CN112953925B

CN112953925B - Real-time audio and video communication system and method based on SIP (Session initiation protocol) and RTC (real time communication) network

Info

Publication number: CN112953925B
Application number: CN202110160573.0A
Authority: CN
Inventors: 王良
Original assignee: Guangzhou Qizhi Information Technology Co ltd
Current assignee: Guangzhou Qizhi Information Technology Co ltd
Priority date: 2021-02-05
Filing date: 2021-02-05
Publication date: 2022-08-16
Anticipated expiration: 2041-02-05
Also published as: CN112953925A

Abstract

The invention discloses a real-time audio and video communication system based on an SIP protocol and an RTC network, which comprises: the signaling module receives a request of a client, judges the type of a call request, packages and combines a call signaling according to the request content if the call request is a call request, and sends the call signaling to the gateway module; the gateway module establishes network connection with the SIP client and establishes connection with the RTC network according to the acquired call signaling content; if the request is an incoming call request, the SIP client side is connected with the gateway module by using an SIP protocol, the gateway module interacts signaling content with the SIP client side, the gateway module packages the request content into the content appointed by the signaling module and the gateway module according to the incoming call request content, sends the content to the signaling module, and then is connected with the RTC network according to the result returned by the signaling module. The system utilizes the existing excellent RTC network to combine with the SIP communication protocol, and realizes the global high-speed effective audio and video communication.

Description

Real-time audio and video communication system and method based on SIP (Session initiation protocol) and RTC (real time communication) network

Technical Field

The invention relates to the technical field of audio and video communication software, in particular to a real-time audio and video communication system and method based on an SIP (session initiation protocol) and an RTC (real time clock).

Background

VoIP voice over IP transmission, i.e., voice over internet protocol; the VoIP function is realized by using SIP (Session Initiation Protocol), H323, skype and other protocols. The method is another good choice for audio-video communication compared with public telephone based on Public Switched Telephone Network (PSTN).

Real-time Communications (RTC) Real-time communication networks bring much extensibility for Real-time applications, while VoIP requires excellent networks for support, and seamless combination of RTC Real-time communication networks and VoIP will bring very good application prospects for audio-video communication.

Audio and video can bring many changes to life and work. Meanwhile, higher requirements are also put forward on the transmission of audio and video data, for example, smaller time delay is required, smaller bandwidth is required to obtain higher-quality audio and video watching effect, and clients are more diversified to facilitate increasing demands.

At present, due to the existence of different operators in China, the problems of interconnection and intercommunication exist, such as network delay; communication between south and north, and different distances between regions, may also cause delays in data communication. These will affect the experience of real-time audio/video communication and also cause production failures. To solve these problems, a network needs to be constructed, but a good network construction requires a huge investment in manpower and material resources. It is more difficult to scale from country to country, and even globally.

The method for solving the network call and the scheme for the intercommunication of the network telephone and the PSTN/VoLTE public network are provided at present, and the real-time audio and video problem is solved without depending on the existing network condition. Either the business capabilities are reduced to accomplish part of the functionality. For example:

1. an IP-PBX is created, the problem of audio and video conversation of a small local area network is effectively solved, or limited external network extension is solved, the realized function is to upgrade the traditional telephone line, and the factors such as the scale of a service station, the network extension speed and the like have to be considered along with the use extension. If the site size grows too fast, it may exceed the traffic capacity capability of the scheme itself. For large companies, the problem of retaining core business applications is solved by full-time telecommunication/IT departments, but for general companies, third parties still have to deal with the problem, and the uncertainty factor is increased accordingly. There is also a clear disadvantage in that it is not possible to implement applications across regions, even globally.

2. The existing network operation calling system is utilized to solve the problems from the client-side level, only the functional interface is applied, only limited promotion on experience is realized, and the related problems such as transmission and the like cannot be fundamentally solved and optimized.

3. Development time and period of the client. How to reduce the development time and reduce the repetitive work of constantly manufacturing "wheels".

4. The access of various platforms Linux, Windows and MacOS is adapted. Various client iOS, Android, application of PC scene, various Web browser and PC application program based use. Various conditions bring inconvenience to the implementation and are problems to be faced in practice.

