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webrtc: Fix whipclientsink name in README

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The element name was changed, but the documentation wasn't updated to
match.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1764>
This commit is contained in:
Arun Raghavan 2024-09-03 16:43:43 -04:00
parent 871756bb70
commit e72db57179
1 changed files with 3 additions and 3 deletions
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6
net/webrtc/README.md
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@ -296,7 +296,7 @@ Testing the whip client as the signaller can be done by setting up janus and
``` shell
gst-launch-1.0 -e uridecodebin uri=file:///home/meh/path/to/video/file ! \
videoconvert ! video/x-raw ! queue ! \
whipwebrtcsink name=ws signaller::whip-endpoint="http://127.0.0.1:7080/whip/endpoint/room1234"
whipclientsink name=ws signaller::whip-endpoint="http://127.0.0.1:7080/whip/endpoint/room1234"
```
You should see a second video displayed in the videoroomtest web page.
@ -310,7 +310,7 @@ The WHIP Server as the signaller can be tested in two ways.
Note: The initial version of `whipserversrc` does not check any auth or encryption.
Host application using `whipserversrc` behind an HTTP(s) proxy to enforce the auth and encryption between the WHIP client and server
#### 1. Using the Gstreamer element `whipwebrtcsink`
#### 1. Using the GStreamer element `whipclientsink`
a. In one tab of the terminal start the WHIP server using the below command
@ -322,7 +322,7 @@ b. In the second tab start the WHIP Client by sending a test video as shown in t
``` shell
RUST_BACKTRACE=full GST_DEBUG=webrtc*:6 GST_PLUGIN_PATH=target/x86_64-unknown-linux-gnu/debug:$GST_PLUGIN_PATH gst-launch-1.0 videotestsrc ! videoconvert ! video/x-raw ! queue ! \
whipwebrtcsink name=ws signaller::whip-endpoint="http://127.0.0.1:8190/whip/endpoint"
whipclientsink name=ws signaller::whip-endpoint="http://127.0.0.1:8190/whip/endpoint"
```
#### 2. Using Meetecho's `simple-whip-client`