KR102546662B1 - System for providing conference connecting between public switched telephone network communication and web real-time communication technology, and method thereof - Google Patents

Abstract

The present invention relates to a system and method for providing a conference linked to public switched telephone network communication and web real-time communication, and more particularly, to a conference providing system and method for providing a conference through a public switched telephone network (PSTN) and at least one voice user participating in a conference through an Internet network. Provides a conference platform that supports multiple users, including at least one user participating in a conference, to participate in a real-time conference (meeting) without limiting the number of people, providing a real-time conference environment regardless of the user's access path Provided is a public switched telephone network communication and web real-time communication-linked conference providing system and method.

Description

Public switched telephone network communication and web real-time communication linked conference providing system and method thereof

The present invention relates to a system and method for providing a conference linked to public switched telephone network communication and web real-time communication, and more particularly, to a conference providing system and method for providing a conference through a public switched telephone network (PSTN) and at least one voice user participating in a conference through an Internet network. Provides a conference platform that supports multiple users, including at least one user participating in a conference, to participate in a real-time conference (meeting) without limiting the number of people, providing a real-time conference environment regardless of the user's access path It relates to a public switched telephone network communication and web real-time communication interlocking conference providing system and method thereof.

Web Real-Time Communication (WebRTC) is an API (Application Programming Interface) designed to allow web browsers to communicate with each other without the help of a plug-in. It is a draft proposed by the World Wide Web Consortium (W3C), an international standardization organization, and can be used for voice calls, video calls, and P2P file sharing.

A video conferencing system is a system that conducts a face-to-face meeting through a video camera rather than a direct face-to-face meeting through a camera, speaker, microphone, and video conferencing software for video conferencing.

WebRTC (Web Real-Time Communication) enables voice calls, video chatting, video calls (Video Telephone: VT, or Video Call), and video calls through browser applications without installing additional plug-ins. It is a technology that provides video conference, file transfer, and the like.

WebRTC-based video chatting and video conferences are implemented in a P2P manner without an intermediary such as a server, and the number of people who can participate in one video conference is about 10, which is insufficient to build a high-definition video conference or multiple video conference systems. many.

To solve this problem, a multi-point control unit (MCU) or selective forwarding unit (SFU) capable of providing a server-based multi-party video conference service supporting WebRTC is required. .

In addition, a conventional conference system uses a phone (wired/wireless phone) using another communication method while making a phone call through one communication channel (Network, wired/wireless communication) in the conference system to implement a video or audio conference. In order to participate, a method of making a call using a separate gateway or using a product that uses a heterogeneous communication channel within the same product must be built-in.

That is, there is a disadvantage in that a separate equipment or dedicated line is required in order to additionally participate in heterogeneous terminals in the conventional conference system.

Korean Registered Patent [10-1447438] discloses a communication system using heterogeneous networks.

Korean Registered Patent [10-1555855] discloses a video conference method and system of various participating devices.

Korean Patent Publication [10-2013-0132040] discloses a conference system using heterogeneous terminals and a control method thereof.

Korean Patent Publication [10-2017-0071251] discloses a multi-access control device providing a conference service.

Korean Registered Patent [10-1447438] (registration date: 2014. 09. 29) Korean Registered Patent [10-1555855] (registration date: 2015. 09. 21) Korea Patent Publication [10-2013-0132040] (Publication date: 2013. 12. 04) Korean Patent Publication [10-2017-0071251] (Publication date: 2017. 06. 23)

Therefore, the present invention has been made to solve the above problems, and an object of the present invention is to participate in a video conference through at least one voice user participating in a conference through a public switched telephone network (PSTN) and an Internet network. Provides a conference platform that supports multiple users, including at least one user, to participate in a real-time conference (meeting) without limiting the number of people, so that users can receive a real-time conference environment regardless of their access path To provide a public switched telephone network communication and web real-time communication linked conference provision system and method.

Objects of the embodiments of the present invention are not limited to the above-mentioned purposes, and other objects not mentioned above will be clearly understood by those skilled in the art from the description below. .

To achieve the above object, a conference providing system linked to public switched telephone network communication and web real-time communication according to an embodiment of the present invention includes at least one first user terminal 100 participating in a conference through an Internet network. ; at least one second user terminal 200 participating in the conference through a telephone network; a web server 101 for providing a conference environment to the first user terminal through a web browser; A load balance server for selecting a media server according to an outgoing request from the first user terminal or the second user terminal, requesting creation of an audio bridge room to the media server, and requesting resource allocation to the created audio bridge room. (103); The audio bridge room is created according to a request from the load balance server, a first audio bridge between the first user terminal and the audio bridge room is connected, and a second audio bridge between the second user terminal and the audio bridge room is created. a plurality of media servers 108 for connecting the first user terminal and the second user terminal to the same audio bridge room by connecting bridges; and converting a Web Real-Time Communication (WebRTC) protocol signal into a Session Initiation Protocol (SIP) signal, which is a communication standard of the telephone network, and transmitting the signal to the telephone network or a SIP protocol signal received from the telephone network. It includes a telecom server 106 for converting a WebRTC protocol signal into a corresponding media server.

