CN106850399B - Communication method based on WebRTC technology instant message - Google Patents
Communication method based on WebRTC technology instant message Download PDFInfo
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- CN106850399B CN106850399B CN201611262789.3A CN201611262789A CN106850399B CN 106850399 B CN106850399 B CN 106850399B CN 201611262789 A CN201611262789 A CN 201611262789A CN 106850399 B CN106850399 B CN 106850399B
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L51/00—User-to-user messaging in packet-switching networks, transmitted according to store-and-forward or real-time protocols, e.g. e-mail
- H04L51/04—Real-time or near real-time messaging, e.g. instant messaging [IM]
- H04L51/046—Interoperability with other network applications or services
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L63/00—Network architectures or network communication protocols for network security
- H04L63/04—Network architectures or network communication protocols for network security for providing a confidential data exchange among entities communicating through data packet networks
- H04L63/0428—Network architectures or network communication protocols for network security for providing a confidential data exchange among entities communicating through data packet networks wherein the data content is protected, e.g. by encrypting or encapsulating the payload
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/1066—Session management
- H04L65/1101—Session protocols
- H04L65/1104—Session initiation protocol [SIP]
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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
- H04L65/60—Network streaming of media packets
- H04L65/65—Network streaming protocols, e.g. real-time transport protocol [RTP] or real-time control protocol [RTCP]
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- 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]
Abstract
The invention mainly discloses a communication method based on WebRTC technology instant message, which can realize instant message communication between a WebRTC terminal and an SIP terminal, and mainly comprises the following steps: the WebRTC terminal collects an ICE (Internet protocol chip) address and sends an offer SDP (service provider protocol) carrying an ICE candidate address to the server; the server analyzes the SDP, when the requested party is found to be the SIP terminal, the server starts a transfer service function, generates an answer SDP carrying an ICE candidate address to respond to the request of the WebRTC terminal, and generates a new offer SDP to the SIP terminal; the WebRTC terminal establishes connection of a DataChannel channel with a server; the WebRTC terminal sends the instant message to the SIP terminal through the SIP Session mechanism. By the scheme, gapless communication between the WebRTC terminal and the SIP terminal is achieved, communication cost is reduced, diversified communication modes between different types of terminals are increased, and communication efficiency of enterprise users is improved.
Description
Technical Field
The invention relates to the field of computer communication, in particular to a method and a system for realizing instant message IM based on a WebRTC (Web real-time communication) DataChannel technology.
Background
WebRTC (Web Real-Time Communication) is a Real-Time multimedia Communication technology based on a browser. The technology aims to enable the Web browser to have real-time communication capacity, and meanwhile, the capacity is packaged and opened to Web application developers in a JavaScript API mode, so that the Web application developers can rapidly develop real-time audio and video applications based on the Web browser through HTML tags and the JavaScript API without depending on any third party plug-in. A terminal capable of WebRTC communication as referred to herein is referred to as a WebRTC terminal. Therefore, the WebRTC technology enables direct Web communication between browsers supporting the WebRTC function, thereby changing the original Client/Server working mode of a terminal browser and a Server, enabling the Web to carry out end-to-end communication and being a great innovation for the Web technology. The WebRTC terminal adopts stream Control Transmission protocol sctp (stream Control Transmission protocol) as a data Transmission protocol, and the protocol can perform flow Control and congestion Control on Transmission data.
The technology creates a Web browser into a universal platform which is suitable for real-time audio and video and data communication between all user equipment, such as a mobile intelligent terminal and a personal computer. In the description of the present invention, we focus on the data communication part only.
An IP Multimedia Subsystem (IMS) is a network architecture supporting Multimedia services based on an IP network, and performs end-to-end call control using a Session Initiation Protocol (SIP), and a terminal conforming to the SIP is called an SIP terminal. The SIP terminal may perform instant messaging by the SIP Message method.
In a VoIP system, many SIP terminals have been deployed, and a newly added WebRTC terminal can perform voice and video communication with the SIP terminal, and cannot implement instant message delivery. A transfer server is needed between the two networks to realize the conversion of data carrying protocols of the two networks and the encryption/decryption of data, and the like, thereby realizing the intercommunication of instant messages of the WebRTC terminal and the SIP terminal.
