CN107682657A - A kind of multi-person speech video call method and system based on WebRTC - Google Patents
A kind of multi-person speech video call method and system based on WebRTC Download PDFInfo
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- CN107682657A CN107682657A CN201710822604.8A CN201710822604A CN107682657A CN 107682657 A CN107682657 A CN 107682657A CN 201710822604 A CN201710822604 A CN 201710822604A CN 107682657 A CN107682657 A CN 107682657A
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
- H04N7/00—Television systems
- H04N7/14—Systems for two-way working
- H04N7/141—Systems for two-way working between two video terminals, e.g. videophone
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
- H04N7/00—Television systems
- H04N7/14—Systems for two-way working
- H04N7/141—Systems for two-way working between two video terminals, e.g. videophone
- H04N7/142—Constructional details of the terminal equipment, e.g. arrangements of the camera and the display
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
- H04N7/00—Television systems
- H04N7/14—Systems for two-way working
- H04N7/141—Systems for two-way working between two video terminals, e.g. videophone
- H04N7/147—Communication arrangements, e.g. identifying the communication as a video-communication, intermediate storage of the signals
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Abstract
A kind of multi-person speech video call method based on WebRTC disclosed by the invention, comprises the following steps：User can initiate multi-way call request simultaneously by specifying connection room number Room ID and room-size n；By establishing P2P connections two-by-two between user, more people's video speech calls are finally realized；Communication mechanism based on WebRTC so that the system has the characteristics of operation is flexible, response is fast, low latency；, it is necessary to carry out a series of signaling exchange and SDP negotiation during P2P connections foundation；In face of the network environment of complexity, ripe NAT through-transmission techniques can make user's direct communication in non-same LAN.Exploitation of the invention based on Android platform, can more be widely used in a variety of mobile devices, improve applicability and flexibility, suitable for small-sized multi-person speech video calling.
Description
Technical field
The present invention relates to video calling field, more particularly to a kind of multi-person speech video call method based on WebRTC and
System.
Background technology
With the fast development of Internet technology and the communication technology, the exchange way of people has obtained greatly with exchanging content
Abundant and development.In the rhythm increasingly faster information age, not only efficiency seems have to traditional exchange way based on word
It is a little low, and the original idea of people can not accurately be expressed sometimes.So support the communication mode gradually prevailing of voice and video
Come.For the webpage instant communication of early stage, it is still desirable to download too fat to move and and unsafe plug-in unit.But WebRTC appearance,
It compensate for the deficiency of traditional instant messaging.
WebRTC, i.e. Web Real-Time Communication, it is that a supported web page browser carries out real-time voice
The technology of video calling.Directly application is exactly to allow developer to realize video calling or other point-to-point data for it
Transmission.WebRTC has a whole set of audio frequency and video solution, and code is increased income.In addition, WebRTC is also full platform branch
Hold.It is not only limited to page end, also provides the interface that Mobile Development uses.This is provided for the application of secondary development mobile terminal
Basis.
The content of the invention
The shortcomings that it is an object of the invention to overcome prior art and deficiency, there is provided a kind of multi-person speech based on WebRTC
Video call method, this method discard tradition streaming media communication mode, reduce communication delay, tackle complex network environment, improve
Consumer's Experience, user's form of communication is enriched, reduce system maintenance cost；And it is simple to operate flexible, there is higher information to pass
The speed passed and responded, suitable for most of mobile terminals.
Another object of the present invention is to provide a kind of multi-person speech video call system based on WebRTC.
The purpose of the present invention is realized by following technical scheme：
A kind of multi-person speech video call method based on WebRTC, is comprised the steps of：
Step 1, first user specify connection room number Room ID and room-size n, initiate the call of n-1 roads；
Step 2, first user gather the network information and local session information, and these connection data are sent
To server, receiving terminal is waited to establish connection therewith；
Step 3, second user specify same room number Room ID and room-size n, initiate the call of n-1 roads, simultaneously
Second user gathers the network information and local session information, as connection data；
Step 4, the second user, which choose wherein to converse all the way, is used as receiving terminal, and the call for responding first user please
Ask, establish connection therewith；Remaining n-1-1 roads call waits new receiving terminal to establish connection therewith simultaneously；
Step 5, repeat step 3,4, the 3rd user establish with first and second user connect respectively, circulate successively, finally
Realize interconnecting two-by-two between n client.