The web server 101 is characterized in that it separates the audio signal and the video signal received from the first user terminal and transmits them to the load balance server.

The telecom server 106 includes Legacy Session Description Protocol (SDP), a media exchange technology supported by the telephone network, and WebRTC SDP (Web Real-Time Communication session Description Protocol), a media exchange technology supported by the Internet network. )), the WebRTC SDP signal is converted into a legacy SDP signal to enable media exchange between the telephone network and the Internet network and transmitted to the telephone network, and the legacy SDP signal received from the telephone network is converted into a WebRCT SDP signal. .

Since the voice codec supported by the telephone network (SIP protocol) and the voice codec supported by the Internet network (WebRTC protocol) are different in the telecom server 106, when receiving or transmitting a voice packet, after converting to the other party's voice codec characterized by transmission.

In the first audio bridge, an audio stream (OPUS) of a first voice codec is transmitted between the web server and the corresponding media server, and in the second audio bridge, between the telecom server and the media server, the audio stream (OPUS) is transmitted. An audio stream (OPUS) of one voice codec is transmitted, but an audio stream (G.711) of a second voice codec is transmitted between the telecom server and the telephone network.

In addition, a method for providing a public switched telephone network communication and web real-time communication-linked conference according to an embodiment of the present invention for achieving the above object is to transmit a call from a first user terminal using the Internet network and use the telephone network. A first call processing step in which an incoming call to a second user terminal is processed (S100); A second call processing step (S200) of processing an incoming call from a third user terminal using a telephone network to a fourth user terminal using an Internet network; And a conference support step of supporting a conference between at least one user terminal using the telephone network connected according to the first call processing step or / and the second call processing step and at least one user terminal using the Internet network ( S300) is included.

In the first call processing step (S100), the web server transfers the first SDP request signal (WebRTC Offer SDP1) to the load balance server according to the call from the first user terminal of the Internet network, and the audio signal and Separating and transmitting video signals (S301); The load balance server selects a media server and requests the corresponding media server to create an audio bridge room (S302); When the audio bridge room is created, the load balance server transmits a first SDP response signal (WebRTC Answer SDP1) to the web server (S303); connecting a first audio bridge between the web server and the media server (S304) and transmitting an audio stream between the first user terminal and the media server through the web server; Transmitting, by the load balance server, a second SDP request signal (WebRTC Offer SDP2) requesting an audio bridge resource for a called user of the telephone network to the media server (S305); The load balance server transmitting the second SDP request signal (WebRTC Offer SDP2) requesting a phone connection to the telecom server (S306); The telecom server converts the second SDP request signal (WebRTC Offer SDP2) into a SIP signal and transmits a session initiation protocol request signal (SIP Offer) to the telephone network (S307); Converting, by the telecom server, a session initiation protocol response signal (SIP Answer) received from the telephone network into a WebRTC signal according to the incoming call response of the second user terminal (S308); Transmitting, by the telecom server, a second SDP response signal (WebRTC Answer SDP2) responding to a phone connection to the load balance server (S309); Connecting the Internet network and the telephone network as the second SDP response signal (WebRTC Answer SDP2) is transmitted (S310); transmitting, by the media server, the second SDP response signal (WebRTC Answer SDP2) to the load balance server as a response to an audio bridge resource for the called user of the telephone network (S311); and connecting a second audio bridge between the telecom server and the media server (S312), and transmitting an audio stream between the second user terminal of the telephone network and the media server.

In the first audio bridge, an audio stream (OPUS) of a first voice codec is transmitted between the web server and the media server, and in the second audio bridge, between the telecom server and the media server, the audio stream (OPUS) is transmitted. An audio stream (OPUS) of one voice codec is transmitted, but an audio stream (G.711) of a second voice codec is transmitted between the telecom server and the telephone network.

