Skip to content
New issue

Have a question about this project? Sign up for a free GitHub account to open an issue and contact its maintainers and the community.

By clicking “Sign up for GitHub”, you agree to our terms of service and privacy statement. We’ll occasionally send you account related emails.

Already on GitHub? Sign in to your account

WebRTC SDP Config #1514

Open
Minims opened this issue Dec 17, 2024 · 1 comment
Open

WebRTC SDP Config #1514

Minims opened this issue Dec 17, 2024 · 1 comment
Labels
question Further information is requested

Comments

@Minims
Copy link

Minims commented Dec 17, 2024

Hello,

I'm trying to configure go2rtc to read a WebRTC stream from a Somfy Visiophone.
I will use echo plugin as my source is changing on every stream request.

I'm receiving a SDP like this, how can i setup the stream in go2rtc ? Thanks !

v=0
o=- * * IN IP4 0.0.0.0
s=-
t=0 0
a=fingerprint:sha-256 *
a=extmap-allow-mixed
a=group:BUNDLE 0 1
m=video 9 UDP/TLS/RTP/SAVPF 96
c=IN IP4 0.0.0.0
a=setup:actpass
a=mid:0
a=ice-ufrag:*
a=ice-pwd:*
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42e01f;packetization-mode=1;
a=ssrc:* cname:pion
a=ssrc:* msid:pion video
a=ssrc:* mslabel:pion
a=ssrc:* label:video
a=msid:pion video
a=sendrecv
a=candidate:* 1 udp * IP_INTERNE PORT typ host
a=candidate:* 2 udp * IP_INTERNE PORT typ host
a=candidate:* 1 udp * IP_EXTERNE PORT typ srflx raddr 0.0.0.0 rport PORT
a=candidate:* 2 udp * IP_EXTERNE PORT typ srflx raddr 0.0.0.0 rport PORT
a=candidate:* 1 udp * IP_VPN PORT typ relay raddr 0.0.0.0 rport PORT
a=candidate:* 2 udp * IP_VPN PORT typ relay raddr 0.0.0.0 rport PORT
a=end-of-candidates
m=audio 9 UDP/TLS/RTP/SAVPF 111
c=IN IP4 0.0.0.0
a=setup:actpass
a=mid:1
a=ice-ufrag:*
a=ice-pwd:*
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:111 opus/48000/2
a=ssrc:* cname:pion
a=ssrc:* msid:pion audio
a=ssrc:* mslabel:pion
a=ssrc:* label:audio
a=msid:pion audio
a=sendrecv
@AlexxIT AlexxIT added the question Further information is requested label Dec 18, 2024
@AlexxIT
Copy link
Owner

AlexxIT commented Dec 18, 2024

There is no way to add a new webrtc source without making edits to the go2rtc source code.
I just haven't figured out how to implement it. The echo source won't help here.
WebRTC connection is an interactive process.

Sign up for free to join this conversation on GitHub. Already have an account? Sign in to comment
Labels
question Further information is requested
Projects
None yet
Development

No branches or pull requests

2 participants