and the third party application sends the interface-related parameters to sdk-ui, and after the registration is successful in step S102, sdk-ui creates a visual interface based on the interface-related parameters.
Further, the visualization page includes the following three modules:
(1) An incoming call window module: the method comprises the steps of converting data in Session into interface display when an incoming call is received;
(2) And a call interface module: the display content comprises call user identity information, audio information and/or video information;
(3) A dial module: the method is used for inputting and dialing numbers, and dialing a call by calling a calling method in a call example.
Further, the call related method may further include any of a method for answering, a method for hanging up, a method for blind forwarding, a method for consultation forwarding, a method for holding, a method for canceling holding, a method for transmitting DTMF, a method for muting, a method for canceling muting, a method for starting a recording, a method for suspending a recording, a method for audio-to-video-call, a method for video-to-audio-call, and a method for renegotiating a call.
Further, the method further comprises the following steps: when different call functions are required to be realized, the corresponding method in the call related method in the call instance phone is called, and sdk-core calls the corresponding method in the JsSIP based on the called method to realize the required call function.
The WebRTC audio-video call function integrated terminal device comprises a processor, a memory and a computer program stored in the memory and capable of running on the processor, wherein the steps of the method according to the embodiment of the invention are realized when the processor executes the computer program.
A computer readable storage medium storing a computer program which, when executed by a processor, implements the steps of the method described above for embodiments of the present invention.
According to the technical scheme, the connection is established between the SIP packet and the IP-PBX server through the WebSocket technology, the SIP instance registration and the SIP message receiving and sending between the client and the IP-PBX server are realized, and the collected call data are converted into the visual interface, so that interaction convenience is provided for users.
Drawings
FIG. 1 is a diagram showing the internal relationship of sdk-core in a first embodiment of the present invention.
Fig. 2 is a schematic diagram of the integrated sdk communication method in this embodiment.
Fig. 3 is a flow chart showing the call stage in this embodiment.
Detailed Description
For further illustration of the various embodiments, the invention is provided with the accompanying drawings. The accompanying drawings, which are incorporated in and constitute a part of this disclosure, illustrate embodiments and together with the description, serve to explain the principles of the embodiments. With reference to these matters, one of ordinary skill in the art will understand other possible embodiments and advantages of the present invention.
The invention will now be further described with reference to the drawings and detailed description.
Embodiment one:
the embodiment of the invention provides a WebRTC audio and video call function integration method, which comprises the following steps:
the WebRTC-based construction sdk, sdk is divided into two parts: kernel modules sdk-core and interface modules sdk-ui.
sdk-core mainly processes call related logic, receives incoming calls and sends outgoing calls, modifies call state, and exposes call related methods to external program calls, such as sdk-ui, third party applications. sdk-core is a stand-alone module, and if sdk-ui is not intended to be used by third party applications, sdk-core can be used alone to implement interface interaction by itself.
sdk-ui will be used to visualize talk state events, providing a convenient humanized interactive interface. sdk-ui is used in combination with sdk-core.
sdk supports npm (nodejs package manager) installation, umd mode import, and the user can select the appropriate method for a particular scenario. The user may use the integrated sdk (including sdk-core and sdk-ui) directly or may use a scheme that includes only sdk-core alone. The main differences are that:
(1) In the case of less user specific custom call scenarios, sdk integrates sdk-core and provides a rich visual interface, the user only needs to configure initialization parameters, and other problems are presented to sdk for completion.
(2) The user has more special requirements on the conversation scene, the existing interface interaction of sdk-ui cannot be satisfied, and sdk-core can be used for development.
sdk-core internally includes a call instance phone. The call instance phone includes a call related method and an on method for registering a listening event, which is implemented based on events. sdk-core implements registration and interception of events by on method, the registered events include: incoming call, registration state change, call start, call end, and changes to the functional view components are made within the visual interface based on the changes to these events.
The call related method in this embodiment includes an outgoing method call, an answering method answer, a method hang up for hang up, a method blindtranfer for blind transfer, a method attendedTransfer for consultative transfer, a method hold for hold, a method unhhold for release, a method DTMF for sending DTMF, a method mute for mute, a method unmute for cancel mute, a method startRecord for start recording, a method pause record for pause recording, a method audioToVideo for audio-to-video-to-audio-to-renego. The person skilled in the art can adjust the call related method included in the call instance phone according to the actual use requirement. The call related methods are all encapsulated by the corresponding methods in the JsSIP tool library and are used for calling the corresponding methods in the JsSIP tool library, such as calling a call method call for calling a call method in the JsSIP tool library.
sdk-core when creating a call method in a call related method, creates a Session object, and stores call related data and other methods other than the call method in the call related method into the Session, as shown in fig. 1.