5. Security issues such as encrypted transmission of voice video also need to be considered.

6. Inter-conversion between protocols.

7. The problem of deployment in public networks and internal networks.

8. And selecting the direction of incoming and outgoing calls, and calling from inside to outside.

9. The system calls out multiple parties.

10. How to implement a conference call, and the management of the conference call.

Disclosure of Invention

Aiming at the defects in the prior art, the invention provides a real-time audio and video communication system and method based on an SIP (Session initiation protocol) and an RTC (real-time communication), which realize high-speed and effective audio and video communication by using an RTC network and combining SIP communication protocol software.

In a first aspect, a real-time audio and video communication system based on an SIP protocol and an RTC network provided in an embodiment of the present invention includes: the system comprises a signaling module and a gateway module, wherein the signaling module is used for receiving and dispatching a request of a client, the content of the request describes keywords by using a character string, the request comprises a calling number, a called number, a used media type, a current state code, a channel number and a calling request type, the signaling module judges the type of the calling request, if the calling request is a calling request, the signaling module packages and combines calling signaling according to the obtained request content, and the signaling module sends the calling signaling to the gateway module;

the gateway module actively utilizes an SIP protocol to establish a network connection relation with a client side for realizing the SIP according to the calling signaling content acquired from the signaling module, and simultaneously establishes network connection with an RTC network;

if the call request is an incoming call request, the SIP client firstly utilizes the SIP protocol request to establish a connection relation with the gateway module, the gateway module and the SIP client interact signaling content, the gateway module packages the request content into the content appointed by the signaling module and the gateway module according to the incoming call request content, and sends the content to the signaling module, and then establishes corresponding network connection with the RTC network according to the result returned by the signaling module.

In a second aspect, the method for real-time audio/video communication based on an SIP protocol and an RTC provided in the embodiments of the present invention includes the following steps:

a signaling module receives a request of a client and establishes a connection relation; the content of the request describes the key words by using character strings, wherein the request comprises a calling number, a called number, a media type, a current state code, a channel number and a calling request type;

extracting a call request type through a key character string acquired by the call request, wherein the call request type comprises an outgoing call request and an incoming call request;

if the call request is a call request, the signaling module encapsulates the negotiated content as a call signaling, and the signaling module sends the call signaling to the gateway module;

the gateway module analyzes and processes the call signaling, interacts data with a client in the RTC network, establishes connection with an SIP client requested by the signaling and transmits data to the two clients;

if the call request is an incoming call request, the gateway module acquires request content according to the incoming call request of the SIP client and packages the request content into an incoming call signaling, transmits the incoming call signaling to the signaling module and receives subsequent scheduling of the signaling module, and the scheduling result also establishes connection relations with the SIP client and the client in the RTC network respectively and provides data intercommunication for the SIP client and the RTC network;

and the signaling module schedules the gateway module, and realizes the data intercommunication between the SIP client and the client in the RTC network through the gateway module.

The invention has the beneficial effects that:

the embodiment of the invention provides a real-time audio and video communication system and method based on an SIP protocol and an RTC network, which provides a high-speed and effective interactive means for client software applying the RTC network and software applying the SIP protocol; the method solves the problem that the real-time transmission network client is connected with a common (SIP) client in the Internet and also obtains good experience effect for the SIP client to access the RTC network.

Drawings

In order to more clearly illustrate the detailed description of the invention or the technical solutions in the prior art, the drawings that are needed in the detailed description of the invention or the prior art will be briefly described below. Throughout the drawings, like elements or portions are generally identified by like reference numerals. In the drawings, elements or portions are not necessarily drawn to scale.

Fig. 1 shows a frame diagram of a real-time audio/video communication system based on a SIP protocol and an RTC network according to a first embodiment of the present invention;

FIG. 2 is a schematic diagram showing interaction between a client and a system in a first embodiment of the invention;

FIG. 3 shows a schematic diagram of customer interaction in a first embodiment of the invention;

fig. 4 shows a flowchart of a real-time audio-video communication method based on the SIP protocol and the RTC network according to a second embodiment of the present invention.

Detailed Description

The technical solutions in the embodiments of the present invention will be clearly and completely described below with reference to the drawings in the embodiments of the present invention, and it is obvious that the described embodiments are some, not all, embodiments of the present invention. All other embodiments, which can be derived by a person skilled in the art from the embodiments given herein without making any creative effort, shall fall within the protection scope of the present invention.

