webrtc: Change H264 examples to use aggregate-mode=zero-latency for best compatibility

The default changed back to none because it broke existing code.

See gst-plugins-good#749

Part-of: <!22>

webrtc/janus/janusvideoroom.py

@@ -78,7 +78,7 @@ from gi.repository import GstSdp
 if DO_VP8:
     ( encoder, payloader, rtp_encoding) = ( "vp8enc target-bitrate=100000 overshoot=25 undershoot=100 deadline=33000 keyframe-max-dist=1", "rtpvp8pay picture-id-mode=2", "VP8" )
 else:
-    ( encoder, payloader, rtp_encoding) = ( "x264enc", "rtph264pay", "H264" )
+    ( encoder, payloader, rtp_encoding) = ( "x264enc", "rtph264pay aggregate-mode=zero-latency", "H264" )
 
 PIPELINE_DESC = '''
  webrtcbin name=sendrecv stun-server=stun://stun.l.google.com:19302

webrtc/janus/rust/src/janus.rs

@@ -66,7 +66,7 @@ const VP8: VideoParameter = VideoParameter {
 const H264: VideoParameter = VideoParameter {
     encoder: "x264enc tune=zerolatency",
     encoding_name: "H264",
-    payloader: "rtph264pay",
+    payloader: "rtph264pay aggregate-mode=zero-latency",
 };
 
 impl std::str::FromStr for VideoParameter {

webrtc/sendonly/webrtc-unidirectional-h264.c

@@ -179,7 +179,7 @@ create_receiver_entry (SoupWebsocketConnection * connection)
       gst_parse_launch ("webrtcbin name=webrtcbin stun-server=stun://"
       STUN_SERVER " "
       "v4l2src ! videorate ! video/x-raw,width=640,height=360,framerate=15/1 ! videoconvert ! queue max-size-buffers=1 ! x264enc bitrate=600 speed-preset=ultrafast tune=zerolatency key-int-max=15 ! video/x-h264,profile=constrained-baseline ! queue max-size-time=100000000 ! h264parse ! "
-      "rtph264pay config-interval=-1 name=payloader ! "
+      "rtph264pay config-interval=-1 name=payloader aggregate-mode=zero-latency ! "
       "application/x-rtp,media=video,encoding-name=H264,payload="
       RTP_PAYLOAD_TYPE " ! webrtcbin. ", &error);
   if (error != NULL) {