paper deeply analyzes the implementation details of WEBRTC internal about packet loss retransmission (NACK), makes an in-depth study of NACK's SDP negotiation, packet loss determination and retransmission, and lays the foundation for continuing to learn the QoS mechanism of WEBRTC.Reference documents[1] RFC5109-RTP Payload Format for Generic Forward Error Correction.[2] rfc5104-rfc 5104-codec Control Messages in the RTP audio-visual profiles with Fee
QuickTime plugin, but we want a pure browser video stream.Another notable option is Flash Player, which can receive rtmp streams obtained through Wowza conversion rtsp/rtp/h.264. But Flash Player is also a browser plugin, although it is more popular than VLC and QuickTime.In our scenario, we test the same RTSP/RTP stream, but use the WEBRTC compatible browser as the player to play the video without any additional plugins. We set up a conversion serve
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
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 allo
ObjectiveBefore writing the article "separate compilation using WEBRTC audio processing module-Android", I have been trying to compile the WEBRTC audio processing engine Voe the whole to use for his project, but limited to the poor technology, time is tight, so did not succeed. The AECM, AGC, NS, and VAD modules in the engine were compiled separately to make it w
Compile WebRTC For Android code in Ubuntu 14.04
Recently, a real-time communication project for audio and video chats was developed based on Google's open-source WebRTC project. Some problems were encountered during the download of WebRTC code, which was recorded here, we also hope to help the children's shoes who enco
Recently, our team is developing a app to help people solve problem face to face.We Choose WEBRTC Protocol as our bridge among different platform (Android, IOS, browser etc).But there are a hole in Android 6.0 system, the protocol can not support Android 6.0 system.As we known, Libjingle (Http://mvnrepository.com/artif
Ubuntu14.04 compile WebRTC For Android code 2014-07-24, ubuntu14.04webrtc
I haven't written a blog for almost a year. Recently, I developed an instant messaging project based on Google's open-source WebRTC project. During this project, I encountered some problems when downloading WebRTC code, this is a record here, an
This article mainly introduces WEBRTC in each platform debug or log viewing mode, to facilitate troubleshooting, including Bs,pc,android,ios (this series of articles reproduced please indicate the source, blog Park rtc.blacker).1, Browser development:This development method does not need to download and compile WEBRTC source code (many people are "dead" here, but
video and audio separately?Answer 8: Universal coding scheme is: Video using H264, audio using AAC; If the end-to-end is controllable,It is recommended to use H265 for higher compression rate;Question 9: What is the recommended video conferencing system in the third scenario?Answer 9: If you are interested, you can see Licode.Question 10: How many people does the development team of the third scenario have, and how long is the development cycle generallyAnswer 10: This is not a lot of people, t
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 WebRT
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 WEB
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 wind
This article original from Http://blog.csdn.net/voipmaker reprint annotated source.The latest version of TELEMCU adds Android phone-side WEBRTC video conferencing capabilities, Android phone installs Chrome browser after loading TELEMCU WEBRTC client TeleWeb can directly participate in video conferencing,At the same ti
Continue with the unfinished part of the previous article, including the following three sections:1, extension: WEBRTC multiparty calls.2,mcu Multipoint Control Unit.2, Extension: VOIP, telephone, message communication.Note: Translation is not verbatim, but in accordance with their own understanding of the translation, at the same time, in order to facilitate understanding, but also to join some of their own organization language.Reprint please indica
The most recent time you've been doing WEBRTC-based Android apps has encountered some problems releasing resources, which are now recorded for memos.The official Apprtcdemo is too simplistic and many questions are not involved.1. Releasing the peerconnection resource problem.Scenario: A and B make a call (Video call)Now stop the call in B.Error: After B terminates the call, the terminal a program will exit
1, about WEBRTCWebRTC is a very popular project. The first problem encountered is the WEBRTC compilation problem.Fortunately, a company has helped compile and put it in Maven's repo.Address:Http://mvnrepository.com/artifact/io.pristine/libjingleThe update is very fast, and WEBRTC the official Basic sync update.2,android DemoThe project is also within the pristine
This article is mainly their own previous research WEBRTC code structure when some information (including ANDROID,IOS,PC), the article from the blog Garden Rtc.blacker, reproduced please explain the source.1, WEBRTC module: Audio data acquisition, sending, receiving, playback call process:2, WEBRTC module: Video data a
, the receiver side decoding good performance, no mosaic phenomenon.3.2, adding the QoS module will bring a certain delay and lag, because packet retransmission is time-required.3.3, the above plan is WEBRTC inside the nack concrete realization way.The above scheme is provided by Peng Zuyuan, a senior audio and video expert from the ring, with some adjustments, and Kelly for editing and finishing.Peng has many years of audio and video codec developmen
This article mainly introduces to help a programmer solve WEBRTC doubt process, the article from the blog Garden Rtc.blacker, support original, reprint please explain the source (www.rtc.help)This article mainly comes from the mail, why I will be specially organized into essays, mainly based on the following reasons:1, the author email me The purpose is to ask questions, but he asked questions in a way worthy of praise, asked very specific (if asked t
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