CN113746808A

CN113746808A - Converged communication method for online conference, gateway, electronic device, and storage medium

Info

Publication number: CN113746808A
Application number: CN202110922251.5A
Authority: CN
Inventors: 官仕富; 阮良; 陈功; 朱振昊; 陈丽
Original assignee: Hangzhou Netease Zhiqi Technology Co Ltd
Current assignee: Hangzhou Netease Zhiqi Technology Co Ltd
Priority date: 2021-08-12
Filing date: 2021-08-12
Publication date: 2021-12-03
Anticipated expiration: 2041-08-12
Also published as: CN113746808B

Abstract

The present disclosure provides a converged communication method, a gateway, an electronic device and a storage medium for an online conference, wherein the method includes: the method comprises the steps that a converged communication gateway receives a call request which is sent by a Session Initiation Protocol (SIP) server and carries a call identifier of an SIP terminal, and acquires an RTC conference identifier of a real-time communication RTC conference bound with the call identifier; sending a RTC conference joining request of the SIP terminal to the RTC server, and establishing a first media transmission channel with the RTC server after determining that the RTC server joins the SIP terminal to the RTC conference; the RTC conference joining request comprises an RTC conference identifier; and sending the first media stream information of the SIP terminal sent by the SIP server to the RTC server through the first media transmission channel, so that the RTC server sends the first media stream information to the corresponding RTC terminal in the RTC conference. The method and the device can enable the SIP terminal to be added into the RTC audio-video conference, and realize communication between the SIP terminal and the RTC terminal in the RTC audio-video conference.

Description

Converged communication method for online conference, gateway, electronic device, and storage medium

Technical Field

The present disclosure relates to the field of communications technologies, and in particular, to a converged communication method for online conferences, a gateway, an electronic device, and a storage medium.

Background

Currently, an audio/video call system generally includes a Real-Time Communication (RTC) audio/video conference system and a Session Initiation Protocol (SIP) audio/video conference system. In the RTC audio and video conference system, a signaling control protocol adopted by the RTC terminal to access the RTC audio and video conference is a private protocol, that is, the RTC terminal needs to integrate Software Development Kits (SDKs) implemented by manufacturers, and can only access the RTC audio and video conference system; in the SIP audio and video conference system, a signaling control protocol adopted by the SIP terminal to access the SIP audio and video conference is a standard protocol RFC 3261, and the SIP terminal can be accessed to the SIP audio and video conference system as long as the SIP terminal follows the standard protocol.

Since the SIP terminal in the SIP audio-video conference is different from the RTC terminal in the RTC audio-video conference in signaling control protocol, the SIP terminal and the RTC terminal can not be communicated with each other; however, in some application scenarios, the SIP terminal may need to communicate with the RTC terminal in the RTC audio-video conference, and therefore, how to communicate the SIP terminal with the RTC terminal in the RTC audio-video conference is a problem to be solved.

Disclosure of Invention

The embodiment of the disclosure provides a converged communication method, a gateway, an electronic device and a storage medium for an online conference, which are used for enabling an SIP terminal to join an RTC audio and video conference and realizing communication between the SIP terminal and an RTC terminal in the RTC audio and video conference.

In a first aspect, an embodiment of the present disclosure provides a converged communication method for an online conference, which is applied to a converged communication gateway, and includes:

receiving a call request which is sent by a Session Initiation Protocol (SIP) server and carries a call identifier of an SIP terminal, and acquiring an RTC conference identifier of a real-time communication (RTC) conference bound with the call identifier;

sending a RTC conference joining request of the SIP terminal to an RTC server, and establishing a first media transmission channel with the RTC server after determining that the RTC server joins the SIP terminal to the RTC conference; the RTC conference joining request comprises the RTC conference identifier;

and sending the first media stream information of the SIP terminal sent by the SIP server to the RTC server through the first media transmission channel, so that the RTC server sends the first media stream information to a corresponding RTC terminal in the RTC conference.

In a possible implementation manner, the call request is a request for joining a SIP conference, and the call identifier is a SIP conference identifier;

the receiving a call request carrying a call identifier and sent by a Session Initiation Protocol (SIP) terminal, and acquiring a real-time communication (RTC) conference identifier of a RTC conference bound with the call identifier includes:

responding to a SIP conference adding request which is sent by the SIP terminal and carries an SIP conference identifier, and adding the SIP terminal into the SIP conference corresponding to the SIP conference identifier;

and acquiring the RTC conference identifier of the RTC conference bound with the SIP conference identifier.

In one possible embodiment, the method further comprises:

each RTC terminal in the RTC conference is added into the SIP conference, and a second media transmission channel with the RTC server is established;

receiving second media stream information of each RTC terminal in the RTC conference, which is sent by the RTC server, through the second media transmission channel;

performing fusion processing on the second media stream information of each RTC terminal and the third media stream information sent by each SIP terminal in the SIP conference to obtain target media stream information;

and respectively sending the target media stream information to each SIP terminal in the SIP conference.

In one possible implementation, the second media stream information and the third media stream information each include audio stream information;

the merging the second media stream information of each RTC terminal and the third media stream information sent by each SIP terminal in the SIP conference to obtain the target media stream information includes:

and performing sound mixing processing on the audio stream information in the second media stream information of each RTC terminal and the audio stream information in the third media stream information sent by each SIP terminal in the SIP conference to obtain first target media stream information.

In a possible implementation, the second media stream information and the third media stream information each further include video stream information;

the merging the second media stream information of each RTC terminal and the third media stream information sent by each SIP terminal in the SIP conference to obtain the target media stream information, further includes:

and performing screen mixing processing on the video stream information in the second media stream information of each RTC terminal and the video stream information in the third media stream information sent by each SIP terminal to obtain second target media stream information.

In a possible implementation, the call request is a call RTC terminal request;

the method further comprises the following steps:

establishing a third media transmission channel with the RTC server, and receiving fourth media stream information of an RTC terminal in the RTC conference, which is sent by the RTC server, through the third media transmission channel;

and sending the fourth media stream information of the RTC terminal to the SIP terminal.

In a possible implementation manner, the sending, by the SIP server, first media stream information of the SIP terminal to the RTC server through the first media transmission channel includes:

transcoding first media stream information of the SIP terminal sent by the SIP server into a preset encoding format;

and sending the transcoded first media stream information to the RTC server through the first media transmission channel.

In a possible embodiment, the obtaining an RTC conference identifier of the real-time communication RTC conference bound to the call identifier includes:

and sending an authentication request carrying the call identifier to the RTC server to obtain an RTC conference identifier of the RTC conference bound with the call identifier, which is returned by the RTC server after the RTC server passes the authentication.

