diff --git a/CMakeLists.txt b/CMakeLists.txt
index d0a5a69f..d24ac2a7 100644
--- a/CMakeLists.txt
+++ b/CMakeLists.txt
@@ -159,6 +159,7 @@ set(libHeaders
   ${SOURCE_DIR}/lib/rijndael.h
   ${SOURCE_DIR}/lib/rtmpchunks.h
   ${SOURCE_DIR}/lib/rtp.h
+  ${SOURCE_DIR}/lib/sdp.h
   ${SOURCE_DIR}/lib/shared_memory.h
   ${SOURCE_DIR}/lib/socket.h
   ${SOURCE_DIR}/lib/stream.h
@@ -208,6 +209,7 @@ set(libSources
   ${SOURCE_DIR}/lib/rijndael.cpp
   ${SOURCE_DIR}/lib/rtmpchunks.cpp
   ${SOURCE_DIR}/lib/rtp.cpp
+  ${SOURCE_DIR}/lib/sdp.cpp
   ${SOURCE_DIR}/lib/shared_memory.cpp
   ${SOURCE_DIR}/lib/socket.cpp
   ${SOURCE_DIR}/lib/stream.cpp
@@ -289,6 +291,7 @@ makeAnalyser(MP4 mp4) #LTS
 makeAnalyser(H264 h264) #LTS
 makeAnalyser(HLS hls) #LTS
 makeAnalyser(RIFF riff) #LTS
+makeAnalyser(RTSP rtsp) #LTS
 
 #LTS_START
 ########################################
diff --git a/lib/rtp.cpp b/lib/rtp.cpp
index 821e602d..26c735ed 100644
--- a/lib/rtp.cpp
+++ b/lib/rtp.cpp
@@ -1,132 +1,227 @@
-#include <arpa/inet.h>
 #include "rtp.h"
-#include "timing.h"
 #include "defines.h"
+#include "encode.h"
+#include "timing.h"
+#include "bitfields.h"
+#include "mpeg.h"
+#include <arpa/inet.h>
 
-
-namespace RTP {
+namespace RTP{
   double Packet::startRTCP = 0;
-  unsigned int MAX_SEND = 1500-28;
+  unsigned int MAX_SEND = 1500 - 28;
 
-  unsigned int Packet::getHsize() const {
-    return 12 + 4 * getContribCount();
-  }
+  unsigned int Packet::getHsize() const{return 12 + 4 * getContribCount();}
 
-  unsigned int Packet::getPayloadSize() const {
-    return datalen - getHsize();
-  }
+  unsigned int Packet::getPayloadSize() const{return datalen - getHsize();}
 
-  char * Packet::getPayload() const {
-    return data + getHsize();
-  }
+  char *Packet::getPayload() const{return data + getHsize();}
 
-  unsigned int Packet::getVersion() const {
-    return (data[0] >> 6) & 0x3;
-  }
+  unsigned int Packet::getVersion() const{return (data[0] >> 6) & 0x3;}
 
-  unsigned int Packet::getPadding() const {
-    return (data[0] >> 5) & 0x1;
-  }
+  unsigned int Packet::getPadding() const{return (data[0] >> 5) & 0x1;}
 
-  unsigned int Packet::getExtension() const {
-    return (data[0] >> 4) & 0x1;
-  }
+  unsigned int Packet::getExtension() const{return (data[0] >> 4) & 0x1;}
 
-  unsigned int Packet::getContribCount() const {
-    return (data[0]) & 0xE;
-  }
+  unsigned int Packet::getContribCount() const{return (data[0]) & 0xE;}
 
-  unsigned int Packet::getMarker() const {
-    return (data[1] >> 7) & 0x1;
-  }
+  unsigned int Packet::getMarker() const{return (data[1] >> 7) & 0x1;}
 
-  unsigned int Packet::getPayloadType() const {
-    return (data[1]) & 0x7F;
-  }
+  unsigned int Packet::getPayloadType() const{return (data[1]) & 0x7F;}
 
-  unsigned int Packet::getSequence() const {
-    return (((((unsigned int)data[2]) << 8) + data[3]));
-  }
+  unsigned int Packet::getSequence() const{return (((((unsigned int)data[2]) << 8) + data[3]));}
 
-  unsigned int Packet::getTimeStamp() const {
-    return ntohl(*((unsigned int *)(data + 4)));
-  }
+  unsigned int Packet::getTimeStamp() const{return ntohl(*((unsigned int *)(data + 4)));}
 
-  unsigned int Packet::getSSRC() const {
-    return ntohl(*((unsigned int *)(data + 8)));
-  }
+  unsigned int Packet::getSSRC() const{return ntohl(*((unsigned int *)(data + 8)));}
 
-  char * Packet::getData() {
-    return data + 8 + 4 * getContribCount() + getExtension();
-  }
+  char *Packet::getData(){return data + 8 + 4 * getContribCount() + getExtension();}
 
-  void Packet::setTimestamp(unsigned int t) {
-    *((unsigned int *)(data + 4)) = htonl(t);
-  }
+  void Packet::setTimestamp(unsigned int t){*((unsigned int *)(data + 4)) = htonl(t);}
 
-  void Packet::setSequence(unsigned int seq) {
-    *((short *)(data + 2)) = htons(seq);
-  }
+  void Packet::setSequence(unsigned int seq){*((short *)(data + 2)) = htons(seq);}
 
-  void Packet::setSSRC(unsigned long ssrc) {
-    *((int *)(data + 8)) = htonl(ssrc);
-  }
+  void Packet::setSSRC(unsigned long ssrc){*((int *)(data + 8)) = htonl(ssrc);}
 
-  void Packet::increaseSequence() {
-    *((short *)(data + 2)) = htons(getSequence() + 1);
-  }
+  void Packet::increaseSequence(){*((short *)(data + 2)) = htons(getSequence() + 1);}
 
-  void Packet::sendH264(void * socket, void callBack(void *, char *, unsigned int, unsigned int), const char * payload, unsigned int payloadlen, unsigned int channel) {
+  void Packet::sendH264(void *socket, void callBack(void *, char *, unsigned int, unsigned int),
+                        const char *payload, unsigned int payloadlen, unsigned int channel){
     /// \todo This function probably belongs in DMS somewhere.
-    if (payloadlen+getHsize()+2 <= maxDataLen) {
-      data[1] |= 0x80;//setting the RTP marker bit to 1
+    if (payloadlen + getHsize() + 2 <= maxDataLen){
+      data[1] |= 0x80; // setting the RTP marker bit to 1
       memcpy(data + getHsize(), payload, payloadlen);
       callBack(socket, data, getHsize() + payloadlen, channel);
       sentPackets++;
-      sentBytes += payloadlen+getHsize();
+      sentBytes += payloadlen + getHsize();
       increaseSequence();
-    } else {
-      data[1] &= 0x7F;//setting the RTP marker bit to 0
+    }else{
+      data[1] &= 0x7F; // setting the RTP marker bit to 0
       unsigned int sent = 0;
-      unsigned int sending = maxDataLen-getHsize()-2;//packages are of size MAX_SEND, except for the final one
+      unsigned int sending =
+          maxDataLen - getHsize() - 2; // packages are of size MAX_SEND, except for the final one
       char initByte = (payload[0] & 0xE0) | 0x1C;
-      char serByte = payload[0] & 0x1F; //ser is now 000
+      char serByte = payload[0] & 0x1F; // ser is now 000
       data[getHsize()] = initByte;
-      while (sent < payloadlen) {
-        if (sent == 0) {
-          serByte |= 0x80;//set first bit to 1
-        } else {
-          serByte &= 0x7F;//set first bit to 0
+      while (sent < payloadlen){
+        if (sent == 0){
+          serByte |= 0x80; // set first bit to 1
+        }else{
+          serByte &= 0x7F; // set first bit to 0
         }
-        if (sent + sending >= payloadlen) {
-          //last package
+        if (sent + sending >= payloadlen){
+          // last package
           serByte |= 0x40;
           sending = payloadlen - sent;
-          data[1] |= 0x80;//setting the RTP marker bit to 1
+          data[1] |= 0x80; // setting the RTP marker bit to 1
         }
         data[getHsize() + 1] = serByte;
         memcpy(data + getHsize() + 2, payload + 1 + sent, sending);
         callBack(socket, data, getHsize() + 2 + sending, channel);
         sentPackets++;
-        sentBytes += sending+getHsize()+2;
+        sentBytes += sending + getHsize() + 2;
         sent += sending;
         increaseSequence();
       }
     }
   }
 