(one) sdk registration procedure:
s101: after the third party application imports sdk, call related parameters are sent to sdk-core.
The third party application may be a browser or desktop application supporting WebRTC.
The call related parameters in this embodiment include at least a user name, a signing key and an IP-PBX address.
When the third party application is imported as the integrated sdk, the communication mode is shown in fig. 2, and the third party application client terminal imports sdk-ui, sdk-ui and establishes communication with the IP-PBX through sdk-core. During parameter transfer, the third party application needs to firstly send interface related parameters and call related parameters to sdk-ui; sdk-ui generates a corresponding visual interface based on the interface-related parameters and sends the call-related parameters to sdk-core. When the third party application imports the independent sdk-core, the third party application directly transmits the call related parameters to sdk-core, and the required visual interface is built by the third party application.
S102: sdk-core registers with the IP-PBX based on the received call related parameters, returns to the call instance phone after registration is successful, and listens for a call event based on the on method provided by the call instance phone.
The method for registering sdk-core with the IP-PBX in this embodiment is: sdk-core creates websocket and connects, after connection is successful, the websocket instance is set as a parameter into a constructor for initializing JsSIP and creates the JsSIP instance, and a registration request is initiated to the SIP server by calling a start method of the JsSIP instance.
When the third party application imports sdk which is integrated, call examples phone to sdk-ui are returned; when the third party application imports itself as sdk-core, the call instance phone is returned to the third party application.
When the third party application imports sdk as integrated, sdk-ui begins to create a visual interface after registration is successful. The successful registration determination is sdk-ui receiving the returned call instance phone.
The visual interface created by sdk-ui in this embodiment includes the following three modules:
(1) An incoming call window module: when receiving an incoming call, the method is used for converting data in Session into interface display, and provides a button method for answering the incoming call and hanging up the incoming call by audio and video.
(2) And a call interface module: for display while talking, the display content includes talking user identity information, audio information, and/or video information, and may also include common talking operation function buttons.
(3) A dial module: the method is used for inputting and dialing numbers, and dialing a call by calling a calling method in a call example.
(II) call flow (as shown in FIG. 3):
s201: when the call method provided by the call instance phone is invoked, sdk-core invokes the call method in JsSIP based on the call method.
Before sdk-core calls the call method in JsSIP, it further includes: sdk-core checks the number when called and the state of the audio and video device of the terminal corresponding to the third party application, and calls the call method in JsSIP after the check passes.
S202: the JsSIP creates rtcpieerconnection instance of WebRTC by call method, creates an offer according to the type of call by createOffer method of rtcpieerconnection instance.
When creating an offer, the offer of the corresponding type is created according to the type of the call, and the type of the call includes an audio type (voice call) and a video type (video call).
S203: after the JsSIP obtains the SDP (session descriptor, describing the call media information, such as whether it is an audio or video call, and what kind of audio/video coding is used) in the offer, the SDP is encapsulated in the body of the INVITE packet in SIP protocol format, and sent to the IP-PBX through websocket (a protocol for full duplex communication over a single TCP connection).
S204: after the JsSIP receives the SIP packet of 200OK replied by the IP-PBX, the call is established successfully, after SDP carried in the SIP packet is analyzed, SDP is set as answer to RTCPeerConnection instance, and WebRTC carries out ICE connectivity check according to the information carried in the SDP.
After receiving the INVITE, the IP-PBX resumes according to the order Trying, ringing and 200OK, and when replying to 200OK, it indicates that the opposite terminal has answered and the call is established successfully.
S205: after ICE connectivity check is completed, the address and the port of the receiving and transmitting flows of the two parties of the call are determined, and based on the address and the port of the receiving and transmitting flows of the two parties of the call, the WebRTC and the IP-PBX mutually send flows, and users at the two ends can hear sound or see video.
After determining the address and port of the sending and receiving flow of the two parties of the call, the call timing can be started in the call window on the visual interface.
S206: when the hang-up method in Session is invoked, sdk-core sends BYE packets to the IP-PBX based on the terminate method of the hang-up method invoking JsSIP.
S207: the IP-PBX replies 200OK after receiving the BYE packet.
S208: after the JsSIP receives the SIP packet of the IP-PBX reply 200OK, the call ending event is triggered, the flow is removed, the rtcp eerconnection is disconnected, and the call ends.
When a user needs to execute a certain call function, the call function can be realized through the selection of a corresponding function button on a visual interface provided by a third party application or sdk-ui, when the certain function button is selected, the third party application or sdk-ui calls a corresponding method in a call related method in a call instance phone, and sdk-core calls the corresponding method in the JsSIP based on the called method to realize the required call function. The call method or hang-up method mentioned in the call flow can also be any call related method such as calling a receiving method when receiving an incoming call event, calling a call for blind turning, consulting turning, holding releasing, muting), canceling muting, starting recording, suspending recording, etc. after the call is established.