It will be understood that the terms "comprises" and/or "comprising," when used in this specification and the appended claims, specify the presence of stated features, integers, steps, operations, elements, and/or components, but do not preclude the presence or addition of one or more other features, integers, steps, operations, elements, components, and/or groups thereof.

It is also to be understood that the terminology used in the description of the invention herein is for the purpose of describing particular embodiments only and is not intended to be limiting of the invention. As used in the specification of the present invention and the appended claims, the singular forms "a," "an," and "the" are intended to include the plural forms as well, unless the context clearly indicates otherwise.

It should be further understood that the term "and/or" as used in this specification and the appended claims refers to and includes any and all possible combinations of one or more of the associated listed items.

As used in this specification and the appended claims, the term "if" may be interpreted contextually as "when", "upon" or "in response to a determination" or "in response to a detection". Similarly, the phrase "if it is determined" or "if a [ described condition or event ] is detected" may be interpreted contextually to mean "upon determining" or "in response to determining" or "upon detecting [ described condition or event ]" or "in response to detecting [ described condition or event ]".

It is to be noted that, unless otherwise specified, technical or scientific terms used herein shall have the ordinary meaning as understood by those skilled in the art to which the invention pertains.

Fig. 1 shows a frame diagram of a real-time audio and video communication system (RTC2SIP) based on an SIP protocol and an RTC network according to a first embodiment of the present invention, where the system includes a signaling module and a gateway module. The signaling module realizes scheduling of the service of the whole system according to a strategy, controls the development of service functions in a mode of providing an application program interface, simply and quickly specifies signaling content according to standardized interface requirements, completes bottom layer complex processing such as signaling encapsulation, network communication and the like by the signaling module, and transmits the signaling to the gateway module to complete a substantial calling function by the gateway module. The gateway module analyzes and processes the signaling of the signaling module, and interacts with public Telephone of PSTN (public Switched Telephone network), VoLTE (voice over long term evolution) network or client sides which communicate by protocols such as SIP (session initiation protocol), H232 and the like to realize the call communication of audio and video. The signaling module is used for receiving a request of a client, the content of the request describes keywords by using a character string, the request comprises a calling called identifier, a calling type, a media type, a calling state, an RTC channel number and other key contents, and is divided into a calling type and a calling type according to the type of the calling request, if the calling request is the calling request, the signaling module packages and combines calling signaling according to the type of the calling request and the acquired key information, and the signaling module sends the calling signaling to the gateway module; the gateway module is used for analyzing and processing the call signaling and interacting with a client end for communicating with a call network or a communication protocol; if the call request is an incoming call request, the gateway module acquires an incoming call signaling according to the content of the incoming call request, analyzes the incoming call signaling and sends the content of the incoming call signaling to the signaling module; the signaling module is also used for scheduling the answering client.

The real-time audio and video communication system based on the SIP protocol and the RTC network of the embodiment realizes multi-party audio and video call by using the RTC, intercommunicates with a public switched network (commonly called telecommunication network), makes call with PSTN and VoLTE phones, and realizes software and hardware real-time audio and video call by using the SIP protocol.

As shown in fig. 2, an interaction diagram of a client and an RTC2SIP system is shown, and the system is responsible for accessing a client request and initiating a conference request through a signaling module. The gateway module is responsible for realizing the calling process and forwarding the incoming call request to the signaling module when the call is incoming. The signaling module and the gateway module called by the client adopt reliable network connection to ensure the transmission effectiveness, the signaling transmission and the signaling reception of the client and the gateway module both have definite interactive confirmation processes, and a loss retransmission and signaling check CRC mechanism is adopted.

As shown in FIG. 3, a customer interaction diagram is shown. The requests sent by the client include outgoing requests and incoming requests. The client is connected with the signaling module in real time through a network, detects the network call request received by the port in a circulating mode in real time, detects the validity of the request content, and replies corresponding response content to the request of the client. The contents of the valid request are logged. The valid call related fields are written to the database. The signaling request is a single call, the signaling module sends signaling to the gateway module, and the gateway module calls the appointed client according to the signaling content. The signaling request is a multi-person call, the signaling module designates the signaling as the multi-person call, and the gateway module calls the multi-person at the same time. The signaling request is a conference call, all clients designated by the conference call are added into the same management channel, and the gateway module performs audio mixing processing on audio in the channel, and performs screen dissolving processing on video if the audio is available.