-  void Packet::sendData(void * socket, void callBack(void *, char *, unsigned int, unsigned int), const char * payload, unsigned int payloadlen, unsigned int channel, std::string codec) {
+  void Packet::sendH265(void *socket, void callBack(void *, char *, unsigned int, unsigned int),
+                        const char *payload, unsigned int payloadlen, unsigned int channel){
     /// \todo This function probably belongs in DMS somewhere.
-    data[1] |= 0x80;//setting the RTP marker bit to 1
+    if (payloadlen + getHsize() + 3 <= maxDataLen){
+      data[1] |= 0x80; // setting the RTP marker bit to 1
+      memcpy(data + getHsize(), payload, payloadlen);
+      callBack(socket, data, getHsize() + payloadlen, channel);
+      sentPackets++;
+      sentBytes += payloadlen + getHsize();
+      increaseSequence();
+    }else{
+      data[1] &= 0x7F; // setting the RTP marker bit to 0
+      unsigned int sent = 0;
+      unsigned int sending =
+          maxDataLen - getHsize() - 3; // packages are of size MAX_SEND, except for the final one
+      char initByteA = (payload[0] & 0x81) | 0x62;
+      char initByteB = payload[1];
+      char serByte = (payload[0] & 0x7E) >> 1; // SE is now 00
+      data[getHsize()] = initByteA;
+      data[getHsize()+1] = initByteB;
+      while (sent < payloadlen){
+        if (sent == 0){
+          serByte |= 0x80; // set first bit to 1
+        }else{
+          serByte &= 0x7F; // set first bit to 0
+        }
+        if (sent + sending >= payloadlen){
+          // last package
+          serByte |= 0x40;
+          sending = payloadlen - sent;
+          data[1] |= 0x80; // setting the RTP marker bit to 1
+        }
+        data[getHsize() + 2] = serByte;
+        memcpy(data + getHsize() + 3, payload + 2 + sent, sending);
+        callBack(socket, data, getHsize() + 3 + sending, channel);
+        sentPackets++;
+        sentBytes += sending + getHsize() + 3;
+        sent += sending;
+        increaseSequence();
+      }
+    }
+  }
+
+  void Packet::sendMPEG2(void *socket, void callBack(void *, char *, unsigned int, unsigned int),
+                        const char *payload, unsigned int payloadlen, unsigned int channel){
+    /// \todo This function probably belongs in DMS somewhere.
+    if (payloadlen + getHsize() + 4 <= maxDataLen){
+      data[1] |= 0x80; // setting the RTP marker bit to 1
+      Mpeg::MPEG2Info mInfo = Mpeg::parseMPEG2Headers(payload, payloadlen);
+      MPEGVideoHeader mHead(data+getHsize());
+      mHead.clear();
+      mHead.setTempRef(mInfo.tempSeq);
+      mHead.setPictureType(mInfo.frameType);
+      if (mInfo.isHeader){
+        mHead.setSequence();
+      }
+      mHead.setBegin();
+      mHead.setEnd();
+      memcpy(data + getHsize() + 4, payload, payloadlen);
+      callBack(socket, data, getHsize() + payloadlen + 4, channel);
+      sentPackets++;
+      sentBytes += payloadlen + getHsize() + 4;
+      increaseSequence();
+    }else{
+      data[1] &= 0x7F; // setting the RTP marker bit to 0
+      unsigned int sent = 0;
+      unsigned int sending =
+          maxDataLen - getHsize() - 4; // packages are of size MAX_SEND, except for the final one
+      Mpeg::MPEG2Info mInfo;
+      MPEGVideoHeader mHead(data+getHsize());
+      while (sent < payloadlen){
+        mHead.clear();
+        if (sent + sending >= payloadlen){
+          mHead.setEnd();
+          sending = payloadlen - sent;
+          data[1] |= 0x80; // setting the RTP marker bit to 1
+        }
+        Mpeg::parseMPEG2Headers(payload, sent+sending, mInfo);
+        mHead.setTempRef(mInfo.tempSeq);
+        mHead.setPictureType(mInfo.frameType);
+        if (sent == 0){
+          if (mInfo.isHeader){
+            mHead.setSequence();
+          }
+          mHead.setBegin();
+        }
+        memcpy(data + getHsize() + 4, payload + sent, sending);
+        callBack(socket, data, getHsize() + 4 + sending, channel);
+        sentPackets++;
+        sentBytes += sending + getHsize() + 4;
+        sent += sending;
+        increaseSequence();
+      }
+    }
+  }
+
+  void Packet::sendData(void *socket, void callBack(void *, char *, unsigned int, unsigned int),
+                        const char *payload, unsigned int payloadlen, unsigned int channel,
+                        std::string codec){
+    if (codec == "H264"){
+      unsigned long sent = 0;
+      while (sent < payloadlen){
+        unsigned long nalSize = ntohl(*((unsigned long *)(payload + sent)));
+        sendH264(socket, callBack, payload + sent + 4, nalSize, channel);
+        sent += nalSize + 4;
+      }
+      return;
+    }
+    if (codec == "HEVC"){
+      unsigned long sent = 0;
+      while (sent < payloadlen){
+        unsigned long nalSize = ntohl(*((unsigned long *)(payload + sent)));
+        sendH265(socket, callBack, payload + sent + 4, nalSize, channel);
+        sent += nalSize + 4;
+      }
+      return;
+    }
+    if (codec == "MPEG2"){
+      sendMPEG2(socket, callBack, payload, payloadlen, channel);
+      return;
+    }
+    /// \todo This function probably belongs in DMS somewhere.
+    data[1] |= 0x80; // setting the RTP marker bit to 1
     long offsetLen = 0;
     if (codec == "AAC"){
       *((long *)(data + getHsize())) = htonl(((payloadlen << 3) & 0x0010fff8) | 0x00100000);
       offsetLen = 4;
-    }else if (codec == "MP3"){
-      *((long *)(data + getHsize())) = 0;//this is MBZ and Frag_Offset, which is always 0
+    }else if (codec == "MP3" || codec == "MP2"){
+      //See RFC 2250, "MPEG Audio-specific header"
+      *((long *)(data + getHsize())) = 0; // this is MBZ and Frag_Offset, which are always 0
+      if (payload[0] != 0xFF){
+        FAIL_MSG("MP2/MP3 data does not start with header?");
+      }
       offsetLen = 4;
     }else if (codec == "AC3"){
-      *((short *)(data + getHsize())) = htons(0x0001) ;//this is 6 bits MBZ, 2 bits FT = 0 = full frames and 8 bits saying we send 1 frame
+      *((short *)(data + getHsize())) = htons(0x0001); // this is 6 bits MBZ, 2 bits FT = 0 = full
+                                                       // frames and 8 bits saying we send 1 frame
       offsetLen = 2;
     }
     if (maxDataLen < getHsize() + offsetLen + payloadlen){
@@ -135,10 +230,10 @@ namespace RTP {
         return;
       }
       uint32_t newMaxLen = getHsize() + offsetLen + payloadlen;
-      char * newData = new char[newMaxLen];
+      char *newData = new char[newMaxLen];
       if (newData){
         memcpy(newData, data, maxDataLen);
-        delete [] data;
+        delete[] data;
         data = newData;
         maxDataLen = newMaxLen;
       }
@@ -146,13 +241,14 @@ namespace RTP {
     memcpy(data + getHsize() + offsetLen, payload, payloadlen);
     callBack(socket, data, getHsize() + offsetLen + payloadlen, channel);
     sentPackets++;
-    sentBytes += payloadlen;
+    sentBytes += payloadlen + offsetLen + getHsize();
     increaseSequence();
   }
-  
 
