CN106850399A - A kind of communication means based on WebRTC technology instant messages - Google Patents
A kind of communication means based on WebRTC technology instant messages Download PDFInfo
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- CN106850399A CN106850399A CN201611262789.3A CN201611262789A CN106850399A CN 106850399 A CN106850399 A CN 106850399A CN 201611262789 A CN201611262789 A CN 201611262789A CN 106850399 A CN106850399 A CN 106850399A
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- server
- webrtc
- communication means
- sip terminal
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Classifications
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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]
-
- 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 present invention mainly discloses a kind of communication means based on WebRTC technology instant messages, it is possible to achieve the instant messaging between WebRTC terminals and sip terminal, the communication means is mainly included the following steps that：WebRTC terminals carry out ICE and collect address, and the offer SDP for carrying ICE candidate sites are sent into server；Server parses SDP, and when it is sip terminal to find Requested Party, startup of server transfer service function produces the answer SDP for carrying ICE candidate sites to respond the request of WebRTC terminals, and produce new offer SDP to sip terminal；WebRTC terminals set up the connection of DataChannel passages with server；WebRTC terminals send instant message by SIP Session mechanism to sip terminal.Gapless communication between WebRTC terminals and sip terminal is got through by foregoing invention scheme, communication cost has been reduced, while the diversified communication modes between also increasing different types of terminals, improve the communication efficiency of enterprise customer.
Description
Technical field
The present invention relates to computer communication field, more specifically to the DataChannel technology realities based on WebRTC
The method and system of existing instant message IM.
Background technology
WebRTC（Web Real-Time Communication, real-time Communication for Power）, it is one real-time many based on browser
Media communication technology.This technology is intended to make Web browsers possess real-time Communication for Power ability, meanwhile, sealed by by these abilities
Fill and Web application developers are given in opening in the way of JavaScript API so that Web application developers can
The real-time audio and video application of Web browsers is rapidly developed based on by HTML labels and JavaScript API,
And without relying any third party's plug-in unit.The terminal that can the carry out WebRTC communications referred to as WebRTC terminals being mentioned above.So
WebRTC technologies cause support WebRTC functions browser between can carry out direct web communication, so as to change terminal
Browser and the original Client/Server mode of operations of server, can allow Web to carry out end to end communication, be to Web technologies
A big innovation.WebRTC terminals use SCTP SCTP（Stream Control Transmission
Protocol）Used as the host-host protocol of data, the agreement can carry out flow control and congestion control to transmission data.
Web browsers are forged into one and are applied in all user equipmenies, such as mobile intelligent terminal by this technology
And PC etc. between, the general-purpose platform of real-time audio frequency and video and data communication is carried out.In this explanation of the invention, we
It is concerned only with data communication part.
IP multimedia subsystem（IP Multimedia Subsystem, IMS）It is that a kind of IP based network supports multimedia
The network architecture of business, using SIP session initiation protocols（Session Initiation Protocol）Exhaled end to end
Control is cried, meets the terminal referred to as sip terminal of Session Initiation Protocol.Sip terminal can immediately be disappeared by SIP Message methods
Breath transmission.
In voip systems, many sip terminals have been deployed, and can between the WebRTC terminals that newly increase and sip terminal
To carry out voice and video communication, the transmission for instant message cannot be realized.A transfer server is needed between them
The conversion of two encryption/decryption of Deta bearer agreement and data of network etc. is realized, so as to realize WebRTC terminals and SIP ends
Hold the intercommunication of instant message.
The transmission of two instant messages of WebRTC terminal rooms by a signal server, it is necessary to complete.Signaling
Server provides a bridge for the foundation of the DataChannel between two WebRTC, by the signal server, both sides
Interaction SDP information, finally so that two WebRTC terminals set up end-to-end DataChannel connections, by this
DataChannel passages, they just can mutually send reliably, encryption instant message.
The content of the invention
It is an object of the invention to realize a kind of communication means based on WebRTC technology instant messages, signaling clothes can be collected
The service end that business device is integrated with transfer server, one kind side is provided for WebRTC terminals and sip terminal send instant message
Method and system.
In order to realize foregoing invention purpose, technical scheme is as follows：
A kind of communication means based on WebRTC technology instant messages, can realize between WebRTC terminals and sip terminal i.e.