The signaling module combines signaling according to the type of the request. The signaling module comprises a call signaling forming unit, and the call signaling forming unit is used for judging the call type: PSTN, VoLTE network number (mobile phone number or fixed number), or SIP calling type, respectively adopting character string to identify calling type; creating a unique identification character string of the calling system; acquiring a calling identification character string and a called identification character string; acquiring the calling AUDIO and VIDEO type, and adopting a character 'AUDIO' or 'VIDEO' identifier; appointing a management identifier of the call and a room number identifier to be added in the call; acquiring a caller identification and a called identification of the call; obtaining identifiers of different calling stages, and using character strings of sip _ calling, sip _ ringing, sip _ connected, sip _ disconnected and sip _ cancel to represent the current service processing stage; combining the identifications into outgoing signaling.

The gateway module comprises a call signaling processing unit, the call signaling processing unit automatically selects SIP software for calling PSTN, VoLTE telephone or IP network according to the call number character string, if the call is the PSTN, VoLTE telephone number call, the unique specific character string is added in front of the number and the number is sent to SBC (conference boundary controller) to be processed together with related information, and the gateway module feeds back the processing result to the signaling module. If the call is an SIP call of the IP network, the call destination number is added with other unique specific character strings and other related information to be sent to the SBC, so that the call of the SIP is realized.

For the incoming call request, the client firstly accesses to the real-time audio/video communication system based on the SIP protocol and the RTC network described in this embodiment by using protocols and systems such as SIP, H232, and the like through a registration manner. The calling client calls the system described in this embodiment in a manner of initiating a call by entering a number or a character string through an agreed PSTN, VoLTE phone number, or software such as SIP.

The gateway module detects the incoming call connection request of the client in real time or transfers the client request to the incoming call request through SBC third-party software, the client sends the request to the system in a standardized protocol, and the gateway module can seamlessly receive the request of the client and perform corresponding processing according to the content of the communication protocol.

The gateway module obtains the content of the incoming call request, analyzes the effectiveness of the protocol, decomposes the text content of the protocol content, and extracts key content or other appointed command content, such as a calling number identifier, a called number identifier, an IP address and a network port number of a requester, a media list, a media identifier in the list and media coding and decoding parameters.

Specifically, the gateway module acquires an incoming signaling and analyzes the signaling content; and setting the state of the call as an initial state. In the initial state, a globally unique session identification character string (which can be generated or designated by a system) is created, a conference identification of a calling end is acquired, a media handle is created, and a call flag (audio, video call or simultaneous audio and video call) is set.

The gateway module combines new signaling according to the signaling of the incoming client and transfers the new signaling to the signaling module, and the signaling module schedules the answering client. The client is developed by using the SDK provided by the RTC network, and the signaling module inquires whether the client to be answered is logged in or accessed to the system or not and returns a related state to the gateway module. And if the client to be answered logs in the system, the signaling module tries to send a self-defined signaling to the client, and returns a result according to the response condition. If the client to be answered logs in the system, the signaling module sends the result to the gateway module, the signaling module tries to send a self-defined signaling to the client to be answered, and a success state signaling is returned according to the defined signaling; the signaling module sends a successful state signaling to the gateway module, the gateway module sets the state of the call as media exchange after receiving the successful state signaling and returns the set result to the incoming client, and the gateway module receives the incoming client and the answering client and simultaneously establishes a media handle required by the gateway module to complete initialization information of media selection; if the gateway module receives the information that the incoming client and the answering client can not use the same media type, the gateway module converts the media. And the client enables the audio-visual service according to the signaling. If the client is not connected or logged in the system, the signaling module directly returns the state of not logging in or not being connected to the gateway module. And the calling client and the answering client continuously interact audio and video, the interactive audio and video is carried out in a real-time mode, and the signaling module and the gateway module exchange the current state with each other at regular time in the interaction process. And the signaling module and the gateway module continuously exchange the call state result. The states of the incoming call client and the called call answering client are changed, such as answering and hang-up, and the state of the signaling module and the gateway module can be changed along with the change of the network state or the self state. The interaction between the signaling module and the gateway module can be continuously carried out along with the change of the state until the call is completed, the resource recovery is successful and the whole process is ended.