-  void Packet::sendRTCP(long long & connectedAt, void * socket, unsigned int tid , DTSC::Meta & metadata, void callBack(void *, char *, unsigned int, unsigned int)) {
-    void * rtcpData = malloc(32);
+  void Packet::sendRTCP(long long &connectedAt, void *socket, unsigned int tid,
+                        DTSC::Meta &metadata,
+                        void callBack(void *, char *, unsigned int, unsigned int)){
+    void *rtcpData = malloc(32);
     if (!rtcpData){
       FAIL_MSG("Could not allocate 32 bytes. Something is seriously messed up.");
       return;
@@ -160,62 +256,63 @@ namespace RTP {
     ((int *)rtcpData)[0] = htonl(0x80C80006);
     ((int *)rtcpData)[1] = htonl(getSSRC());
     // unsigned int tid = packet["trackid"].asInt();
-    //timestamp in ms
+    // timestamp in ms
     double ntpTime = 2208988800UL + Util::epoch() + (Util::getMS() % 1000) / 1000.0;
-    if (startRTCP < 1 && startRTCP > -1) {
-      startRTCP = ntpTime;
-    }
+    if (startRTCP < 1 && startRTCP > -1){startRTCP = ntpTime;}
     ntpTime -= startRTCP;
-    
-    ((int *)rtcpData)[2] = htonl(2208988800UL + Util::epoch()); //epoch is in seconds
-    ((int *)rtcpData)[3]  = htonl((Util::getMS() % 1000) * 4294967.295);
-    if (metadata.tracks[tid].codec == "H264") {
-      ((int *)rtcpData)[4] = htonl((ntpTime - 0) * 90000); //rtpts
-    } else {
-      ((int *)rtcpData)[4] = htonl((ntpTime - 0) * metadata.tracks[tid].rate); //rtpts
+
+    ((int *)rtcpData)[2] = htonl(2208988800UL + Util::epoch()); // epoch is in seconds
+    ((int *)rtcpData)[3] = htonl((Util::getMS() % 1000) * 4294967.295);
+    if (metadata.tracks[tid].codec == "H264"){
+      ((int *)rtcpData)[4] = htonl((ntpTime - 0) * 90000); // rtpts
+    }else{
+      ((int *)rtcpData)[4] = htonl((ntpTime - 0) * metadata.tracks[tid].rate); // rtpts
     }
-    //it should be the time packet was sent maybe, after all?
+    // it should be the time packet was sent maybe, after all?
     //*((int *)(rtcpData+16) ) = htonl(getTimeStamp());//rtpts
-    ((int *)rtcpData)[5] = htonl(sentPackets);//packet
-    ((int *)rtcpData)[6] = htonl(sentBytes);//octet
-    callBack(socket, (char*)rtcpData , 28 , 0);
+    ((int *)rtcpData)[5] = htonl(sentPackets); // packet
+    ((int *)rtcpData)[6] = htonl(sentBytes);   // octet
+    callBack(socket, (char *)rtcpData, 28, 0);
     free(rtcpData);
   }
 
-  Packet::Packet() {
+  Packet::Packet(){
     managed = false;
     data = 0;
   }
 
-  Packet::Packet(unsigned int payloadType, unsigned int sequence, unsigned int timestamp, unsigned int ssrc, unsigned int csrcCount) {
+  Packet::Packet(unsigned int payloadType, unsigned int sequence, unsigned int timestamp,
+                 unsigned int ssrc, unsigned int csrcCount){
     managed = true;
-    data = new char[12 + 4 * csrcCount + 2 + MAX_SEND]; //headerSize, 2 for FU-A, MAX_SEND for maximum sent size
+    data = new char[12 + 4 * csrcCount + 2 +
+                    MAX_SEND]; // headerSize, 2 for FU-A, MAX_SEND for maximum sent size
     if (data){
       maxDataLen = 12 + 4 * csrcCount + 2 + MAX_SEND;
-      data[0] = ((2) << 6) | ((0 & 1) << 5) | ((0 & 1) << 4) | (csrcCount & 15); //version, padding, extension, csrc count
-      data[1] = payloadType & 0x7F; //marker and payload type
+      data[0] = ((2) << 6) | ((0 & 1) << 5) | ((0 & 1) << 4) |
+                (csrcCount & 15);   // version, padding, extension, csrc count
+      data[1] = payloadType & 0x7F; // marker and payload type
     }else{
       maxDataLen = 0;
     }
-    setSequence(sequence - 1); //we automatically increase the sequence each time when p
+    setSequence(sequence - 1); // we automatically increase the sequence each time when p
     setTimestamp(timestamp);
     setSSRC(ssrc);
     sentBytes = 0;
     sentPackets = 0;
   }
 
-  Packet::Packet(const Packet & o) {
+  Packet::Packet(const Packet &o){
     managed = true;
     maxDataLen = 0;
-    if (o.data && o.maxDataLen) {
-      data = new char[o.maxDataLen]; //headerSize, 2 for FU-A, MAX_SEND for maximum sent size
-      if (data) {
+    if (o.data && o.maxDataLen){
+      data = new char[o.maxDataLen]; // headerSize, 2 for FU-A, MAX_SEND for maximum sent size
+      if (data){
         maxDataLen = o.maxDataLen;
         memcpy(data, o.data, o.maxDataLen);
       }
-    } else {
-      data = new char[14 + MAX_SEND];//headerSize, 2 for FU-A, MAX_SEND for maximum sent size
-      if (data) {
+    }else{
+      data = new char[14 + MAX_SEND]; // headerSize, 2 for FU-A, MAX_SEND for maximum sent size
+      if (data){
         maxDataLen = 14 + MAX_SEND;
         memset(data, 0, maxDataLen);
       }
@@ -224,23 +321,21 @@ namespace RTP {
     sentPackets = o.sentPackets;
   }
 
-  void Packet::operator=(const Packet & o) {
+  void Packet::operator=(const Packet &o){
     managed = true;
     maxDataLen = 0;
-    if (data && managed) {
-      delete[] data;
-    }
+    if (data && managed){delete[] data;}
     data = 0;
 
