webrtc demo

"Getting Started with WebRTC" The first chapter WebRTC introduction?This chapter is a conceptual introduction to WEBRTC.after reading this chapter. You will have a clear understanding of the following: . What is WEBRTC . How to use it . which browsers support1.1. WEBRTC IntroductionWorld Wide Web (WWW) is the early day

one of the objects.3. JavaScript Proposal/Response negotiation controlThe local browser only focuses on two specific calls:// 将我的会话描述告知我的浏览器pc.setLocalDescription(mySessionDescription)...// 将对等端的会话描述告知我的浏览器pc.setRemoteDescription(yourSessionDescription)Generate a proposal, answer://Generate proposalsPC.Createoffer(Gotoffer,Didntgetoffer)function Gotoffer(asessiondescription){ setlocaldescription(asessiondescription) ...the session description (proposed offer) can now be sent to the peer so t

-point and do not need to be brokered by the server. But this does not mean that we can abandon the server, we still need it to pass signaling (signaling) to build this channel. WEBRTC there is no protocol defined for signaling to establish a channel: signaling is not part of the Rtcpeerconnection API SignalingSince there is no protocol to define the specific signaling, we can choose any way (AJAX, WebSocket), use arbitrary protocol (SIP, XMPP) to pas

This paper mainly introduces the RTP/RTCP protocol in WEBRTC, Weizhenwei, the earliest published articles in the Wind network, ID:BEFOIOSupport original, reprint must indicate the source, welcome attention to my public number blacker (Id:blackerteam or WEBRTCORGCN).a prefaceThe RTP/RTCP protocol is the cornerstone of streaming media communications. The RTP protocol defines the packet format for streaming media data over the Internet, while the RTCP pr

lot of time this way the effect is very good,This only takes up a lot of bandwidth and CPU, especially for mobile phones.Complete mesh topology: Everyone is connected to each otherIn addition, the WEBRTC client can select a client to send streaming data directly to other clients, in this star network structure, you can directly do a publishing service side, the client will stream to the server,The server is then pushed to each client to relieve clien

first, the network topology structureWEBRTC can also be used as multiparty calls, such as video conferencing, in addition to peer-to-peer communication. When it comes to multi-party calls, we need to select a schema for our application. This is a very important decision, because how to organize users is related to the scale of the conference system. Corresponding to WEBRTC, there are two common network topologies: Mesh networks and star-shaped netwo

this This paper mainly introduces the realization of WEBRTC in Nack, Weizhenwei, the article was first published in the Wind network , Id:befoioSupport original, reprint must indicate the source, welcome attention to my public number blacker (Id:blackerteam or WEBRTCORGCN).In WEBRTC, forward error correction (FEC) and packet loss retransmission (NACK) are important methods to resist network errors. FEC adds

This article mainly introduces WEBRTC (we translate and collation, translator: Weizhenwei, check: Blacker), the earliest published in the "Weaving wind net"Support original, reprint must indicate the source, welcome attention to my public number blacker (Id:blackerteam or WEBRTCORGCN).Technically speaking, using a webcam for online broadcasting does not require WEBRTC. The camera itself is a server that can

Which framework or library is the best for use WebRTCHttp://stackoverflow.com/questions/24857637/current-state-of-javascript-webrtc-librariesI want to know the which framework or library is the best for use WebRTC. Here are a small and incomplete list of libraries/sdk out there. Any lib that I forgot, feel free to let me know:Libraries: Simplertc Rtcmulticonnection Crocodilertc Lynckia/

, which are point-to-point and do not need to be brokered by the server. But this does not mean that we can abandon the server, we still need it to pass signaling (signaling) to build this channel. WEBRTC there is no protocol defined for signaling to establish a channel: signaling is not part of the Rtcpeerconnection APISignalingSince there is no protocol to define the specific signaling, we can choose any way (AJAX, WebSocket), use arbitrary protocol

, the audio effect of the demo can be, the video effect is still not ideal.III) csipsimple1) SIP stack is used Pjsip, audio and video codec used in the third-party library has ffmpeg (video), silk (audio), WebRTC. The WEBRTC echo algorithm is used by default. supports the ICE protocol.2) Advantages and disadvantages:The Csipsimple architecture is clear, the SIP p

monitoring, so as to completely discard desktop software. The development of WebRTC applications requires a complete solution, including signaling servers, STUN servers and TURN servers, development libraries for mobile phones and web pages, etc. Currently, this is the best technology import period. Look at our products: http://yacamera.com We have previously prepared the doubango set, WebRTC and SIP, all

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