The instant message transmission between two WebRTC terminals needs to be completed by a signaling server. The signaling server provides a bridge for establishing the DataChannel between the two WebRTCs, the two WebRTCs interact SDP information through the signaling server, finally, the two WebRTCs establish end-to-end DataChannel connection, and the two WebRTCs can mutually send reliable and encrypted instant messages through the DataChannel channel.
Disclosure of Invention
The invention aims to realize a communication method based on the WebRTC technology instant message, which can integrate a signaling server and a transfer server into a whole and provides a method and a system for a WebRTC terminal and an SIP terminal to send the instant message.
In order to achieve the above object, the technical solution of the present invention is as follows:
a communication method based on instant message of WebRTC technique can realize instant message communication between WebRTC terminal and SIP terminal, the communication method mainly includes the following steps: the WebRTC terminal collects an ICE (Internet protocol chip) address and sends an offer SDP (service provider protocol) carrying an ICE candidate address to the server; the server analyzes the SDP, when the requested party is found to be the SIP terminal, the server starts a transfer service function, generates an answer SDP carrying an ICE candidate address to respond to the request of the WebRTC terminal, and generates a new offer SDP to the SIP terminal; the WebRTC terminal establishes connection of a DataChannel channel with a server; the WebRTC terminal sends the instant message to the SIP terminal through the SIP Session mechanism.
Preferably, the communication method further includes: and ICE verification is required between the WebRTC terminal and the server to verify the connectivity of the ip and the port.
Preferably, in the above communication method, the server decrypts the SCTP-encrypted message, and then sends the instant message to the SIP terminal through an SIP Session mechanism.
Preferably, the communication method further includes: the WebRTC terminal and the SIP terminal are both registered to the server.
Preferably, in the process of implementing the communication method, the SIP terminal only needs to register to the server, and can work in a non-call-on state or a call-on state.
Preferably, in the above communication method, the server encrypts an unencrypted SIP message and then sends the encrypted SIP message to the WebRTC terminal through the DataChannel.
Preferably, the communication method further includes: the SIP terminal and the server establish a voice connection path.
Preferably, the communication method further includes: the RTP data packet of the SIP terminal is analyzed and processed in the server, the voice of the SIP terminal is converted into text information through the voice recognition module, and finally the instant text information is sent to the WebRTC terminal through the DataChannel channel.
Preferably, the communication method further includes: the WebRTC terminal sends the text information to the server through the SCTP protocol, and the server decrypts the SCTP encrypted message.
Preferably, the communication method further includes: the server converts the text information into voice data through the voice recognition module, and then sends the voice data to the SIP terminal through the voice connection channel.
In addition, the invention also provides a communication method based on the WebRTC technology instant message, which can realize that an SIP terminal is used as a request direction to establish communication with the WebRTC terminal, and is characterized in that the communication method mainly comprises the following steps:
the SIP terminal dials a call to the WebRTC terminal, and the SIP terminal firstly sends a request signaling to a server; the server will collect ICE address, and refill the offer SDP carrying the ICE candidate address and send to the WebRTC terminal; after receiving the offer, the WebRTC terminal generates an answer SDP carrying an ICE candidate address to respond to the server; establishing a DataChannel channel between the server and the WebRTC terminal, and establishing a common SIP session between the SIP terminal and the server; the server analyzes and processes the RTP data packet of the SIP terminal, converts the voice of the SIP terminal into text information through a voice recognition module, and finally sends the instant text information to the WebRTC terminal through the DataChannel channel; the server converts the text information into voice data through a voice recognition module, and then sends the voice data to the SIP terminal through the voice connection channel.
The communication method of the instant message based on the WebRTC technology can provide the function of the instant message for the ongoing voice conversation, video conversation or conference, add the data communication function of the instant message to the audio and video communication, and also can create an independent DataChannel channel without audio or video communication, thereby greatly expanding the existing application mode and greatly enriching the diversified choices of users.
Drawings
FIG. 1 is a schematic diagram of an embodiment of the present invention;
fig. 2 is a schematic signaling interaction diagram according to a first embodiment of the present invention;
FIG. 3 is a schematic diagram of a second embodiment of the present invention;
fig. 4 is a schematic signaling interaction diagram according to a second embodiment of the present invention;
FIG. 5 is a third schematic view of an embodiment of the present invention;
fig. 6 is a diagram illustrating an internal structure of a signaling/relay server according to an embodiment of the present invention.