-    if (o.data && o.maxDataLen) {
-      data = new char[o.maxDataLen]; //headerSize, 2 for FU-A, MAX_SEND for maximum sent size
-      if (data) {
+    if (o.data && o.maxDataLen){
+      data = new char[o.maxDataLen]; // headerSize, 2 for FU-A, MAX_SEND for maximum sent size
+      if (data){
         maxDataLen = o.maxDataLen;
         memcpy(data, o.data, o.maxDataLen);
       }
-    } else {
-      data = new char[14 + MAX_SEND];//headerSize, 2 for FU-A, MAX_SEND for maximum sent size
-      if (data) {
+    }else{
+      data = new char[14 + MAX_SEND]; // headerSize, 2 for FU-A, MAX_SEND for maximum sent size
+      if (data){
         maxDataLen = 14 + MAX_SEND;
         memset(data, 0, maxDataLen);
       }
@@ -249,15 +344,68 @@ namespace RTP {
     sentPackets = o.sentPackets;
   }
 
-  Packet::~Packet() {
-    if (managed) {
-      delete [] data;
-    }
+  Packet::~Packet(){
+    if (managed){delete[] data;}
   }
-  Packet::Packet(const char * dat, unsigned int len) {
+  Packet::Packet(const char *dat, unsigned int len){
     managed = false;
     datalen = len;
-    data = (char *) dat;
+    data = (char *)dat;
+  }
+
+  MPEGVideoHeader::MPEGVideoHeader(char *d){
+    data = d;
+  }
+
+  uint16_t MPEGVideoHeader::getTotalLen() const{
+    uint16_t ret = 4;
+    if (data[0] & 0x08){
+      ret += 4;
+      if (data[4] & 0x40){
+        ret += data[8];
+      }
+    }
+    return ret;
+  }
+
+  std::string MPEGVideoHeader::toString() const{
+    std::stringstream ret;
+    uint32_t firstHead = Bit::btohl(data);
+    ret << "TR=" << ((firstHead & 0x3FF0000) >> 16);
+    if (firstHead & 0x4000000){ret << " Ext";}
+    if (firstHead & 0x2000){ret << " SeqHead";}
+    if (firstHead & 0x1000){ret << " SliceBegin";}
+    if (firstHead & 0x800){ret << " SliceEnd";}
+    ret << " PicType=" << ((firstHead & 0x700) >> 8);
+    if (firstHead & 0x80){ret << " FBV";}
+    ret << " BFC=" << ((firstHead & 0x70) >> 4);
+    if (firstHead & 0x8){ret << " FFV";}
+    ret << " FFC=" << (firstHead & 0x7);
+    return ret.str();
+  }
+
+  void MPEGVideoHeader::clear(){
+    ((uint32_t*)data)[0] = 0;
+  }
+
+  void MPEGVideoHeader::setTempRef(uint16_t ref){
+    data[0] |= (ref >> 8) & 0x03;
+    data[1] = ref & 0xff;
+  }
+
+  void MPEGVideoHeader::setPictureType(uint8_t pType){
+    data[2] |= pType & 0x7;
+  }
+
+  void MPEGVideoHeader::setSequence(){
+    data[2] |= 0x20;
+  }
+  void MPEGVideoHeader::setBegin(){
+    data[2] |= 0x10;
+  }
+  void MPEGVideoHeader::setEnd(){
+    data[2] |= 0x8;
   }
 
 }
+
diff --git a/lib/rtp.h b/lib/rtp.h
index 495a4540..1849e6cf 100644
--- a/lib/rtp.h
+++ b/lib/rtp.h
@@ -1,65 +1,91 @@
 #pragma once
-#include <string>
-#include <vector>
-#include <set>
-#include <iostream>
-#include <iomanip>
-#include <cstdio>
-#include <stdint.h>
-#include <sstream>
-#include <deque>
-#include <algorithm>
-#include "socket.h"
-#include "json.h"
 #include "dtsc.h"
+#include "json.h"
 #include "mp4.h"
 #include "mp4_generic.h"
+#include "socket.h"
+#include <algorithm>
+#include <cstdio>
+#include <deque>
+#include <iomanip>
+#include <iostream>
+#include <set>
+#include <sstream>
+#include <stdint.h>
+#include <string>
+#include <vector>
 
 /// This namespace holds all RTP-parsing and sending related functionality.
-namespace RTP {
+namespace RTP{
 
   extern unsigned int MAX_SEND;
 
-  /// This class is used to make RTP packets. Currently, H264, and AAC are supported. RTP mechanisms, like increasing sequence numbers and setting timestamps are all taken care of in here.
-  class Packet {
-    private:
-      bool managed;
-      char * data; ///<The actual RTP packet that is being sent
-      uint32_t maxDataLen;///< Amount of reserved bytes for the packet(s)
-      unsigned int datalen;  ///<Size of rtp packet
-      int sentPackets;
-      int sentBytes;//Because ugly is beautiful
+  /// This class is used to make RTP packets. Currently, H264, and AAC are supported. RTP
+  /// mechanisms, like increasing sequence numbers and setting timestamps are all taken care of in
+  /// here.
+  class Packet{
+  private:
+    bool managed;
+    char *data;           ///<The actual RTP packet that is being sent
+    uint32_t maxDataLen;  ///< Amount of reserved bytes for the packet(s)
+    unsigned int datalen; ///<Size of rtp packet
+    int sentPackets;
+    int sentBytes; // Because ugly is beautiful
+  public:
+    static double startRTCP;
+    unsigned int getHsize() const;
+    unsigned int getPayloadSize() const;
+    char *getPayload() const;
+    unsigned int getVersion() const;
+    unsigned int getPadding() const;
+    unsigned int getExtension() const;
+    unsigned int getContribCount() const;
+    unsigned int getMarker() const;
+    unsigned int getPayloadType() const;
+    unsigned int getSequence() const;
+    unsigned int getTimeStamp() const;
+    void setSequence(unsigned int seq);
+    unsigned int getSSRC() const;
+    void setSSRC(unsigned long ssrc);
+
+    void setTimestamp(unsigned int t);
+    void increaseSequence();
+    void sendH264(void *socket, void callBack(void *, char *, unsigned int, unsigned int),
+                  const char *payload, unsigned int payloadlen, unsigned int channel);
+    void sendH265(void *socket, void callBack(void *, char *, unsigned int, unsigned int),
+                  const char *payload, unsigned int payloadlen, unsigned int channel);
+    void sendMPEG2(void *socket, void callBack(void *, char *, unsigned int, unsigned int),
+                  const char *payload, unsigned int payloadlen, unsigned int channel);
+    void sendData(void *socket, void callBack(void *, char *, unsigned int, unsigned int),
+                  const char *payload, unsigned int payloadlen, unsigned int channel,
+                  std::string codec);
+    void sendRTCP(long long &connectedAt, void *socket, unsigned int tid, DTSC::Meta &metadata,
+                  void callBack(void *, char *, unsigned int, unsigned int));
+
+    Packet();
+    Packet(unsigned int pt, unsigned int seq, unsigned int ts, unsigned int ssr,
+           unsigned int csrcCount = 0);
+    Packet(const Packet &o);
+    void operator=(const Packet &o);
+    ~Packet();
+    Packet(const char *dat, unsigned int len);
+    char *getData();
+  };
+
+  class MPEGVideoHeader{
     public:
-      static double startRTCP;
-      unsigned int getHsize() const;
-      unsigned int getPayloadSize() const;
-      char * getPayload() const;
-      unsigned int getVersion() const;
-      unsigned int getPadding() const;
-      unsigned int getExtension() const;
-      unsigned int getContribCount() const;
-      unsigned int getMarker() const;
-      unsigned int getPayloadType() const;
-      unsigned int getSequence() const;
-      unsigned int getTimeStamp() const;
-      void setSequence(unsigned int seq);
-      unsigned int getSSRC() const;
-      void setSSRC(unsigned long ssrc);
-
-      void setTimestamp(unsigned int t);
-      void increaseSequence();
-      void sendH264(void * socket, void callBack(void *, char *, unsigned int, unsigned int), const char * payload, unsigned int payloadlen, unsigned int channel);
-      void sendData(void * socket, void callBack(void *, char *, unsigned int, unsigned int), const char * payload, unsigned int payloadlen, unsigned int channel, std::string codec);
-      void sendRTCP(long long & connectedAt, void * socket,  unsigned int tid, DTSC::Meta & metadata, void callBack(void *, char *, unsigned int, unsigned int));
-
-
-      Packet();
-      Packet(unsigned int pt, unsigned int seq, unsigned int ts, unsigned int ssr, unsigned int csrcCount = 0);
-      Packet(const Packet & o);
-      void operator=(const Packet & o);
-      ~Packet();
-      Packet(const char * dat, unsigned int len);
-      char * getData();
+      MPEGVideoHeader(char * d);
+      void clear();
+      uint16_t getTotalLen() const;
+      std::string toString() const;
+      void setTempRef(uint16_t ref);
+      void setPictureType(uint8_t pType);
+      void setSequence();
+      void setBegin();
+      void setEnd();
+    private:
+      char * data;
   };
 