The gateway module is used as a core processing module, plays a role in mutual conversion between RTC and SIP (namely RTC2SIP) in the system and maintains a call state machine; the gateway module is applied as a platform module which can be re-loaded and repeatedly loaded, is dynamically loaded in hot running, and flexibly loads and modifies application parameters in a non-stop mode, so that the addition and deletion of various service capabilities can be realized; the gateway module has audio and video coding and decoding processing capacity, provides format conversion when a calling opposite terminal uses different audio and video formats, and realizes audio and video intercommunication between the RTC network client and the SIP client.

The real-time audio and video communication system based on the SIP protocol and the RTC network provided by the embodiment has the following advantages:

1. the RTC network is fully utilized, and the time delay caused by network difference between different ISP telecommunication operators is reduced to the minimum.

2. On the premise of utilizing excellent RTC basic network, the same transmission effect can be realized in different countries and different regions in the global range, and the voice communication and portrait display quality can be ensured.

3. The system can be accessed by software and hardware equipment of a protocol (SIP) for realizing the public unified standard.

4. And a special network is not required to be repeatedly deployed, so that the operation cost is reduced.

5. The RTC2SIP has strong expandability and can realize multi-person conference and call center; and connecting PSTN and VoLTE networks.

6. Based on the mature SIP protocol, the client can answer the call by using the existing SIP software without modifying the software.

7. Audio or video communication may be implemented.

8. The audio supports multiple coding formats, such as PCMA, PCMU, G729, etc.

9. And various audio sampling rates are supported, such as 8000, 16000, 32000, 48000 and the like.

10. Video supports the native format YUV, and also supports the H264(AVC) coding format.

11. Realizing calling to the global operator number and supporting the external calling of the global operator mobile phone number or fixed phone number;

12. the outbound realizes the audio equipment of the standard SIP communication protocol, supports the real-time communication with the SIP equipment and carries out the audio conversation.

13. And calling out the video equipment supporting the SIP protocol in a real-time mode, and communicating with the SIP equipment and carrying out audio and video conversation.

14. And calling the specified enterprise fixed telephone number to join the conference.

15. An audio device supporting the SIP protocol is used for calling into the system, and an external SIP device directly calls into the RTC network through the TCP/IP network and carries out conference audio conversation with other clients in the network (the standard SIP protocol needs to be supported).

16. And an external standard SIP protocol device is supported to call the RTC network and participate in the RTC network teleconference, and the teleconference supports audio and video calls.

17. The support to the original SIP system is realized, and the client self-line is simply connected (the intercommunication of the new system and the old system is realized).

18. The method realizes that the user key-press message (DTMF) in the call process is monitored in an out-of-band and in-band mode, the message is pushed through an HTTP interface, and the SIP user key-press code is transmitted to the client in the RTC network through the HTTP interface.

19. The method comprises the steps of pushing a calling state in real time, and enabling a calling party and a calling party to obtain a result state in real time, wherein the state comprises establishment, initialization, searching and matching, result execution, media exchange, waiting, media processing, dormancy, reset, hang-up, ending and destruction.

The following conference may be a public network teleconference, a VoIP teleconference, or a mixed voice, video conference of the two.

20. The signaling module initiates the conference and requests the client to join the conference in real time in a calling mode.

21. And releasing the initiated conference, finishing pushing and releasing the initiated conference resources.

22. And reserving the conference, and supporting the reservation to initiate the conference.

23. And conference management, which supports that the conference can be added only by inputting and checking a conference password.

24. And conference management for prohibiting the SIP user of the designated user from speaking in the conference by temporarily interrupting the stream data transmission of the designated user.

25. And conference management, namely forbidding all SIP users to speak by temporarily interrupting all user data transmission modes.

26. And conference management, namely releasing the forbidden statement for the specified user to recover the streaming data transmission mode and allowing the specified SIP user to speak again.

27. And (4) conference management, namely all the users in the designated conference are released from being forbidden to speak, and all the SIP users can speak again.

28. The user applies for the speech function and applies for the speech by pressing a designated (like numbers and symbols on a telephone dial) button. The application result can be passed or maintained.

29. And the user applies for speech auditing, and sends a speech applying request to the background system for auditing through the button to achieve a speech control function.