Detailed Description
The basic principle of the invention is as follows: a method and a system for realizing instant message IM communication based on the DataChannel technology of WebRTC establish end-to-end DataChannel connection between two WebRTC terminals through a signaling server. And the instant message is encrypted or decrypted between the WebRTC terminal and the SIP terminal through the transfer server, and then the conversion between the SCTP and the SIP protocol is carried out to realize the instant message technology between two different terminals. The scheme of the invention can provide the function of instant message between the WebRTC terminals and also can provide the function of transferring the instant message between the WebRTC terminals and the SIP terminal.
In order to more clearly illustrate the technical solutions of the embodiments of the present invention, the following detailed description of the embodiments of the present invention is provided in conjunction with the accompanying drawings, and the following description is only some embodiments of the present invention. It should be understood that the specific embodiments described herein are merely illustrative of the invention and are not intended to limit the invention. Other embodiments of the invention can be obtained according to these examples without inventive effort for a person skilled in the art.
A specific embodiment is an implementation of a DataChannel connection between two WebRTC terminals in the specific embodiment of the present invention. Fig. 1 is a schematic diagram of an embodiment of the present invention. Fig. 2 is a schematic signaling interaction diagram according to a first embodiment of the present invention. Referring to fig. 1 and fig. 2, it can be known that the signaling server mentioned therein is also our UCM here, and this UCM only provides the function of signaling channel for the extension of the WebRTC terminal, and does not participate in the link of instant message delivery.
And both the two WebRTC terminals are registered in the signaling server, wherein one WebRTC terminal can request a DataChannel connection from the other WebRTC terminal.
First, the requester calls the PeerConnection interface of the WebRTC module to create a PeerConnection object, and then calls the createDataChannel method of the object to create a DataChannel.
And then, the WebRTC terminal of the requester starts to collect ICE candidate addresses, and when the stun server is deployed in the system, a binding request is sent to the stun server to acquire the public network address of the WebRTC terminal. After the ICE address collection is completed, the offer SDP carrying the ICE candidate address is sent to the signaling/relay server, and then the server passes through the SDP to the WebRTC terminal of the requested party.
The WebRTC terminal of the requested party generates an answer SDP carrying the ICE candidate address, and sends the answer SDP to the signaling/transfer server, and then the server transmits the SDP to the WebRTC terminal of the requesting party.
After receiving the answer SDP, the WebRTC terminal of the requesting party carries out ICE verification with the requested party to verify whether a UDP port negotiated by the SDP is available, and if the UDP port negotiated by the SDP is available, a DataChannel channel is established between the two WebRTC terminals.
Finally, any one party WebRTC terminal can send an instant message to the other party.
The signaling/relay server mentioned in the embodiments of the present invention is specifically referred to as UCM, and it is used as both a signaling server and a relay server. When the signaling server is used, the signaling sent by the WebRTC and the SIP terminal is analyzed, and corresponding actions are carried out; when the system is used as a transfer server, the system bears the gateway service function of audio and video and messages, and encrypts and decrypts the messages of the WebRTC terminal and the SIP terminal.
The second embodiment is an implementation scheme of the WebRTC terminal requesting an instant message function from the SIP terminal in the second embodiment of the present invention. FIG. 3 is a diagram illustrating a second embodiment of the present invention. Fig. 4 is a schematic signaling interaction diagram according to a second embodiment of the present invention. Referring to fig. 3 and fig. 4, it can be seen that the scheme at this time mainly includes the following implementation operations:
the WebRTC terminal and the SIP terminal are both registered to the signaling/transfer server.
After the WebRTC terminal finishes the ICE collection of the addresses, the offer SDP carrying the ICE candidate addresses is sent to the signaling/transfer server, and the signaling/transfer server analyzes the SDP to find that the requested party is the SIP terminal.
At the moment, the signaling/transfer server needs the function of a transfer server and generates an answer SDP response request WebRTC terminal carrying an ICE candidate address; and generating a new offer SDP to be sent to the SIP terminal;
ICE verification is needed between the WebRTC terminal and the signaling/transfer server to verify the connectivity of IP and PORT.
After the ICE verification is completed, a connection of a DataChannel channel is established between the WebRTC terminal and the signaling/transfer server.
A common SIP Session is between the SIP terminal and the signaling/relay server.
