CN112822169B - Integration system and method suitable for seat Web telephone application system - Google Patents

Integration system and method suitable for seat Web telephone application system Download PDF

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Publication number
CN112822169B
CN112822169B CN202011616732.5A CN202011616732A CN112822169B CN 112822169 B CN112822169 B CN 112822169B CN 202011616732 A CN202011616732 A CN 202011616732A CN 112822169 B CN112822169 B CN 112822169B
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browser
seat
voice
media stream
media
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CN112822169A (en
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李虎
曾毅峰
王之良
刘书题
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Shanghai Pudong Development Bank Co Ltd
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Shanghai Pudong Development Bank Co Ltd
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066—Session management
    • H04L65/1069—Session establishment or de-establishment
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066—Session management
    • H04L65/1101—Session protocols
    • H04L65/1104—Session initiation protocol [SIP]
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/60—Network streaming of media packets
    • H04L65/75—Media network packet handling
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L67/00—Network arrangements or protocols for supporting network services or applications
    • H04L67/01—Protocols
    • H04L67/02—Protocols based on web technology, e.g. hypertext transfer protocol [HTTP]
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L69/00—Network arrangements, protocols or services independent of the application payload and not provided for in the other groups of this subclass
    • H04L69/16—Implementation or adaptation of Internet protocol [IP], of transmission control protocol [TCP] or of user datagram protocol [UDP]
    • H04L69/161—Implementation details of TCP/IP or UDP/IP stack architecture; Specification of modified or new header fields
    • H04L69/162—Implementation details of TCP/IP or UDP/IP stack architecture; Specification of modified or new header fields involving adaptations of sockets based mechanisms

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  • Engineering & Computer Science (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Signal Processing (AREA)
  • Multimedia (AREA)
  • Business, Economics & Management (AREA)
  • General Business, Economics & Management (AREA)
  • Computer Security & Cryptography (AREA)
  • Telephonic Communication Services (AREA)

Abstract

The invention relates to an integrated system and method suitable for a seat Web telephone application system, which is realized by a frame consisting of a seat browser, a message server, a WebRTC gateway and a voice soft switch platform, wherein the WebRTC gateway is used as an intermediate transmission medium of an SIP soft phone and the voice soft switch platform and is responsible for mutually converting a media stream based on Webscolet and the SIP media stream; the message server acquires SIP signaling of the voice soft switching platform, analyzes the signaling in the background, converts the signaling into a message event, and sends the message event to the seat browser through the websocket to realize telephone operation; and the seat browser is used for creating two web browser connection objects when a web phone page is opened, and the two connection objects are respectively connected with the message server and the WebRTC gateway. Compared with the prior art, the method has the advantages of saving labor and time cost, stronger compatibility, wider application range and the like.

