we are fiddling around with WebRTC in our company. And i stumbled upon a weird thing, which i'm not sure is by design in WebRTC or an implementaiton error on our side. We currently have a simple W...

How to implement SIP protocol in Android ? there is any SDK or library available to implement it easily into Android?

I have a server which receives OFFER, ANSWER, CANDIDATE-s from web browser for a WebRTC session and passes to its peer. Later the data is also passed from the same server. Now to implement our own ...

I've been googling a way to change codec in Chrome's implementation of WebRTC, but there doesn't seem to be a way. How can I change the default codec used(audio or video) in a WebRTCpeer connectio...

I get RTP stream from WebRTC server (I used mediasoup) using node.js and I get the decrypted RTP packets raw data from the stream. I want to forward this RTP data to ffmpeg and from there I can sav...

How to enable H264 on Android WebRTC. PeerConnection to createOffer there was no h264 description in SDP.

In a SIP video call, the receiver of the video stream respond with the capabilities of its decoder. The parameter which defines that is the profile-level-id. Here is an example value of the profil...

Identify h264 profile and level from profile-level-id in sdp? How does one identify what the constraints actually mean? For example I have a profile-type-id: 42801e that translates to: How am I...

the gstreamer webrtc demo works fine.but all demo has a small problem: all webrtcbin that created offer must have some video/audio data to send. i want use webrtcbin create offer,and only receive v...

I want to have a method at the end that can set VP9 or H.264 as preferred video codec in a SDP. So I am looking for the m line in my SDP: m=video 9 UDP/TLS/RTP/SAVPF 96 98 100 102 127 97 99 101 1...

I hope there is no flaw in the logic. Step 1: caller creates offer Step 2: caller sets localDescription Step 3: caller sends the description to the callee //------------------------------------...

Is it possible to use same SDP in multiple peer connections? I'm building video conference using WebRTC. The idea is that caller, using some signaling mechanism, send broadcast message to all othe...

I would like to know what are the meaning of this SDP lines as I am trying to get the smoothest framerate posible with 5% to 10% packet losses. The lines i don´t know are: a=rtcp-fb:100 goog-remb ...

I'm using webRTC to build a system which supports audio calls. Here's how it works: - User A createOffer, then setLocalDescription with the offer - User B receiveOffer, then setRemoteDescription wi...

So, I have been working with FFMPEG on a project that involves streaming video from one computer to another across the internet with RTP. I want to take that into ffmpeg and use ffserver to display...

I am dealing with a RTCPeerConnection (pc) which has an event handler named onnegotiationneeded. The "onnegotiationneeded" is triggered when a complete media stream is added or removed with pc.add...

Now I test webrtc communicate with SIP Client(sx20) I send invite message with webrtc sdp. but sip client answer has not finger print, and sip client answer is not SRTP just RTP. So I need to t...

Is it possible to force a TCP tunneled (TLS) connection with WebRTC? We are developing a WebRTC application for our business, but we are experiencing some major issues with incoming UDP streams ca...

I am trying to setup video chat where two peer connections exchange video. This happens after creating a data channel. So this is the process of events: offerer creates data channel, offerer crea...

I am trying to get the audio and video from a WebRTC stream and handle it (transcode or dump) with ffmpeg on ubuntu server. I have naively expected it to simply interpret the sdp offered by WebRTC,...

I am working on a WebRTC client and I would like to allow the clients to modify the ongoing audio/video session to add or remove an audio or video stream. So for instance if there is an ongoing a...

I'm trying to understand what is the required parameter in SDP to be able to decode H264 from RTP packets. This is an related to this question, for the answer to that one works only in small numbe...

I'm looking for an example of a minimum necessary SDP for setting up a H264 video stream.| The assumption is that the receiver can play H264 as long as it gets the required parameters through SDP. ...

I'm trying to use WebRTC for purely decentralised and peer-to-peer communications. I'm trying to build a P2P overlay network, wherein nodes exchange details of other nodes so that they may connect ...

I'm using ffmpeg to create a streaming. It works fine. I have a server and with ffplay I can watch my stream. My only (big) constraint is real time. I have to embed it into an HTML page accessible...