That doesn’t come without downsides though. We need a way to make it smaller, and the answer is video compression. With those numbers, a 4-person conference call isn’t going to happen. To store 30 minutes of uncompressed 720 8-bit video you need about 110 GB. They are all characteristics of your network that you will need to overcome. These constraints are all caused by the limitations of the real world. The more latency you are willing to tolerate, the higher quality video you can expect. Real-time media is about making trade-offs between latency and quality. It gives you the bi-directional communication necessary to respond to changing network conditions. It is also used to handle packet loss and to implement congestion control. This is used to communicate statistics about the call. The format is very flexible and allows you to add any metadata you want. RTCP (RTP Control Protocol) is the protocol that communicates metadata about the call. It also gives you the timing and ordering information you need to feed a media pipeline. ![]() RTP gives you streams, so you can run multiple media feeds over one connection. It does not stipulate any rules around latency or reliability, but gives you the tools to implement them. It was designed to allow for real-time delivery of video. RTP (Real-time Transport Protocol) is the protocol that carries the media. WebRTC uses two preexisting protocols RTP and RTCP, both defined in RFC 1889. WebRTC responds to network conditions and tries to give you the best experience possible with the resources available. The protocol is designed to handle all of this. Maybe you suddenly experience lots of packet loss. During a call your bandwidth might increase, or decrease. ![]() WebRTC is also designed to handle dynamic network conditions. The underlying transport supports everything, even things that don’t exist yet! However, the WebRTC Agent you are communicating with may not have the necessary tools to accept it. These streams could all be independent, or they could be bundled together! You could send a video feed of your desktop, and then include audio and video from your webcam. You can add and remove these streams at anytime during a call. WebRTC allows you to send and receive an unlimited amount of audio and video streams. # What do I get from WebRTC’s media communication? TMMBR, TMMBN, REMB and TWCC, paired with GCC.Identifying and Communicating Network Status.Adaptive Bitrate and Bandwidth Estimation.Full INTRA-frame Request (FIR) and Picture Loss Indication (PLI).What do I get from WebRTC’s media communication?.
0 Comments
Leave a Reply. |