 }
+
diff --git a/lib/sdp.cpp b/lib/sdp.cpp
new file mode 100644
index 00000000..502d6e46
--- /dev/null
+++ b/lib/sdp.cpp
@@ -0,0 +1,901 @@
+#include "sdp.h"
+#include "adts.h"
+#include "defines.h"
+#include "encode.h"
+#include "h264.h"
+#include "h265.h"
+#include "util.h"
+
+namespace SDP{
+  Track::Track(){
+    rtcpSent = 0;
+    channel = -1;
+    firstTime = 0;
+    packCount = 0;
+    cPort = 0;
+    rtpSeq = 0;
+    fpsTime = 0;
+    fpsMeta = 0;
+    fps = 0;
+  }
+
+  /// Extracts a particular parameter from the fmtp string. fmtp member must be set before calling.
+  std::string Track::getParamString(const std::string &param) const{
+    if (!fmtp.size()){return "";}
+    size_t pos = fmtp.find(param);
+    if (pos == std::string::npos){return "";}
+    pos += param.size() + 1;
+    size_t ePos = fmtp.find_first_of(" ;", pos);
+    return fmtp.substr(pos, ePos - pos);
+  }
+
+  /// Extracts a particular parameter from the fmtp string as an integer. fmtp member must be set
+  /// before calling.
+  uint64_t Track::getParamInt(const std::string &param) const{
+    return atoll(getParamString(param).c_str());
+  }
+
+  /// Gets the SDP contents for sending out a particular given DTSC::Track.
+  std::string mediaDescription(const DTSC::Track &trk){
+    std::stringstream mediaDesc;
+    if (trk.codec == "H264"){
+      MP4::AVCC avccbox;
+      avccbox.setPayload(trk.init);
+      mediaDesc << "m=video 0 RTP/AVP 97\r\n"
+                   "a=rtpmap:97 H264/90000\r\n"
+                   "a=cliprect:0,0,"
+                << trk.height << "," << trk.width << "\r\n"
+                                                     "a=framesize:97 "
+                << trk.width << '-' << trk.height
+                << "\r\n"
+                   "a=fmtp:97 packetization-mode=1;profile-level-id="
+                << std::hex << std::setw(2) << std::setfill('0') << (int)trk.init.data()[1]
+                << std::dec << "E0" << std::hex << std::setw(2) << std::setfill('0')
+                << (int)trk.init.data()[3] << std::dec << ";"
+                                                          "sprop-parameter-sets="
+                << Encodings::Base64::encode(std::string(avccbox.getSPS(), avccbox.getSPSLen()))
+                << ","
+                << Encodings::Base64::encode(std::string(avccbox.getPPS(), avccbox.getPPSLen()))
+                << "\r\n"
+                   "a=framerate:"
+                << ((double)trk.fpks) / 1000.0 << "\r\n"
+                                                  "a=control:track"
+                << trk.trackID << "\r\n";
+    }else if (trk.codec == "HEVC"){
+      h265::initData iData(trk.init);
+      mediaDesc << "m=video 0 RTP/AVP 104\r\n"
+                   "a=rtpmap:104 H265/90000\r\n"
+                   "a=cliprect:0,0,"
+                << trk.height << "," << trk.width << "\r\n"
+                                                     "a=framesize:104 "
+                << trk.width << '-' << trk.height << "\r\n"
+                << "a=fmtp:104 sprop-vps=";
+      const std::set<std::string> & vps = iData.getVPS();
+      if (vps.size()){
+       for (std::set<std::string>::iterator it = vps.begin(); it != vps.end(); it++){
+         if (it != vps.begin()){mediaDesc << ",";}
+         mediaDesc << Encodings::Base64::encode(*it);
+       }
+      }
+      mediaDesc << "; sprop-sps=";
+      const std::set<std::string> & sps = iData.getSPS();
+      if (sps.size()){
+       for (std::set<std::string>::iterator it = sps.begin(); it != sps.end(); it++){
+         if (it != sps.begin()){mediaDesc << ",";}
+         mediaDesc << Encodings::Base64::encode(*it);
+       }
+      }
+      mediaDesc << "; sprop-pps=";
+      const std::set<std::string> & pps = iData.getPPS();
+      if (pps.size()){
+       for (std::set<std::string>::iterator it = pps.begin(); it != pps.end(); it++){
+         if (it != pps.begin()){mediaDesc << ",";}
+         mediaDesc << Encodings::Base64::encode(*it);
+       }
+      }
+      mediaDesc << "\r\na=framerate:"
+                << ((double)trk.fpks) / 1000.0 << "\r\n"
+                                                  "a=control:track"
+                << trk.trackID << "\r\n";
+    }else if (trk.codec == "MPEG2"){
+      mediaDesc << "m=video 0 RTP/AVP 32\r\n"
+                   "a=cliprect:0,0,"
+                << trk.height << "," << trk.width << "\r\n"
+                                                     "a=framesize:32 "
+                << trk.width << '-' << trk.height << "\r\n"
+                << "a=framerate:" << ((double)trk.fpks) / 1000.0 << "\r\n"
+                << "a=control:track" << trk.trackID << "\r\n";
+    }else if (trk.codec == "AAC"){
+      mediaDesc << "m=audio 0 RTP/AVP 96"
+                << "\r\n"
+                   "a=rtpmap:96 mpeg4-generic/"
+                << trk.rate << "/" << trk.channels
+                << "\r\n"
+                   "a=fmtp:96 streamtype=5; profile-level-id=15; config=";
+      for (unsigned int i = 0; i < trk.init.size(); i++){
+        mediaDesc << std::hex << std::setw(2) << std::setfill('0') << (int)trk.init[i] << std::dec;
+      }
+      // these values are described in RFC 3640
+      mediaDesc << "; mode=AAC-hbr; SizeLength=13; IndexLength=3; IndexDeltaLength=3;\r\n"
+                   "a=control:track"
+                << trk.trackID << "\r\n";
+    }else if (trk.codec == "MP3" || trk.codec == "MP2"){
+      mediaDesc << "m=" << trk.type << " 0 RTP/AVP 14"
+                << "\r\n"
+                   "a=rtpmap:14 MPA/90000/"
+                << trk.channels << "\r\n"
+                                   "a=control:track"
+                << trk.trackID << "\r\n";
+    }else if (trk.codec == "AC3"){
+      mediaDesc << "m=audio 0 RTP/AVP 100"
+                << "\r\n"
+                   "a=rtpmap:100 AC3/"
+                << trk.rate << "/" << trk.channels << "\r\n"
+                                                      "a=control:track"
+                << trk.trackID << "\r\n";
+    }else if (trk.codec == "ALAW"){
+      if (trk.channels == 1 && trk.rate == 8000){
+        mediaDesc << "m=audio 0 RTP/AVP 8"
+                  << "\r\n";
+      }else{
+        mediaDesc << "m=audio 0 RTP/AVP 101"
+                  << "\r\n";
+        mediaDesc << "a=rtpmap:101 PCMA/" << trk.rate << "/" << trk.channels << "\r\n";
+      }
+      mediaDesc << "a=control:track" << trk.trackID << "\r\n";
+    }else if (trk.codec == "PCM"){
