webrtc browser

The text of this text connection is: http://blog.csdn.net/freewebsys/article/details/47174209 not allowed to reprint without the Bo master.1, Encounter problemsFirst of all, WEBRTC is a very good open source project, it is a company that specializes in this, was acquired by Google and then open source projects.You can quickly build a video chat project, and you can compile it yourself.Https://github.com/pristineio/

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

original articles, Forbidden reprint. otherwise pursued. The information parsing of RTP header in WebRTC has been explained before. Here to explain the WEBRTC in the RTP parsing, here is the main explanation of h264 analysis; About class implementations and related test files that are relevant in VP8 and VP9,WEBRTC; Regarding the RTP file parsing of H264, t

reproduced in the original: Http://www.cnblogs.com/mod109/p/5767867.html#top thank you very much. The WEBRTC's audio processing module is divided into noise reduction ns (NSX), echo cancellation AEC (Echo control Acem), Audio gain AGC, and Mute detection section . In addition WEBRTC has encapsulated a set of audio processing module APM, if it is not a special need, if users want to use the echo cancellation and other more complex modules, it is best

Several questions1, WEBRTC transmit bandwidth is estimated for each stream or the total bandwidth2, WebRTC Remb is the overall bandwidth of statistics.3, if WEBRTC at the same time to watch the multi-channel flow, how to for each stream feedback bandwidth, packet loss and other information5, if the WEBRTC simultaneousl

[Transfer to]WEBRTC Learning: Deploying Stun and turn serversHttp://www.cnblogs.com/lingdhox/p/4209659.htmlThe WEBRTC-to-peer penetration part is implemented by Libjingle.The sequence of steps is probably this:1. Try direct Connect.2. Penetrate through the stun server3. Cannot penetrate through the turn server relay.Stun server is relatively simple. There are also many public stun servers available for test

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.Based on WEBRTC technology, you can achieve point-to-point audio and video, instant messaging, video conferencing, and the latest system components include:Teleice NAT Traversal server:Standard-based NAT traversal protocol ICE for NAT traversal, audio-video-to-peer transmissionSingle machine supports tens of thousands of concurrentTELEMCU Video Conferencing Server:Imple

Very interesting website http://io13webrtc.appspot.com/#1HTML5 about using WEBRTC http://www.html5rocks.com/en/tutorials/getusermedia/intro/HTML5 example of setting resolution https://simpl.info/getusermedia/constraints/HTML5 examples of various special effects http://webcamtoy.com/zh/app/Examples of HTML5 recordings http://www.webaudiodemos.appspot.com/AudioRecorder/index.htmlGoogle's latest HTML5-based code HTTPS://GITHUB.COM/GOOGLECHROME/WEBRTCSIPM

016-11-25 DORAWEBRTC series Wind net WEBRTC series Wind Net Source: Wind NET Series Author: Weizhenwei, fan network columnist Audio and video synchronization is the most intuitive user experience for multimedia products, and is the most basic quality guarantee for audio and video media data transmission and rendering playback. If the audio and video is not synchronized, it is possible to cause delay, lag, etc. that affect the user experience very much

It is not recommended to compile individual modules in the WEBRTC separately for use. Yesterday, I was fortunate enough to ask the Google forum about the delay in computing the AECM module, and Project member said churn this delay actually didn't help the AECM effect, which only sped up the convergence of the built-in latency estimator when the AECM started, and if the delay in the update was incorrect, it would even make AE The CM built-in delay

1, first look at the simplest SSE:Only use the SSE-enabled browser (most), the browser built-in EventSource object, the object by default three seconds to refresh the response data.HTML code (taken from W3cschool):DOCTYPE HTML>HTML>Head>Metahttp-equiv= "Content-type"content= "text/html; charset=utf-8" />Head>Body>H1>Get server-side update dataH1>DivID= "Result">Div>Script>if(typeof(EventSource)!=="undefined

Crosswalk QuickStart, using WEBRTC (HTML) to start developing video callsInstall PythonDownload the installer from http://www.python.org/downloads/After the installation is complete, add the environment variable again.Installing Oracle JDK Download page:http://www.oracle.com/technetwork/java/javase/downloads/Select the Java version to download (recommended Java 7). Select a JDK to download and accept the license agreement. Once downlo

Some time ago in the audio and video version of iOS, so the title changed to Android IOS WebRTC audio and Video development summary, the following summarizes some of the experience in the development process:1. iOS WEBRTC audio and video compilation and download: have android WebRTC compile download experience and then go to get IOS, you will find a lot easier, a

Some time ago in the audio and video version of iOS, so the title changed to the Android iOS WEBRTC audio and Video development summary, the following summary of some experience in the development process: 1. iOS WEBRTC audio and video compilation and download: Have the android WEBRTC compile download experience to get IOS, you will find that more simple, then th

When deploying WEBRTC or SIP-to-peer scenarios, you will often encounter environments that are not penetrated by peerThis is where the tunserver comes in.Here we use turnserver-0.7.3Download confuse dependent librarieswget http://savannah.nongnu.org/download/confuse/confuse-2.7.tar.gzTar zxvf confuse-2.7.tar.gzCD confuse*./configureMake make installDownloadwget http://downloads.sourceforge.net/project/turnserver/turnserver-0.7.3.tar.bz2Tar jxvf turns

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

H264 code Stream parsing, online has a lot of open source files; The general analysis is: Obtain Nalu,sps,pps,nalu type,slice type, obtain QP and so on; The computation can be obtained by the bitwise operation of C + +, but the structure can be obtained directly. Here is the WEBRTC in the H264 parsing Related: In the WEBRTC, about the H264 related source files in: webrtc58\src\

reproduced in the original: http://www.cnblogs.com/mod109/p/5469799.html thank you very much. The WEBRTC audio processing module is divided into noise reduction ns, Echo cancellation AEC(Echo control Acem), Automatic control gain AGC, Mute detection section. In addition WEBRTC has encapsulated a set of audio processing module APM, if it is not a special need, if users want to use the echo cancellation and

gcc-c++ gcc-g77 Flex Bison autoconf automake bzip2-devel zlib-devel ncurses-devel libjpeg-devel Libpng-dev El libtiff-devel freetype-devel pam-devel openssl-devel libxml2-devel gettext-devel pcre-devel3. Installation dependencies3.1 mac./scripts/installmacdeps.sh3.2 Ubuntu./licode/scripts/installubuntudeps.sh4. Installing Licode./scripts/installerizo.sh. /scripts/installnuve.sh5. Mounting Base Example./scripts/installbasicexample.sh6. Run Licode and examples, run at two terminals, or run in the

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