Debugging WebRTC media stream and codec issues
I wrote a WebRTC server and tested it using a WebRTC client on Edge, playing through the WHEP protocol. The SDP exchange for media information is working correctly, as are the STUN and DTLS handshakes, with no issues. I use libsrtp to encrypt and decrypt RTP packets.
Debugging WebRTC media stream and codec issues
I wrote a WebRTC server and tested it using a WebRTC client on Edge, playing through the WHEP protocol. The SDP exchange for media information is working correctly, as are the STUN and DTLS handshakes, with no issues. I use libsrtp to encrypt and decrypt RTP packets.
Debugging WebRTC media stream and codec issues
I wrote a WebRTC server and tested it using a WebRTC client on Edge, playing through the WHEP protocol. The SDP exchange for media information is working correctly, as are the STUN and DTLS handshakes, with no issues. I use libsrtp to encrypt and decrypt RTP packets.
How to search for WebRTC issues on the web?
I wrote a WebRTC server and tested it using a WebRTC client on Edge, playing through the WHEP protocol. The SDP exchange for media information is working correctly, as are the STUN and DTLS handshakes, with no issues. I use libsrtp to encrypt and decrypt RTP packets.