In the second call processing step (S200), when the telephone network transmits a session initiation protocol request signal (SIP Offer) to the telecom server according to an outgoing call from the third user terminal of the telephone network, the telecom server Converting the session initiation protocol request signal (SIP Offer) received from the telephone network into a WebRTC signal (S401); transmitting, by the telecom server, a third SDP request signal (WebRTC Offer SDP3) requesting a phone connection to the load balance server (S402); requesting, by the load balance server, the media server to create an audio bridge room (S403); Transmitting a fourth SDP request signal (WebRTC Offer SDP4) requesting an audio bridge resource for the Internet network called user between the load balance server and the media server as the audio bridge room is created (S404); The load balancing server transmitting the fourth SDP request signal (WebRTC Offer SDP4) to the web server (S405); Transmitting, by the web server, a fourth SDP response signal (WebRTC Answer SDP4) to the load balance server according to the incoming call response of the fourth user terminal (S406); transmitting, by the media server, the fourth SDP response signal (WebRTC Answer SDP4) to the load balance server as a response to an audio bridge resource for the Internet network called user (S407); connecting a third audio bridge between the web server and the media server (S408) and transmitting an audio stream between the fourth user terminal and the media server through the web server; The load balance server transmitting a third SDP response signal (WebRTC Answer SDP3) responding to the phone connection to the telecom server (S409); When the telecom server receives the third SDP response signal (WebRTC Answer SDP3), converting the WebRTC signal into a SIP signal and transmitting a session initiation protocol response signal (SIP Answer) to the telephone network (S410); and connecting a fourth audio bridge between the telecom server and the media server (S411), and transmitting an audio stream between the third user terminal of the telephone network and the media server.

In the third audio bridge, an audio stream (OPUS) of the first voice codec is transmitted between the web server and the media server, and in the fourth audio bridge, between the telecom server and the media server, the audio stream (OPUS) is transmitted. An audio stream (OPUS) of the first voice codec is transmitted, but an audio stream (G.711) of the second voice codec is transmitted between the telecom server and the telephone network.

In addition, according to an embodiment of the present invention, a computer readable recording medium storing a program for implementing the public switched telephone network communication and web real-time communication linked conference providing method is provided.

In addition, according to an embodiment of the present invention, a program stored in a computer readable recording medium is provided to implement the public switched telephone network communication and web real-time communication linked conference providing method.

According to a conference providing system and method linked to public switched telephone network communication and web real-time communication according to an embodiment of the present invention, participation in a conference through a public switched telephone network (PSTN) through an audio bridge connection to a corresponding audio bridge room There is an effect in which a plurality of users including at least one voice user and at least one user participating in a video conference through an Internet network can participate in a real-time conference (meeting) without limiting the number of people.

According to a conference providing system and method linked with public switched telephone network communication and web real-time communication according to an embodiment of the present invention, at least one voice user participating in a conference through a public switched telephone network (PSTN) and a video image through an Internet network are provided. There is an effect that a plurality of users including at least one user participating in a conference can receive a real-time conference environment regardless of their access paths.

1 is a configuration diagram of a public switched telephone network communication and web real-time communication linked conference providing system according to an embodiment of the present invention.
2 is a flowchart of a method for providing a conference linked to public switched telephone network communication and web real-time communication according to an embodiment of the present invention.
3 is a detailed flowchart of an embodiment of the first call processing step (S100) of FIG. 2;
4 is a detailed flowchart of an embodiment of the second call processing step (S200) of FIG. 2;

Since the present invention can make various changes and have various embodiments, specific embodiments will be illustrated in the drawings and described in detail. However, this is not intended to limit the present invention to specific embodiments, and should be understood to include all modifications, equivalents, or substitutes included in the spirit and scope of the present invention.

It is understood that when an element is referred to as being "connected" or "connected" to another element, it may be directly connected or connected to the other element, but other elements may exist in the middle. It should be.

On the other hand, when an element is referred to as “directly connected” or “directly connected” to another element, it should be understood that no other element exists in the middle.

Terms used in this specification are only used to describe specific embodiments, and are not intended to limit the present invention. Singular expressions include plural expressions unless the context clearly dictates otherwise. In this application, the terms "include" or "have" are intended to designate that there is a feature, number, process, operation, component, part, or combination thereof described in the specification, but one or more other features It should be understood that the presence or addition of numbers, processes, operations, components, parts, or combinations thereof is not precluded.

Unless defined otherwise, all terms used herein, including technical or scientific terms, have the same meaning as commonly understood by one of ordinary skill in the art to which the present invention belongs. Terms such as those defined in commonly used dictionaries should be interpreted as having meanings consistent with the meanings in the context of the related art, and unless explicitly defined in this application, they should not be interpreted in ideal or excessively formal meanings. don't

Hereinafter, the present invention will be described in more detail with reference to the accompanying drawings. Prior to this, the terms or words used in this specification and claims should not be construed as being limited to ordinary or dictionary meanings, and the inventor appropriately uses the concept of terms in order to explain his/her invention in the best way. Based on the principle that it can be defined, it should be interpreted as meaning and concept consistent with the technical spirit of the present invention. In addition, unless there is another definition in the technical terms and scientific terms used, they have meanings commonly understood by those of ordinary skill in the art to which this invention belongs, and the gist of the present invention is described in the following description and accompanying drawings. Descriptions of well-known functions and configurations that may be unnecessarily obscure are omitted. The drawings introduced below are provided as examples to sufficiently convey the spirit of the present invention to those skilled in the art. Accordingly, the present invention may be embodied in other forms without being limited to the drawings presented below. Also, like reference numerals denote like elements throughout the specification. It should be noted that like elements in the drawings are indicated by like numerals wherever possible.

