CN109995734A - A kind of communication means of the WebRTC based on Session Initiation Protocol - Google Patents
A kind of communication means of the WebRTC based on Session Initiation Protocol Download PDFInfo
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- CN109995734A CN109995734A CN201711492037.0A CN201711492037A CN109995734A CN 109995734 A CN109995734 A CN 109995734A CN 201711492037 A CN201711492037 A CN 201711492037A CN 109995734 A CN109995734 A CN 109995734A
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- sip
- webrtc
- signaling
- called end
- calling terminal
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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
- H04L65/10—Architectures or entities
- H04L65/1045—Proxies, e.g. for 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
- H04L65/1066—Session management
-
- 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
- H04L67/00—Network arrangements or protocols for supporting network services or applications
- H04L67/01—Protocols
- H04L67/10—Protocols in which an application is distributed across nodes in the network
- H04L67/104—Peer-to-peer [P2P] networks
-
- 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
- H04L67/14—Session management
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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)
- Telephonic Communication Services (AREA)
Abstract
The present invention relates to the communication means of WebRTC based on Session Initiation Protocol a kind of, including calling terminal and called end to be acted on behalf of by ICE and collect candidate site, generates offer signaling and answer signaling.Calling terminal and called end are by being built in the SIPRTC local gateway realization WebRTC signaling of client and mutually converting for SIP signaling, then calling terminal and called end establish P2P or carry out Media Stream transfer by media relay servers by sip server forwarding SIP signaling, final calling terminal and called end.The present invention realizes WebRTC signaling and SIP signaling mutually converting in client-side, and is handled by the PeerConnection layer of WebRTC real time flow medium, realizes Session Initiation Protocol and the efficient fusion of WebRTC.
Description
Technical field
The present invention relates to WebRTC application field, the communication means of especially a kind of WebRTC based on Session Initiation Protocol.
Background technique
SIP (Session Initiation Protocol) is the signaling control protocol of an application layer, be the core of IMS,
Technology that is mature, being used widely, for creating, modifying and discharging the session of one or more participants.
Web real time communication (WebRTC) is a kind of real-time audio and video communication technology of the building on Web browser.WebRTC is provided
A whole set of the audio/video communication solution party such as audio-video collection, network transmission, audio/video encoding/decoding, signal optimization and processing
Case.Due to the powerful multi-media processing engine of WebRTC, WebRTC Chrome, Firefox, Opera, Android,
It is supported on the browsers such as iOS and platform.
The JSEP that WebRTC is provided is a kind of weak signaling, must be by WebRTC and reality in the converged communication application of enterprise-level
The signaling protocol on border combines.Instantly there are mainly two types of SIP and WebRTC Interworking Schemes: one is realized with JavaScript
Session Initiation Protocol constructs WebRTC application on the basis of this protocol stack.Another kind is to develop the SIP/WebRTC based on server to turn
Draping is closed.Two schemes are JavaScript interface (WebRTC interface) development and application provided using browser, certain
The operational efficiency of application can be reduced in degree, secondly because WebRTC interface is in standard formulation, so utilizing
JavaScript interface can be inconvenient, and last two schemes are required to server and do opening for compatible WebRTC in various degree
Hair.
Summary of the invention
The present invention is directed to the deficiency of existing WebRTC and Session Initiation Protocol integration program, proposes a kind of server and does not need to do compatibility
WebRTC exploitation and greater efficiency develop the communication means of audio/video communication software using WebRTC and Session Initiation Protocol.
Present invention technical solution used for the above purpose is: a kind of communication means of the WebRTC based on Session Initiation Protocol,
Include:
Step 1: calling terminal collects candidate site and generates offer signaling；Candidate site and offer signaling are converted to by calling terminal
Sip request message is sent to sip server；Sip server forwards sip request message to called end；
Step 2: after called end receives sip request message, sip request message is converted to WebRTC signaling and candidate ground by called end
Location；Called end generates answer signaling；Candidate site and answer signaling are converted to SIP request again and receiveed the response by called end,
It is sent to sip server；Sip server forwarding SIP request is receiveed the response to calling terminal；
Step 3: after calling terminal reception SIP request is receiveed the response, after SIP transmission message is directly replied to called end, two clients
P2P or the Multimedia session by media relay servers are established in end；
Step 4: giving another client when one of client sends SIP termination messages, it is whole that another client receives SIP
Only message, and reply SIP termination and receive the response to starting client, then terminate the session of two clients.