The step 1 is specially：First user specifies connection room number Room ID and room-size n, and is used as n-1
The originating end of road call, initiates call, waits receiving terminal to establish connection therewith；First user forms a Client,
Identified by unique Client Id；It is referred to as Instance per call all the way, is identified by unique Instance ID, be P2P
The least unit of connection, while according to prescribed coding form, start to gather local audio, video data.
The step 2 is specially：First user passes through NAT through-transmission techniques in per the Instance that converses all the way
The network information is gathered, as cross-network segment communicating；Local media description is gathered simultaneously, and by the network information and local media of collection
Description is sent to server, by transit server, waits receiving terminal to obtain.
The local media description includes the relevant parameter of audio frequency and video.
The step 3 is specially：Second user is by specifying connection room number Room ID and room-size n, from service
Device obtains connection data, and the user Client numbers m to be connected such as current (at present due to above only having a Client to initiate
Request, so m=1), while gather the network information and local media description information.
The step 4 is specially：Second user Client is in local n-1 Instance, random selection m
Instance is used as receiving terminal, and m Client is responded, and establishes P2P connections；Simultaneously again by remaining n-1-m
Instance initiates call request, waits new receiving terminal to establish connection therewith as originating end.
The step 5 is specially：Repeat step 3 and step 4, the 3rd user Client obtain first connection data and
The user Client numbers m (now m=2) to be connected such as current, respectively with first, second Client of locality connection data response,
P2P connections are established therewith；By that analogy, when n user Client connects room number Room ID and room by identical successively
After size n establishes connection, it will realize that n user interconnects two-by-two.
Another purpose of the present invention is realized by following technical scheme：
A kind of multi-person speech video call system based on WebRTC, including local audio video flowing acquisition module, P2P connect
Connect management module and more people's management modules；Wherein
Local audio video flowing acquisition module, it is responsible for collection local audio video flowing, and is encoded；Client terminal start-up is taken the photograph
As head and microphone, audio/video flow is obtained by prescribed coding form, and create audio track and track of video；
P2P connection management modules, the Signalling exchange being responsible between two clients, and then establish WebRTC connections；Often
One P2P connection is managed by an Instance, has unique Instance ID；
More people's management modules, it is responsible for coordinating under more people's environment, the Organization And Management of n-1 roads call；In current Client
In, there is n-1 Instance by being asked to server, the number of users m that room is had been added in current context can be obtained, its
Middle m<n；Then conversed on local Client Zhong n-1 roads, i.e., in n-1 Instance, randomly select the call conduct of m roads
The receiving terminal of P2P connections, meets simultaneously：For each Client, only response one of Instance；Remaining n-1-m
Road call waits receiving terminal to establish connection therewith as new originating end.
The present invention compared with prior art, has the following advantages that and beneficial effect：
1st, Android operation system is widely used, not only in mobile phone terminal, also ten on tablet personal computer and wearable device
Divide welcome.WebRTC is used for the exploitation of Android applications by the present invention, has both enriched Android communication exchanges means, again
So that WebRTC is promoted well.
2nd, NAT technologies, i.e. Network Address Translation technology, it is that a kind of internal network private IP address is converted to can be
The technology of the outside network address of Internet valid propagations.Because public network IP address is in short supply, so most of computer is place
In the network environment after NAT.But for WebRTC agreements, NAT can cause it to be failed in the environment of cross-network segment.Cause
This, the present invention carries out free communication on the outside of NAT, it is necessary to realize in order to solve the problem with inner side, i.e. NAT is penetrated, worn by NAT
This saturating technological means, preferably solves above-mentioned technical problem.Simple Traversal of UDP Through Network Address Translators or TURN agreements can be very good solve big portion
Divide NAT penetration problems.ICE agreements are exactly that the comprehensive NAT formed with reference to Simple Traversal of UDP Through Network Address Translators and this two parts of TURN agreements is penetrated
Solution.
Multimedia session describes SDP, be mainly used in WebRTC conversation initialization between client and client with
And Signalling exchange.The SDP information of collection is sent to server by both sides, then by transit server to other side.This process is logical
The offer/answer for crossing a pair of PeerConnection objects is operated to complete.