The international standard communication technology of the Public Switched Telephone Network (PSTN) is SIP RFC 3261 (https://www.rfc-editor.org/rfc/rfc3261), and the international standard real-time communication technology of the Internet network is WebRTC RFC8825. , 8835, 8826, 8827, etc.

Hereinafter, in this specification, the public switched telephone network (PSTN) will be briefly referred to as a 'telephone network'.

However, since SIP (session initiation protocol) and WebRTC (web real-time communication) technologies have different signaling standards for call processing, a signaling conversion technology is required.

In addition, the media exchange standard technology commonly used by the telephone network and the Internet network is SDP (https://www.rfc-editor.org/rfc/rfc4566.html), but the telephone network provides only legacy SDP technology, Since it is not compatible with the WebRTC SDP that supports the specification, a media conversion technology is required to convert it.

In addition, the main voice codecs supported by the SIP protocol (telephone network) are G.711u and G.711a, and the main voice codecs supported by the WebRTC protocol (internet network) are Opus and G.711u. When transmitting, it must be converted into a codec that the peer can understand before transmitting.

Therefore, the public switched telephone network communication and web real-time communication-linked conference providing system according to the present invention relates to a signaling technology for interlocking call processing between a telephone network and an Internet network multi-party, and an audio bridge technology for multi-party voice connection between a telephone network and an Internet network. .

1 is a configuration diagram of a public switched telephone network communication and web real-time communication linked conference providing system according to an embodiment of the present invention.

As shown in FIG. 1, a public switched telephone network communication and web real-time communication linking conference providing system according to an embodiment of the present invention includes a first user terminal 100, a second user terminal 200, a web server ( 101), a load balance server 103, a plurality of media servers 108, and a telecom server 106.

The first user terminal 100 is a terminal participating in a conference through an internet network (web browser).

The second user terminal 200 is a terminal participating in the conference through a telephone network (PSTN).

The web server 101 provides a conference environment to the first user terminal 100 through a web browser.

In addition, the web server 101 separates the audio signal and the video signal received from the first user terminal 100 and transmits them to the load balance server 103 .

The reason why the web server 101 separates audio and video signals and transmits them to the load balance server 103 is to connect audio traffic with a separate audio bridge and distribute the traffic with optimal server resources. .

The load balance server 103 selects a media server according to an outgoing request from the first user terminal 100 or the second user terminal 200 and creates an audio bridge room in the corresponding media server. and requests resource allocation to the created audio bridge room.

The load balance server 103 selects an optimal media server in consideration of traffic in the current system.

The corresponding media server 108 creates the audio bridge room according to a request from the load balance server, connects a first audio bridge between the first user terminal and the audio bridge room, and connects the second user terminal and the audio bridge room. A second audio bridge between the audio bridge rooms is connected to connect the first user terminal and the second user terminal to the same audio bridge room.

The telecom server 106 converts a Web Real-Time Communication (WebRTC) protocol signal into a Session Initiation Protocol (SIP) signal, which is a communication standard of the telephone network, and transmits the signal to the telephone network 102 or , The SIP protocol signal received from the telephone network 102 is converted into a WebRTC protocol signal and transmitted to the corresponding media server.

In addition, the telecom server 106 includes Legacy Session Description Protocol (SDP), a media exchange technology supported by the telephone network 102, and WebRTC SDP (Web Real-Time Protocol), a media exchange technology supported by the Internet network. Time Communication session Description Protocol)), the WebRTC SDP signal is converted into a legacy SDP signal to enable media exchange between the telephone network and the Internet network, and the legacy SDP signal is transmitted to the telephone network 102 and received from the telephone network 102. is converted into a WebRCT SDP signal and transmitted to the load balance server 103 or the media server 108.

In addition, since the voice codec supported by the telephone network (SIP protocol) is different from the voice codec supported by the Internet network (WebRTC protocol), the telecom server 106 uses the other party's voice codec when receiving or transmitting a voice packet. Transmit and forward.