When message communicating, the communication means mainly includes the following steps that：WebRTC terminals carry out ICE and collect address, and will carry ICE
The offer SDP of candidate site are sent to server；Server parses SDP, when it is sip terminal to find Requested Party, service
Device starts transfer service function, produces the answer SDP for carrying ICE candidate sites to respond the request of WebRTC terminals, and produce
New offer SDP are to sip terminal；WebRTC terminals set up the connection of DataChannel passages with server；WebRTC ends
End sends instant message by SIP Session mechanism to sip terminal.
Preferably, above-mentioned communication means also includes：, it is necessary to carry out ICE verifications between WebRTC terminals and the server, come
Verify the connectedness of ip, port.
Preferably, in above-mentioned communication means, after the message decryption that server can encrypt SCTP, by SIP Session
The instant message is sent to sip terminal by mechanism.
Preferably, above-mentioned communication means also includes：WebRTC terminals are registered to server with sip terminal.
Preferably, during communication means is realized, sip terminal only needs to be registered to server, can be operated in not
Telephone state is connected, connection telephone state can be also operated in.
Preferably, in above-mentioned communication means, after server can be by the encryption of the sip message of unencryption, then pass through
DataChannel is sent to WebRTC terminals.
Preferably, above-mentioned communication means also includes：Sip terminal sets up voice connecting path with server.
Preferably, above-mentioned communication means also includes：The RTP resolve packets of sip terminal are processed in server internal, then
The voice of sip terminal is converted into by text information by sound identification module, will be instant finally by DataChannel passages
Text information is sent to WebRTC terminals.
Preferably, above-mentioned communication means also includes：Text information is sent to service by WebRTC terminals by Stream Control Transmission Protocol
Device, server is decrypted to the message that SCTP is encrypted.
Preferably, above-mentioned communication means also includes：Text information is converted into voice by server by sound identification module
Data, then the sip terminal is sent to by the voice connecting path.
Additionally, the present invention also provides a kind of communication means based on WebRTC technology instant messages, it is possible to achieve sip terminal
Communication is set up as request direction WebRTC terminals, it is characterised in that the communication means is mainly included the following steps that：
The sip terminal is called to the WebRTC terminals, and the sip terminal sends demand signalling to server first；
The server can collect ICE addresses, and refill carries the ICE candidate sites offer SDP be sent to described in
WebRTC terminals；After the WebRTC terminals receive offer, the answer SDP for carrying ICE candidate sites are produced to respond described
Server；DataChannel passages are set up with the WebRTC terminal rooms in the server, in the sip terminal and service
Common SIP session are set up between device；The server internal processes the RTP resolve packets of the sip terminal, then leads to
Cross sound identification module and the voice of the sip terminal is converted into text information, will finally by the DataChannel passages
Instant text information is sent to the WebRTC terminals；Be converted into for text information by sound identification module by the server
Speech data, then the sip terminal is sent to by the voice connecting path.
A kind of communication means based on WebRTC technology instant messages of the present invention, can for ongoing voice conversation,
Video session or meeting provide the function of instant message, and the data communication function of instant message is increased to audio/video communication, also may be used
To create independent DataChannel passages, without audio or video communication, existing application model has been expanded significantly, greatly
Enrich the diversified selection of user.
Brief description of the drawings
Fig. 1 is the schematic diagram of the specific embodiment of the invention one；
Fig. 2 is the Signalling exchange schematic diagram of the specific embodiment of the invention one；
Fig. 3 is the schematic diagram of the specific embodiment of the invention two；
Fig. 4 is the Signalling exchange schematic diagram of the specific embodiment of the invention two；
Fig. 5 is the schematic diagram of the specific embodiment of the invention three；
Fig. 6 is signaling/transfer server cut-away view in the specific embodiment of the invention.
Specific embodiment
General principle of the invention：The method that DataChannel technologies based on WebRTC realize instant message IM communications
And system, end-to-end DataChannel is set up by signal server in two WebRTC terminal rooms and is connected.And in WebRTC
Between terminal and sip terminal, by transfer server by immediate news enciphering or decryption, then turn between SCTP and Session Initiation Protocol
Change the instant message technology realized between two different terminals.The present invention program can provide the work(of instant message for WebRTC terminal rooms
Energy, or WebRTC terminals provide the function of transfer instant message with sip terminal.
