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rpicamsrc: webrtc example: Add a STUN server to the configuration
To let the webrtc example work through NAT firewalls
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1 changed files with 5 additions and 3 deletions
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@ -15,6 +15,7 @@
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#define RTP_PAYLOAD_TYPE "96"
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#define RTP_PAYLOAD_TYPE "96"
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#define SOUP_HTTP_PORT 57778
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#define SOUP_HTTP_PORT 57778
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#define STUN_SERVER "stun.l.google.com:19302"
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@ -150,7 +151,8 @@ const gchar *html_source = " \n \
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\n \
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\n \
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window.onload = function() { \n \
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window.onload = function() { \n \
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var vidstream = document.getElementById(\"stream\"); \n \
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var vidstream = document.getElementById(\"stream\"); \n \
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playStream(vidstream, null, null, null, null, function (errmsg) { console.error(errmsg); }); \n \
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var config = { 'iceServers': [{ 'urls': 'stun:" STUN_SERVER "' }] }; \n\
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playStream(vidstream, null, null, null, config, function (errmsg) { console.error(errmsg); }); \n \
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}; \n \
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}; \n \
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\n \
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\n \
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</script> \n \
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</script> \n \
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@ -182,8 +184,8 @@ create_receiver_entry (SoupWebsocketConnection * connection)
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G_CALLBACK (soup_websocket_message_cb), (gpointer) receiver_entry);
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G_CALLBACK (soup_websocket_message_cb), (gpointer) receiver_entry);
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error = NULL;
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error = NULL;
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receiver_entry->pipeline = gst_parse_launch ("webrtcbin name=webrtcbin "
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receiver_entry->pipeline = gst_parse_launch ("webrtcbin name=webrtcbin stun-server=stun://" STUN_SERVER " "
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"rpicamsrc bitrate=300000 annotation-mode=12 ! video/x-h264,profile=baseline,width=640,height=480 ! queue max-size-time=100000000 ! h264parse ! "
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"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 ! "
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"rtph264pay config-interval=-1 name=payloader ! "
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"rtph264pay config-interval=-1 name=payloader ! "
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"application/x-rtp,media=video,encoding-name=H264,payload="
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"application/x-rtp,media=video,encoding-name=H264,payload="
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RTP_PAYLOAD_TYPE " ! webrtcbin. ", &error);
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RTP_PAYLOAD_TYPE " ! webrtcbin. ", &error);
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