Various fixes, among which:

- Fixed segfault when attempting to initialseek on disconnected streams
- Fix 100% CPU bug in controller's stats code
- WebRTC UDP bind socket improvements
- Several segfault fixes
- Increased packet reordering buffer size from 30 to 150 packets
- Tweaks to default output/buffer behaviour for incoming pushes
- Added message for load balancer checks
- Fixed HLS content type
- Stats fixes
- Exit reason fixes
- Fixed socket IP address detection
- Fixed non-string arguments for stream settings
- Added caching for getConnectedBinHost()
- Added WebRTC playback rate control
- Added/completed VP8/VP9 support to WebRTC/RTSP
- Added live seek option to WebRTC
- Fixed seek to exactly newest timestamp
- Fixed HLS input

# Conflicts:
#	lib/defines.h
#	src/input/input.cpp

lib/rtp.h

@@ -77,6 +77,7 @@ namespace RTP{
     Packet(const char *dat, uint64_t len);
     const char *getData();
     char *ptr() const{return data;}
+    std::string toString() const;
   };
 
   /// Sorts RTP packets, outputting them through a callback in correct order.
@@ -163,6 +164,9 @@ namespace RTP{
     h265::initData hevcInfo;            ///< For HEVC init parsing
     Util::ResizeablePointer fuaBuffer;  ///< For H264/HEVC FU-A packets
     Util::ResizeablePointer packBuffer; ///< For H264/HEVC regular and STAP packets
+    uint64_t currH264Time;//Time of the DTSC packet currently being built (pre-conversion)
+    Util::ResizeablePointer h264OutBuffer; ///< For collecting multiple timestamps into one packet
+    bool h264BufferWasKey;
     void handleH264(uint64_t msTime, char *pl, uint32_t plSize, bool missed, bool hasPadding);
     void handleH264Single(uint64_t ts, const char *buffer, const uint32_t len, bool isKey);
     void handleH264Multi(uint64_t ts, char *buffer, const uint32_t len);