WebSocket agreements, it is a kind of instant messaging agreement.Its Socket being substantially built upon on Transmission Control Protocol connects
Connect, encapsulated in application layer, simplify interface and calling interface.WebSocket agreements can make client and service
The high-speed data channel of full duplex is established between device.In communication process, use is text-based message transmission data.With poll
Compared with long connection, also there is very big advantage in terms of transmission stability and transmitted data amount.
Signaling, apply during Coordinated Communication.In order to establish WebRTC communications, client both sides need to carry out a system
The Signalling exchange of row.Although two main frames in network can be enable directly to be communicated using WebRTC agreements, i.e., P2P leads to
Letter.But this does not imply that WebRTC does not need server.During the channel of data transfer is established, it is necessary to have server
Participated in.And signaling just plays such effect.
2nd, the present invention is to realize multi-person speech video calling based on WebRTC Android platform, by participating in conversing
User establish P2P connections two-by-two, given up high complicated streaming media server, reduced maintenance cost.
3rd, status is identical between each user of the present invention, and the mechanism for adding and leaving call is flexible.
4th, technical scheme has a low latency, it is simple to operate flexibly, cross-network segment, suitable for Android platform
Feature.In small-sized call scene, there is good Consumer's Experience.
Brief description of the drawings
Fig. 1 is more people's communication plan schematic diagrames.
Fig. 2 is P2P connection building process figures.
Fig. 3 is that local audio/video flow captures flow chart.
Fig. 4 is more people's call management structural representations.
Embodiment
With reference to embodiment and accompanying drawing, the present invention is described in further detail, but embodiments of the present invention are unlimited
In this.
A kind of multi-person speech video call method based on WebRTC, specific implementation comprise the following steps：
Step 1, first user specify connection room number Room ID and room-size n, initiate the call of n-1 roads.
Step 2, the user gather the network information and local session information, and these connection data are sent to service
Device, receiving terminal is waited to establish connection therewith.
Step 3, second user specify same room number Room ID and room-size n, initiate the call of n-1 roads.Simultaneously
The network information and local session information are gathered, as connection data.
Step 4, second user, which choose wherein to converse all the way, is used as receiving terminal, responds the call request of first user,
Connection is established therewith.Remaining n-1-1 roads call waits new receiving terminal to establish connection therewith simultaneously.
It is final real that step 5, repeat step 3, step 4, the 3rd user establish with first and second user connect respectively ...
Interconnecting two-by-two between existing multiple client.
Further, the step 1 is specially further：First user specifies connection room number Room ID and room
Size n, and initiate n-1 calls.The originating end of P2P connections is now appointed as per conversing all the way, waits receiving terminal to establish therewith
Connection.The user forms a Client, is identified by unique Client Id.It is referred to as Instance per call all the way, by only
One Instance ID marks, it is the least unit of P2P connections.Instance can both be used as originating end, can also be used as and connect
Receiving end, the role of originating end or receiving terminal is appointed as by more people's management modules.Simultaneously according to prescribed coding form, start camera
With microphone, local audio, video data is gathered.And renderer is loaded, local video data is showed.
Further, the step 2 is specially further：In per the calls of Instance all the way：Start and the service of burrowing
Device communicates, and by NAT through-transmission techniques, gathers the network information, such as place public network IP address and port information.And save as
ICECandidate examples, used cross-network segment communicating.Local media description information SDP is gathered simultaneously, such as the correlation ginseng of audio frequency and video
Number.This two parts information needs to call setLocalICECandidate () and setLocalSDP () method to be saved in local
In PeerConnection examples；HTTP POST requests are sent to server simultaneously, this two parts data is saved in service
Device, receiving terminal is waited to obtain.
Further, the step 3 is specially further：Second user Client is by specifying connection room number Room
ID and room-size n, initiate the call of n-1 roads, i.e. n-1 Instance.The user for currently having been added to the room is obtained first
Client numbers m.Because now only have first user to add room, m=1.According to prescribed coding form, the user starts
Camera and microphone, gather local audio, video data.And renderer is loaded, local video data is showed.Visit simultaneously
The server that burrows is asked, obtains the network information；Gather local video information.This two parts information save as ICECandidate examples and
SDP examples, it is stored in as locality connection data in each Instance PeerConnection examples.