In order to illustrate more clearly of the technical scheme of present example, below in conjunction with exemplary plot to it is of the invention implement into
The detailed introduction of row, description below is only some embodiments of the present invention.It should be appreciated that specific implementation described herein
Example is only used to explain the present invention, is not intended to limit the present invention.For those of ordinary skill in the art, wound is not being paid
On the premise of the property made is worked, other embodiment of the invention can also be obtained according to these embodiments.
Specific embodiment one is two realities of the DataChannel connections of WebRTC terminal rooms in the specific embodiment of the invention
Apply scheme.Fig. 1 is the schematic diagram of the specific embodiment of the invention one.Fig. 2 is the Signalling exchange schematic diagram of the specific embodiment of the invention one.
With reference to Fig. 1 and Fig. 2, we understand that the signal server referred in its Chinese is also herein our UCM, and the UCM is only
The extension set of WebRTC terminals provides the function of signalling path, cannot participate in the link of the transmission of instant message.
Two WebRTC terminals are registered in signal server, wherein, a WebRTC terminal can be to another
WebRTC terminal requests DataChannel is connected.
First, requesting party can call the PeerConnection interfaces of WebRTC modules, be created that peerconnection
Object, then calls the createDataChannel methods of the object to create a DataChannel.
Then, requesting party WebRTC terminals start to collect ICE candidate sites, when stun server are deployed in system,
Binding can be sent to stun server to ask, obtain the public network address of WebRTC terminals.When completing ICE address collections
Afterwards, the offer SDP that can will carry ICE candidate sites are sent to signaling/transfer server, and then server can be by the SDP
Pass through Requested Party WebRTC terminals.
Requested Party WebRTC terminals can produce the answer SDP that carry ICE candidate sites, be sent to signaling/transfer
Server, then server the SDP can be passed through requesting party's WebRTC terminals.
After requesting party's WebRTC terminals receive answer SDP, can carry out ICE verifications with Requested Party to verify SDP
Whether the udp port of negotiation can use, if can use, can set up DataChannel passages in two WebRTC terminal rooms.
Finally, any one party WebRTC terminals can send instant message to the opposing party.
The equal spy of signaling/transfer server referred in the specific embodiment of the invention refers to UCM, and it both does signalling service
Device, also does transfer server.When it does signal server, the signaling that WebRTC and sip terminal are sent is parsed, and do phase
The action answered；When it does transfer server, the gateway service function of audio frequency and video and message is undertaken, to WebRTC terminals and SIP ends
The message at end such as is encrypted, decrypts at the work.
Specific embodiment two is that WebRTC terminals ask instant message function to sip terminal in the specific embodiment of the invention
Embodiment.Fig. 3 is the schematic diagram of the specific embodiment of the invention two.Fig. 4 illustrates for the Signalling exchange of the specific embodiment of the invention two
Figure.With reference to Fig. 3 and Fig. 4, it is known that now scheme mainly includes following some realization operations：
WebRTC terminals are registered to signaling/transfer server with sip terminal.
WebRTC terminals complete ICE collect address after, by carry ICE candidate sites offer SDP be sent to signaling/in
Turn server, SDP is parsed by it, discovery Requested Party is sip terminal.
Now, signaling/transfer server needs the function of transfer server, produces the answer for carrying ICE candidate sites
SDP responds request WebRTC terminals；And produce new offer SDP to be sent to sip terminal；
, it is necessary to carry out ICE verifications to verify the connectedness of IP, PORT between WebRTC terminals and signaling/transfer server.
After completing ICE verifications, DataChannel passages are established between WebRTC terminals and signaling/transfer server
Connection.
It is exactly common SIP Session between sip terminal and signaling/transfer server.
When WebRTC terminals send instant message to sip terminal, the message that signaling/transfer server can encrypt SCTP
After decryption, sip terminal is sent to by SIP Session mechanism.Sip terminal now only needs to be registered to signaling/transfer clothes
Business device, can be operated in and disconnect telephone state, can also be operated in connection telephone state.The message mechanism of SIP Session
In, sip terminal can transmit text information in voice connecting path by the message mechanism and UCM of SIP Session；
Text information can mutually be passed by the message mechanism and UCM of SIP Session when voice call is not got through
When sip terminal sends instant message to WebRTC terminals, signaling/transfer server can add the sip message of unencryption
After close, then WebRTC terminals are sent to by DataChannel.
, it is necessary to signaling/transfer server is assisted to Deta bearer in the instant message function of WebRTC terminals and sip terminal
Negotiation encryption type etc. is changed.