Description

Integration system and method suitable for seat Web telephone application system
Technical Field
The invention relates to the technical field of communication, in particular to an integration system and method suitable for a position Web telephone application system.
Background
At present, most call center softphone application systems realize transmission of voice stream data with a voice platform by installing SIP client software (such as SIP client, xlite and the like) on a PC-end computer, interact with a background signaling system, and maintain relationship binding between desktop end software and seat personnel of the call center softphone application system. However, the implementation of the conventional call center softphone application system has the following disadvantages: (1) A PC terminal computer of a seat needs to be provided with desktop SIP telephone software, the desktop SIP telephone software is separated from a call center soft phone application system, and a browser of the seat needs to realize signaling interaction with a voice platform by calling a plug-in; (2) SIP client software of different manufacturers is only adapted to the voice platform of the user, and the compatibility to other voice platforms is poor; (3) the traditional soft phone is not applicable at the mobile end; (4) The seat can dial the number through the SIP client software, but the operation of the seat on the SIP client cannot be set.
Disclosure of Invention
The invention aims to provide an integration system and method suitable for a seat Web telephone application system, which at least partially solve the problems in the prior art and have the advantages of labor and time cost saving, stronger compatibility, wider application range and the like.
The purpose of the invention can be realized by the following technical scheme:
an integrated system for a seat Web phone application, the system comprising:
a voice soft switch platform;
the WebRTC gateway is used for providing a transmission medium between the SIP soft phone and the voice soft switch platform and converting the media stream based on Webscolet and the SIP media stream into each other;
the message server is connected with the voice soft switch platform and used for acquiring the SIP signaling of the voice soft switch platform, analyzing the signaling at the background and converting the signaling into a message event;
and the seat browser is used for creating two webscoket connection objects when a web phone page is opened, and the two webscoket connection objects are respectively connected with the message server and the WebRTC gateway. The agent side browser adopts a browser supporting WebRTC.
An integration method suitable for a position Web telephone application system relates to a position browser, a message server, a WebRTC gateway and a voice soft switch platform, and comprises the following steps:
s1, starting a seat browser, opening a web phone application system to execute phone operation, and displaying different phone states on an event message sent by a message server.
And S2, establishing a media stream connection between the seat browser and the WebRTC gateway, wherein the media stream connection is used for carrying out voice media stream transmission between the seat browser and the WebRTC gateway by the seat end and the client.
Further, the seat browser establishes a media stream connection with the WebRTC gateway based on a webscolet connection with two-way transmission of json-format telephony event messages. The WebRTC gateway and the voice soft switch platform establish the connection and transmission of media streams through the SIP protocol.
And S3, opening a seat browser of the web phone application system to establish connection with a message server, establishing signaling interaction between the message server and the voice soft switch platform, and performing signaling conversion on the operation sent by the seat browser of the web phone application system and then sending the operation to the voice soft switch platform by the message server.
The message server establishes signaling interaction of an SIP protocol with the voice soft switch platform, converts operation sent by the Web telephone of the seat end into an SIP signaling and sends the SIP signaling to the voice soft switch platform.
And the message server receives SIP signaling returned by the voice soft switch platform and converts the SIP signaling into a general event message in a json format, and the message server pushes the event message in the json format to the seat end browser through Websocket connection.
And S4, the seat browser analyzes the received media stream information pushed by the WebRTC gateway, converts voice media data into voice streams and outputs the voice streams to local audio output equipment, and simultaneously converts media stream data collected by the local media input equipment into media streams and sends the media streams to the WebRTC gateway.
And the seat browser converts the media stream data collected by the local media input equipment into the media stream of the Websocket and sends the media stream of the WebRTC gateway. Furthermore, the seat browser calls a MediaStream interface of the WebRTC API to analyze and receive media stream information pushed by the WebRTC gateway, converts voice media data into voice streams and outputs the voice streams to the local audio output device, and simultaneously converts media stream data collected by the local media input device into Websocket media streams through the MediaStream interface and sends the WebRTC media streams to the WebRTC gateway.
And S5, displaying the current state of the telephone in real time by the Web telephone page of the seat browser according to the message event pushed by the message server.
Compared with the prior art, the integration system and the method for the seat Web telephone application system at least have the following beneficial effects:
1) According to the invention, the function of making a call can be realized through the Web page without installing client software and browser plug-ins, so that the labor and time cost are saved;
2) The method can be applied to a soft phone application system operated in a browser of a plurality of terminals (including a PC terminal and a PAD terminal, but not limited to the terminal system), and has stronger compatibility and wide application range;
3) The invention integrates the seat telephone and the service application system, the seat can make and receive calls through the browser, the browser only needs to process complex media streams, does not need to consume more memory to process SIP signaling, reduces the memory utilization rate of the browser at the seat end, and improves the running reliability of the telephone application system at the browser at the seat end.