Further, the step 4 is specially further：Second user Client in local n-1 Instance,
M Instance of random selection is appointed as receiving terminal role.And sound is done to m Client respectively using the connection data of local
Should, and the connection data that other side is stored in server are obtained, it is allowed to establish P2P companies with an Instance in each Client
Connect.Then, in the call of the n-1 roads of second user, choose without the n-1-m bar circuits for responding receiving terminal, i.e., remaining n-
1-m Instance.These Instance are respectively designated as to the role of transmitting terminal.More than collect locality connection data with
Same mode is sent to server, waits receiving terminal to establish connection therewith.
Further, the step 5 is specially further：Repeat step 3 and step 4.3rd user Client is same
Ground obtains the Client quantity m for having been added to the room, now m=2.After successful collection locality connection data, the Client with
M Instance of machine selection responds an Instance in m Client respectively, is allowed to establish P2P connections respectively.Equally
Ground, as n user, i.e. n Client, room number Room ID and room-size n are connected by identical successively and establish connection
Afterwards, it will realize that n user interconnects two-by-two.
A kind of multi-person speech video call method based on WebRTC, it is that WebRTC is based on based on one kind such as Fig. 1,4
Multi-person speech video call system realize, a kind of multi-person speech video call system based on WebRTC includes following mould
Block：
Local audio video flowing acquisition module, it is responsible for collection local audio video flowing, and carries out the work such as encoding.Such as Fig. 3,
Client terminal start-up camera and microphone, audio/video flow AudioSource/VideoSource is obtained by prescribed coding form.
And create audio track and track of video AudioTrack/VideoTrack.This two tracks need to be stored in same
In MediaStream examples.It is last again that MediaStream examples is associated with PeerConnection examples.
P2P connection management modules, the Signalling exchange being responsible between two clients, and then establish WebRTC connections.Often
One P2P connection is managed by an Instance, has unique Instance ID.Such as Fig. 2, it is following to establish P2P connections needs
Step：
The first step：Call request is initiated from originating end to server.Device inspection connection room number Room ID to be serviced and phase
After related parameter is errorless, originating end Instance will create PeerConnection examples.PeerConnection examples are to realize
P2P core instances, and Instance key component, save all information on connection.
Second step：Originating end Instance is collected and is preserved local SDP information.
3rd step：SDP information is sent to server --- This move is referred to as Offer.
4th step：Receiving terminal Instance obtains the SDP information of originating end, simultaneously with identical parameter access server
Collect local SDP information.Similarly, these information are all stored in receiving terminal Instance PeerConnection examples.
5th step：Receiving terminal Instance is by local SDP information response to originating end --- and This move is referred to as
answer。
6th step：After originating end and receiving terminal possess the SDP information of other side respectively, the P2P connections based on WebRTC with
Foundation.
Receiving terminal Instance obtains connection data from server, can be asked by Http Post, be one " drawing "
Action.And receiving terminal Instance responds originating end Instance, it is actively to be pass by data-pushing by WebSocket, is
One action of " pushing away ".If both ends are respectively in different networks, it is necessary to access STUN/TURN/ICE services first
Device, obtain oneself and be exposed to the information such as Internet IP address and port, save as ICE Candidate examples, as
A part for Signalling exchange, make to be in directly foundation communication between the main frame in different network environments.
More people's management modules, it is responsible for coordinating under more people's environment, the Organization And Management of n-1 roads call.In current Client
In, there is n-1 Instance.By being asked to server, the number of users m (m that room is had been added in current context can be obtained
<n).Then conversed on local Client Zhong n-1 roads, be i.e. in n-1 Instance, randomly select the call of m roads and connect as P2P
The receiving terminal connect.And need to ensure：For each Client, only response one of Instance.Remaining n-1-m roads
Call waits receiving terminal to establish connection therewith as new originating end.The module is responsible for specifying each in this Client
Instance role, ensure orderly between multidigit user to establish P2P connections.When certain P2P disconnecting all the way, the module
Connection resource will be reclaimed, angle setting of laying equal stress on color, and then ensure that user's exits reconnection.