It is to act on behalf of to collect by ICE that calling terminal, which collects candidate site, in the step 1；
It is by SIPRTC local network that candidate site and offer signaling are converted to sip request message by calling terminal in the step 1
Close realization.
Sip request message is converted to WebRTC signaling by called end in the step 2 and candidate site is by the local SIPRTC
What gateway was realized；
It is by calling the WebRTCPeerConnection layer of called end to connect that called end, which generates answer signaling, in the step 2
Cause for gossip is existing；
It is to pass through calling that candidate site and answer signaling are converted to SIP request and receiveed the response by called end again in the step 2
What SIPRTC local gateway was realized.
The step 4 specifically:
When one of client call WebRTC PeerConnection layer interface and SIP termination messages are sent to another
Client, another client receive SIP termination messages, and call WebRTCPeerConnection layer interface to terminate media and pass
It is defeated, it then replys SIP termination and receives the response to starting client, then terminate the session of two clients.
The calling terminal, called end pass through the SIPRTC local gateway for turning mutually between SIP SDP and WebRTCSDP
Change and be mapped to sip message PeerConnection layers of WebRTC the corresponding interface.
The SIPRTC local gateway is embedded in client, and the client includes calling terminal and called end.
The SIPRTC local gateway uses PeerConnection layer interface and Session Initiation Protocol based on WebRTC C++API.
The invention has the following beneficial effects and advantage:
1. the communication means of the WebRTC proposed by the present invention based on Session Initiation Protocol is suitable for Windows, iOS and Android client
Hold platform, and the specific implementation no requirement (NR) to Session Initiation Protocol.
2. SIP signaling and the conversion of WebRTC signaling are transparent to sip server in the present invention, it is not necessarily to sip server intervention.
3. the non-Web of present invention application is applied, operational efficiency is high.
4. the present invention realizes the compatibility with traditional IMS network by client SIPRTC local gateway.
Detailed description of the invention
Integration program of the PeerConnection layer and Session Initiation Protocol that Fig. 1 is the WebRTC of the embodiment of the present invention one in client
Architecture diagram；
Fig. 2 is the PeerConnetion layer interface and SIP protocol mapping flow chart of the WebRTC of the embodiment of the present invention two.
Specific embodiment
The present invention is described in further detail with reference to the accompanying drawings and embodiments.
Basic principle of the invention: WebRTC (webpage real time communication) signaling and SIP (session are realized based on SIPRTC local gateway
Initiation protocol) signaling conversion, i.e. the PeerConnection of WebRTC (peer to peer connection) layer and Session Initiation Protocol melting in client
It closes.A kind of server may be implemented and do not need to do opening using WebRTC and Session Initiation Protocol for compatible WebRTC exploitation and greater efficiency
Pronunciation video communication applications, the application scheme are as follows:
A) calling terminal is acted on behalf of by the ICE collects candidate site and generates offer signaling, and then calling terminal passes through described
SIPRTC local gateway is converted to sip message, and is sent to the sip server；
B) after called end receives sip message, after called end is converted by the SIPRTC local gateway, WebRTC is called
PeerConnection layer interface come collect candidate site and generate answer signaling, then recall the SIPRTC local network
Pass is converted to sip message and is sent to sip server；
C) after calling terminal receives sip message, 200OK message is replied to called end, latter two right client establishes P2P (point-to-point)
Or the Multimedia session by the media relay servers；
D) PeerConnection layers of api interface of one of client call WebRTC and send sip message to another visitor
Family end, another client receive sip message, reply sip message and WebRTCPeerConnection layer interface is called to terminate
The session of two clients.