When the WebRTC terminal sends the instant message to the SIP terminal, the signaling/transfer server decrypts the SCTP encrypted message and sends the decrypted message to the SIP terminal through an SIP Session mechanism. At this time, the SIP terminal only needs to register to the signaling/relay server, and can work in a non-call-on state or a call-on state. In the message mechanism of the SIP Session, the SIP terminal can transmit the text information with the UCM through the message mechanism of the SIP Session when the voice connection is accessed; or when the voice call is not made, the text information can be mutually transmitted with UCM through the message mechanism of SIP Session
When the SIP terminal sends the instant message to the WebRTC terminal, the signaling/transfer server encrypts the unencrypted SIP message and then sends the encrypted SIP message to the WebRTC terminal through the DataChannel.
In the instant messaging function of the WebRTC terminal and the SIP terminal, a signaling/relay server is required to convert a data bearer protocol, an encryption type, and the like.
Similarly, the SIP terminal may also request the WebRTC terminal for the implementation of the instant messaging function.
The third specific embodiment is an implementation scheme for realizing voice and message interaction between the SIP terminal and the WebRTC terminal in the specific embodiment of the present invention. Fig. 5 is a third schematic diagram of the embodiment of the invention. The scheme mainly comprises the following steps to realize the operation:
the WebRTC terminal and the SIP terminal are both registered to the signaling/transfer server.
The registration and ICE candidate address collection flow of the scheme is the same as that of the specific embodiment, and meanwhile, after ICE verification is completed, a DataChannel channel connection is established between the WebRTC terminal and the signaling/transit server.
A common SIP session is between the SIP terminal and the signaling/transit server.
Different from the second embodiment, the SIP terminal in the third embodiment must make a voice call with the signaling/forwarding server, that is, an audio RTP voice connection path is to be established. The RTP data packet of the SIP terminal is analyzed and processed in the signaling/transfer server, the voice of the SIP terminal is converted into text information through the voice recognition module, and finally the instant text information is sent to the WebRTC terminal through a DataChannel module channel. Fig. 6 is a diagram illustrating an internal structure of a signaling/relay server according to an embodiment of the present invention. As shown in the figure, in the third embodiment of the present invention, the signaling/relay server internally includes: the device comprises a signaling processing module, an encryption/decryption module, a bridging module, a voice recognition module, a WebRTC DataChannel module and an SIP module, wherein the voice recognition module is used for realizing the function of converting the voice into words.
The WebRTC terminal can send the text information to the signaling/transfer server through the SCTP, the SCTP encrypted message is decrypted on the side of the signaling/transfer server, the text information is converted into voice data through the voice recognition module, and the voice data is sent to the SIP terminal through audio RTP.
Therefore, the real-time interaction of the messages and the voice of the SIP terminal and the WebRTC terminal can be realized.
The above description is only for the purpose of illustrating the preferred embodiments of the present invention and is not to be construed as limiting the invention, and any modifications, equivalents and improvements made within the spirit and principle of the present invention are intended to be included within the scope of the present invention.
Claims (11)
1. A communication method based on instant message of WebRTC technique can realize instant message communication between WebRTC terminal and SIP terminal, characterized in that, the communication method mainly includes the following steps:
the WebRTC terminal and the SIP terminal are both registered to a signaling/transfer server; the WebRTC terminal collects an ICE (Internet protocol chip) address and sends an offer SDP (service provider protocol) carrying an ICE candidate address to the signaling/transfer server;
the signaling/transfer server analyzes the SDP, when finding that the requested party is the SIP terminal, the signaling/transfer server starts a transfer service function, generates an answer SDP carrying an ICE candidate address to respond to the request of the WebRTC terminal, and generates a new offer SDP to be sent to the SIP terminal;
performing ICE verification between the WebRTC terminal and the signaling/transfer server to verify the connectivity of IP and PORT;
after the ICE verification is completed, the WebRTC terminal establishes connection of a DataChannel channel with the server;
the WebRTC terminal sends an instant message to the SIP terminal through an SIP Session mechanism, and the signaling/transfer server decrypts the SCTP encrypted message and sends the decrypted message to the SIP terminal through the SIP Session mechanism;
when the SIP terminal sends the instant message to the WebRTC terminal, the signaling/transfer server encrypts the unencrypted SIP message and then sends the encrypted SIP message to the WebRTC terminal through the DataChannel.