30. The administrator can customize and configure the appointed speaking key, and the self-defined configuration of the speaking key is realized.

31. The in-conference user may be prohibited from receiving speech. When the SIP user is prohibited from listening to the in-conference voice, the connection can be maintained and then the post-conference voice is permitted to listen.

32. All SIP users in the conference are prohibited from listening to the conference voice.

33. And releasing the forbidden listening of the appointed user and supporting releasing the forbidden listening of the appointed SIP user.

34. And releasing the forbidden listening of all the users and supporting releasing the forbidden listening of all the SIP users.

35. And after the appointed user is terminated to continue participating in the conference, the client receives the on-hook signaling and closes the connection after the appointed SIP is terminated to continue participating in the conference.

36. And terminating all the users to continue participating in the conference, and ending the connection of all the SIP users participating in the conference.

37. And querying all SIP users in the specified conference room, querying all SIP users in the room and returning the user ID character strings as results.

38. And (4) pushing the user to join the conference, and sending user UID (user identifier) and mobile phone number conference joining information to all users in the same conference room in a broadcasting mode after supporting the SIP user to join the conference.

39. And (4) the user exits the conference pushing, and after the SIP user exits the conference, the user exit UID and the mobile phone number conference entering information are sent to all users in the same conference room in a broadcasting mode.

40. The conference log pushing supports that after the SIP user exits the conference, the user participation information is sent;

41. and (4) customizing the flow recording configuration, and supporting the user-defined configuration of the whole flow recording file.

In the specific implementation process, the method comprises the following steps:

1. obtaining PSTN and VoLTE access qualification.

The method mainly acquires the line landing qualification, namely realizes the qualification of calling the number in the PSTN network by the RTC. The PSTN operator assigns numbers or sets of numbers for incoming and outgoing calls, using these numbers as the identification number for each call. Meanwhile, some special signaling requirements are required for the access of the PSTN operator, and the parameter set needs to be modified correspondingly to adapt to the requirement.

2. The RTC network access qualification is obtained.

Access credentials, including account numbers, some application identification of the access, are obtained from the RTC network operator.

An RTC network operator provides development kits (Application SDKs) supporting different platforms, and software of Web, PC or mobile platform is developed by the development kits, so that the RTC2SIP can be accessed very easily.

3. And (3) setting up a server environment, wherein the server meets certain conditions so as to meet the hardware environment matched with the required functions and performances, including the CPU computing capacity, the memory size, the network bandwidth and the hard disk size.

4. Deploying key module software: a signaling module and a gateway module.

Copying the signaling module to a specified directory, modifying corresponding configuration parameters, and operating the signaling module; the signaling module provides a log file, prompts the running state of the module, outputs various prompt messages and can analyze the conditions of the module and a module scheduling system in real time. And checking the log output result and checking that the corresponding port is intercepted, and ensuring the normal operation of the signaling module.

Copying the gateway module to a specified directory, configuring related parameters such as calls, media and the like, and adapting to different project requirements through parametric change; the gateway module provides log output to prompt the running state of the module, the log content comprises the state change value of the call connection process and various key contents of data communication, and the log can be used for conveniently removing faults; and simultaneously provides various real-time commands to check the running state of the gateway module.

5. And developing a client (Android, iOS client software, PC client software and Web client software) developed by an application development SDK provided by an RTC network operator, connecting the RTC network and the RTC2SIP system, and acquiring audio and video data in real time through the scheduling of a signaling module of the RTC2SIP system.

6. The RTC network client and the SIP client call the SIP number or account number according to the standard protocol provided by the RTC2SIP, and connect the designated client.

7. The RTC network client or the SIP client calls PSTN, VoLTE public network mobile phone numbers or fixed numbers by using a standard protocol provided by RTC2 SIP.

8. The RTC network client or the SIP client calls the SIP number or the account number and transfers the SIP number or the account number to the Web client for answering.