In a second aspect, an embodiment of the present disclosure further provides a converged communication gateway, including a session initiation protocol SIP signaling proxy module, at least one SIP media service module, and at least one protocol conversion module;

the SIP signaling proxy module is used for receiving a call request which is sent by a Session Initiation Protocol (SIP) server and carries a call identifier of an SIP terminal, and distributing the call request to a corresponding SIP media service module;

the SIP media service module is used for establishing a calling event carrying a calling identifier according to the calling request and sending the calling event to a corresponding protocol conversion module;

the protocol conversion module is used for acquiring an RTC conference identifier of a real-time communication RTC conference bound with the call identifier, sending an RTC conference joining request carrying the RTC conference identifier of the SIP terminal to an RTC server, establishing a first media transmission channel between the SIP service module and the RTC server after determining that the RTC server joins the SIP terminal to the RTC conference, and sending a media forwarding instruction to the SIP media service module;

the SIP media service module is further configured to respond to the media forwarding instruction, and send first media stream information of the SIP terminal, which is sent by the SIP server, to the RTC server through the first media transmission channel, so that the RTC server sends the first media stream information to a corresponding RTC terminal in the RTC conference.

the SIP media service module further comprises:

the SIP conference management submodule is used for responding to a SIP conference adding request which is sent by the SIP terminal and carries an SIP conference identifier, and adding the SIP terminal into the SIP conference corresponding to the SIP conference identifier;

the protocol conversion module further comprises:

and the acquisition submodule is used for acquiring the RTC conference identifier of the RTC conference bound with the SIP conference identifier.

In one possible implementation, the protocol conversion module further includes:

the first channel establishing submodule is used for adding each RTC terminal in the RTC conference into the SIP conference and establishing a second media transmission channel with the RTC server;

the SIP media service module further comprises:

the receiving submodule is used for receiving second media stream information of each RTC terminal in the RTC conference, which is sent by the RTC server through the second media transmission channel;

the merging submodule is used for merging the second media stream information of each RTC terminal and the third media stream information sent by each SIP terminal in the SIP conference to obtain target media stream information;

and the first sending submodule is used for respectively sending the target media stream information to each SIP terminal in the SIP conference.

the fusion submodule is further configured to:

In a possible implementation, the call request is a call RTC terminal request;

the protocol conversion module further comprises:

the second channel establishing submodule is used for establishing a third media transmission channel with the RTC server and receiving fourth media stream information of one RTC terminal in the RTC conference, which is sent by the RTC server through the third media transmission channel;

and the second sending submodule is used for sending the fourth media stream information of the RTC terminal to the SIP terminal.

In one possible implementation, the SIP media service module further includes:

the transcoding submodule is used for transcoding the first media stream information of the SIP terminal sent by the SIP server into a preset coding format;

and the third sending submodule is used for sending the transcoded first media stream information to the RTC server through the first media transmission channel.

and the request verification submodule is used for sending a verification request carrying the call identifier to the RTC server so as to obtain the RTC conference identifier of the RTC conference bound with the call identifier, which is returned by the RTC server after the RTC server passes the verification.

In a third aspect, the present disclosure also provides an electronic device, including a memory and a processor, where the memory stores a computer program operable on the processor, and when the computer program is executed by the processor, the processor is caused to implement the steps of the converged communication method for online conferencing according to any one of the first aspect.

In a fourth aspect, the present disclosure further provides a computer-readable storage medium, in which a computer program is stored, and when the computer program is executed by a processor, the steps of the converged communication method for online conferences according to any one of the first aspect are implemented.

The converged communication method for the online conference provided by the embodiment of the disclosure at least has the following beneficial effects:

according to the scheme provided by the embodiment of the disclosure, when receiving a call request of an SIP terminal sent by an SIP server, a convergence communication gateway acquires an RTC conference identifier of an RTC conference corresponding to the call request, sends a request of the RTC conference joining of the SIP terminal to the RTC server, and establishes a first media transmission channel with the RTC server after determining that the RTC server joins the SIP terminal to the RTC conference; and then the first media stream information of the SIP terminal sent by the SIP server is sent to the RTC server through the first media transmission channel, so that the RTC server sends the first media stream information to the corresponding RTC terminal in the RTC conference. Therefore, the converged communication gateway serves as a communication link between the SIP server and the RTC server, on one hand, signaling interaction can be carried out with the SIP server, and on the other hand, signaling interaction can be carried out with the RTC server, so that conversion between the SIP protocol and the RTC protocol is realized, further, the SIP terminal can be added into the RTC audio-video conference, and communication between the SIP terminal and the RTC terminal in the RTC audio-video conference is realized.

Additional features and advantages of the disclosure will be set forth in the description which follows, and in part will be obvious from the description, or may be learned by the practice of the disclosure. The objectives and other advantages of the disclosure may be realized and attained by the structure particularly pointed out in the written description and claims hereof as well as the appended drawings.

Drawings

In order to more clearly illustrate the embodiments of the present disclosure or the technical solutions in the prior art, the drawings used in the description of the embodiments or the prior art will be briefly described below, it is obvious that the drawings in the following description are only some embodiments of the present disclosure, and other drawings can be obtained by those skilled in the art without creative efforts.

Fig. 1 is a schematic view of an application scenario of a converged communication method for an online conference according to an embodiment of the present disclosure;

fig. 2 is a flowchart of a converged communication method for an online conference according to an embodiment of the present disclosure;

fig. 3 is a flowchart of another converged communication method for an online conference according to an embodiment of the present disclosure;

fig. 4A is a schematic view of a conference interface of an SIP terminal according to an embodiment of the present disclosure;

fig. 4B is a schematic view of a conference interface of an RTC terminal according to an embodiment of the present disclosure;

fig. 5 is a flowchart of another converged communication method for an online conference according to an embodiment of the present disclosure;

fig. 6 is a schematic diagram of a converged communication system for online conferences according to an embodiment of the present disclosure;

fig. 7 is a block diagram of a converged communication gateway according to an embodiment of the present disclosure;

fig. 8 is a schematic structural diagram of an electronic device according to an embodiment of the present disclosure.

Detailed Description

To make the objects, technical solutions and advantages of the present disclosure clearer, the present disclosure will be described in further detail with reference to the accompanying drawings, and it is apparent that the described embodiments are only a part of the embodiments of the present disclosure, not all of the embodiments. All other embodiments, which can be derived by a person skilled in the art from the embodiments disclosed herein without making any creative effort, shall fall within the protection scope of the present disclosure.

It is noted that the terms "first," "second," and the like in the description and in the claims of the present disclosure are used for distinguishing between similar elements and not necessarily for describing a particular sequential or chronological order. It is to be understood that the data so used is interchangeable under appropriate circumstances such that the embodiments of the disclosure described herein are capable of operation in sequences other than those illustrated or otherwise described herein.

Furthermore, the terms "comprises," "comprising," and "having," and any variations thereof, are intended to cover a non-exclusive inclusion, such that a process, method, system, article, or apparatus that comprises a list of steps or elements is not necessarily limited to those steps or elements expressly listed, but may include other steps or elements not expressly listed or inherent to such process, method, article, or apparatus.

For convenience of understanding, some concepts involved in the embodiments of the present disclosure are explained below.

Session Initiation Protocol (SIP): it belongs to standard protocol RFC (Request For CommisssFC, is a series of files scheduled by numbers) 3261, and is a signaling control protocol of application layer. SIP is used to create new, manage, or terminate existing one or more-party sessions as defined by RFC 3261.