Calling terminal, called end be required to be used for by the local gateway of the SIPRTC SIP SDP (Session Description Protocol) and
Between WebRTC SDP mutual inversion of phases and by SIP protocol mapping at WebRTCPeerConnection layers the corresponding interface.
The SIPRTC local gateway is embedded in client.
The exploitation of the SIPRTC local gateway is based on WebRTC C++API and Session Initiation Protocol is developed.SIPRTC local gateway is opened
Hair is developed based on WebRTC C++API, but is not limited to C++API, Objective-C, Java language generated including means such as compilings
The WebRTC API of speech.The exploitation of SIPRTC local gateway is developed based on Session Initiation Protocol, but does not limit the specific implementation of Session Initiation Protocol
Mode.
Calling terminal, called end can establish P2P or the Multimedia session by the media relay servers.Calling terminal is called
End can act on behalf of collection candidate site by the ICE and establish P2P Multimedia session, also can establish through the media relays
The Multimedia session of server.The ICE agency and the media relay servers are not one during a session establishment
It is fixed to need to exist simultaneously.
Calling terminal, called end be required to by the sip server exchange SIP signaling, but the sip server do not need to do it is simultaneous
Hold the exploitation of WebRTC.
For the clearer technical solution for illustrating present example, implementation of the invention is carried out below in conjunction with exemplary diagram detailed
Thin introduction, description below are only some embodiments of the present invention.It should be appreciated that specific embodiment described herein is only
Only to explain the present invention, it is not intended to limit the present invention.To those skilled in the art, creative labor is not being paid
Under the premise of dynamic, the other embodiment of the present invention mode can also be obtained according to these embodiments.
Specific embodiment first is that WebRTC PeerConnection layer and Session Initiation Protocol client integration program overall architecture
Embodiment.Integration program framework of the PeerConnection layer and Session Initiation Protocol that Fig. 1 is WebRTC of the present invention in client
Figure.With reference to Fig. 1, sip server namely our OpenSIPS server, the media relays service referred in text are referred in text
Device namely our Asterisk server, the ICE agency referred in text namely our STUN Server server.
Sip server only transfer sip message, so sip server does not need to do the exploitation for compatible WebRTC.
Client can act on behalf of the candidate site information for collecting oneself from ICE first, can be by candidate ground during SDP negotiation below
Location information is sent together.End-Customer end can establish P2P Multimedia session according to the candidate site information got, if
P2P Multimedia session establishes failure, it will establishes and carries out media relays by Asterisk server.
Specific embodiment is second is that the embodiment that SIPRTC local gateway does WebRTC signaling and the conversion of SIP signaling.Fig. 2 is this hair
Bright WebRTC PeerConnetion interface and SIP protocol mapping flow chart.With reference to Fig. 2, it is known that at this time scheme mainly include with
Lower following implemented operation:
Calling terminal calls PeerConnection layers of createOffer interface, generates WebRTC SDP message and passes through ICE generation
Reason collects candidate site information, and then calling terminal calls SIPRTC local gateway to be converted to sip message (INVITE message) transmission
To sip server.
Sip server transfer sip message is to called end, after called end receives the sip message, calls SIPRTC local gateway, and
It is converted to WebRTC signaling, PeerConnection layers of setRemoteDescription interface is then called, then calls
PeerConnection layers of createAnswer interface finally calls SIPRTC local gateway by WebRTC SDP and candidate ground
Location message transformation is sent to sip server at sip message (180,200OK message).
Sip server transfer sip message is to calling terminal, after calling terminal receives the sip message, calls SIPRTC local gateway, and
It is converted to WebRTC signaling, then calls PeerConnection layers of setRemoteDescription interface.
Calling terminal replys sip message (ACK message) to called end.Two clients first attempt to establish P2P multimedia meeting at this time
Words, if failure will establish the Multimedia session by Asterisk media relay servers.The wherein multimedia of WebRTC
Stream is sent and received by SRTP (Security Real Time Protocol).
One of client call PeerConnection layers of removeRemoteMedia interface, and send sip message
(BYE message) gives another client.