4) The agent browser comprises and is not limited to all browsers supporting WebRTC;
5) The Web telephone can be used for self-defining development according to the service requirement, and further, the operation of setting a seat on the SIP client side can be realized.
Drawings
FIG. 1 is a schematic diagram of a framework of an integrated system suitable for a seat Web phone application system in an embodiment;
fig. 2 is a schematic flowchart of an integration method of a suitable location Web phone application system in the embodiment.
The reference numbers in the figures indicate:
1. a seat browser 2, a message server 3, a WebRTC gateway 4 and a voice soft switch platform.
Detailed Description
The invention is described in detail below with reference to the figures and specific embodiments. It should be apparent that the described embodiments are only some of the embodiments of the present invention, and not all of the embodiments. All other embodiments, which can be obtained by a person skilled in the art without any inventive step based on the embodiments of the present invention, shall fall within the scope of protection of the present invention.
Examples
The invention relates to an integrated system suitable for a seat Web telephone application system, which is realized by a frame consisting of a seat browser, a message server, a WebRTC gateway and a voice soft switch platform, wherein the frame is shown in figure 1. Wherein:
1) WebRTC gateway and voice soft switch platform
The invention takes the WebRTC gateway as an intermediate transmission medium of the SIP soft phone and the voice soft switch platform and is mainly responsible for converting the Webscolet-based media stream and the SIP media stream into each other.
The voice soft switch platform of the embodiment may adopt currently mainstream voice soft switch platforms such as Freeswitch and Asterisk, and is not limited to the above voice soft switch platforms.
2) Message server
And the system is connected with the voice soft switching platform and used for acquiring the SIP signaling of the voice soft switching platform, analyzing the signaling at the background, converting the signaling into a message event, and sending the message event to the seat browser through the websocket to realize telephone operations such as registration, ringing, calling, consultation, transfer, conference and the like.
3) Seat browser
The method is used for creating two webscoket connecting objects when a web phone page is opened, and the two webscoket connecting objects are respectively connected with a message server and a WebRTC gateway.
Based on the framework with the structure, the invention also provides an integration method suitable for the seat Web telephone application system, which specifically comprises the following steps:
s1: and starting the seat browser, opening a web phone application system (namely a phone call interface provided by the voice platform externally), performing phone operations such as sign-on, dialing, answering, consulting, transferring and the like, and simultaneously displaying different phone states (for example, the page of the seat browser can be customized and developed according to business needs in readiness, conversation, summary, consultation and the like) according to the event message sent by the message server.
In this embodiment, a Chrome browser may be preferably used as the seat browser.
S2: and the seat browser establishes a media stream connection based on Webscolet connection with the WebRTC gateway, and is used for transmitting the voice media stream between the seat browser and the WebRTC gateway for the seat and the client. Meanwhile, the WebRTC gateway and the voice soft switch platform establish the connection and transmission of media streams through an SIP protocol; and establishing a common Webscolet connection between the agent browser and the message server, and using the Webscolet connection for bidirectional transmission of the json-format telephone event messages.
S3: the method comprises the steps that a Web telephone (a Web telephone operation executed by a Web telephone application system opened by a seat browser) at a seat end is connected with a message server, the message server simultaneously establishes signaling interaction of an SIP protocol with a voice soft switch platform, the message server converts the operation sent by the Web telephone at the seat end into an SIP signaling and sends the SIP signaling to the voice soft switch platform, the SIP signaling returned by the voice soft switch platform is received and converted into a general event message in a json format, and the message server pushes the event message in the json format to the browser at the seat end through Websocket connection.
S4: the seat browser calls MediaStream of the WebRTC API to analyze and receive media stream information pushed by the WebRTC gateway, voice media data are converted into voice streams to be output to local audio output equipment, meanwhile, the MediaStream converts media stream data collected by the local media input equipment into media streams of Websocket to be sent to the WebRTC gateway, and at the moment, a call is established between the seat and a client, and the sound of the seat and the voice of the client can be heard.
S5: the Web telephone page of the seat browser can show the current state of the telephone in real time according to the message event pushed by the message server, if the client hangs up the telephone, the message server pushes a message of which the on-hook party is the client to the seat browser, and the seat browser updates the state of the Web telephone from the conversation to the conversation end after receiving the message.
According to the invention, the function of making a call can be realized through the Web page without installing client software and a browser plug-in, so that the labor and time cost are saved; the soft phone application system can be operated in a browser of multiple terminals (including a PC terminal and a PAD terminal, but not limited to the terminal system), has stronger compatibility and wide application range; the invention integrates the seat telephone and the service application system, the seat can make and receive calls through the browser, the browser only needs to process complex media streams, does not need to consume more memory to process SIP signaling, reduces the memory utilization rate of the browser at the seat end, and improves the running reliability of the telephone application system at the browser at the seat end.
While the invention has been described with reference to specific embodiments, it will be understood by those skilled in the art that various changes in form and details may be made therein without departing from the spirit and scope of the invention. Therefore, the protection scope of the present invention shall be subject to the protection scope of the claims.