For example, when the voice codec supported by the telephone network 102 is G.711 and the voice codec supported by the Internet network is OPUS, the telecom server 106 transmits an audio stream to the telephone network 102. In the case of transmission, the voice codec must be converted from OPUS to G.711, and in the case of receiving the audio stream from the telephone network 102, the voice codec must be converted from G.711 to OPUS.

That is, in the first audio bridge, the audio stream (OPUS) of the first voice codec is transmitted between the web server 101 and the corresponding media server 108, and in the second audio bridge, the telecom 106 server The audio stream (OPUS) of the first voice codec is transmitted between the first voice codec and the media server 108, but the audio stream (G.711) of the second voice codec is transmitted between the telecom server 106 and the telephone network 102. do.

In addition, the public switched telephone network communication and web real-time communication linked conference providing system according to an embodiment of the present invention includes a billing server 104, an authentication server 105, a Redis server, an RDBMS server 107, and a recording It further includes a server 109.

The billing server 104 performs processing related to charges, the authentication server 105 performs processing related to user authentication, the RDBMS server 107 is a relational database management system, and the recording server 109 records and stores the contents of multilateral conferences.

The Redis server is located between each server, and is an open source-based non-relational database management system (DBMS) for storing and managing unstructured data with a key and value structure. , which is used as a message broker and is a store with in-memory data structures.

2 is a flowchart of a method for providing a conference linked to public switched telephone network communication and web real-time communication according to an embodiment of the present invention.

In the method for providing a conference linked to public switched telephone network communication and web real-time communication according to the present invention, first, a call is received from a first user terminal 100 using an Internet network to a second user terminal 200 using a telephone network. The call is processed (S100).

Meanwhile, an incoming call from a third user terminal using the telephone network to a fourth user terminal using the Internet network is processed (S200).

Thereafter, a conference between at least one user terminal using the telephone network connected according to the first call processing step (S100) or/and the second call processing step (S200) and at least one user terminal using the Internet network is in progress. Support as much as possible (S300).

In FIG. 2, it is exemplified that the first call processing step (S100) is performed first and the second call processing step (S200) is performed after the first call processing step (S100) is performed, but the present invention is limited thereto. It does not, and the second call processing step (S200) may be performed first, followed by the first call processing step (S100).

In addition, the first call processing step (S100) is performed alone, or the second call processing step (S200) is performed alone, or an additional user terminal at any time during the conference (meeting) after the conference (meeting) has already started ( It is also possible that the first call processing step (S100) or the second call processing step (S200) for the additional conference participant) is performed.

In addition, in the present invention, for convenience of description, it has been described that after two users (a first user terminal and a second user terminal) are connected, two additional users (a third user terminal and a fourth user terminal) are connected. Simultaneous access to a plurality of user terminals using the Internet network is also possible, and additionally, access to user terminals using the Internet network or user terminals using the telephone network is also possible.

3 is a detailed flowchart of an embodiment of the first call processing step S100 of FIG. 2 .

In the first call processing step (S100), first, in response to an outgoing call from the first user terminal 100 of the Internet network, the web server 101 establishes a Load Balance Server (LBS) 103 The first SDP request signal (WebRTC Offer SDP1) is transmitted to , and the audio signal and the video signal are separated and transmitted (S301).

Thereafter, the load balance server 103 selects a media server (MS: Media Server) 108 and requests the corresponding media server to create an audio bridge room (S302).

Thereafter, as the audio bridge room is created, the load balance server 103 transfers a first SDP response signal (WebRTC Answer SDP1) to the web server 101 (S303).

Thereafter, the first audio bridge between the web server 101 and the media server 108 is connected (S304), and the first user terminal 100 and the media server 108 via the web server 101 The audio stream between them is transmitted.

Thereafter, the load balance server 103 transfers a second SDP request signal (WebRTC Offer SDP2) requesting audio bridge (AB) resources for the called user of the telephone network 102 to the media server 108. Do (S305).

Thereafter, the load balance server 103 transmits the second SDP request signal (WebRTC Offer SDP2) requesting a phone connection to the telecom server 106 (S306).

Thereafter, the telecom server 106 converts the second SDP request signal (WebRTC Offer SDP2) into a legacy SDP signal and transmits a session initiation protocol request signal (SIP Offer) to the telephone network (S307).

Then, according to the incoming call response of the second user terminal 200, the telecom server 106 converts the session initiation protocol response signal (SIP Answer) received from the telephone network 102 into a WebRTC signal (S308).

Thereafter, the telecom server 106 transmits a second SDP response signal (WebRTC Answer SDP2) responding to the phone connection to the load balance server 103 (S309).

As the second SDP response signal (WebRTC Answer SDP2) is transmitted, the Internet network and the telephone network are connected (S310).