After another client receives the sip message, also PeerConnection layers of removeRemoteMedia is called to connect
Mouthful, and send a sip message (200OK message).A Multimedia session is just completely completed at this time.
Claims (7)
1. a kind of communication means of the WebRTC based on Session Initiation Protocol characterized by comprising
Step 1: calling terminal collects candidate site and generates offer signaling；Candidate site and offer signaling are converted to by calling terminal
Sip request message is sent to sip server；Sip server forwards sip request message to called end；
Step 2: after called end receives sip request message, sip request message is converted to WebRTC signaling and candidate ground by called end
Location；Called end generates answer signaling；Candidate site and answer signaling are converted to SIP request again and receiveed the response by called end,
It is sent to sip server；Sip server forwarding SIP request is receiveed the response to calling terminal；
Step 3: after calling terminal reception SIP request is receiveed the response, after SIP transmission message is directly replied to called end, two clients
P2P or the Multimedia session by media relay servers are established in end；
Step 4: giving another client when one of client sends SIP termination messages, it is whole that another client receives SIP
Only message, and reply SIP termination and receive the response to starting client, then terminate the session of two clients.
2. the communication means of WebRTC based on Session Initiation Protocol described in accordance with the claim 1 a kind of, it is characterised in that:
It is to act on behalf of to collect by ICE that calling terminal, which collects candidate site, in the step 1；
It is by SIPRTC local network that candidate site and offer signaling are converted to sip request message by calling terminal in the step 1
Close realization.
3. the communication means of WebRTC based on Session Initiation Protocol described in accordance with the claim 1 a kind of, it is characterised in that:
Sip request message is converted to WebRTC signaling by called end in the step 2 and candidate site is by the local SIPRTC
What gateway was realized；
It is by calling the WebRTCPeerConnection layer of called end to connect that called end, which generates answer signaling, in the step 2
Cause for gossip is existing；
It is to pass through calling that candidate site and answer signaling are converted to SIP request and receiveed the response by called end again in the step 2
What SIPRTC local gateway was realized.
4. the communication means of WebRTC based on Session Initiation Protocol described in accordance with the claim 1 a kind of, which is characterized in that the step
Rapid 4 specifically:
When one of client call WebRTC PeerConnection layer interface and SIP termination messages are sent to another
Client, another client receive SIP termination messages, and call WebRTCPeerConnection layer interface to terminate media and pass
It is defeated, it then replys SIP termination and receives the response to starting client, then terminate the session of two clients.
5. according to the communication means of WebRTC based on Session Initiation Protocol described in claim 2-3 any one a kind of, feature exists
In the calling terminal, called end pass through the SIPRTC local gateway for turning mutually between SIP SDP and WebRTC SDP
Change and be mapped to sip message PeerConnection layers of WebRTC the corresponding interface.
6. according to the communication means of WebRTC based on Session Initiation Protocol described in claim 5 a kind of, which is characterized in that described
SIPRTC local gateway is embedded in client, and the client includes calling terminal and called end.
7. according to the communication means of WebRTC based on Session Initiation Protocol described in claim 6 a kind of, which is characterized in that described
SIPRTC local gateway uses PeerConnection layer interface and Session Initiation Protocol based on WebRTC C++API.
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| CN201711492037.0A CN109995734A (en) | 2017-12-30 | 2017-12-30 | A kind of communication means of the WebRTC based on Session Initiation Protocol |
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| CN201711492037.0A CN109995734A (en) | 2017-12-30 | 2017-12-30 | A kind of communication means of the WebRTC based on Session Initiation Protocol |
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Cited By (5)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| 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 |
| CN112769818A (en) * | 2021-01-05 | 2021-05-07 | 武汉球之道科技有限公司 | Video processing method based on webpage instant messaging and IP communication |
| 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 |
| CN113746808A (en) * | 2021-08-12 | 2021-12-03 | 杭州网易智企科技有限公司 | Converged communication method for online conference, gateway, electronic device, and storage medium |
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Application publication date: 20190709 |