Above-described embodiment is the preferable embodiment of the present invention, but embodiments of the present invention are not by above-described embodiment
Limitation, other any Spirit Essences without departing from the present invention with made under principle change, modification, replacement, combine, simplification,
Equivalent substitute mode is should be, is included within protection scope of the present invention.
Claims (8)
1. a kind of multi-person speech video call method based on WebRTC, it is characterised in that comprise the steps of：
Step 1, first user specify connection room number Room ID and room-size n, initiate the call of n-1 roads；
Step 2, first user gather the network information and local session information, and these connection data are sent to clothes
Business device, waits receiving terminal to establish connection therewith；
Step 3, second user specify same room number Room ID and room-size n, initiate the call of n-1 roads, while second
Position user gathers the network information and local session information, as connection data；
Step 4, the second user, which choose wherein to converse all the way, is used as receiving terminal, responds the call request of first user,
Connection is established therewith；Remaining n-1-1 roads call waits new receiving terminal to establish connection therewith simultaneously；
Step 5, repeat step 3,4, the 3rd user establish with first and second user connect respectively, circulate successively, final to realize
Interconnecting two-by-two between n client.
2. the multi-person speech video call method based on WebRTC according to claim 1, it is characterised in that the step 1
Specially：First user specifies connection room number Room ID and room-size n, and as the originating end of n-1 roads call, hair
Call is played, waits receiving terminal to establish connection therewith；First user forms a Client, by unique Client Id
Mark；It is referred to as Instance per call all the way, is identified by unique Instance ID, is the least unit of P2P connections, together
When according to prescribed coding form, start to gather local audio, video data.
3. the multi-person speech video call method based on WebRTC according to claim 1, it is characterised in that the step 2
Specially：First user gathers the network information in per the Instance that converses all the way, by NAT through-transmission techniques, is used as
Cross-network segment communicating；Local media description is gathered simultaneously, and the network information of collection and local media description are sent to server,
By transit server, receiving terminal is waited to obtain.
4. the multi-person speech video call method based on WebRTC according to claim 3, it is characterised in that the local matchmaker
Body describes the relevant parameter for including audio frequency and video.
5. the multi-person speech video call method based on WebRTC according to claim 1, it is characterised in that the step 3
Specially：Second user obtains connection data by specifying connection room number Room ID and room-size n, from server, with
And the user Client number m to be connected such as current, while gather the network information and local media description information.
6. the multi-person speech video call method based on WebRTC according to claim 1, it is characterised in that the step 4
Specially：For second user Client in local n-1 Instance, m Instance of random selection is used as receiving terminal, right
M Client is responded, and establishes P2P connections；Initiate call using remaining n-1-m Instance as originating end again simultaneously
Request, waits new receiving terminal to establish connection therewith.
7. the multi-person speech video call method based on WebRTC according to claim 1, it is characterised in that the step 5
Specially：Repeat step 3 and step 4, the 3rd user Client obtain connection data and the user to be connected such as current first
Client number m, respectively with first, second Client of locality connection data response, P2P connections are established therewith；By that analogy, when
After n user Client connects room number Room ID and room-size n foundation connections by identical successively, n use will be realized
Interconnect two-by-two at family.
A kind of 8. multi-person speech video call system based on WebRTC, it is characterised in that：Gathered including local audio video flowing
Module, P2P connection managements module and more people's management modules；Wherein
Local audio video flowing acquisition module, it is responsible for collection local audio video flowing, and is encoded；Client terminal start-up camera
With microphone, audio/video flow is obtained by prescribed coding form, and create audio track and track of video；
P2P connection management modules, the Signalling exchange being responsible between two clients, and then establish WebRTC connections；Each
P2P connections are managed by an Instance, have unique Instance ID；
More people's management modules, it is responsible for coordinating under more people's environment, the Organization And Management of n-1 roads call；In current Client, have
N-1 Instance can obtain the number of users m that room is had been added in current context, wherein m by being asked to server<
n；Then conversed on local Client Zhong n-1 roads, i.e., in n-1 Instance, randomly select the call of m roads and connect as P2P
The receiving terminal connect, meets simultaneously：For each Client, only response one of Instance；Remaining n-1-m roads call
As new originating end, receiving terminal is waited to establish connection therewith.
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