The embodiment of the invention has the following beneficial effects:
(1) Can talk with various types of terminals: because of the development based on the SIP protocol, a call can be made with a communication device supporting the SIP protocol.
(2) The universality is good: sdk may be imported for use in current web applications.
(3) The development period is short: sdk is out of box and is used, so that the visualization of audio and video calls and common call scenes is realized, a third party user does not need to know the WebRTC technology, SIP and related protocols in depth, a call visualization interface is not required to be realized, and development time is saved.
(4) The development threshold is low: sdk the call function is highly packaged, so that a user does not need to pay attention to initialization processes such as SIP registration and the like, the initialization can be completed by only inputting necessary three parameters when the application is initialized, and the call can be used by using visualized initialization without any operation after the initialization is completed, thereby greatly reducing development burden.
(5) Support configuration UI styles: the third party user may modify the display position of the visual interface and the partial page UI style according to the usage scenario.
(6) Timeliness and security: sdk establishes WebSocket by using encrypted TLS connection, and ensures the timeliness of the call message and the security of the message.
(7) The application range is wide: sdk is a scheme of combining SIP and WebRTC communication, forwarding through an IP-PBX, so that it supports not only network telephony communication but also communication with conventional IP telephones. In addition, sdk has super-strong cross-platform capability, and can be applied to mobile phones (apples and android) and computers (window, mac, ubuntu) only by providing an environment capable of running js and a webrtc module.
Embodiment two:
the invention also provides WebRTC audio-video call function integrated terminal equipment, which comprises a memory, a processor and a computer program stored in the memory and capable of running on the processor, wherein the steps in the method embodiment of the first embodiment of the invention are realized when the processor executes the computer program.
Further, as an executable scheme, the WebRTC audio-video call function integrated terminal device may be a computing device such as a desktop computer, a notebook computer, a palm computer, and a cloud server. The WebRTC audio-video call function integrated terminal device may include, but is not limited to, a processor and a memory. It will be appreciated by those skilled in the art that the above-mentioned composition structure of the WebRTC audio-video call function integrated terminal device is merely an example of the WebRTC audio-video call function integrated terminal device, and does not constitute limitation of the WebRTC audio-video call function integrated terminal device, and may include more or fewer components than the above-mentioned components, or may combine some components, or different components, for example, the WebRTC audio-video call function integrated terminal device may further include an input-output device, a network access device, a bus, and the embodiment of the present invention does not limit the above.
Further, as an executable scheme, the processor may be a central processing unit (Central Processing Unit, CPU), other general purpose processor, digital signal processor (Digital Signal Processor, DSP), application specific integrated circuit (Application Specific Integrated Circuit, ASIC), field programmable gate array (Field-Programmable Gate Array, FPGA) or other programmable logic device, etc. The general processor may be a microprocessor or any conventional processor, and the processor is a control center of the WebRTC audio-video call function integrated terminal device, and connects various parts of the whole WebRTC audio-video call function integrated terminal device by using various interfaces and lines.
The memory may be used to store the computer program and/or module, and the processor may implement various functions of the WebRTC audio-video call function integrated terminal device by running or executing the computer program and/or module stored in the memory and invoking data stored in the memory. The memory may mainly include a storage program area and a storage data area, wherein the storage program area may store an operating system, at least one application program required for a function; the storage data area may store data created according to the use of the cellular phone, etc. In addition, the memory may include high-speed random access memory, and may also include non-volatile memory, such as a hard disk, memory, plug-in hard disk, smart Media Card (SMC), secure Digital (SD) Card, flash Card (Flash Card), at least one disk storage device, flash memory device, or other volatile solid-state storage device.
The present invention also provides a computer readable storage medium storing a computer program which when executed by a processor implements the steps of the above-described method of an embodiment of the present invention.
The modules/units integrated by the WebRTC audio-video call function integrated terminal device may be stored in a computer-readable storage medium if implemented in the form of software functional units and sold or used as independent products. Based on such understanding, the present invention may implement all or part of the flow of the method of the above embodiment, or may be implemented by a computer program to instruct related hardware, where the computer program may be stored in a computer readable storage medium, and when the computer program is executed by a processor, the computer program may implement the steps of each of the method embodiments described above. Wherein the computer program comprises computer program code which may be in source code form, object code form, executable file or some intermediate form etc. The computer readable medium may include: any entity or device capable of carrying the computer program code, a recording medium, a U disk, a removable hard disk, a magnetic disk, an optical disk, a computer Memory, a Read-Only Memory (ROM), a random access Memory (RAM, random Access Memory), a software distribution medium, and so forth.