Internet Protocol Private Branch Exchange (IPPBX): a corporate SIP telephony system based on the Internet protocol IP.

Real-Time Communication (RTC): common video telephone and voice call belong to the technology. The method has the greatest characteristic of bidirectional low-delay communication, and the delay time is generally 300-500 milliseconds.

Real-Time Transport Protocol (RTP): the application layer transport Protocol is the most commonly used application layer transport Protocol in the real-time communication technology, and generally operates on a User Datagram Protocol (UDP) or a Transmission Control Protocol (TCP), and mainly carries media data Transmission.

Selective Forwarding Unit (SFU): a real-time audio and video server architecture. The basic principle is as follows: and the central server forwards the media streams of all the sending ends to the receiving end.

MultiPoint Control Unit (MCU): an audio and video server architecture for mixing sound and screen. The basic principle is as follows: the server receives the uplink media streams of each end, decodes the audio and video streams in each media stream, then performs audio mixing and screen mixing processing, re-encodes the mixed audio and video, and forwards the re-encoded audio and video to the downlink users of each end.

The following outlines the design ideas of the present disclosure.

At present, in an RTC audio and video conference system, a signaling control protocol adopted by an RTC terminal to access an RTC audio and video conference is a private protocol; in the SIP audio and video conference system, a signaling control protocol adopted by the SIP terminal to access the SIP audio and video conference is a standard protocol RFC 3261. Since the SIP terminal in the SIP audio-video conference is different from the RTC terminal in the RTC audio-video conference in signaling control protocol, the SIP terminal and the RTC terminal can not be communicated with each other; however, in some application scenarios, the SIP terminal may need to communicate with the RTC terminal in the RTC audio-video conference, and therefore, how to communicate the SIP terminal with the RTC terminal in the RTC audio-video conference is a problem to be solved.

In view of this, the embodiments of the present disclosure provide a converged communication method, a gateway, an electronic device, and a storage medium for an online conference, where the converged communication gateway is used as a communication link between an SIP server and an RTC server, and on one hand, the converged communication gateway can perform signaling interaction with the SIP server and on the other hand, the converged communication gateway can perform signaling interaction with the RTC server, so as to implement conversion between an SIP protocol and an RTC protocol, and further enable an SIP terminal to join an RTC audio/video conference, thereby implementing communication between the SIP terminal and an RTC terminal in the RTC audio/video conference.

An application scenario of the embodiments of the present disclosure is described below with reference to the drawings.

Fig. 1 is a schematic view of an application scenario of a converged communication method for an online conference according to an embodiment of the present disclosure. The application scenario includes an SIP terminal 100, an SIP server 200, a converged communication gateway 300, an RTC server 400, and an RTC terminal 500; the SIP terminal 100 and the SIP server 200 may be connected through a communication network to form an SIP audio/video conference system; the RTC server 400 and the RTC terminal 500 may be connected through a communication network to form an RTC audio and video conference system; the converged communication gateway 300 may be connected to the SIP server 200 and the RTC server 400 via communication networks, respectively. Optionally, the communication network may be a wired network or a wireless network, and the embodiments of the present disclosure are not limited in particular herein.

The SIP terminal 100 may include, but is not limited to, an electronic device such as a mobile phone, a fixed phone, a desktop computer, a mobile computer, a tablet computer, a media player, a smart wearable device, a smart television, a vehicle-mounted device, a Personal Digital Assistant (PDA), and the like; the RTC terminal 500 may include, but is not limited to, a desktop computer, a mobile phone, a mobile computer, a tablet computer, a media player, a smart wearable device, a smart television, a vehicle mounted device, a PDA, and other electronic devices.

The converged communication gateway may be deployed on a server, and the converged communication gateway, the SIP server 200, and the RTC server 400 may be independent physical servers, may also be a server cluster or distributed system formed by a plurality of physical servers, and may also be a cloud server that provides basic cloud computing services such as cloud services, a cloud database, cloud computing, cloud storage, a cloud function, Network services, cloud communication, middleware services, domain name services, security services, a Content Delivery Network (CDN), and a big data and artificial intelligence platform.

The SIP terminal 100 and the SIP server 200 perform signaling interaction through an SIP protocol, the RTC server 400 and the RTC terminal 500 perform signaling interaction through an RTC protocol, and the converged communication gateway can be used as a communication link between the SIP server and the RTC server, on one hand, the SIP protocol can be adopted to perform signaling interaction with the SIP server, on the other hand, the RTC protocol can be adopted to perform signaling interaction with the RTC server, so that conversion between the SIP protocol and the RTC protocol is realized, further, the SIP terminal can join in an RTC audio-video conference created by the RTC server, and communication between the SIP terminal and the RTC terminal in the RTC audio-video conference is realized.

The converged communication method of the online conference according to the exemplary embodiment of the present disclosure is described below with reference to an application scenario of fig. 1. The above application scenarios are merely illustrative for facilitating an understanding of the spirit and principles of the present disclosure, and embodiments of the present disclosure are not limited in any way in this respect. Rather, embodiments of the present disclosure may be applied to any scenario where applicable.

Referring to fig. 2, an embodiment of the present disclosure provides a converged communication method for an online conference, which may be applied to a converged communication gateway, such as converged communication gateway 300 shown in fig. 1; the converged communication method for the online conference can comprise the following steps:

step S201, receiving a call request carrying a call identifier of the SIP terminal sent by the SIP server, and acquiring an RTC conference identifier of the RTC conference bound to the call identifier.

The SIP terminal may be a telephony device supporting an SIP protocol, such as an SIP hardware phone, an electronic device installed with an SIP softphone, where the SIP hardware phone includes a PSTN (Public Switched Telephone Network) mobile phone, a PSTN fixed phone, and the like, and the electronic device may include but is not limited to a desktop computer, a mobile computer, a tablet computer, a media player, an intelligent wearable device, an intelligent television, a vehicle-mounted device, a PDA, and the like. The SIP terminal and the SIP server may form a SIP audio video conference system, for example: the SIP system of an enterprise includes SIP terminals and an ip pbx system, which may be understood as a SIP server, for managing access of the SIP terminals.

The SIP terminal can send a call request carrying a call identifier to the SIP server, wherein the call identifier can be an SIP conference identifier, an SIP call number and the like and can be determined according to an actual application scene; the SIP server may then send the call request of the SIP terminal to the converged communication gateway.

Further, the acquiring, by the converged communication gateway, the RTC conference identifier of the RTC conference bound to the SIP conference identifier may specifically include the following steps:

and sending a verification request carrying the SIP conference identifier to the RTC server to obtain the RTC conference identifier of the RTC conference bound with the SIP conference identifier, which is returned by the RTC server after the verification is passed.

In the embodiment of the disclosure, after receiving a call request of an SIP terminal, the converged communication gateway may send a verification request carrying a call identifier to the RTC server, and if the RTC server matches the call identifier from a binding relationship between the pre-created call identifier and the RTC conference identifier, the RTC server passes the verification and sends the RTC conference identifier bound to the call identifier to the converged communication gateway.

