webrtc server

attend the meeting2, A and B establish A connection3, B and C establish the connection4, B forward a audio and video to c,b forward C audio and video to aThis situation in the case of B equipment performance is high, and a and C performance is weak, with B as a bridge to achieve 3-party calls, thus reducing the burden on the server. applicable Scenario : This model is only suitable for meetings of 3 people.B. forwarding via

A recent study on how WebRTC speech runs on iOS found that the voice_engine of WebRTC has implemented iOS-related classes, but encountered a series of problems in specific applications. After several days of hard work, finally, we solved a series of problems and successfully realized recording and playing local loop in the simulator. Compile the testProgramIn the process, we plan to use the libjingle Libr

I have only recently started to study webrtc deeply. If I have any questions, please leave a message.How to generate WEBRTC vs engineering under Windows see my last article.But when I modify the project, such as adding cc and H files, adding a third-party dependency library, you will find that the VS modification is useless. VS can only be used when a code reader and editor are available.This time we need t

Turn from: http://blog.csdn.net/lixiaowei16/article/details/53407010 Audio and video synchronization is related to the most intuitive user experience of multimedia products, audio and video media data transmission and rendering playback of the most basic quality assurance. If the audio and video is not synchronized, it may cause delays, such as cotton, etc. very affect the user experience phenomenon. Therefore, it is very important. Generally speaking, the audio and video synchronization maint

The bandwidth assessment (BWE) is perhaps the most critical module in the WEBRTC video engine, which determines the amount of video data that can be generated when network congestion is not raised in video traffic. Early bandwidth assessment algorithms are relatively primitive, mostly based on packet loss estimation, the basic strategy is to gradually increase the amount of data sent, until the loss of packets detected. In order for the sender to lear

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

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

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.When using WEBRTC Base library to develop Android,ios native application, because the mobile terminal is not like PC side, in bandwidth stability, system performance is very big difference, so for mobile device WEBRTC need to do some optimization to improve the call effect,For example, WE