Webrtc ios framework compilation and Webrtcios framework Compilation1. WebRTC iOS framework Selection
Currently, two active open-source WebRTC implementations are available.
Google WebRTC
WEBRTC IOS Framework compilation http://www.th7.cn/Program/IOS/201502/390418.shtml
WebRTC in webkit:http://www.webrtcinwebkit.org/
OPENWEBRTC is designed for flexibility and modularity. The bulk of the API layer is implemented in JavaScript, making it super fast to modify and extend with new functionality. Below is a simplified sketch of the architecture.
OPE
Selection of iOS framework for 1.WebRTCCurrently two more active open source WEBRTC implementations.
Google WebRTC:
Project address is: https://code.google.com/p/webrtc/
Ericsson OPENWEBRTC:
Project address is: HTTPS://GITHUB.COM/ERICSSONRESEARCH/OPENWEBRTCOur Camp David Education is designe
In fact, as early as June 2 ago, a friend who worked at Google told me this information. I also got all the source code from WebRTC for the first time, but since my recent work was really busy, this information was not immediately reproduced here. Now, I want to keep an eye on the multimedia applications, learn the technologies in the WebRTC Framework earlier, an
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
WEBRTC Voice Overall framework
Figure One voice overall frame diagram
As shown above, the entire processing frame of the audio is responsible for the transmission of the peer data in addition to the Ligjingle, mainly the Voe (Voice Engine) and the channel adaptation layer
Figure II Creating a data communication channel timing diagramThe image above is the local sideComplete process, Voe is created by Creat
extension protocols, including the RTP protocol-based scaling, and the RTCP protocol based on the message type extension, and so on. [2] The details can be found in the reference literature.three WEBRTC thread relationships and data flowWEBRTC provides two threads: signal and worker, the former responsible for signaling data processing and transmission, the latter responsible for the processing and transmission of media data. Within
node:pc2 = new webkitRTCPeerConnection(servers);pc2.onaddstream = gotRemoteStream;//...function gotRemoteStream(e){ vid2.src = URL.createObjectURL(e.stream);}Rtcpeerconnection and serversIn the real world, WebRTC needs servers, but when it's simple, the following may be true:
You can find each other by user name.
WebRTC clients to Exchange network information.
Exchange media data informat
must be strictly licensed and can only be called when the user interface is displayed.
A detailed discussion of WebRTC security is beyond the scope of this article, and if you want to learn more about this, take a look at the WebRTC security Architecture provided by the IETF.Developer Tools
when the WEBRTC session is created, Chrome://
must be strictly licensed and can only be called when the user interface is displayed.
A detailed discussion of WebRTC security is beyond the scope of this article, and if you want to learn more about this, take a look at the WebRTC security Architecture provided by the IETF.Developer Tools
when the WEBRTC session is created, Chrome://
ability to create and manage sessions. This layer protocol is left to the application developer to customize the implementation. (5)VoiceengineThe audio engine is a framework that includes a range of audio multimedia processing, ranging from video capture cards to network-based transmission solutions. Ps:voiceengine is one of WEBRTC's most valuable technologies and is open source for Google's acquisition of Gips company. On VoIP, the technology indus
"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
Google open real-time communication framework WEBRTC source code
In fact, as early as June 2, in Google work friends told me this message, I also first time to get WEBRTC all source code, but because the recent work is busy, not the first time in this reprint this information. Now, I hope to pay attention to the peer of multimedia application, can learn the techn
Uncover the mystery of WEBRTC Media server--WEBRTC Media Server Open Source project IntroductionThe WEBRTC ecosystem is very large. When I first tried to understand WEBRTC, the number of network resources was unbelievable. This article provides some introduction to WEBRTC m
WEBRTC Introduction and simple Application
WebRTC, web Real-time communication, Web real-time communication technology. In short, the introduction of real-time communication in a Web browser, including audio and video calls.
WEBRTC Real-time communication technology Introduction
How to use
Media Introduction
Signaling
Stun
Introduction:First declare I was just a small intern, if there is not correct, I hope you help correct me.First, WEBRTC basic structureFigure A WEBRTC overall structure, from Baidu EncyclopediaFirst of all, WEBRTC the general realization of the idea: we create a web app, and then call in the app's JS Api,js API will invoke the C + + layer API in the browser, the
WEBRTC provides a point-to-point channel between browsers for data transmission, it is necessary for the server to be involved in establishing this channel. WEBRTC requires the server to support four aspects of its functionality:1. User Discovery and communication2. Signaling Transmission3. nat/Firewall traversal4. If the point-to-point communication fails to establish, it can be used as a transit server n
communicate between the browser and the server.Rtcdatachannel is a completely different approach:* It can establish point-to-point interconnection through the rtcpeerconnection API. Because there is no need for a mediation server, the median "hop count" is reduced and the latency is lower.* Rtcdatachannel uses the stream Control transmission Protocol (SCTP) protocol, allowing us to configure delivery semantics: We can configure the order of packet transfers and provide some configuration for re
, it is not necessary.The corresponding signaling server also needs to do a little bit of setup: Edit Collider/collidermain/main.go, modify the settings of your own room server URL://var roomSrv = flag.String("room-server", "https://apprtc.appspot.com", "The origin of the room server")var roomSrv = flag.String("room-server", "http://apprtc.diveinedu.com:8080/", "The origin of the room server")After these simple room servers and the custom settings of the signaling server, we built a simple servi
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