Claims (5)

1. An integration method suitable for a seat Web telephone application system is characterized by comprising the following steps:
a voice soft switch platform;
the WebRTC gateway is used for providing a transmission medium between the SIP soft phone and the voice soft switch platform and mutually converting the media stream based on the Webscolet and the SIP media stream;
the message server is connected with the voice soft switch platform and used for acquiring the SIP signaling of the voice soft switch platform, analyzing the signaling at the background and converting the signaling into a message event;
the seat browser is used for creating two webscolet connection objects when a web phone page is opened, and the two webscolet connection objects are respectively connected with the message server and the WebRTC gateway;
the method comprises the following implementation steps:
1) Starting a seat browser, opening a web phone application system to execute phone operation, and displaying different phone states on an event message sent by a message server;
2) The seat browser and the WebRTC gateway establish media stream connection for performing voice media stream transmission between the seat browser and the WebRTC gateway by a seat end and a client; the seat browser and the WebRTC gateway establish media stream connection based on Webscolet connection for bidirectional transmission of json format telephone event messages;
3) The method comprises the steps that a seat browser of a web phone application system is opened to establish connection with a message server, meanwhile, the message server establishes signaling interaction with a voice soft switch platform, and the message server performs signaling conversion on operation sent by the seat browser of the web phone application system and then sends the operation to the voice soft switch platform; the message server establishes signaling interaction of an SIP protocol with the voice soft switch platform, converts the operation sent by the Web telephone of the seat end into an SIP signaling and sends the SIP signaling to the voice soft switch platform; the message server receives SIP signaling returned by the voice soft switch platform and converts the SIP signaling into a general event message in a json format, and the message server pushes the event message in the json format to a seat end browser through Websocket connection;
4) The agent browser analyzes the received media stream information pushed by the WebRTC gateway, converts voice media data into voice streams and outputs the voice streams to local audio output equipment, and simultaneously converts media stream data collected by the local media input equipment into media streams and sends the media streams to the WebRTC gateway; the agent browser converts media stream data collected by local media input equipment into media streams of Websocket and sends the media streams of WebRTC gateway;
5) And the Web telephone page of the seat browser displays the current state of the telephone in real time according to the message event pushed by the message server.
2. The method of claim 1, wherein the agent-side browser is a WebRTC-enabled browser.
3. The method of claim 1, wherein the WebRTC gateway and the voice softswitch platform establish connection and transmission of media streams via SIP protocol.
4. The integration method of the application system of the Web phone with the seat as claimed in claim 1, wherein in step 4), the seat browser calls the WebRTC API interface to analyze and receive the media stream information pushed by the WebRTC gateway, and simultaneously, the WebRTC API interface is called to convert the media stream data collected by the local media input device into the media stream of the Websocket and send the media stream to the WebRTC gateway.
5. The integration method of the application system of the Web phone with the agent as claimed in claim 1, wherein the agent browser calls a MediaStream interface of the WebRTC API to parse and receive the media stream information pushed by the WebRTC gateway, convert the voice media data into a voice stream and output the voice stream to the local audio output device, and simultaneously convert the media stream data collected by the local media input device into a media stream of the Websocket through the MediaStream interface and send the media stream of the WebRTC gateway.
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CN113923306B (en) * 2021-10-13 2023-10-20 上海淇馥信息技术有限公司 Internet online voice communication method and device and electronic equipment
CN114070939A (en) * 2021-12-28 2022-02-18 宝东信息技术有限公司 Network voice call method, system, storage medium and server

Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN103001814A (en) * 2011-09-09 2013-03-27 湖南神州祥网科技有限公司 Method for describing network flow characteristic statistics
CN110417805A (en) * 2019-08-07 2019-11-05 上海浦东发展银行股份有限公司信用卡中心 The seat soft telephone system and method for more voice integrated platform based on WebSocket agreement
CN111083295A (en) * 2019-12-31 2020-04-28 北京讯鸟软件有限公司 Telephone communication method and equipment based on browser

Family Cites Families (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN111885272B (en) * 2020-07-24 2021-11-16 南京易米云通网络科技有限公司 Intelligent call-out method for supporting telephone by call center seat and intelligent call center system

Patent Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN103001814A (en) * 2011-09-09 2013-03-27 湖南神州祥网科技有限公司 Method for describing network flow characteristic statistics
CN110417805A (en) * 2019-08-07 2019-11-05 上海浦东发展银行股份有限公司信用卡中心 The seat soft telephone system and method for more voice integrated platform based on WebSocket agreement
CN111083295A (en) * 2019-12-31 2020-04-28 北京讯鸟软件有限公司 Telephone communication method and equipment based on browser

Non-Patent Citations (1)

* Cited by examiner, † Cited by third party
Title
一种高效的网络流记录表示方法;张广兴等;《计算机研究与发展》;20130415(第04期);全文 *

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