+      if (trk.size == 16 && trk.channels == 2 && trk.rate == 44100){
+        mediaDesc << "m=audio 0 RTP/AVP 10"
+                  << "\r\n";
+      }else if (trk.size == 16 && trk.channels == 1 && trk.rate == 44100){
+        mediaDesc << "m=audio 0 RTP/AVP 11"
+                  << "\r\n";
+      }else{
+        mediaDesc << "m=audio 0 RTP/AVP 103"
+                  << "\r\n";
+        mediaDesc << "a=rtpmap:103 L" << trk.size << "/" << trk.rate << "/" << trk.channels
+                  << "\r\n";
+      }
+      mediaDesc << "a=control:track" << trk.trackID << "\r\n";
+    }else if (trk.codec == "opus"){
+      mediaDesc << "m=audio 0 RTP/AVP 102"
+                << "\r\n"
+                   "a=rtpmap:102 opus/"
+                << trk.rate << "/" << trk.channels << "\r\n"
+                                                      "a=control:track"
+                << trk.trackID << "\r\n";
+    }
+    return mediaDesc.str();
+  }
+
+  /// Sets the TCP/UDP connection details from a given transport string.
+  /// Sets the transportString member to the current transport string on success.
+  /// \param host The host connecting to us.
+  /// \source The source identifier.
+  /// \return True if successful, false otherwise.
+  bool Track::parseTransport(const std::string &transport, const std::string &host,
+                             const std::string &source, const DTSC::Track &trk){
+    unsigned int SSrc = rand();
+    if (trk.codec == "H264"){
+      pack = RTP::Packet(97, 1, 0, SSrc);
+    }else if (trk.codec == "HEVC"){
+      pack = RTP::Packet(104, 1, 0, SSrc);
+    }else if (trk.codec == "MPEG2"){
+      pack = RTP::Packet(32, 1, 0, SSrc);
+    }else if (trk.codec == "AAC"){
+      pack = RTP::Packet(96, 1, 0, SSrc);
+    }else if (trk.codec == "AC3"){
+      pack = RTP::Packet(100, 1, 0, SSrc);
+    }else if (trk.codec == "MP3" || trk.codec == "MP2"){
+      pack = RTP::Packet(14, 1, 0, SSrc);
+    }else if (trk.codec == "ALAW"){
+      if (trk.channels == 1 && trk.rate == 8000){
+        pack = RTP::Packet(8, 1, 0, SSrc);
+      }else{
+        pack = RTP::Packet(101, 1, 0, SSrc);
+      }
+    }else if (trk.codec == "PCM"){
+      if (trk.size == 16 && trk.channels == 2 && trk.rate == 44100){
+        pack = RTP::Packet(10, 1, 0, SSrc);
+      }else if (trk.size == 16 && trk.channels == 1 && trk.rate == 44100){
+        pack = RTP::Packet(11, 1, 0, SSrc);
+      }else{
+        pack = RTP::Packet(103, 1, 0, SSrc);
+      }
+    }else if (trk.codec == "opus"){
+      pack = RTP::Packet(102, 1, 0, SSrc);
+    }else{
+      ERROR_MSG("Unsupported codec %s for RTSP on track %u", trk.codec.c_str(), trk.trackID);
+      return false;
+    }
+    if (transport.find("TCP") != std::string::npos){
+      std::string chanE =
+          transport.substr(transport.find("interleaved=") + 12,
+                           (transport.size() - transport.rfind('-') - 1)); // extract channel ID
+      channel = atol(chanE.c_str());
+      rtcpSent = 0;
+      transportString = transport;
+    }else{
+      channel = -1;
+      size_t port_loc = transport.rfind("client_port=") + 12;
+      cPort = atol(transport.substr(port_loc, transport.rfind('-') - port_loc).c_str());
+      uint32_t portA, portB;
+      // find available ports locally;
+      int sendbuff = 4 * 1024 * 1024;
+      data.SetDestination(host, cPort);
+      portA = data.bind(0);
+      setsockopt(data.getSock(), SOL_SOCKET, SO_SNDBUF, &sendbuff, sizeof(sendbuff));
+      rtcp.SetDestination(host, cPort + 1);
+      portB = rtcp.bind(0);
+      setsockopt(rtcp.getSock(), SOL_SOCKET, SO_SNDBUF, &sendbuff, sizeof(sendbuff));
+      std::stringstream tStr;
+      tStr << "RTP/AVP/UDP;unicast;client_port=" << cPort << '-' << cPort + 1 << ";";
+      if (source.size()){tStr << "source=" << source << ";";}
+      tStr << "server_port=" << portA << "-" << portB << ";ssrc=" << std::hex << SSrc << std::dec;
+      transportString = tStr.str();
+      INFO_MSG("Transport string: %s", transportString.c_str());
+    }
+    return true;
+  }
+
+  /// Gets the rtpInfo for a given DTSC::Track, source identifier and timestamp (in millis).
+  std::string Track::rtpInfo(const DTSC::Track &trk, const std::string &source,
+                             uint64_t currentTime){
+    std::stringstream rInfo;
+    rInfo << "url=" << source << "/track" << trk.trackID
+          << ";"; // get the current url, not localhost
+    rInfo << "sequence=" << pack.getSequence() << ";rtptime=" << currentTime * getMultiplier(trk);
+    return rInfo.str();
+  }
+
+  void State::parseSDP(const std::string &sdp){
+    std::stringstream ss(sdp);
+    std::string to;
+    uint64_t trackNo = 0;
+    bool nope = true; // true if we have no valid track to fill
+    DTSC::Track *thisTrack = 0;
+    while (std::getline(ss, to, '\n')){
+      if (!to.empty() && *to.rbegin() == '\r'){to.erase(to.size() - 1, 1);}
+
+      // All tracks start with a media line
+      if (to.substr(0, 2) == "m="){
+        nope = true;
+        ++trackNo;
+        thisTrack = &(myMeta->tracks[trackNo]);
+        std::stringstream words(to.substr(2));
+        std::string item;
+        if (getline(words, item, ' ') && (item == "audio" || item == "video")){
+          thisTrack->type = item;
+          thisTrack->trackID = trackNo;
+        }else{
+          WARN_MSG("Media type not supported: %s", item.c_str());
+          continue;
+        }
+        getline(words, item, ' ');
+        if (!getline(words, item, ' ') || item != "RTP/AVP"){
+          WARN_MSG("Media transport not supported: %s", item.c_str());
+          continue;
+        }
+        if (getline(words, item, ' ')){
+          uint64_t avp_type = JSON::Value(item).asInt();
+          switch (avp_type){
+          case 8: // PCM A-law
+            INFO_MSG("PCM A-law payload type");
+            nope = false;
+            thisTrack->codec = "ALAW";
+            thisTrack->rate = 8000;
+            thisTrack->channels = 1;
+            INFO_MSG("Incoming track %s", thisTrack->getIdentifier().c_str());
+            break;
+          case 10: // PCM Stereo, 44.1kHz
+            INFO_MSG("Linear PCM stereo 44.1kHz payload type");
+            nope = false;
+            thisTrack->codec = "PCM";
+            thisTrack->size = 16;
+            thisTrack->rate = 44100;
+            thisTrack->channels = 2;
+            INFO_MSG("Incoming track %s", thisTrack->getIdentifier().c_str());