The real-time audio and video communication system based on the SIP protocol and the RTC network provided by the embodiment of the invention fully utilizes the existing excellent RTC network to solve the problem of data transmission; the excellent RTC network covers countries and regions above 200 of the world, easily solves the basic problem of implementing services in any country, and realizes more extension possibility of services. For example, SD-RTN (Software Defined Real-time Network) of the sound Network is directly utilized, cross-country Network access optimization is performed in North America, southeast Asia, south Africa and the like, Network access optimization is performed in small and medium-sized cities in China, the problem of packet loss is solved, the video experience can be kept smooth under 70% of packet loss, and the audio experience can be kept smooth under 80% of packet loss rate. For audio, there may be a high fidelity, 3D surround sound experience, and for video, an immersive visual experience may be enjoyed. A developer tool Set (SDK) suitable for various platforms and oriented to clients and easily adapted is provided. Effectively combines VoIP audio and video communication based on SIP, H232 and more possible protocols for expansion, and creates a powerful platform for global high-speed effective audio and video communication.

In the first embodiment, a real-time audio and video communication system based on the SIP protocol and the RTC network is provided, and correspondingly, the present application also provides a real-time audio and video communication method based on the SIP protocol and the RTC network. Please refer to fig. 4, which is a flowchart of a real-time audio/video communication method based on the SIP protocol and the RTC network according to a second embodiment of the present invention. Since the method embodiment is basically similar to the device embodiment, the description is simple, and the relevant points can be referred to the partial description of the device embodiment. The method embodiments described below are merely illustrative.

As shown in fig. 4, a flowchart of a real-time audio/video communication method based on an SIP protocol and an RTC network according to an embodiment of the present invention is shown, where the method includes the following steps:

a signaling module receives a request of a client and establishes a connection relation; the content of the request describes the keyword using a character string, the request including a calling number, a called number, a usage media type, a current status code, a channel number, and a call request type.

And extracting the call request type through the key character string acquired by the call request, wherein the call request type comprises an outgoing call request and an incoming call request.

If the call request is a call request, the signaling module encapsulates the negotiated content as a call signaling, and the signaling module sends the call signaling to the gateway module.

The method for encapsulating the negotiated content into the outgoing signaling by the signaling module specifically comprises the following steps:

judging the calling type, and respectively adopting characters to mark the calling type;

creating a 32-bit unique identification character string of the whole calling system;

acquiring a calling identifier and a called identifier character string;

acquiring a calling AUDIO and VIDEO type, and representing by adopting an AUDIO/VIDEO character string;

appointing a management identifier of the call;

acquiring a caller identification and a called identification of the call;

acquiring identifiers of different stages of calling;

combining the identifications into outgoing signaling.

The gateway module analyzes and processes the call signaling, interacts data with the client in the RTC network, and establishes connection with the SIP client requested by the signaling to mutually transmit data for the two clients.

Wherein, gateway module analysis processes the signaling of calling out specifically includes:

the gateway module selects a calling network according to the type of the network number;

the unique identification prefix is added to the calling network number and sent to the conference border controller.

If the call request is an incoming call request, the gateway module acquires request content according to the incoming call request of the SIP client and packages the request content into an incoming call signaling, the incoming call signaling is transmitted to the signaling module and receives subsequent scheduling of the signaling module, the scheduling result also establishes connection relations with the SIP client and the client in the RTC network respectively, and data intercommunication is provided for the SIP client and the client in the RTC network.

The gateway module obtains request content and packages the request content into an incoming signaling according to the incoming request of the SIP client, and specifically comprises the following steps:

acquiring request content according to the incoming request, analyzing the effectiveness of the communication protocol, decomposing the protocol content, and taking out key content or appointed content;

setting the calling state as an initial state;

in the initial state, a global displacement session identification character string is created, a session identification of a call-in end is obtained, a media handle is created, and a call mark of the time is set.

Specifically, the signaling module scheduling gateway module specifically includes:

the signaling module judges whether the client end of the call to be answered is logged in the system or accessed into the system,

if the client to be answered logs in the system, the signaling module tries to send a self-defined signaling to the client to be answered, and returns a success state signaling according to the defined signaling;

the signaling module sends the success state signaling to the gateway module;

and if the client to be answered is not connected or the system is not logged in, the signaling module sends the state of no logging in or no connection to the gateway module.

The above is a description of an embodiment of a real-time audio/video communication method based on an SIP protocol and an RTC network according to a second embodiment of the present invention.

The real-time audio and video communication method based on the SIP protocol and the RTC network provided by the invention and the real-time audio and video communication system based on the SIP protocol and the RTC network have the same inventive concept and the same beneficial effects, and are not described again here.

