webrtc to rtmp

Take a UAV project, responsible for video playback this piece, choose is video.js this video plug-in, this thought can open happy heart development, who how material online about this part of the information so little, give me this hand party gave a great pressure. Okay, don't talk nonsense.The project needs to achieve a four-way player, can play the drone shot real-time screen, and can do local refresh without affecting the entire page, and to be able to dynamically switch the source of the vid

In the RTMP Audio Video preview transmission, the video to the Crtmpserver server, and then play with the Flashbuilder, playback failure, and then use the FMS server for transmission, on the flashbuilder can play normally, So suspect that the problem is on the Crtmpserver server side, open the log of the Crtmpserver server, found this line of warning: Default implementation of Processinvokegeneric:request: _ CHECKBW, immediately found in the source of

RTMP is designed for transport network streaming, requires support from servers such as Fms,awaza, and provides better copyright protection for streaming media content, and it also needs to pay royalties to adobe itself. First, the two work differently: RTMP data requires a dedicated server to receive, such as FMS, Awazal, etc., and then play through the local Flash player. The HTTP protocol can transf

Since the beginning of WEBRTC development, often asked by others, Safari browser can support WEBRTC? I also want Safari to support WEBRTC so you don't have to write native WebRTC apps or Safari plugins.Happily, Apple recently announced that WEBRTC will enter Safari and iO

Reproduced in the original: http://blog.csdn.net/shichaog/article/details/52399354 thank you very much. Vad (voice Activity Detection) algorithm is to detect the voice , in the far-field speech interaction scenario, VAD faces two challenges: 1. How to successfully detect the lowest energy voice (sensitivity).2. How to successfully detect in the multi-noise environment (detection rate and false detection rate).The missed response is originally the voice but not detected, and the virtual detection

http://blog.csdn.net/nonmarking/article/details/47910043 This series is currently a total of three articles, follow up will also update WebRTC Videoengine Ultra-Detailed tutorial (i)--the basic process of video Call WebRTC Videoengine Ultra-Detailed tutorial (ii)--integrated OPENH264 codec WebRTC Videoengine Ultra-Detailed tutorial (iii)-- Overview of Integrate

In WebRTC's example project, there is an Android project called Apprtcdemo, which enables video calling (VoIP) on a wide area network. This article is intended to demonstrate the compilation of Apprtcdemo, with Windows as an example, but also for Mac and Linux. Switch to a Linux environment please specify what platform you are currently using, and if it is Linux, you can ignore this step; otherwise, you will need a virtual machine. I'm using damn windows, and I recommend vagrant, a lightweight v

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

Author: Gustavo Garcia (original link) Translation: Liu Tong Bandwidth estimation is probably the most important part of the WEBRTC video engine. The task of the Bandwidth estimation (BWE) module is to determine how much video stream you can send and not network congestion to ensure that video quality is not reduced. In the previous bandwidth estimation algorithm is very basic, in general, based on the design of packet loss. Usually we start to slowl

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

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

: This article mainly introduces how to use nginx + nginx-rtmp-module + ffmpeg to build a streaming media server (6). If you are interested in the PHP Tutorial, refer to it. Part 6 This problem has been plagued by the nginx startup problem that has been transplanted to the ARM Development board a few days ago. it does not occur when started on the ARM Development Board. nginx: [emerg] getgrnam("nogroup") failed (2: No such file or directory) Is to a

Recently made a RTMP transit service program, through the practice, familiar with rtmp play and push in various formats, summarized here.Program GitHub Address: Https://github.com/runner365/rtmp_relayRTMP Play Receive Message analysisMessage received at first frame:1) 0x46 4c: Refer to the following article:Character FLV Header2) 0x01 05Version typeflagsreserved Typeflagsaudio typeflagsreserved typeflagsvi

Reprint Please specify source: http://www.cnblogs.com/fangkm/p/4374668.htmlThe previous blog introduces the local video collection, this article introduces the audio capture process, but also introduces WEBRTC native audio collection, and then introduces chromium source to its customization.1. WebRTC native Audio captureLet's first introduce the interface concepts in WE