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webrtc: fix documentation after signaller interface changes

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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1175>
This commit is contained in:
Loïc Le Page 2023-04-12 20:19:22 +02:00
parent 8f2273328b
commit dba91bceca
2 changed files with 2 additions and 2 deletions
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2
net/webrtc/README.md
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@ -132,7 +132,7 @@ can be accessed through the `gst::ChildProxy` interface, for example
with gst-launch:
``` shell
gst-launch-1.0 webrtcsink signaller::address="ws://127.0.0.1:8443" ..
gst-launch-1.0 webrtcsink signaller::uri="ws://127.0.0.1:8443" ..
```
### Enable 'navigation' a.k.a user interactivity with the content

2
net/webrtc/gstwebrtc-api/README.md
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@ -130,7 +130,7 @@ $ gst-launch-1.0 videotestsrc ! agingtv ! webrtcsink meta="meta,name=native-stre
By default *webrtcsink* element uses *ws://127.0.0.1:8443* for the signalling server address, so there is no need
for more arguments. If you're hosting the signalling server elsewhere, you can specify its address by adding
`signaller::address="ws[s]://[signalling server]"` to the list of *webrtcsink* properties.
`signaller::uri="ws[s]://[signalling server]"` to the list of *webrtcsink* properties.
Once the GStreamer pipeline launched, you will see the registration of a new producer in the logs of the signalling
server and a new remote stream, with the name *native-stream*, will appear on the webpage.