webRTC is implemented PeerConnection as per https://apprtc.appspot.com/
How does webRTC implement synchronization of the their audio and video streams from remote?
Normal RTP a/v sync done using RTCP SR/RR reports and the timestamps in each SRTP packet.
See any VoIP application.
Related
How to capture rtp stream from webrtc then convert it to hls to broadcast to client ?
I want to receive rtp from webrtc in browser via media server (eg kurento ... ) then convert it to hls stream. User can use hlsEndpoint to play.
WebRTC -> RTP -> HLS
What is the correct way?
My aim is to create a live stream app that supports push streams using webrtc , i'm working with rtmp , i want webrtc as an additional option.
Tks all.
Just use a media server to covert WebRTC to Live Streaming like RTMP, HTTP-FLV or HLS, please read this wiki.
Because the WebRTC is not only RTP, but also need to transcode the audio from opus to aac, and there is something like the jitter-buffer, NACK or packet out-of-order to handle.
For live streaming, the RTMP is the de-facto standard in live streaming industry, so if you covert WebRTC to RTMP, you got everything, like transcoding by FFmpeg, forwarding to YouTube, or DVR to file, etc.
If you need to convert WebRTC to HLS or RTMP you may check Ant Media Server
The community edition also provides this.
How can i make an audio stream captured in browser using WebRTC to be streamed live via icecast/shoutcast protocols?
Use liquidsoap + webcaster.js:
https://github.com/webcast/webcaster
Currently, I'm using open source code from OBS multiplatform for my video streaming application. But OBS uses RTMP for streaming, which only works on TCP. I want to use UDP since this is a live streaming scenario where I don't desire latency.
What is the best way to move to UDP? I read that protocols like RTP support UDP. Is this the best option to use? How about writing a simple UDP socket code to send and receive video data? Please guide.
The reason for trying to move to UDP is to ensure minimum latency between client and server. I'm looking for a quickest way to move to a udp based streaming than the current RTMP which uses TCP. I'm not very experienced in networking related coding. Will a simple socket program suffice to stream a video data continuously or should I implement a protocol like RTP? Or any other alternative? Ultimately, streaming should be smooth with minimum latency.
can anyone help me what is the technical difference between WebRTC communication and the VoIP communication?
The question doesn't exactly make sense because it makes the assumption that VoIP is a technical stack, but it's not - it's a concept. The concept of sending Voice (V) over (o) Internet Protocols (IP). This means that different technology stacks can be used for accessing/capturing the media, establishing connections, negotiating streams, and transmitting streams.
WebRTC is one such stack (set of APIs, methods, and standards) for VoIP.
VOIP - Voice over Internet Protocol was a concept which came with popularity of internet. This involved using the internet to route voice telephony data, basically using existing IP infrastructure to transport audio streams without having dedicated circuit switched telephony. Over the time popular VOIP applications like Skype, Vonage and many in enterprise telephony came in.
VOIP had two parts one signalling , basically controller part and other actual media.
Actual media usually but not necessarily followed RTP (Real Time) protocol. RTP could carry both voice and video. Problem with RTP has been that browsers don't support it natively and it is not secure. You usually needed some sort of plugin to have VOIP work inside browser.
With WebRTC now popular browsers like FF, Chrome and Opera support a variation of RTP which is secure and can be natively invoked. Using WebRTC and browser Javascript you can send Voice, Video and Screen (it's video only) data to any other browser, which is cool.
VOIP : Voice over internet protocol uses DSL/Cable Modem voice over Wi-Fi/3G (VoWiFi/3G), voice over LTE (VoLTE), and Rich Communication Suite (RCS). VoIP is cloud-based, calls are sent as digital data and no cables are needed to send the call so any kind of Internet connection can be used to make calls and from a plethora of devices.
Web RTC: Web Real time communication use only OS browsers to communicate.
WebRTC requires the use of two main component JavaScript APIs.
WebRTC is an extension of VoIP to the browser world. It can reuse the existing VoIP infrastructure with incremental upgrades. This is good news for VoIP, as adoption of WebRTC only serves to increase overall VoIP proliferation.
Also, WebRTC is ideal for low-cost browser-based contact center applications. VoIP can serve embedded operator-driven VoLTE applications. Consequently, between WebRTC and VoIP, they can support wide range of consumer and enterprise applications.
For a research purpose, I developed and app to control a wheeled mobile robot using the gyro and the accelerometer of an iPhone. The robot has a IP address, and I control it by sending messages through a socket. Since the robot has to be controlled from anywhere in the world, I mounted a camera on top of it. I tried to stream the video from the camera using the http live streaming protocol and vlc, but the latency is too high (15-30sec) to properly control it.
Now, vlc has the possibility to stream over udp or http, but the point is how do I decode the stream on the iPhone? How should I treat the data coming into the socket in order to visualize them as a continuous live video?