Thereafter, the media server 108 transmits the second SDP response signal (WebRTC Answer SDP2) to the load balance server 103 as a response to the audio bridge resource for the called user of the telephone network (S311).

Thereafter, a second audio bridge between the telecom server 106 and the media server 108 is connected (S312), and an audio stream between the second user terminal 200 and the media server 108 of the telephone network is transmitted. do.

In the first audio bridge, an audio stream (OPUS) of a first voice codec is transmitted between the web server 101 and the media server 108, and in the second audio bridge, the telecom server 106 and the The audio stream (OPUS) of the first voice codec is transmitted between the media servers 108, but the audio stream (G.711) of the second voice codec is transmitted between the telecom server 106 and the telephone network 102.

4 is a detailed flowchart of an embodiment of the second call processing step S200 of FIG. 2 .

In the second call processing step (S200), first, according to an outgoing call from the third user terminal of the telephone network 102, the telephone network 102 transmits a session initiation protocol request signal (SIP) to the telecom server 106. Offer), the telecom server 106 converts the session initiation protocol request signal (SIP Offer) received from the telephone network into a WebRTC signal (S401).

The telecom server 106 includes Legacy Session Description Protocol (SDP), a media exchange technology supported by the telephone network, and WebRTC SDP (Web Real-Time Communication session Description Protocol), a media exchange technology supported by the Internet network. )), the legacy SDP signal received from the telephone network is converted into a WebRCT SDP signal so that media exchange between the telephone network and the Internet network is possible.

Thereafter, the telecom server 106 transmits a third SDP request signal (WebRTC Offer SDP3) requesting a phone connection to the load balance server 103 (S402).

Then, the load balance server 103 requests the media server 108 to create an audio bridge room (S403).

As the audio bridge room is created, a fourth SDP request signal (WebRTC Offer SDP4) requesting audio bridge resources for the Internet network called user is transmitted between the load balance server 103 and the media server 108 ( S404).

Thereafter, the load balance server 103 transmits the fourth SDP request signal (WebRTC Offer SDP4) to the web server 101 (S405).

According to the incoming call response of the fourth user terminal, the web server 101 transfers a fourth SDP response signal (WebRTC Answer SDP4) to the load balance server 103 (S406).

Thereafter, the media server 108 transmits the fourth SDP response signal (WebRTC Answer SDP4) to the load balance server 103 as a response to the audio bridge resource for the Internet network called user (S407).

Thereafter, a third audio bridge between the web server 101 and the media server is connected (S408), and an audio stream between the fourth user terminal and the media server 108 is transmitted through the web server 101. .

Thereafter, the load balance server 103 transmits a third SDP response signal (WebRTC Answer SDP3) responding to the phone connection to the telecom server 106 (S409).

Thereafter, as the telecom server 106 receives the third SDP response signal (WebRTC Answer SDP3), it converts the WebRTC signal into a SIP signal and transmits the session initiation protocol response signal (SIP Answer) to the telephone network 102. Do (S410).

In addition, the telecom server 106 converts the WebRTC SDP signal into a legacy SDP signal and transmits it to the telephone network 102.

Thereafter, a fourth audio bridge between the telecom server 106 and the media server 108 is connected (S411), and an audio stream is transmitted between the third user terminal of the telephone network and the media server 108.

In the third audio bridge, the audio stream (OPUS) of the first voice codec is transmitted between the web server 101 and the media server 108, and in the fourth audio bridge, the telecom server 106 and The audio stream (OPUS) of the first voice codec is transmitted between the media servers 108, but the audio stream (G.711) of the second voice codec is transmitted between the telecom server 108 and the telephone network 102. is sent

In FIGS. 3 and 4, one media server is taken as an example, but the present invention is not limited thereto. It is possible to have a conference without limiting access path.

Although the public switched telephone network communication and web real-time communication-linked conference providing method according to an embodiment of the present invention has been described above, the computer readable program storing the program for implementing the public switched telephone network communication and web real-time communication-linked conference providing method has been described. Of course, a program stored in a computer-readable recording medium for realizing a possible recording medium and a method of providing a conference linked to public switched telephone network communication and web real-time communication can be implemented.

That is, those skilled in the art can realize that the above-described public switched telephone network communication and web real-time communication-linked conference providing method may be provided by being included in a computer-readable recording medium by tangibly implementing a program of instructions for implementing the same. You will understand easily. In other words, it may be implemented in the form of program instructions that can be executed through various computer means and recorded on a computer-readable recording medium. The computer readable recording medium may include program instructions, data files, data structures, etc. alone or in combination. Program instructions recorded on the computer-readable recording medium may be specially designed and configured for the present invention, or may be known and usable to those skilled in computer software. Examples of the computer-readable recording medium include magnetic media such as hard disks, floppy disks and magnetic tapes, optical media such as CD-ROMs and DVDs, and floptical disks. Included are hardware devices specially configured to store and execute program instructions, such as magneto-optical media, and ROM, RAM, flash memory, USB memory, and the like. Examples of program instructions include high-level language codes that can be executed by a computer using an interpreter, as well as machine language codes such as those produced by a compiler. The hardware device may be configured to act as one or more software modules to perform the operations of the present invention and vice versa.