+            break;
+          case 11: // PCM Mono, 44.1kHz
+            INFO_MSG("Linear PCM mono 44.1kHz payload type");
+            nope = false;
+            thisTrack->codec = "PCM";
+            thisTrack->rate = 44100;
+            thisTrack->size = 16;
+            thisTrack->channels = 1;
+            INFO_MSG("Incoming track %s", thisTrack->getIdentifier().c_str());
+            break;
+          case 14: // MPA
+            INFO_MSG("MPA payload type");
+            nope = false;
+            thisTrack->codec = "MP3";
+            thisTrack->rate = 0;
+            thisTrack->size = 0;
+            thisTrack->channels = 0;
+            INFO_MSG("Incoming track %s", thisTrack->getIdentifier().c_str());
+            break;
+          case 32: // MPV
+            INFO_MSG("MPV payload type");
+            nope = false;
+            thisTrack->codec = "MPEG2";
+            INFO_MSG("Incoming track %s", thisTrack->getIdentifier().c_str());
+            break;
+          default:
+            // dynamic type
+            if (avp_type >= 96 && avp_type <= 127){
+              INFO_MSG("Dynamic payload type (%llu) detected", avp_type);
+              nope = false;
+              continue;
+            }else{
+              FAIL_MSG("Payload type %llu not supported!", avp_type);
+              continue;
+            }
+          }
+        }
+        continue;
+      }
+      if (nope){continue;}// ignore lines if we have no valid track
+      // RTP mapping
+      if (to.substr(0, 8) == "a=rtpmap"){
+        std::string mediaType = to.substr(to.find(' ', 8) + 1);
+        std::string trCodec = mediaType.substr(0, mediaType.find('/'));
+        // convert to fullcaps
+        for (unsigned int i = 0; i < trCodec.size(); ++i){
+          if (trCodec[i] <= 122 && trCodec[i] >= 97){trCodec[i] -= 32;}
+        }
+        if (thisTrack->type == "audio"){
+          std::string extraInfo = mediaType.substr(mediaType.find('/') + 1);
+          if (extraInfo.find('/') != std::string::npos){
+            size_t lastSlash = extraInfo.find('/');
+            thisTrack->rate = atoll(extraInfo.substr(0, lastSlash).c_str());
+            thisTrack->channels = atoll(extraInfo.substr(lastSlash + 1).c_str());
+          }else{
+            thisTrack->rate = atoll(extraInfo.c_str());
+            thisTrack->channels = 1;
+          }
+        }
+        if (trCodec == "H264"){
+          thisTrack->codec = "H264";
+          thisTrack->rate = 90000;
+        }
+        if (trCodec == "H265"){
+          thisTrack->codec = "HEVC";
+          thisTrack->rate = 90000;
+        }
+        if (trCodec == "OPUS"){
+          thisTrack->codec = "opus";
+          thisTrack->init = std::string("OpusHead\001\002\170\000\200\273\000\000\000\000\000", 19);
+        }
+        if (trCodec == "PCMA"){thisTrack->codec = "ALAW";}
+        if (trCodec == "L8"){
+          thisTrack->codec = "PCM";
+          thisTrack->size = 8;
+        }
+        if (trCodec == "L16"){
+          thisTrack->codec = "PCM";
+          thisTrack->size = 16;
+        }
+        if (trCodec == "L20"){
+          thisTrack->codec = "PCM";
+          thisTrack->size = 20;
+        }
+        if (trCodec == "L24"){
+          thisTrack->codec = "PCM";
+          thisTrack->size = 24;
+        }
+        if (trCodec == "MPEG4-GENERIC"){thisTrack->codec = "AAC";}
+        if (!thisTrack->codec.size()){
+          ERROR_MSG("Unsupported RTP mapping: %s", mediaType.c_str());
+        }else{
+          INFO_MSG("Incoming track %s", thisTrack->getIdentifier().c_str());
+        }
+        continue;
+      }
+      if (to.substr(0, 10) == "a=control:"){
+        tracks[trackNo].control = to.substr(10);
+        continue;
+      }
+      if (to.substr(0, 7) == "a=fmtp:"){
+        tracks[trackNo].fmtp = to.substr(7);
+        if (thisTrack->codec == "AAC"){
+          if (tracks[trackNo].getParamString("mode") != "AAC-hbr"){
+            // a=fmtp:97
+            // profile-level-id=1;mode=AAC-hbr;sizelength=13;indexlength=3;indexdeltalength=3;
+            // config=120856E500
+            FAIL_MSG("AAC transport mode not supported: %s",
+                     tracks[trackNo].getParamString("mode").c_str());
+            nope = true;
+            myMeta->tracks.erase(trackNo);
+            tracks.erase(trackNo);
+            continue;
+          }
+          thisTrack->init = Encodings::Hex::decode(tracks[trackNo].getParamString("config"));
+          // myMeta.tracks[trackNo].rate = aac::AudSpecConf::rate(myMeta.tracks[trackNo].init);
+        }
+        if (thisTrack->codec == "H264"){
+          // a=fmtp:96 packetization-mode=1;
+          // sprop-parameter-sets=Z0LAHtkA2D3m//AUABqxAAADAAEAAAMAMg8WLkg=,aMuDyyA=;
+          // profile-level-id=42C01E
+          std::string sprop = tracks[trackNo].getParamString("sprop-parameter-sets");
+          size_t comma = sprop.find(',');
+          tracks[trackNo].spsData = Encodings::Base64::decode(sprop.substr(0, comma));
+          tracks[trackNo].ppsData = Encodings::Base64::decode(sprop.substr(comma + 1));
+          updateH264Init(trackNo);
+        }
+        if (thisTrack->codec == "HEVC"){
+          tracks[trackNo].hevcInfo.addUnit(Encodings::Base64::decode(tracks[trackNo].getParamString("sprop-vps")));
+          tracks[trackNo].hevcInfo.addUnit(Encodings::Base64::decode(tracks[trackNo].getParamString("sprop-sps")));
+          tracks[trackNo].hevcInfo.addUnit(Encodings::Base64::decode(tracks[trackNo].getParamString("sprop-pps")));
+          updateH265Init(trackNo);
+        }
+        continue;
+      }
+      // We ignore bandwidth lines
+      if (to.substr(0, 2) == "b="){continue;}
+      // we ignore everything before the first media line.
+      if (!trackNo){continue;}
+      // at this point, the data is definitely for a track
+      INFO_MSG("Unhandled SDP line for track %llu: %s", trackNo, to.c_str());
+    }
+  }
+
+  /// Calculates H265 track metadata from sps and pps data stored in tracks[trackNo]
+  void State::updateH265Init(uint64_t trackNo){
+    DTSC::Track &Trk = myMeta->tracks[trackNo];
+    SDP::Track &RTrk = tracks[trackNo];
+    if (!RTrk.hevcInfo.haveRequired()){
+      MEDIUM_MSG("Aborted meta fill for hevc track %lu: no info nal unit", trackNo);
+      return;
+    }
+    Trk.init = RTrk.hevcInfo.generateHVCC();
+
+    h265::metaInfo MI = tracks[trackNo].hevcInfo.getMeta();
+
+    RTrk.fpsMeta = MI.fps;
+    Trk.width = MI.width;