Similarly, sip terminal also can be the implementation to WebRTC terminal request instant message functions.
Specific embodiment three is that sip terminal realizes voice and interacting message with WebRTC terminals in the specific embodiment of the invention
Embodiment.Fig. 5 is the schematic diagram of the specific embodiment of the invention three.Now scheme mainly includes following some steps to realize behaviour
Make：
WebRTC terminals are registered to signaling/transfer server with sip terminal.
The registration of the program is identical with specific embodiment two with ICE candidate sites collection flow, while completing ICE verifications
Afterwards, the connection of DataChannel passages is established between WebRTC terminals and signaling/transfer server.
It is exactly universal SIP sessions between sip terminal and signaling/transfer server.
And from unlike specific embodiment two, the sip terminal of this side of specific embodiment three must be got through and signaling/transfer
The voice call of server, that is to say, that to set up audio RTP voice connecting paths.Will inside signaling/transfer server
The RTP resolve packets treatment of sip terminal, then the voice of sip terminal is converted into by text information by sound identification module, most
Instant text information is sent to WebRTC terminals by DataChannel module channels afterwards.Fig. 6 is embodied for the present invention
Signaling/transfer server cut-away view in example.As illustrated, in the specific embodiment of the invention three, signaling/transfer server
Inside includes：Signaling processing module, encrypting-decrypting module, bridge module, sound identification module, WebRTC DataChannle
Module, SIP module, wherein sound identification module are used to realize above-mentioned voice conversion literal function.
And text information can be sent to signaling/transfer server by Stream Control Transmission Protocol in WebRTC terminals, signaling/
Message of the transfer server here first to SCTP encryptions is decrypted, then text information is converted into language by sound identification module
Sound data, sip terminal is sent to by audio RTP.
This way it is possible to realize sip terminal and the message of WebRTC terminals and the real-time, interactive of voice.
Presently preferred embodiments of the present invention is the foregoing is only, is not intended to limit the invention, it is all in essence of the invention
Any modification, equivalent and improvement made within god and principle etc., should be included within the scope of the present invention.
Claims (11)
1. a kind of communication means based on WebRTC technology instant messages, can realize between WebRTC terminals and sip terminal
Instant messaging, it is characterised in that the communication means is mainly included the following steps that：
The WebRTC terminals carry out ICE and collect address, and the offer SDP for carrying ICE candidate sites are sent into server；
The server parses SDP, and when it is the sip terminal to find Requested Party, the startup of server transfer services work(
Can, produce the answer SDP for carrying ICE candidate sites to respond the request of the WebRTC terminals, and produce new offer
SDP is sent to the sip terminal；
The WebRTC terminals set up the connection of DataChannel passages with the server；
The WebRTC terminals send instant message by SIP Session mechanism to the sip terminal.
2. communication means according to claim 1, it is characterised in that the communication means also includes：The WebRTC ends
, it is necessary to carry out ICE verifications to verify the connectedness of ip, port between end and the server.
3. communication means according to claim 2, it is characterised in that the communication means also includes：The server meeting
After the message decryption that SCTP is encrypted, the instant message is sent to by the sip terminal by SIP Session mechanism.
4. communication means according to claim 3, it is characterised in that the communication means also includes：The WebRTC ends
End is registered to the server with the sip terminal.
5. communication means according to claim 4, it is characterised in that during the communication means is realized, the SIP
Terminal only needs to be registered to the server, can be operated in and disconnect telephone state, can also be operated in connection telephone state.
6. communication means according to claim 5, it is characterised in that the server can add the sip message of unencryption
After close, then the WebRTC terminals are sent to by the DataChannel.
7. communication means according to claim 6, it is characterised in that the communication means also includes：The sip terminal with
The server sets up voice connecting path.
8. communication means according to claim 7, it is characterised in that the communication means also includes：In the server
The internal RTP resolve packets by the sip terminal are processed, then are turned the voice of the sip terminal by sound identification module
Text information is turned to, instant text information is sent to the WebRTC terminals finally by the DataChannel passages.
9. communication means according to claim 8, it is characterised in that the communication means also includes：The WebRTC ends
Text information is sent to the server by end by Stream Control Transmission Protocol, and message of the server first to SCTP encryptions is solved
It is close, then text information is converted into by speech data by sound identification module, it is sent to by the voice connecting path described
Sip terminal.