Step S202, sending an RTC conference joining request of the SIP terminal to the RTC server, and establishing a first media transmission channel with the RTC server after determining that the RTC server joins the SIP terminal to the RTC conference; the RTC conference joining request comprises an RTC conference identifier.

In specific implementation, after acquiring an RTC conference identifier of an RTC conference, the converged communication gateway simulates a user at an SIP terminal to an RTC user, sends a request for joining the RTC conference from the SIP terminal to the RTC server, and after joining the RTC conference from the SIP terminal, the RTC server returns a successful message for joining the RTC conference to the converged communication gateway.

In addition, the converged communication gateway can negotiate with the RTC server about a preset encoding format of the media stream information transmitted by the first media transmission channel while establishing the first media transmission channel with the RTC server, so as to ensure that the media stream information transmitted by the converged communication gateway can be decoded by the RTC server.

Step S203, the first media stream information of the SIP terminal sent by the SIP server is sent to the RTC server through the first media transmission channel, so that the RTC server sends the first media stream information to the corresponding RTC terminal in the RTC conference.

In this step, the first media stream information of the SIP terminal may include one or both of video stream information and audio stream information. If the coding format of the first media stream information does not conform to the corresponding preset coding format, transcoding the first media stream information into the preset coding format; if the coding format of the first media stream information conforms to the corresponding preset coding format, transcoding is not required.

Therefore, in a possible implementation manner, the step S202 of sending the first media stream information of the SIP terminal sent by the SIP server to the RTC server through the first media transmission channel may further include the following steps:

b1, transcoding the first media stream information of the SIP terminal sent by the SIP server into a preset encoding format;

and B2, sending the transcoded first media stream information to the RTC server through the first media transmission channel.

Optionally, considering that the video encoding formats supported by the SIP terminal and the RTC terminal may be the same and the performance overhead of video transcoding is relatively large, for the video stream information, the preset encoding format corresponding to the video stream information may be set to the encoding format supported by both the SIP terminal and the RTC terminal, for example, the H264 encoding format, so that transcoding is not required for the video stream information, and the additional performance overhead may be avoided.

In addition, for the audio stream information, because the audio encoding capabilities of different terminals are different, for example, the PSTN mobile phone uses G711, G729 and other encoding formats, and the encoding format supported by the RTC terminal is the OPUS encoding format, at this time, the preset encoding format is set as the OPUS encoding format, so for the audio stream information of the SIP terminal, if the encoding format does not conform to the preset encoding format, the audio stream information of the SIP terminal needs to be transcoded into the preset encoding format to ensure that the transmitted audio stream information can be decoded by the RTC server.

In the embodiment of the disclosure, the converged communication gateway can perform signaling interaction with the SIP server on the one hand and can perform signaling interaction with the RTC server on the other hand, thereby realizing conversion between the SIP protocol and the RTC protocol, further enabling the SIP terminal to join the RTC conference, and transmitting media stream information of the SIP terminal to the corresponding RTC terminal, thereby realizing communication between the SIP terminal and the RTC terminal in the RTC conference.

The method of the embodiment of the present disclosure can be applied to various converged communication scenarios, for example: the method comprises the following steps that the SIP terminal joins an RTC conference scene, the SIP terminal calls the RTC terminal (namely point-to-point calling) scene and the like. The two scenarios are described below as examples.

In some embodiments, for the SIP terminal joining the RTC conference scenario, the call request in step S201 may be a request for joining an SIP conference, and the call identifier is an SIP conference identifier; in the step S201, receiving a call request carrying a call identifier sent by the SIP terminal, and acquiring an RTC conference identifier of the real-time communication RTC conference bound to the call identifier, may include the following steps:

a1, responding to a SIP conference adding request which is sent by the SIP terminal and carries the SIP conference identifier, and adding the SIP terminal into the SIP conference corresponding to the SIP conference identifier;

a2, acquiring RTC conference identification of the RTC conference bound with the SIP conference identification.

For example, the SIP conference identity may be a SIP conference number, which may be obtained by: the RTC terminal requests to create an RTC conference from the RTC server, when the RTC server creates the RTC conference, the RTC server generates an RTC conference identifier and a corresponding SIP conference identifier, and the RTC conference identifier and the SIP conference identifier are bound and then returned to the RTC terminal, and then the RTC terminal can send the SIP conference identifier to an SIP terminal needing to join the RTC conference through a setting mode (such as mail, short message and the like), wherein the RTC conference can be an RTC audio conference or an RTC video conference, and the RTC conference identifier can include RTC conference room information and the like.

After the SIP terminal acquires the SIP conference identifier, a call request carrying the SIP conference identifier can be sent to the SIP server, the call request can also carry the identifier of the SIP terminal and the like, then the SIP server sends the call request to the converged communication gateway, the converged communication gateway can add the SIP terminal into the corresponding SIP conference according to the SIP conference identifier, and acquire the RTC conference identifier of the RTC conference bound with the SIP conference identifier.

In some possible embodiments, the acquiring, by the converged communication gateway, the RTC conference identifier of the RTC conference bound to the SIP conference identifier may include the following steps:

In this embodiment, after receiving a call request from the SIP terminal, the converged communication gateway may send a verification request carrying a call identifier to the RTC server, and if the RTC server matches the call identifier from a binding relationship between the pre-created call identifier and the RTC conference identifier, the RTC server passes the verification and sends the RTC conference identifier bound to the call identifier to the converged communication gateway.

Further, the steps S202 to S203 may be continuously performed, which is not described herein again.

In specific implementation, the SIP terminal can join in an RTC conference scene, where multiple SIP terminals can join in one RTC conference at the same time, the converged communication gateway can forward media stream information of the SIP terminals to the RTC terminals based on an SFU architecture, under the SFU architecture, the RTC terminals can selectively subscribe to media stream information of the SIP terminals and other RTC terminals, and after receiving the media stream information of a certain SIP terminal, the RTC server can forward the media stream information of the SIP terminal to the RTC terminals subscribing to the SIP terminal, and at the same time, can also forward media stream information of other RTC terminals subscribed by each RTC terminal to the corresponding RTC terminals. It can be understood that the RTC terminal receives multiple media stream information.

In the embodiment of the disclosure, in a scenario that the SIP terminal joins the RTC conference, the converged communication gateway can perform signaling interaction with the SIP server on the one hand and perform signaling interaction with the RTC server on the other hand, so as to realize conversion between the SIP protocol and the RTC protocol, further enable the SIP terminal to join the RTC conference, transmit media stream information of the SIP terminal to the corresponding RTC terminal, and realize communication between the SIP terminal and each RTC terminal in the RTC conference.

In other embodiments, for the scenario that the SIP terminal calls the RTC terminal, the call request in step S201 may be a call RTC terminal request, and the call identifier may be a SIP call number.