The present invention is not limited to the above embodiments, and the scope of application is diverse, and various modifications and implementations are possible without departing from the gist of the present invention claimed in the claims.

100: first user terminal
200: second user terminal
101: web server
102: telephone network (PSTN)
103: load balance server
104: billing server
105: authentication server
106: telecom server
107: Relational database management system (RDBMS) server
108: media server
109: recording server
S100: First call processing step
S200: Second call processing step
S300: Conference support step

Claims (10)

Public switched telephone network communication and web real-time communication In the interlocking conference providing system,
at least one first user terminal 100 participating in a conference through an Internet network;
at least one second user terminal 200 participating in the conference through a telephone network;
a web server 101 for providing a conference environment to the first user terminal through a web browser;
In response to an outgoing request from the first user terminal or the second user terminal, a media server is selected in consideration of traffic in the conference providing system, an audio bridge room creation request is requested to the media server, and the audio bridge room is created. Load balance server 103 for requesting resource allocation to;
The audio bridge room is created according to a request from the load balance server, a first audio bridge between the first user terminal and the audio bridge room is connected, and a second audio bridge between the second user terminal and the audio bridge room is created. a plurality of media servers 108 for connecting the first user terminal and the second user terminal to the same audio bridge room by connecting bridges; and
A Web Real-Time Communication (WebRTC) protocol signal is converted into a Session Initiation Protocol (SIP) signal, which is the communication standard of the telephone network, and transmitted to the telephone network, or the SIP protocol signal received from the telephone network is converted into a Session Initiation Protocol (SIP) signal. Telecom server 106 for converting into a WebRTC protocol signal and forwarding it to the corresponding media server
including,
The web server 101,
Characterized in that the audio signal and the video signal received from the first user terminal are separated and transmitted to the load balance server,
In addition to the first user terminal and the second user terminal, it is possible to access a user terminal using an internet network or a user terminal using a telephone network additionally, and even if the media server to which each user terminal accesses is different, it is connected to the audio bridge room Characterized in that a plurality of users are provided with a conference environment through each audio bridge,
billing server 104 for performing processing related to charges;
authentication server 105 for performing processing related to user authentication;
RDBMS server 107, which is a relational database management system;
a recording server 109 for recording and storing multi-party conference contents; and
Redis Server, an open source-based non-relational database management system for storing and managing unstructured data with key and value structures located between each server.
Public switched telephone network communication and web real-time communication further comprising Linked conference provision system.
delete According to claim 1,
The telecom server 106,
Regarding Legacy Session Description Protocol (SDP), a media exchange technology supported by the telephone network, and WebRTC SDP (Web Real-Time Communication session Description Protocol), a media exchange technology supported by the Internet network, the telephone network and the Internet Public switched telephone network communication and web real-time communication characterized by converting WebRTC SDP signals into legacy SDP signals and transmitting them to the telephone network to enable media exchange between networks, and converting legacy SDP signals received from the telephone network into WebRCT SDP signals Linked conference provision system.
According to claim 3,
The telecom server 106,
Since the voice codec supported by the telephone network (SIP protocol) and the voice codec supported by the Internet network (WebRTC protocol) are different, when a voice packet is received or transmitted, it is converted to the other party's voice codec and then transmitted. Telephone network communication and web real-time communication linked conference provision system.
According to claim 4,
In the first audio bridge, an audio stream (OPUS) of a first voice codec is transmitted between the web server and the corresponding media server,
In the second audio bridge, an audio stream (OPUS) of the first voice codec is transmitted between the telecom server and the media server,
A conference providing system linked to public switched telephone network communication and web real-time communication, characterized in that an audio stream (G.711) of a second voice codec is transmitted between the telecom server and the telephone network.
A method for providing a public switched telephone network communication and web real-time communication-linked conference,
A first call processing step (S100) of processing an incoming call from a first user terminal using the Internet network to a second user terminal using the telephone network;
A second call processing step (S200) of processing an incoming call from a third user terminal using a telephone network to a fourth user terminal using an Internet network; and
A conference support step of supporting a conference between at least one user terminal using the telephone network connected to the first call processing step or/and the second call processing step and at least one user terminal using the Internet network (S300). )
including,
In the second call processing step (S200),