10. communication means according to claim 9, it is characterised in that the communication means also includes：The server leads to
Cross sound identification module and text information is converted into speech data, then the SIP ends are sent to by the voice connecting path
End.
A kind of 11. communication means based on WebRTC technology instant messages, can realize between WebRTC terminals and sip terminal
Instant messaging, it is characterised in that the communication means is mainly included the following steps that：
The sip terminal is called to the WebRTC terminals, and the sip terminal sends demand signalling to server first；
The server can collect ICE addresses, and refill carries the ICE candidate sites offer SDP be sent to described in
WebRTC terminals；After the WebRTC terminals receive offer, the answer SDP for carrying ICE candidate sites are produced to respond described
Server；DataChannel passages are set up with the WebRTC terminal rooms in the server, in the sip terminal and service
Common SIP session mechanism is set up between device；The server internal processes the RTP resolve packets of the sip terminal,
The voice of the sip terminal is converted into by text information by sound identification module again, it is logical finally by the DataChannel
Instant text information is sent to the WebRTC terminals by road；The server is turned text information by sound identification module
Speech data is turned to, then the sip terminal is sent to by the voice connecting path.
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| CN201611262789.3A CN106850399B (en) | 2016-12-30 | 2016-12-30 | Communication method based on WebRTC technology instant message |
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| CN201611262789.3A CN106850399B (en) | 2016-12-30 | 2016-12-30 | Communication method based on WebRTC technology instant message |
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Cited By (11)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN109995734A (en) * | 2017-12-30 | 2019-07-09 | 中国科学院沈阳计算技术研究所有限公司 | A kind of communication means of the WebRTC based on Session Initiation Protocol |
| CN111092802A (en) * | 2018-10-23 | 2020-05-01 | 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 |
| CN111447236A (en) * | 2020-04-03 | 2020-07-24 | 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 |
| CN113014544A (en) * | 2021-01-25 | 2021-06-22 | 阳光凯讯(北京)科技有限公司 | Method and device for establishing centerless media link based on webRtc |
| CN113114702A (en) * | 2021-05-13 | 2021-07-13 | 上海井星信息科技有限公司 | WebRTC communication method and system based on SIP protocol interaction at IOS (input/output System) end |
| CN113726893A (en) * | 2021-08-31 | 2021-11-30 | 广州艾美网络科技有限公司 | Method and device for updating song library, KTV box, song server and network KTV system |
| CN114785729A (en) * | 2022-03-30 | 2022-07-22 | 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 |
| CN116048668A (en) * | 2023-03-30 | 2023-05-02 | 麒麟软件有限公司 | Application plug-in communication system based on web operating system |
| WO2023160076A1 (en) * | 2022-02-25 | 2023-08-31 | 腾讯科技（深圳）有限公司 | Information interaction method and apparatus, computer device, storage medium and program product |
Citations (10)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN1738243A (en) * | 2004-08-16 | 2006-02-22 | 梁松 | Search engine call communication system and method |
| CN102790710A (en) * | 2011-05-16 | 2012-11-21 | 北京新媒传信科技有限公司 | Method and device for audio and video communication between PC (personal computer) terminal and cell phone |
| CN104052732A (en) * | 2013-03-14 | 2014-09-17 | 阿瓦亚公司 | Managing identity provider (IdP) identifiers for web real-time communications (webrtc) interactive flows, and related methods, systems |
| CN104283760A (en) * | 2013-07-04 | 2015-01-14 | 华为技术有限公司 | Method, related device and system for WebRTC communication |
| US20150127709A1 (en) * | 2013-11-05 | 2015-05-07 | Avaya Inc. | Providing reliable session initiation protocol (sip) signaling for web real-time communications (webrtc) interactive flows, and related methods, systems, and computer-readable media |
| CN104704795A (en) * | 2012-10-19 | 2015-06-10 | 统一有限责任两合公司 | Method and system for creating a virtual sip user agent by use of a webrtc enabled web browser |