For example, in a scenario where an SIP terminal calls an RTC terminal, the SIP server may be a call center server, and the call center server may send a call request carrying an SIP call number to a converged communication gateway after obtaining the SIP call number, where the SIP call number may be obtained by:

when the SIP terminal needs to call the RTC terminal, the SIP terminal firstly sends a call request to the call center server, the call center server designates the corresponding RTC terminal according to the call request, and sends the called request to the corresponding RTC terminal, so that the RTC terminal creates an RTC conference through the RTC server and generates an RTC conference identifier and a corresponding SIP call number, the RTC conference only comprises one RTC terminal, and the RTC server can send the SIP call number to the call center server. The call center server can construct a call command, and send a call request carrying an SIP call number to the convergence communication gateway, wherein the call request can also carry an identification of an SIP terminal and the like.

In some possible embodiments, after receiving a call request carrying an SIP call number, the converged communication gateway may acquire an RTC conference identifier of an RTC conference bound to the SIP call number, and may specifically perform the following steps:

and sending a verification request carrying the SIP calling number to the RTC server to obtain the RTC conference identifier of the RTC conference bound with the SIP calling number, which is returned by the RTC server after the verification is passed.

In the embodiment of the disclosure, in a scenario where the SIP terminal calls the RTC terminal, the convergence communication gateway can perform signaling interaction with the SIP server on the one hand and perform signaling interaction with the RTC server on the other hand, so as to realize conversion between the SIP protocol and the RTC protocol, and further transmit media stream information of the SIP terminal to the corresponding RTC terminal, thereby realizing point-to-point communication between the SIP terminal and the RTC terminal.

Based on the above embodiments of the present disclosure, the converged communication gateway can send the media stream information of the SIP terminal to the corresponding RTC terminal, and can also obtain the media stream information of the RTC terminal, and send the media stream information of the RTC terminal to the SIP terminal, and the following introduces the two scenarios, namely adding the RTC conference scenario to the SIP terminal and calling the RTC terminal by the SIP terminal, respectively.

In some embodiments, for a SIP terminal joining an RTC conference scenario, as shown in fig. 3, the converged communication gateway may further perform the following steps:

and step S301, adding each RTC terminal in the RTC conference into the SIP conference, and establishing a second media transmission channel with the RTC server.

The SIP conference can include a plurality of SIP terminals, the converged communication gateway can simulate a false RTC terminal (namely a virtual terminal) to join the RTC conference, then list information of all RTC terminals in the RTC conference returned by the RTC server is obtained, then each RTC terminal is joined in the SIP conference, and a second media transmission channel with the RTC server is established, wherein the second media transmission channel is used for obtaining media stream information of the RTC terminal transmitted by the RTC server.

In addition, the converged communication gateway can negotiate with the RTC server about a preset encoding format of the media stream information transmitted by the second media transmission channel while establishing a second media transmission channel with the RTC server, so as to ensure that the media stream information transmitted by the RTC server can be decoded by the converged communication gateway; the preset encoding format of the media stream information transmitted by the second media transmission channel may be the same as or different from the preset encoding format of the media stream information transmitted by the first media transmission channel, and the embodiment of the disclosure is not limited thereto.

Step S302, receiving, by a second media transmission channel, second media stream information of each RTC terminal in the RTC conference, which is sent by the RTC server.

Wherein the second media stream information may include one or both of video stream information and audio stream information. After the RTC server acquires the respective second media stream information of each RTC terminal, if the coding format of each second media stream information does not conform to the corresponding preset coding format, transcoding the second media stream information into the preset coding format; and if the coding format of each piece of second media stream information conforms to the corresponding preset coding format, transcoding is not required.

Step S303, performing fusion processing on the second media stream information of each RTC terminal and the third media stream information sent by each SIP terminal in the SIP conference to obtain target media stream information.

Wherein the third media stream information includes one or both of audio stream information and video stream information.

In the embodiment of the present disclosure, the converged communication gateway may perform audio mixing processing and/or screen mixing processing on the second media stream information of each RTC terminal and the third media stream information of each SIP terminal based on an MCU architecture, and then encode the mixed media stream information to obtain the target media stream information.

In one possible implementation, the second media stream information and the third media stream information each include audio stream information; in the step S303, the merging the second media stream information of each RTC terminal and the third media stream information sent by each SIP terminal in the SIP conference to obtain the target media stream information may include the following steps:

and audio stream information in the second media stream information of each RTC terminal and audio stream information in the third media stream information sent by each SIP terminal in the SIP conference are subjected to audio mixing processing to obtain first target media stream information.

The audio mixing process may include three processes of decoding, mixing and encoding, and the audio mixing process includes decoding each piece of audio stream information, mixing each piece of decoded audio stream information, and encoding the mixed audio stream information to obtain the first target media stream information.

In another possible implementation, the second media stream information and the third media stream information each further include video stream information; in the step S303, the second media stream information of each RTC terminal and the third media stream information sent by each SIP terminal in the SIP conference are fused to obtain the target media stream information, which may further include the following steps:

The screen mixing processing may include three processes of decoding, screen mixing and encoding, and the screen mixing processing includes decoding each piece of video stream information, then mixing the screen of each piece of decoded video stream information, and then encoding the video stream information after screen mixing to obtain second target media stream information.

And step S304, respectively sending the target media stream information to each SIP terminal in the SIP conference.

It can be understood that each SIP terminal receives one path of media stream information after mixing and mixing, and the RTC terminal receives multiple paths of media stream information.

For example, the terminals participating in the RTC conference include an RTC terminal 1, an RTC terminal 2, an SIP terminal 1, and an SIP terminal 2, and the RTC terminal 1 can receive the multi-path media stream information of the RTC terminal 2, the SIP terminal 1, and the SIP terminal 2, at this time, the display screen is as shown in fig. 4A, the local screen refers to the local display screen of the RTC terminal 1, and the RTC2, the SIP1, and the SIP2 are respectively display screens of the media stream information of the RTC terminal 2, the media stream information of the SIP terminal 1, and the media stream information of the SIP terminal 2. The SIP terminal 1 may receive one path of media stream information obtained by mixing multiple paths of media stream information of the RTC terminal 1, the RTC terminal 2, the SIP terminal 1, and the SIP terminal 2, and at this time, the display screen is as shown in fig. 4B.

In the embodiment of the disclosure, the SIP terminal can join the RTC conference by fusing the communication gateways, which not only can send the media stream information of the SIP terminal to the corresponding RTC terminal, but also can acquire the media stream information of the RTC terminal and send the media stream information of the RTC terminal to the SIP terminal, thereby realizing the intercommunication between the SIP terminal and the RTC terminal.

In addition, when a plurality of SIP terminals join in one RTC conference at the same time, for the video stream information, because the conference pictures displayed by each SIP terminal are the same, the fusion communication gateway only needs to mix the screen for the video stream information of each terminal in the RTC conference once, and the performance overhead can be greatly reduced; for the audio stream information, each SIP terminal needs to exclude the sound collected by the SIP terminal, so the convergence communication gateway needs to perform audio mixing processing N-1 times on the audio stream information of each terminal in the RTC conference, where N is the number of each terminal, and the performance overhead of the audio mixing processing is relatively small compared with that of the screen mixing processing, and thus, a large performance overhead is not caused.

