In my webrtc-applications (peers in javascript, Windows, Firefox, my signalling server in C++):
peer1 starts the connection (sends offer) add tracks to the connection, peer2 gets offer and sends answer, adds own track.
All works good if user grants both peers all right to media (camera, mic) – getUserMedia resolved OK.
I’d like to create a “limited” connection: peer1 will not have rights for camera and mic, perr2 will have.

The problem is: peer1 can not call addTrack(), because it has not rights and tracks to send. So, negotiationneeded is not fired for it.

The question is: how can peer1 can force negotiationneeded event to start a connection process – send your offer etc ?

I’m trying to implement screen recording in my React Native app using WebRTC, and I want to upload the recorded video in chunks to AWS S3. I’m facing challenges with both the recording and the chunked upload process. Any guidance or sample code would be appreciated!

I am building WebRTC using version M120 with the command:
gn gen out/clang/win_x64 –args=”target_os=”win” target_cpu=”x64″ is_component_build=false is_clang=true is_debug=false rtc_use_h264=true ffmpeg_branding=”Chrome” rtc_include_tests=false rtc_build_examples=false use_custom_libcxx=true” –ide=vs2022.

According to the MDN, RtcPeerConnection.removeTrack() “Tells the local end of the connection to stop
sending media from the specified track, without actually removing the corresponding RTCRtpSender
from the list of senders as reported by getSenders()”.
First I add video and audio track, webrtc connection works fine.

Error reporting member RTCIceServer. When using WebRTC on a Mac computer, the URL is required and must be an instance of (DOMString or sequence)
I hope to have a normal conversation, but I can’t. This is the most headache inducing problem for me

I am using JSSIP to implement WebRTC phone, and I have an trouble when a video call comes in (the SDP contains m=video). I want create SDP to contain only audio,not m=video.But always createAnswer sdp have m=video and a=recvonly.

have a 2 Google Nest Cam (Battery) and a 2 (Cable) . I would get the stream live of the cam. I use SDM API and Device Access, but I get this error:

I tried to fetch Windows WebRTC using the command:

[Bug]: [2024/05/23 15:19:40:0909] W: [wsicli|5a|WS/h1/default/m-579da5db.kinesisvideo.ap-southeast-]: lws_client_ws_upgrade: got bad HTTP response ‘403’ 2024-05-23 15:19:40.091 VERBOSE lwsWssCallbackRoutine(): WSS callback with reason 1 2024-05-23 15:19:40.091 WARN lwsWssCallbackRoutine(): Client connection failed. Connection error string: HS: ws upgrade response not 101

In our webrtc application we experience too high values of jitterBufferDelay which causes out of sync audio and video.