peerconnection

(traversal Traversal of the Using Relay NAT relay implementation). Ice will first use stun, try to establish a UDP-based connection, if it fails, it will go to TCP (try HTTP first, then try HTTPS), and if it fails, ice will use a trunk turn server.We can use Google's stun server: stun:stun.l.google.com:19302 So, an architecture that integrates the ice framework should look like this.Browser compatibleor a different prefix, using a method similar to the one above:var

information to establish a peer-to connection to deliver audio and video data. Due to the complexity of the network environment, not all clients can establish peer-to connection, in this case, there is a need for a relay server to do audio and video data transfer, this article in line with the analysis of the attitude of the source, this situation is not considered. Here explain, Stun/turn, relay server implementation in WEBRTC source code, there are examples, is really a big treasure trove.In

and video data. Due to the complexity of the network environment, not all clients can establish peer-to connection, in this case, there is a need for a relay server to do audio and video data transfer, this article in line with the analysis of the attitude of the source, this situation is not considered. Here explain, Stun/turn, relay server implementation in WEBRTC source code, there are examples, is really a big treasure trove.In the above sequence, the annotated scene is Clienta to ClientB t

almost all the features that may be used. Combined with the label, you can display any local video streams and remote video streams that you want to display.Fortunately, Android also has the same API, although their names are different. The Android-related APIs are videocapturerandroid, Videorenderer, MediaStream, Peerconnection, and Peerconnectionfactory. Here we will explain each of them individually.Before you start, you need to create a peerconn

peer-to connection to deliver audio and video data.Because of the complexity of the network environment, not all of the client can establish peer-to-peer connection, so that the situation requires a relayserver to do audio and video data transfer, this article is based on the source code analysis of the attitude, this situation is not considered.Here to illustrate, Stun/turn, Relayserver's real today WEBRTC source code has a demo sample, is really a big treasure of the name of the deputy de fac

(event.stream); }; Offer and answer var SENDOFFERFN = function (desc) {pc.setlocaldescription (DESC); Console.log (' offer-----'); Socket.emit (' message ', Json.stringify (DESC)); }, SENDANSWERFN = function (desc) {pc.setlocaldescription (DESC); Console.log (' Answer-----'); Socket.emit (' message ', Json.stringify (DESC)); }; Get local audio and video stream Navigator.webkitgetusermedia ({"Audi

The most recent time you've been doing WEBRTC-based Android apps has encountered some problems releasing resources, which are now recorded for memos.The official Apprtcdemo is too simplistic and many questions are not involved.1. Releasing the peerconnection resource problem.Scenario: A and B make a call (Video call)Now stop the call in B.Error: After B terminates the call, the terminal a program will exit unexpectedly.Analysis: When A and b make a ca

The proxy class for peerconnection is defined in the Trunk\talk\app\webrtc\peerconnectionproxy.h file, class Peerconnection:public Peerconnectioninterface, while # define BEGIN_PROXY_MAP (c) \Class C # #Proxy: Public C # #Interface所以BEGIN_PROXY_MAP (peerconnection) expands toclasspeerconnection:publicpeerconnectioninterface. So the talk\examples\peerconnection\c

is that some core technologies such as HTML, HTTP, and TCP/IP are open and free to implement. At present, there is no free, high quality, complete solution in the field of browser communication. WEBRTC is such a technology. The technology has been integrated with the best audio and video engines and deployed to several millions terminals, after more than 8 years of discipline. Google does not charge for this technology. Includes key NAT and firewall traversal techniques using stun, ICE,

Scene:1, A call B2, B answer3, A connected with BCommon steps:Both A and B need to initialize the WEBRTC module to create the PeerconnectionfactoryStatus of a in step 11. Create Peerconnection instances through Peerconnectionfactory2. Call Peerconnection.createoffer3, PeerConnection.Observer.onCreateSuccess (final sessiondescription ORIGSDP)4. Send SDP to B5, the following is the collection of Icecandidate, send the mobile phone icecandidate informati

src\webrtc\examples\peerconnection\client\conductor.ccboolconductor::initializepeerconnection()1 webrtc::createpeerconnectionfactory ();src\talk\app\webrtc\peerconnectionfactory.cc1.1 New Rtc::refcountedobject1.2 bool Peerconnectionfactory::initialize ()1.2.1 cricket::mediaengineinterface* media_engine =Peerconnectionfactory::createmediaengine_w()src\talk\media\webrtc\webrtcmediaengine.cc1.2.1.1mediaengineinterface* webrtcmediaenginefactory::create (w

WEBRTC reply content: I am in development and have a basic understanding of the WebRTC source code stack. It mainly consists of two key technologies: 1. webRTC Video/Voice Engine, including camera microphone operations, Video preprocessing, VP8 coding/decoding, and streaming media transmission (RTP/RTCP); 2. implement the P2P channel and use libjingle to completely implement the ICE, STUN, and TURN protocols to provide P2P communication capabilities. A set of JS APIs, including getUserMedia and

thread that is Peerconnectionfactory::createpeerconnection_ The Signaling_thread set in the S function. So in the main thread call Peer_connection_->setremotedescription, because of the role of the agent (specifically see Proxy.h file), call this function is to send a message with Signaling_thread, and binds the message to its implementation, the Setremotedescription method of the Peerconnection class associated with the proxy class, so that the mess

("video, audio", onusermediasuccess, onusermediaerror); console. log ("requested access to local media with old syntax. ");} catch (e) {alert (" webkitgetusermedia () failed. is the mediastream flag enabled in about: Flags? "); Console. log ("webkitgetusermedia failed with exception:" + E. message) ;}} function onusermediasuccess (Stream) {console. log ("user has granted access to local media. "); var url = webkiturl. createobjecturl (Stream); localvideo. style. opacity = 1; localvideo. src = U

first understand the WebRTC browser's basic interface and the process of a realtime video call through the browser-side WEBRTC Video Call process.　　WEBRTC structureFirst of all, from the WEB-RTC aspect analysis WEBRTC on the browser interface structure, in the browser side, WEBRTC mainly implemented three interfaces MediaStream, to achieve the encapsulation of local audio and video resources, such as from camera, microphone, remote stream and so on, MediaStream represents a media data

1 SDP negotiation nack and action on video JitterbufferPeerconnection at Createoffer, collect session control information, audio and video codec information and network information on this side. The video codec information is obtained from the WebRtcVideoEngine2. The last offer is to form an SDP message, which is sent to the network via the Peerconnection object.After receiving an offer, the receiver first calls Setremotedescription, creating a Video