2. The communication method according to claim 1, further comprising: and ICE verification is required between the WebRTC terminal and the server to verify the connectivity of the ip and the port.
3. The communication method according to claim 2, further comprising: and the server decrypts the SCTP encrypted message and sends the instant message to the SIP terminal through an SIP Session mechanism.
4. The communication method according to claim 3, further comprising: and the WebRTC terminal and the SIP terminal are both registered to the server.
5. The communication method according to claim 4, wherein the SIP terminal only needs to register with the server during the communication method, and can operate in a non-call-on state or a call-on state.
6. The communication method according to claim 5, wherein the server encrypts an unencrypted SIP message and then sends the unencrypted SIP message to the WebRTC terminal through the DataChannel.
7. The communication method according to claim 6, further comprising: and the SIP terminal and the server establish a voice connection path.
8. The communication method according to claim 7, further comprising: and analyzing and processing the RTP data packet of the SIP terminal in the server, converting the voice of the SIP terminal into text information through a voice recognition module, and finally sending the instant text information to the WebRTC terminal through the DataChannel channel.
9. The communication method according to claim 8, further comprising: the WebRTC terminal sends the text information to the server through the SCTP, the server decrypts the SCTP encrypted message firstly, then converts the text information into voice data through the voice recognition module, and sends the voice data to the SIP terminal through the voice connection path.
10. The communication method according to claim 9, further comprising: the server converts the text information into voice data through a voice recognition module, and then sends the voice data to the SIP terminal through the voice connection channel.
11. A communication method based on instant message of WebRTC technique can realize instant message communication between WebRTC terminal and SIP terminal, characterized in that, the communication method mainly includes the following steps:
the SIP terminal dials a call to the WebRTC terminal, and the SIP terminal firstly sends a request signaling to a signaling/transfer server; the signaling/transfer server will collect the ICE address and re-fill the offer SDP carrying the ICE candidate address and send it to the WebRTC terminal; after receiving the offer, the WebRTC terminal generates an answer SDP carrying an ICE candidate address to respond to the signaling/transfer server; establishing a DataChannel channel between the signaling/transfer server and the WebRTC terminal, and establishing a common SIP session mechanism between the SIP terminal and the signaling/transfer server; the inside of the signaling/transfer server analyzes and processes the RTP data packet of the SIP terminal, converts the voice of the SIP terminal into text information through a voice recognition module, and finally sends the instant text information to the WebRTC terminal through the DataChannel channel; the signaling/transfer server converts the text information into voice data through a voice recognition module, and then sends the voice data to the SIP terminal through the voice connection channel.
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| CN111092802B (en) * | 2018-10-23 | 2022-09-16 | 中国移动通信有限公司研究院 | Message interaction method, sending equipment and receiving equipment |
| CN111147506A (en) * | 2019-12-30 | 2020-05-12 | 武汉兴图新科电子股份有限公司 | Method, system and storage device for playing streaming media data based on HTML5 |
| CN111447236B (en) * | 2020-04-03 | 2023-02-10 | 安康鸿天科技股份有限公司 | Block chain-based communication authentication method and device, terminal equipment and storage medium |
| CN112073423A (en) * | 2020-09-15 | 2020-12-11 | 杭州二更网络科技有限公司 | Browser plug-flow method and system based on WebRTC |
| CN113014544B (en) * | 2021-01-25 | 2023-02-10 | 阳光凯讯(北京)科技有限公司 | Method and device for establishing centerless media link based on webRtc |
| CN113114702B (en) * | 2021-05-13 | 2023-05-09 | 上海井星信息科技有限公司 | WebRTC communication method and system based on SIP protocol interaction at IOS end |
| CN113726893B (en) * | 2021-08-31 | 2024-01-09 | 广州艾美网络科技有限公司 | Method and device for updating music library, KTV box, song server and network KTV system |
| CN116708530A (en) * | 2022-02-25 | 2023-09-05 | 腾讯科技（深圳）有限公司 | Information interaction method, information interaction device, computer equipment, storage medium and program product |
| CN114785729B (en) * | 2022-03-30 | 2023-01-10 | 慧之安信息技术股份有限公司 | Signaling interaction control method and system based on SIP protocol format conversion |
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