In other embodiments, for the scenario that the SIP terminal calls the RTC terminal, as shown in fig. 5, the converged communication gateway may further perform the following steps:

step S401, establishing a third media transmission channel with the RTC server, and receiving fourth media stream information of one RTC terminal in the RTC conference, which is sent by the RTC server, through the third media transmission channel.

In this step, the converged communication gateway establishes a third media transmission channel with the RTC server, and simultaneously negotiates with the RTC server about a preset encoding format of media stream information transmitted by the third media transmission channel to ensure that the media stream information transmitted by the RTC server can be decoded by the converged communication gateway; the preset encoding format of the media stream information transmitted by the third media transmission channel may be the same as or different from the preset encoding format of the media stream information transmitted by the first media transmission channel, and the embodiment of the disclosure is not limited thereto.

Wherein the fourth media stream information may include one or both of video stream information and audio stream information. After the RTC server acquires the fourth media stream information of the RTC terminal, if the coding format of the fourth media stream information does not conform to the corresponding preset coding format, transcoding the fourth media stream information into the preset coding format; and if the coding format of the fourth media stream information conforms to the corresponding preset coding format, transcoding is not required.

Step S402, the fourth media stream information of an RTC terminal is sent to the SIP terminal.

The embodiment of the disclosure can enable the SIP terminal to call the RTC terminal point-to-point, not only can send the media stream information of the SIP terminal to the RTC terminal, but also can acquire the media stream information of the RTC terminal and send the media stream information of the RTC terminal to the SIP terminal, thereby realizing the intercommunication between the SIP terminal and the RTC terminal.

The converged communication system for online conferencing according to the embodiment of the present disclosure is described below with reference to fig. 6.

As shown in fig. 6, the converged communication system for online conferencing according to the embodiment of the present disclosure includes an SIP docking system, a converged communication gateway, and an RTC service.

The SIP docking system comprises an SIP server and SIP terminals (including SIP hardware terminals, PSTN mobile phones, PSTN fixed phones and the like), the SIP server can be an enterprise SIP system, such as an enterprise telephone switch IPPBX system, and the IPPBX is used for managing access of the SIP terminals. The converged communication gateway, namely the converged communication gateway of the above-mentioned embodiment of the present disclosure, can implement RTC/SIP protocol conversion, so that the SIP terminal and the RTC terminal are converged and intercommunicated. The RTC service system and the RTC edge media service in the RTC service may be understood as the RTC server in the above embodiments of the disclosure, and the RTC service joins the RTC conference service for the RTC terminal. The converged communication gateway is described in detail below.

The converged communication gateway includes 3 services: SIP signaling proxy service, SIP media service, Thruster service (SIP/RTC protocol conversion service); the SIP media service can comprise a plurality of SIP media services, and each SIP media service is bound with one thread service; the SIP signaling proxy service can support high availability of a Virtual IP Address (VIP), and master-slave switching of the SIP signaling proxy service is realized.

In a specific implementation, the SIP signaling proxy service, the SIP media services, and the protocol conversion services may be deployed on the same server or on different servers. For example, when deployed on different servers, a SIP signaling proxy service may be deployed on one server, and a SIP media service and a bound protocol conversion service may be deployed on one server.

1) The SIP signaling proxy service mainly plays a role of SIP signaling routing, is butted with an SIP butt joint system, realizes the butt joint of incoming calls and outgoing calls of SIP terminals, dispatches and distributes SIP media services, and intelligently routes call requests of the SIP terminals to the corresponding SIP media services.

When the SIP signaling proxy service is in butt joint with the SIP butt joint system, two SIP standard butt joint modes of SIP registration and IP butt joint are supported, incoming calls and outgoing calls are supported, namely, incoming calls of the SIP terminal and the PSTN mobile fixed telephone can be accepted to join the RTC conference, and the RTC terminal can also actively call the SIP terminal and the PSTN mobile fixed telephone.

When the SIP signaling proxy service intelligently routes the SIP media service, a Load Balance (LBS) strategy can be adopted, after the SIP signaling proxy service receives a call request of an SIP terminal, an available SIP media service can be distributed according to the Load of the SIP media service and the IP proximity principle, and meanwhile, the low delay of an edge node is ensured, but if an SIP conference added by the SIP terminal exists, the SIP media service corresponding to the SIP conference can be distributed.

2) The SIP media service can be based on an open source freeswitch, which is an SIP soft switch system and integrates processing modules such as Sofia, Conference, Codec, RTP, SFU, Verto, ESL and the like.

And the Sofia is a SIP signaling media module used for receiving or initiating SIP terminal call processing.

CCConference: and the conference management module is used for SIP conference room management and audio mixing and screen mixing processing and is based on an MCU framework.

CCCODec, encoding and decoding module, for audio encoding and decoding processing and video encoding and decoding processing.

RTP engine module, mainly used for receiving and transmitting RTP media stream, RTP is a standard output.

SFU media transfer module, to transfer the SIP terminal media flow to RTC edge media service, on RTC service side, RTC terminal can selectively subscribe SIP terminal media flow.

Verto: the Verto signaling media module provides a websocket + rpc (Remote Procedure Call) mode for signaling media negotiation, and is different from the SIP signaling in that the SIP signaling adopts a standard protocol, the SIP signaling carries sdp (Session Description protocol) media stream information, and the websocket + rpc in Verto supports websocket connection, and the websocket transmits sdp media stream information in json (JavaScript Object Notation, JS Object Notation) format.

ESL: the ESL event processing module, the router, can be connected to the SIP media service through the ESL event processing module, the SIP media process the SIP user in the incoming and outgoing call process, the generated event will inform the SIP/RTC conversion service, at the same time, it will also receive and process the control command (call control, SIP conference management) sent from the SIP/RTC conversion service.

3) The cluster service is used for converting SIP and RTC protocols and is a key for fusing and communicating the SIP terminal and the RTC terminal.

Based on the converged communication system, an RTC conference scene added by the SIP terminal and a point-to-point call scene of the SIP terminal and the RTC terminal are introduced respectively.

Scene 1: SIP terminal joining RTC conference

Firstly, an RTC terminal requests an RTC server to create an RTC conference in advance, an RTC conference identifier and an SIP conference short number are generated, and the RTC conference identifier and the SIP conference short number have a binding relationship.

The SIP terminal dials the SIP conference short number, calls in the SIP signaling proxy service in the converged communication gateway through an SIP system of an enterprise, intelligently routes the call request to the corresponding SIP media service, receives the call request by the SIP media service, performs dialing matching, adds the SIP terminal into the SIP conference, and notifies the Thruster service of the event that the SIP terminal is added into the SIP conference.

The method comprises the steps that a Thruster service receives a notice that an SIP terminal joins in an SIP conference event, firstly, login authentication is carried out on an RTC service system, after the authentication is passed, a nearby RTC edge media service is distributed through a scheduling service of the RTC service system, and an IP address of the RTC edge media service and an RTC conference identifier (such as RTC conference room information) are responded to the Thruster service.