While the invention has been particularly shown and described with reference to a preferred embodiment, it will be understood by those skilled in the art that various changes in form and details may be made therein without departing from the spirit and scope of the invention as defined by the appended claims.
Claims (9)
1. The WebRTC audio and video call function integration method is characterized by comprising the following steps of:
building sdk based on WebRTC; sdk includes a kernel module sdk-core; sdk-core internally includes a call instance phone; the call instance phone comprises a call related method and an on method which is realized based on events.eventemitter in nodejs and is used for registering a monitoring event; sdk-core creates a Session object when creating a calling method in a call related method, and stores call related data and other methods except the calling method in the call related method into the Session;
(1) sdk registration procedure:
s101: after the third party application imports sdk, sending call related parameters to sdk-core;
s102: sdk-core registers to IP-PBX based on the received call related parameters, returns to the call instance phone after registration is successful, and monitors call event based on the on method provided by the call instance phone;
(2) The conversation flow is as follows:
s201: when a call method provided by a call instance phone is called, sdk-core calls a call method in JsSIP based on the call method;
s202: the JsSIP creates an RTCPeerConnection instance of the WebRTC through a call method, and creates an offer through a createOffer method of the RTCPeerConnection instance;
s203: after the JsSIP obtains SDP in the offer, the SDP is encapsulated into a body of an INVITE packet in an SIP protocol format and is sent to the IP-PBX through a websocket;
s204: after the JsSIP receives a 200OK reply SIP packet of the IP-PBX, the call is established successfully, after SDP carried in the SIP packet is analyzed, the SDP is set as an answer to an RTCPeerConnection instance, and the WebRTC carries out ICE connectivity check according to the information carried in the SDP;
s205: after ICE connectivity check is completed, determining the address and port of the receiving and transmitting flows of the two parties of the call, and based on the address and port of the receiving and transmitting flows of the two parties of the call, the WebRTC and the IP-PBX mutually transmit flows;
s206: after the hang-up method in the Session is called, sdk-core sends a BYE packet to the IP-PBX based on the terminate method of the hang-up method calling JsSIP;
s207: the IP-PBX replies 200OK after receiving the BYE packet;
s208: after the JsSIP receives the SIP packet of the IP-PBX reply 200OK, the call ending event is triggered, the flow is removed, the rtcp eerconnection is disconnected, and the call ends.
2. The WebRTC audio-video call function integration method of claim 1, wherein: the method for registering sdk-core with the IP-PBX in step S102 is as follows: sdk-core creates websocket and connects, after connection is successful, the websocket instance is set as a parameter into a constructor for initializing JsSIP and creates the JsSIP instance, and a registration request is initiated to the SIP server by calling a start method of the JsSIP instance.
3. The WebRTC audio-video call function integration method of claim 1, wherein: sdk also includes an interface module sdk-ui, and in step S101, the third party application needs to first send the call related parameters to sdk-ui; sdk-ui sends the call related parameters to sdk-core.
4. The WebRTC audio-video call function integration method of claim 1, wherein: sdk also includes an interface module sdk-ui, and the third party application sends the interface-related parameters to sdk-ui, and after successful registration in step S102, sdk-ui creates a visual interface based on the interface-related parameters.
5. The WebRTC audio-video call function integration method of claim 4, wherein: the visualization page includes the following three modules:
(1) An incoming call window module: the method comprises the steps of converting data in Session into interface display when an incoming call is received;
(2) And a call interface module: the display content comprises call user identity information, audio information and/or video information;
(3) A dial module: the method is used for inputting and dialing numbers, and dialing a call by calling a calling method in a call example.
6. The WebRTC audio-video call function integration method of claim 1, wherein: the call related method further includes any of a method for answering, a method for hanging up, a method for blind forwarding, a method for consultation forwarding, a method for holding, a method for canceling holding, a method for transmitting DTMF, a method for muting, a method for canceling muting, a method for starting recording, a method for suspending recording, a method for audio call to video call, a method for video call to audio call, and a method for renegotiating a call.
7. The WebRTC audio-video call function integration method of claim 1, wherein: further comprises: when different call functions are required to be realized, the corresponding method in the call related method in the call instance phone is called, and sdk-core calls the corresponding method in the JsSIP based on the called method to realize the required call function.
8. The WebRTC audio and video call function integrated terminal device is characterized in that: comprising a processor, a memory and a computer program stored in the memory and running on the processor, which processor, when executing the computer program, carries out the steps of the method according to any one of claims 1 to 7.
9. A computer-readable storage medium storing a computer program, characterized in that: the computer program implementing the steps of the method according to any one of claims 1 to 7 when executed by a processor.
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