WebRTC fixes/improvements:

- Added public host setting
- Implemented Sender Report based  track time syncing
- Added 10 second timeout for output connections (no timeout for input connections)
- Timing fixes

lib/rtp.cpp

@@ -616,6 +616,7 @@ namespace RTP{
     if (M.getCodec(tid) == "opus"){
       m = 48.0;
     }
+    bootMsOffset = M.getBootMsOffset();
     setProperties(M.getID(tid), M.getCodec(tid), M.getType(tid), M.getInit(tid), m);
   }
 
@@ -625,6 +626,24 @@ namespace RTP{
     cbInit = cbI;
   }
 
+  /// Improves A/V sync by providing an NTP time source
+  /// msDiff is the amount of millis our current NTP time is ahead of the sync moment NTP time
+  /// May be negative, if we're behind instead of ahead.
+  void toDTSC::timeSync(uint32_t rtpTime, int64_t msDiff){
+    if (!firstTime){return;}
+    uint64_t rtp64Time = rtpTime;
+    if (recentWrap){
+      if (rtpTime > 0x80000000lu){rtp64Time -= 0x100000000ll;}
+    }
+    uint64_t msTime = (rtp64Time - firstTime + 1 + 0x100000000ull * wrapArounds) / multiplier + milliSync;
+    int32_t rtpDiff = (bootMsOffset + msTime) - (Util::bootMS() - msDiff);
+    if (rtpDiff > 25 || rtpDiff < -25){
+      INFO_MSG("RTP difference (%s %s): %" PRId32 "ms, syncing...", type.c_str(), codec.c_str(), rtpDiff);
+      milliSync -= rtpDiff;
+    }
+
+  }
+
   /// Adds an RTP packet to the converter, outputting DTSC packets and/or updating init data,
   /// as-needed.
   void toDTSC::addRTP(const RTP::Packet &pkt){
@@ -636,6 +655,7 @@ namespace RTP{
     // This part isn't codec-specific, so we do it before anything else.
     int64_t pTime = pkt.getTimeStamp();
     if (!firstTime){
+      milliSync = Util::bootMS() - bootMsOffset;
       firstTime = pTime + 1;
       INFO_MSG("RTP timestamp rollover expected in " PRETTY_PRINT_TIME,
                PRETTY_ARG_TIME((0xFFFFFFFFul - firstTime) / multiplier / 1000));
@@ -651,7 +671,7 @@ namespace RTP{
       }
     }
     prevTime = pkt.getTimeStamp();
-    uint64_t msTime = ((uint64_t)pTime - firstTime + 1 + 0x100000000ull * wrapArounds) / multiplier;
+    uint64_t msTime = ((uint64_t)pTime - firstTime + 1 + 0x100000000ull * wrapArounds) / multiplier + milliSync;
     char *pl = (char *)pkt.getPayload();
     uint32_t plSize = pkt.getPayloadSize();
     bool missed = lastSeq != (pkt.getSequence() - 1);

lib/rtp.h

@@ -139,6 +139,7 @@ namespace RTP{
     void setCallbacks(void (*cbPack)(const DTSC::Packet &pkt),
                       void (*cbInit)(const uint64_t track, const std::string &initData));
     void addRTP(const RTP::Packet &rPkt);
+    void timeSync(uint32_t rtpTime, int64_t msDiff);
     virtual void outPacket(const DTSC::Packet &pkt){
       if (cbPack){cbPack(pkt);}
     }
@@ -148,6 +149,7 @@ namespace RTP{
 
   public:
     uint64_t trackId;
+    uint64_t bootMsOffset;
     double multiplier;    ///< Multiplier to convert from millis to RTP time
     std::string codec;    ///< Codec of this track
     std::string type;     ///< Type of this track
@@ -159,6 +161,7 @@ namespace RTP{
     bool recentWrap;      ///< True if a wraparound happened recently.
     uint32_t prevTime;
     uint64_t firstTime;
+    int32_t milliSync;
     void (*cbPack)(const DTSC::Packet &pkt);
     void (*cbInit)(const uint64_t track, const std::string &initData);
     // Codec-specific handlers