When the media stream of the SIP terminal is pushed to the RTC edge media service by the Thruster service, the SIP terminal is simulated into the RTC terminal, the RTC conference is requested to be added to the RTC edge media service, if the RTC conference is successfully added, a media transmission channel is established between the Thruster service and the RTC edge media service, signaling interaction such as media negotiation and the like is carried out, then an SFU pushing control command is sent to the SIP media service through an ESL event processing module, SFU media forwarding forwards the media stream of the SIP terminal to the RTC edge media service, and if the encoding format of the media stream of the SIP terminal is different from the encoding format negotiated by the RTC edge media service, the SFU media forwarding module transcodes the media stream of the SIP terminal firstly and then transmits the media stream to the RTC edge media service through RTP. The RTC edge media service is based on an SFU architecture, the RTC terminal can selectively subscribe the audio stream or the video stream of the SIP terminal and the RTC terminal, and the RTC terminal can receive the media stream of the SIP terminal through the scheme.

Similarly, when the media stream of the RTC terminal is pulled to the SIP media service, the Thruster service simulates a false RTC terminal to join in the RTC conference, the list information of all RTC terminals in the RTC conference is obtained in response, each RTC terminal is simulated to be a Verto terminal to be connected and registered with the SIP media service, after each Verto terminal is successfully registered, the Thruster service firstly establishes a media transmission channel with the RTC edge media service and carries out media negotiation, then calls the SIP media service through Verto signaling, so that the SIP media service joins each Verto terminal in the SIP conference, carries the media stream information of the RTC terminal during calling, and after each Verto terminal is successfully joined in the SIP conference, the SIP media service sends the media address information of the SIP media service to the Thruster service, and then the Thruster service sends the pulled media stream of the RTC terminal to the SIP conference. The SIP conference of the SIP media service is an MCU (micro control unit) framework, carries out audio mixing and screen mixing processing on media streams of all terminals in the SIP conference, and then sends the audio mixing and screen mixing media streams to the SIP terminal, so that the SIP terminal displays one path of screen mixing pictures of the SIP conference, the SIP terminal can receive the media streams of the RTC terminal, and the screen mixing pictures can be managed, adjusted and distributed through meeting control.

In the above scenario, when a plurality of SIP terminals join in one RTC conference at the same time, for video stream information, because the conference pictures displayed by each SIP terminal are the same, the convergence communication gateway only needs to perform screen mixing processing once for the video stream information of each terminal in the RTC conference, which can greatly reduce performance overhead; for the audio stream information, each SIP terminal needs to exclude the sound collected by the SIP terminal, so the convergence communication gateway needs to perform audio mixing processing N-1 times on the audio stream information of each terminal in the RTC conference, where N is the number of each terminal, and the performance overhead of the audio mixing processing is relatively small compared with that of the screen mixing processing, and thus, a large performance overhead is not caused.

Scene 2: point-to-point calling between SIP terminal and RTC terminal

Different from the situation that the SIP terminal joins the RTC conference, the SIP media service does not need to carry out audio mixing and screen mixing processing because 1-to-1 conversation is adopted. When the SIP terminal calls in the converged communication gateway, the Thruster service can simulate the RTC terminal as a Verto terminal, the Verto terminal is connected and registered with the SIP media service, the Verto terminal is called by the SIP terminal through the SIP media service call processing, the Thruster service receives a signaling of calling the Verto terminal by the SIP terminal, the signaling carries a media stream of the SIP terminal, then the Thruster service carries out signaling interaction with an RTC edge media service, an SIP uplink media channel and an RTC downlink media channel are created, and the Verto signaling carrying the RTC media stream is responded to the SIP media service, so that point-to-point intercommunication between the SIP terminal and the RTC terminal is realized. Compared with the situation that a plurality of SIP terminals join the RTC conference, the method has the greatest advantages that the SIP media service is not required to carry out audio mixing and screen mixing processing, and the performance overhead of the SIP media service is greatly reduced.

It should be noted that, in a point-to-point call scenario between the SIP terminal and the RTC terminal, the converged communication gateway supports two modes, namely signaling proxy and media forwarding. In the signaling proxy mode, the converged communication gateway does not receive the media stream of the SIP terminal, namely the media stream of the SIP terminal is directly communicated between the SIP system of an enterprise and RTC edge media service, and the converged communication gateway is only responsible for RTC/SIP signaling conversion, so that the performance overhead can be almost ignored; the media forwarding mode is that the media stream of the SIP terminal firstly passes through the SIP media service of the converged communication gateway, and then the SIP media service is communicated with the RTC edge media service, so that the SIP system of an enterprise can be compatible.

The embodiment of the disclosure gives consideration to two calling scenes of intercommunication between the SIP terminal and the RTC terminal, namely, the SIP terminal joins in the RTC conference and the SIP terminal calls with the RTC terminal point to point. In the scene that the SIP terminals are added into the RTC conference, the RTC conference can be added into a plurality of SIP terminals, and two framework modes of MCU and SFU are compatible through RTC/SIP signaling protocol conversion, so that the SIP terminals and the RTC terminals can be interconnected and intercommunicated. In the SIP and RTC point-to-point calling scene, RTC/SIP signaling protocol conversion is optimized, audio mixing and screen mixing processing of SIP media service are not needed, and performance overhead is greatly reduced.

Based on the same inventive concept, the embodiment of the present disclosure further provides a converged communication gateway, and the principle of the gateway to solve the problem is similar to the method of the above embodiment, so that the implementation of the gateway may refer to the implementation of the method, and repeated details are not repeated.

Referring to fig. 7, a converged communication gateway provided by the embodiment of the present disclosure includes a SIP signaling proxy module 71, at least one SIP media service module 72, and at least one protocol conversion module 73;

the SIP signaling proxy module 71 is configured to receive a call request, which is sent by a session initiation protocol SIP server and carries a call identifier, of an SIP terminal, and allocate the call request to a corresponding SIP media service module 72;

the SIP media service module 72 is configured to establish a call event carrying a call identifier according to the call request, and send the call event to the corresponding protocol conversion module 73;

the protocol conversion module 73 is configured to acquire an RTC conference identifier of the real-time communication RTC conference bound to the call identifier, send an RTC conference joining request carrying the RTC conference identifier of the SIP terminal to the RTC server, establish a first media transmission channel between the SIP service module and the RTC server after determining that the RTC server joins the SIP terminal to the RTC conference, and send a media forwarding instruction to the SIP media service module 72;

the SIP media service module 72 is further configured to respond to the media forwarding instruction, and send the first media stream information of the SIP terminal sent by the SIP server to the RTC server through the first media transmission channel, so that the RTC server sends the first media stream information to the corresponding RTC terminal in the RTC conference.

The SIP signaling proxy module 71, the at least one SIP media service module 72, and the at least one protocol conversion module 73 may be deployed on the same server, or may be deployed on different servers. For example, when deployed on different servers, the SIP signaling proxy module 71 may be deployed on one server, and one SIP media service module 72 and one bound protocol conversion module 73 may be deployed on one server.

The SIP signaling proxy module 71, the SIP media service module 72 and the protocol conversion module 73 can be understood as the SIP signaling proxy service, the SIP media service and the router service in fig. 5 in the above embodiment.