When the telephone network transmits a session initiation protocol request signal (SIP Offer) to the telecom server according to an outgoing call from the third user terminal of the telephone network, the telecom server receives the session initiation protocol request signal (SIP Offer) from the telephone network. ) into a WebRTC signal (S401);
The telecom server transmitting a third SDP request signal (WebRTC Offer SDP3) requesting a phone connection to the load balance server (S402);
requesting, by the load balance server, the media server to create an audio bridge room (S403);
When the audio bridge room is created, transmitting a fourth SDP request signal (WebRTC Offer SDP4) requesting an audio bridge resource for an Internet network called user between the load balance server and the media server (S404);
The load balance server transmitting the fourth SDP request signal (WebRTC Offer SDP4) to the web server (S405);
Transmitting, by the web server, a fourth SDP response signal (WebRTC Answer SDP4) to the load balance server according to the incoming call response of the fourth user terminal (S406);
transmitting, by the media server, the fourth SDP response signal (WebRTC Answer SDP4) to the load balance server as a response to an audio bridge resource for the Internet network called user (S407);
connecting a third audio bridge between the web server and the media server (S408) and transmitting an audio stream between the fourth user terminal and the media server through the web server;
The load balance server transmitting a third SDP response signal (WebRTC Answer SDP3) responding to the phone connection to the telecom server (S409);
When the telecom server receives the third SDP response signal (WebRTC Answer SDP3), converting the WebRTC signal into a SIP signal and transmitting a session initiation protocol response signal (SIP Answer) to the telephone network (S410); and
connecting a fourth audio bridge between the telecom server and the media server (S411) and transmitting an audio stream between the third user terminal and the media server of the telephone network;
It is characterized by including,
In addition to the first to fourth user terminals, it is possible to access a user terminal using an internet network or a user terminal using a telephone network in addition to the first to fourth user terminals, and even if the media server to which each user terminal accesses is different, it is connected to the audio bridge room A conference providing method linked to public switched telephone network communication and web real-time communication, characterized in that a plurality of users are provided with a conference environment through each audio bridge.
According to claim 6,
In the first call processing step (S100),
In response to a call from the first user terminal of the Internet network, the web server forwarding a first SDP request signal (WebRTC Offer SDP1) to the load balance server, and separating and transmitting an audio signal and a video signal ( S301);
The load balance server selects a media server and requests the corresponding media server to create an audio bridge room (S302);
When the audio bridge room is created, the load balance server transmits a first SDP response signal (WebRTC Answer SDP1) to the web server (S303);
connecting a first audio bridge between the web server and the media server (S304) and transmitting an audio stream between the first user terminal and the media server through the web server;
Transmitting, by the load balance server, a second SDP request signal (WebRTC Offer SDP2) requesting an audio bridge resource for a called user of the telephone network to the media server (S305);
The load balance server transmitting the second SDP request signal (WebRTC Offer SDP2) requesting a phone connection to the telecom server (S306);
The telecom server converts the second SDP request signal (WebRTC Offer SDP2) into a SIP signal and transmits a session initiation protocol request signal (SIP Offer) to the telephone network (S307);
Converting, by the telecom server, a session initiation protocol response signal (SIP Answer) received from the telephone network into a WebRTC signal according to the incoming call response of the second user terminal (S308);
Transmitting, by the telecom server, a second SDP response signal (WebRTC Answer SDP2) responding to a phone connection to the load balance server (S309);
Connecting the Internet network and the telephone network as the second SDP response signal (WebRTC Answer SDP2) is transmitted (S310);
transmitting, by the media server, the second SDP response signal (WebRTC Answer SDP2) to the load balance server as a response to an audio bridge resource for the called user of the telephone network (S311); and
connecting a second audio bridge between the telecom server and the media server (S312), and transmitting an audio stream between the second user terminal of the telephone network and the media server;
A method for providing a public switched telephone network communication and web real-time communication-linked conference, comprising:
According to claim 7,
In the first audio bridge, an audio stream (OPUS) of a first voice codec is transmitted between the web server and the media server,
In the second audio bridge, an audio stream (OPUS) of the first voice codec is transmitted between the telecom server and the media server,
A conference providing method linked to public switched telephone network communication and web real-time communication, characterized in that an audio stream (G.711) of a second voice codec is transmitted between the telecom server and the telephone network.
delete According to claim 8,
In the third audio bridge, an audio stream (OPUS) of the first voice codec is transmitted between the web server and the media server,
In the fourth audio bridge, an audio stream (OPUS) of the first voice codec is transmitted between the telecom server and the media server,
711 of the second voice codec is transmitted between the telecom server and the telephone network.

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