| US20150180825A1 (en) * | 2013-12-20 | 2015-06-25 | Futurewei Technologies Inc. | METHOD OF IMS (SIP NETWORK) webRTC OPTIMIZED P2P COMMUNICATION |
| US20150304359A1 (en) * | 2014-04-17 | 2015-10-22 | Avaya Inc. | APPLICATION OF ENTERPRISE POLICIES TO WEB REAL-TIME COMMUNICATIONS (WebRTC) INTERACTIVE SESSIONS USING AN ENTERPRISE SESSION INITIATION PROTOCOL (SIP) ENGINE, AND RELATED METHODS, SYSTEMS, AND COMPUTER-READABLE MEDIA |
| US20150341312A1 (en) * | 2014-05-20 | 2015-11-26 | Avaya, Inc. | Firewall traversal for web real-time communications |
| CN106161179A (en) * | 2015-03-26 | 2016-11-23 | 中兴通讯股份有限公司 | The media processing method of a kind of real-time Communication for Power based on webpage and device |
-
2016
- 2016-12-30 CN CN201611262789.3A patent/CN106850399B/en active Active
Patent Citations (10)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN1738243A (en) * | 2004-08-16 | 2006-02-22 | 梁松 | Search engine call communication system and method |
| CN102790710A (en) * | 2011-05-16 | 2012-11-21 | 北京新媒传信科技有限公司 | Method and device for audio and video communication between PC (personal computer) terminal and cell phone |
| CN104704795A (en) * | 2012-10-19 | 2015-06-10 | 统一有限责任两合公司 | Method and system for creating a virtual sip user agent by use of a webrtc enabled web browser |
| CN104052732A (en) * | 2013-03-14 | 2014-09-17 | 阿瓦亚公司 | Managing identity provider (IdP) identifiers for web real-time communications (webrtc) interactive flows, and related methods, systems |
| CN104283760A (en) * | 2013-07-04 | 2015-01-14 | 华为技术有限公司 | Method, related device and system for WebRTC communication |
| US20150127709A1 (en) * | 2013-11-05 | 2015-05-07 | Avaya Inc. | Providing reliable session initiation protocol (sip) signaling for web real-time communications (webrtc) interactive flows, and related methods, systems, and computer-readable media |
| US20150180825A1 (en) * | 2013-12-20 | 2015-06-25 | Futurewei Technologies Inc. | METHOD OF IMS (SIP NETWORK) webRTC OPTIMIZED P2P COMMUNICATION |
| US20150304359A1 (en) * | 2014-04-17 | 2015-10-22 | Avaya Inc. | APPLICATION OF ENTERPRISE POLICIES TO WEB REAL-TIME COMMUNICATIONS (WebRTC) INTERACTIVE SESSIONS USING AN ENTERPRISE SESSION INITIATION PROTOCOL (SIP) ENGINE, AND RELATED METHODS, SYSTEMS, AND COMPUTER-READABLE MEDIA |
| US20150341312A1 (en) * | 2014-05-20 | 2015-11-26 | Avaya, Inc. | Firewall traversal for web real-time communications |
| CN106161179A (en) * | 2015-03-26 | 2016-11-23 | 中兴通讯股份有限公司 | The media processing method of a kind of real-time Communication for Power based on webpage and device |
Cited By (15)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN109995734A (en) * | 2017-12-30 | 2019-07-09 | 中国科学院沈阳计算技术研究所有限公司 | A kind of communication means of the WebRTC based on Session Initiation Protocol |
| CN111092802A (en) * | 2018-10-23 | 2020-05-01 | 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 |
| CN111447236A (en) * | 2020-04-03 | 2020-07-24 | 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 |
| CN113014544A (en) * | 2021-01-25 | 2021-06-22 | 阳光凯讯(北京)科技有限公司 | Method and device for establishing centerless media link based on webRtc |
| CN113114702A (en) * | 2021-05-13 | 2021-07-13 | 上海井星信息科技有限公司 | WebRTC communication method and system based on SIP protocol interaction at IOS (input/output System) end |
| CN113726893A (en) * | 2021-08-31 | 2021-11-30 | 广州艾美网络科技有限公司 | Method and device for updating song library, KTV box, song server and network KTV system |
| 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 |
| WO2023160076A1 (en) * | 2022-02-25 | 2023-08-31 | 腾讯科技（深圳）有限公司 | Information interaction method and apparatus, computer device, storage medium and program product |
| CN114785729A (en) * | 2022-03-30 | 2022-07-22 | 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 |
| 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 |
| CN116048668A (en) * | 2023-03-30 | 2023-05-02 | 麒麟软件有限公司 | Application plug-in communication system based on web operating system |
| CN116048668B (en) * | 2023-03-30 | 2023-06-23 | 麒麟软件有限公司 | Application plug-in communication system based on web operating system |
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