of authentication methods.Long-term credentialShort-term credentialSpecific reference to stun standard http://tools.ietf.org/html/rfc5389#section-15.4But for WEBRTC. Only long-term is supported.Http://www.ietf.org/proceedings/87/slides/slides-87-behave-10.pdfTURN REST Server APIThis PDF describes the. Turn the interaction process between the server and the client.4. Simple to useTurnserver-o-a-f-v--mobilit
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 technology of
Original link: Introduction to WebRTC on Android
Original Author: Dag-inge Aas
Translated by: appear.in
Translator: Dorisminmin
Status: Complete
WebRTC is regarded as a New of web long-term open source development, and is the most important innovation in web development in recent years. WEBRTC allows web developers to add video chats or point
This article in order to remember for the audio and video communications to make outstanding contributions to the young talent-Lei Hua, really jealous talent!!!A bit sad at the beginning, as a work on the front-line it technical personnel, heard the news is always a bit bad, if you are fortunate to read this article please remember: Pay attention to rest, work is not finished, the body is the capital of the revolution. The last blog interaction with Comrade Ray is as follows:650) this.width=650;
What is WEBRTC?It is well known that the browser itself does not support the direct establishment of communication channels between each other through the server to relay. For example, there are now two clients, A and B, they want to communicate, first need a and server, B and server to establish a channel between. A to send a message to B, a first send the message to the server, the server to a message to relay, sent to B, the reverse is the same. So
implementation program.2. Write the video, voice chat background, using Nodejs as the service side, completed the channel login, signaling exchange, channel establishment and other functions of code writing. and completed the client login request, signaling send, signaling save, channel establishment, camera and microphone call code writing. Passed through the web-side test, but all Android 4.0-4.4 are not available.3. Access the blocked Google Brows
Transferred from: http://www.cnblogs.com/fangkm/p/4370492.htmlReprint Please specify source: http://www.cnblogs.com/fangkm/p/4370492.htmlThe previous article simply introduced the next WEBRTC protocol process, which begins with the introduction of frameworks and interfaces.When it comes to frames, instinctively don't know where to start. Once directly from the chromium project on the integration of the source of W
Switch from using WEBRTC to build front-end video chat room--Data channel ChapterIn two browsers, it is very complex to send messages for chat, games, or file transfers. Usually, we need to set up a server to forward the data, of course, the larger the size of the case, will be expanded into multiple data centers. In this case, there is a high latency and it is difficult to guarantee the privacy of the data.These issues can be addressed through the Rt
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
Previous notes, finishingWEBRTC in the default open RTX for packet loss retransmission, the introduction of RTX can refer to Rfc4588,https://tools.ietf.org/html/rfc4588#section-4RTX uses an additional SSRC transmission, SSRC is identified in the SDP.↵a=rtpmap: rtx/90000↵a2736695910239189782Like this.A RTX packet, in Turnserver, is such that the raw UDP data->turn/stun protocol header->RTP Header1->RTP header2In RTP header1, according to payload type to distinguish RTP, RTX data, if it is rtx, yo
Some personal understanding about WEBRTC
Just participated in the sound network presided over the first WEBRTC conference in Beijing, coupled with reading "hundred asked Freeswtich" written by Daniel, to it has more understanding, record for later review:
1, simple understanding, WEBRTC is a way to achieve web-to-business dial audio and video telephony technolog
We have introduced WebRTC and read the example of https://apprtc.appspot.com/on the Internet (which may need to be accessed through a wall). This example is an application deployed on the Google App Engine and relies on the Gae environment, the background language is Python and also relies on the Google App Engine channel API. Therefore, it cannot be run locally or be expanded. After studying the Python source code in the example, I decided to impleme
This article original from Http://blog.csdn.net/voipmaker reprint annotated source.WEBRTC provides real-time, web-based audio and video data interoperability, but WEBRTC can also run as a native app on a mobile platform, WEBRTC is a set of media frameworks, implemented in C + +, and officially ported to mobile platforms, including Android,ios, Platform-corresponding development language can be directly deve
Google's Turnserver download method:svn checkout http://rfc5766-turn-server.googlecode.com/svn/branches/v3.2/ Rfc5766-turn-server-read-onlyAbout the application of WEBRTC Google gives an example:https://apprtc.appspot.com/(need FQ, sometimes fq may not be able to land, it is estimated that the use of too many people)I was always curious about the way he used turn, and then finally figured out what was going on. Take a look at the following characters:
This article original from Http://blog.csdn.net/voipmaker reprint annotated source.WEBRTC Client access to the IMS network, to achieve interoperability with the IMS client, because the standard difference requires a gateway device, Telemedia Media Server to achieve this function, responsible for WEBRTC media layer conversion, including DTLS-SRTP to RTP, and opus
This article original from Http://blog.csdn.net/voipmaker reprint annotated source.The WEBRTC standard customizes how real-time media is transmitted to and from the web, but there is no specification for multiplayer video and is also a challenge for WEBRTC in enterprise solutions, WEBRTC technical VideoThe conference programme can be summed up in several1. Mesh S
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/
1. Introduction
It is well known that the browser itself does not support the direct establishment of communication channels between each other through the server to relay. For example, there are now two clients, A and B, they want to communicate, first need a and server, B and server to establish a channel between. A to send a message to B, a first send the message to the server, the server to a message to relay, sent to B, the reverse is the same. So a message between A and b through
Transferred from: http://www.cnblogs.com/lingyunhu/p/4232348.htmlToss for one weeks to finally set up the environment of Kurento (development stage of the product, some bugs to solve their own), so write a separate document to introduce.The following starts to introduce Kurento, the article from the blog Garden Rtc.blacker, reproduced please explain the source.What is Kurento?Video conferencing involves a one-to-many, many-to-many, broadcast, transcoding, mixing, co-screen, recording, which requ
WebRTC really is not very good to get, currently only the PC-side web page and mobile phone-side web page video. But there are still some problems. 1, both must use Firefox 2, feel pc-side camera shot out of the screen can also, the phone side a little bit of spending 3, enter the room after a period of time to show two video ~~~~APPRTC demo has not been tuned, the problem in Turnserver , and then sent the article. There are a lot of APPRTC on the Int
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