In one possible implementation, the call request is a request for joining an SIP conference, and the call identifier is an SIP conference identifier;

the SIP media service module 72 further includes:

the SIP conference management submodule is used for responding to a SIP conference adding request which is sent by the SIP terminal and carries the SIP conference identification, and adding the SIP terminal into the SIP conference corresponding to the SIP conference identification;

the protocol conversion module 73 further includes:

In a possible implementation, the protocol conversion module 73 further includes:

the SIP media service module 72 further includes:

the receiving submodule is used for receiving second media stream information of each RTC terminal in the RTC conference, which is sent by the RTC server through a second media transmission channel;

the fusion sub-module is used for carrying out fusion processing on the second media stream information of each RTC terminal and the third media stream information sent by each SIP terminal in the SIP conference to obtain target media stream information;

the fusion submodule is further configured to:

and mixing the second media stream information of each RTC terminal and the third media stream information sent by each SIP terminal in the SIP conference to obtain first target media stream information.

the fusion submodule is further configured to:

In one possible implementation, the call request is a call RTC terminal request;

the protocol conversion module 73 further includes:

In one possible implementation, the SIP media service module 72 further includes:

and the request verification submodule is used for sending a verification request carrying the call identifier to the RTC server so as to obtain the RTC conference identifier of the RTC conference bound with the call identifier, which is returned by the RTC server after the verification is passed.

Based on the same inventive concept, the embodiment of the present disclosure further provides an electronic device, and the principle of the electronic device to solve the problem is similar to the method of the above embodiment, so that the implementation of the electronic device may refer to the implementation of the method, and repeated details are not repeated. Fig. 8 shows a schematic structural diagram of an electronic device provided in an embodiment of the present disclosure.

Referring to fig. 8, an electronic device may include a processor 802 and a memory 801. The memory 801 provides the processor 802 with program instructions and data stored in the memory 801. In the embodiment of the present disclosure, the memory 801 may be used to store a program of converged communication of an online conference in the embodiment of the present disclosure.

The processor 802 is configured to execute the method in any of the above-described method embodiments, such as the converged communication method for online conferencing provided by the embodiment shown in fig. 2, by calling the program instructions stored in the memory 801 by the processor 802.

The specific connection medium between the memory 801 and the processor 802 is not limited in the embodiments of the present disclosure. In fig. 8, the memory 801 and the processor 802 are connected by a bus 803, the bus 803 is represented by a thick line in fig. 8, and the connection manner between other components is merely illustrative and not limited. The bus 803 may be divided into an address bus, a data bus, a control bus, and the like. For ease of illustration, only one thick line is shown in FIG. 8, but this is not intended to represent only one bus or type of bus.

The Memory may include a Read-Only Memory (ROM) and a Random Access Memory (RAM), and may further include a Non-Volatile Memory (NVM), such as at least one disk Memory. Alternatively, the memory may be at least one memory device located remotely from the processor.

The Processor may be a general-purpose Processor, including a central processing unit, a Network Processor (NP), and the like; but may also be a Digital instruction processor (DSP), an application specific integrated circuit, a field programmable gate array or other programmable logic device, discrete gate or transistor logic, discrete hardware components, or the like.

The disclosed embodiment also provides a computer storage medium, in which a computer program is stored, a processor of a computer device reads the computer program from the computer readable storage medium, and the processor executes the computer program, so that the computer device executes the converged communication method for the online conference in any method embodiment.

In particular implementations, computer storage media may include: various storage media capable of storing program codes, such as a Universal Serial Bus Flash Drive (USB), a mobile hard disk, a Read-Only Memory (ROM), a Random Access Memory (RAM), a magnetic disk, or an optical disk.

In some possible embodiments, the aspects of the converged communication method for an online conference provided by the present disclosure may also be implemented in the form of a program product including program code for causing a computer device to perform the steps of the converged communication method for an online conference according to various exemplary embodiments of the present disclosure described above in this specification when the program product is run on the computer device, for example, the computer device may perform a converged communication flow of the online conference in steps S201 to S203 shown in fig. 2.

As will be appreciated by one skilled in the art, embodiments of the present disclosure may be provided as a method, system, or computer program product. Accordingly, the present disclosure may take the form of an entirely hardware embodiment, an entirely software embodiment or an embodiment combining software and hardware aspects. Furthermore, the present disclosure may take the form of a computer program product embodied on one or more computer-usable storage media (including, but not limited to, disk storage, CD-ROM, optical storage, and so forth) having computer-usable program code embodied therein.

The present disclosure is described with reference to flowchart illustrations and/or block diagrams of methods, apparatus (systems), and computer program products according to the present disclosure. It will be understood that each flow and/or block of the flow diagrams and/or block diagrams, and combinations of flows and/or blocks in the flow diagrams and/or block diagrams, can be implemented by computer program instructions. These computer program instructions may be provided to a processor of a general purpose computer, special purpose computer, embedded processor, or other programmable data processing apparatus to produce a machine, such that the instructions, which execute via the processor of the computer or other programmable data processing apparatus, create means for implementing the functions specified in the flowchart flow or flows and/or block diagram block or blocks.

These computer program instructions may also be stored in a computer-readable memory that can direct a computer or other programmable data processing apparatus to function in a particular manner, such that the instructions stored in the computer-readable memory produce an article of manufacture including instruction means which implement the function specified in the flowchart flow or flows and/or block diagram block or blocks.

These computer program instructions may also be loaded onto a computer or other programmable data processing apparatus to cause a series of operational steps to be performed on the computer or other programmable apparatus to produce a computer implemented process such that the instructions which execute on the computer or other programmable apparatus provide steps for implementing the functions specified in the flowchart flow or flows and/or block diagram block or blocks.

It will be apparent to those skilled in the art that various changes and modifications can be made in the present disclosure without departing from the spirit and scope of the disclosure. Thus, if such modifications and variations of the present disclosure fall within the scope of the claims of the present disclosure and their equivalents, the present disclosure is intended to include such modifications and variations as well.

Claims

1. A converged communication method of an online conference is applied to a converged communication gateway, and comprises the following steps:

2. The method of claim 1, wherein the call request is a join SIP conference request, and the call identifier is a SIP conference identifier;

3. The method of claim 2, further comprising:

4. The method of claim 3, wherein the second media stream information and the third media stream information each comprise audio stream information;

5. The method of claim 4, wherein the second media stream information and the third media stream information each further comprise video stream information;

6. The method of claim 1, wherein the call request is a call RTC terminal request;

the method further comprises the following steps:

7. The method according to any one of claims 1 to 6, wherein the sending the first media stream information of the SIP terminal sent by the SIP server to the RTC server through the first media transmission channel comprises:

8. A converged communication gateway, comprising a session initiation protocol, SIP, signalling proxy module, at least one SIP media service module and at least one protocol conversion module;

9. An electronic device, comprising a processor and a memory, wherein the memory stores program code which, when executed by the processor, causes the processor to perform the steps of the method of any of claims 1 to 7.

10. A computer-readable storage medium, characterized in that it comprises program code for causing an electronic device to carry out the steps of the method according to any one of claims 1 to 7, when said program code is run on said electronic device.