Finally, it should be noted that: the above embodiments are only used to illustrate the technical solution of the present invention, and not to limit the same; while the invention has been described in detail and with reference to the foregoing embodiments, it will be understood by those skilled in the art that: the technical solutions described in the foregoing embodiments may still be modified, or some or all of the technical features may be equivalently replaced; such modifications and substitutions do not depart from the spirit and scope of the present invention, and they should be construed as being included in the following claims and description.

Claims

1. A real-time audio and video communication system based on SIP protocol and RTC network is characterized by comprising: the system comprises a signaling module and a gateway module, wherein the signaling module is used for receiving and dispatching a request of a client, the content of the request describes keywords by using a character string, the request comprises a calling number, a called number, a used media type, a current state code, a channel number and a calling request type, the signaling module judges the type of the calling request, if the calling request is a calling request, the signaling module packages and combines calling signaling according to the obtained request content, and the signaling module sends the calling signaling to the gateway module; the signaling module comprises a calling scheduling unit, the calling scheduling unit is used for realizing data interaction between an RTC network client and an SIP client through a key identification information scheduling gateway module, and each key information required by scheduling adopts a character string as an identification; creating a unique identification character string of the system as a keyword of a service process in the system until the service is finished;

the gateway module establishes a network connection relationship with the SIP client by using an SIP protocol according to the calling signaling content acquired from the signaling module, and establishes network connection with the RTC network;

if the call request is an incoming call request, the SIP client utilizes the SIP protocol request to establish a connection relation with the gateway module, the gateway module interacts signaling content with the SIP client, the gateway module packages the request content into the content appointed by the signaling module and the gateway module according to the incoming call request content, sends the content to the signaling module, and establishes corresponding network connection with the RTC network according to the result returned by the signaling module.

2. The system of claim 1, wherein the outgoing call scheduling unit acquires calling information and called information from request information of the client;

acquiring the audio and video type of the service operation from the request information of the client, wherein the audio and video type is audio or audio and video;

appointing a management identification character string of the call, and using a universal unique identification code;

acquiring a caller identification and a called identification of the call;

acquiring processing result identifiers of different stages of calling;

specifying channel number information;

specifying a call type;

specifying a media mode for use by the call;

adding different content information according to the call type or the current state;

and encapsulating and combining the calling person identification, the called person identification and the processing result identification of different calling stages into a calling signaling.

3. The system of claim 2, wherein the gateway module comprises an outgoing signaling processing unit;

and the calling signaling processing unit selects a calling network according to the type of the network number, adds a unique identification prefix to the calling network number and sends the unique identification prefix to the conference boundary controller.

4. The system of claim 1, wherein the gateway module comprises an incoming signaling processing unit, and the incoming signaling processing unit is configured to obtain request content according to an incoming request, analyze validity of a communication protocol, decompose protocol content, extract key content or agreed content, set a current call state to an initial state, create a globally displaced session identifier string in the initial state, obtain a session identifier of an incoming call end, create a media handle, and set a current call flag.

5. A real-time audio and video communication method based on an SIP protocol and an RTC network is characterized by comprising the following steps:

judging the type of the call request;

the signaling module schedules the gateway module, and realizes the data intercommunication between the SIP client and the client in the RTC network through the gateway module; the signaling module comprises a calling scheduling unit, the calling scheduling unit is used for realizing data interaction between an RTC network client and an SIP client through a key identification information scheduling gateway module, and each key information required by scheduling adopts a character string as an identification; and creating a unique identification character string of the system as a keyword of a service process in the system until the service is finished.

6. The method of claim 5, wherein the step of encapsulating the negotiated content as outgoing signaling by the signaling module specifically comprises:

acquiring a calling identifier and a called identifier character string;

acquiring a calling audio and video type, and adopting character identification;

appointing a management identifier of the call;

acquiring a caller identification and a called identification of the call;

acquiring identifiers of different stages of calling;

7. The method of claim 6, wherein the analyzing and processing of the outgoing signaling by the gateway module specifically comprises:

8. The method of claim 5, wherein the gateway module obtaining the request content and encapsulating the request content into the incoming signaling according to the SIP client incoming call request specifically comprises:

setting the calling state as an initial state;

9. The method of claim 8, wherein the signaling module scheduling the gateway module specifically comprises:

the signaling module sends the successful state signaling to the gateway module;