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rpicamsrc: webrtc example: Add a STUN server to the configuration

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To let the webrtc example work through NAT firewalls
This commit is contained in:
Jan Schmidt 2018-06-21 22:50:28 +10:00 committed by Tim-Philipp Müller
parent b333e32e18
commit 41f41f1fdd
1 changed files with 5 additions and 3 deletions
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8
tests/examples/rpicamsrc/webrtc-unidirectional-h264.c
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@ -15,6 +15,7 @@
#define RTP_PAYLOAD_TYPE "96"
#define SOUP_HTTP_PORT 57778
#define STUN_SERVER "stun.l.google.com:19302"
@ -150,7 +151,8 @@ const gchar *html_source = " \n \
\n \
window.onload = function() { \n \
var vidstream = document.getElementById(\"stream\"); \n \
playStream(vidstream, null, null, null, null, function (errmsg) { console.error(errmsg); }); \n \
var config = { 'iceServers': [{ 'urls': 'stun:" STUN_SERVER "' }] }; \n\
playStream(vidstream, null, null, null, config, function (errmsg) { console.error(errmsg); }); \n \
}; \n \
\n \
</script> \n \
@ -182,8 +184,8 @@ create_receiver_entry (SoupWebsocketConnection * connection)
G_CALLBACK (soup_websocket_message_cb), (gpointer) receiver_entry);
error = NULL;
receiver_entry->pipeline = gst_parse_launch ("webrtcbin name=webrtcbin "
"rpicamsrc bitrate=300000 annotation-mode=12 ! video/x-h264,profile=baseline,width=640,height=480 ! queue max-size-time=100000000 ! h264parse ! "
receiver_entry->pipeline = gst_parse_launch ("webrtcbin name=webrtcbin stun-server=stun://" STUN_SERVER " "
"rpicamsrc bitrate=600000 annotation-mode=12 preview=false ! video/x-h264,profile=constrained-baseline,width=640,height=360,level=3.0 ! queue max-size-time=100000000 ! h264parse ! "
"rtph264pay config-interval=-1 name=payloader ! "
"application/x-rtp,media=video,encoding-name=H264,payload="
RTP_PAYLOAD_TYPE " ! webrtcbin. ", &error);