+    Trk.height = MI.height;
+    Trk.fpks = RTrk.fpsMeta * 1000;
+  }
+
+  /// Calculates H264 track metadata from vps, sps and pps data stored in tracks[trackNo]
+  void State::updateH264Init(uint64_t trackNo){
+    DTSC::Track &Trk = myMeta->tracks[trackNo];
+    SDP::Track &RTrk = tracks[trackNo];
+    h264::sequenceParameterSet sps(RTrk.spsData.data(), RTrk.spsData.size());
+    h264::SPSMeta hMeta = sps.getCharacteristics();
+    MP4::AVCC avccBox;
+    avccBox.setVersion(1);
+    avccBox.setProfile(RTrk.spsData[1]);
+    avccBox.setCompatibleProfiles(RTrk.spsData[2]);
+    avccBox.setLevel(RTrk.spsData[3]);
+    avccBox.setSPSNumber(1);
+    avccBox.setSPS(RTrk.spsData);
+    avccBox.setPPSNumber(1);
+    avccBox.setPPS(RTrk.ppsData);
+    RTrk.fpsMeta = hMeta.fps;
+    Trk.width = hMeta.width;
+    Trk.height = hMeta.height;
+    Trk.fpks = hMeta.fps * 1000;
+    Trk.init = std::string(avccBox.payload(), avccBox.payloadSize());
+  }
+
+  uint32_t State::getTrackNoForChannel(uint8_t chan){
+    for (std::map<uint32_t, Track>::iterator it = tracks.begin(); it != tracks.end(); ++it){
+      if (chan == it->second.channel){return it->first;}
+    }
+    return 0;
+  }
+
+  uint32_t State::parseSetup(HTTP::Parser &H, const std::string &cH, const std::string &src){
+    static uint32_t trackCounter = 0;
+    if (H.url == "200"){
+      ++trackCounter;
+      if (!tracks.count(trackCounter)){return 0;}
+      if (!tracks[trackCounter].parseTransport(H.GetHeader("Transport"), cH, src,
+                                               myMeta->tracks[trackCounter])){
+        return 0;
+      }
+      return trackCounter;
+    }
+
+    if (tracks.size()){
+    for (std::map<uint32_t, Track>::iterator it = tracks.begin(); it != tracks.end(); ++it){
+      if (!it->second.control.size()){
+        it->second.control = "track" + JSON::Value((long long)it->first).asString();
+        INFO_MSG("Control track: %s", it->second.control.c_str());
+      }
+      if (H.url.find(it->second.control) != std::string::npos ||
+           H.GetVar("pass").find(it->second.control) != std::string::npos){
+        INFO_MSG("Parsing SETUP against track %lu", it->first);
+        if (!it->second.parseTransport(H.GetHeader("Transport"), cH, src,
+                                       myMeta->tracks[it->first])){
+          return 0;
+        }
+        return it->first;
+      }
+    }
+    }
+    if (H.url.find("/track") != std::string::npos){
+      uint32_t trackNo = atoi(H.url.c_str() + H.url.find("/track") + 6);
+      if (trackNo){
+        INFO_MSG("Parsing SETUP against track %lu", trackNo);
+        if (!tracks[trackNo].parseTransport(H.GetHeader("Transport"), cH, src,
+                                       myMeta->tracks[trackNo])){
+          return 0;
+        }
+        return trackNo;
+      }
+    }
+    return 0;
+  }
+
+  /// Handles a single H264 packet, checking if others are appended at the end in Annex B format.
+  /// If so, splits them up and calls h264Packet for each. If not, calls it only once for the whole
+  /// payload.
+  void State::h264MultiParse(uint64_t ts, const uint64_t track, char *buffer, const uint32_t len){
+    uint32_t lastStart = 0;
+    for (uint32_t i = 0; i < len - 4; ++i){
+      // search for start code
+      if (buffer[i] == 0 && buffer[i + 1] == 0 && buffer[i + 2] == 0 && buffer[i + 3] == 1){
+        // if found, handle a packet from the last start code up to this start code
+        Bit::htobl(buffer + lastStart, (i - lastStart - 1) - 4); // size-prepend
+        h264Packet(ts, track, buffer + lastStart, (i - lastStart - 1),
+                   h264::isKeyframe(buffer + lastStart + 4, i - lastStart - 5));
+        lastStart = i;
+      }
+    }
+    // Last packet (might be first, if no start codes found)
+    Bit::htobl(buffer + lastStart, (len - lastStart) - 4); // size-prepend
+    h264Packet(ts, track, buffer + lastStart, (len - lastStart),
+               h264::isKeyframe(buffer + lastStart + 4, len - lastStart - 4));
+  }
+
+  void State::h264Packet(uint64_t ts, const uint64_t track, const char *buffer, const uint32_t len,
+                         bool isKey){
+    MEDIUM_MSG("H264: %llu@%llu, %lub%s", track, ts, len, isKey ? " (key)" : "");
+    // Ignore zero-length packets (e.g. only contained init data and nothing else)
+    if (!len){return;}
+
+    // Header data? Compare to init, set if needed, and throw away
+    uint8_t nalType = (buffer[4] & 0x1F);
+    switch (nalType){
+    case 7: // SPS
+      if (tracks[track].spsData.size() != len - 4 ||
+          memcmp(buffer + 4, tracks[track].spsData.data(), len - 4) != 0){
+        INFO_MSG("Updated SPS from RTP data");
+        tracks[track].spsData.assign(buffer + 4, len - 4);
+        updateH264Init(track);
+      }
+      return;
+    case 8: // PPS
+      if (tracks[track].ppsData.size() != len - 4 ||
+          memcmp(buffer + 4, tracks[track].ppsData.data(), len - 4) != 0){
+        INFO_MSG("Updated PPS from RTP data");
+        tracks[track].ppsData.assign(buffer + 4, len - 4);
+        updateH264Init(track);
+      }
+      return;
+    default: // others, continue parsing
+      break;
+    }
+
+    double fps = tracks[track].fpsMeta;
+    uint32_t offset = 0;
+    uint64_t newTs = ts;
+    if (fps > 1){
+      // Assume a steady frame rate, clip the timestamp based on frame number.
+      uint64_t frameNo = (ts / (1000.0 / fps)) + 0.5;
+      while (frameNo < tracks[track].packCount){tracks[track].packCount--;}
+      // More than 32 frames behind? We probably skipped something, somewhere...
+      if ((frameNo - tracks[track].packCount) > 32){tracks[track].packCount = frameNo;}
+      // After some experimentation, we found that the time offset is the difference between the
+      // frame number and the packet counter, times the frame rate in ms
+      offset = (frameNo - tracks[track].packCount) * (1000.0 / fps);
+      //... and the timestamp is the packet counter times the frame rate in ms.
+      newTs = tracks[track].packCount * (1000.0 / fps);
+      VERYHIGH_MSG("Packing time %llu = %sframe %llu (%.2f FPS). Expected %llu -> +%llu/%lu", ts,
+                   isKey ? "key" : "i", frameNo, fps, tracks[track].packCount,
+                   (frameNo - tracks[track].packCount), offset);
+    }else{
+      // For non-steady frame rate, assume no offsets are used and the timestamp is already correct
+      VERYHIGH_MSG("Packing time %llu = %sframe %llu (variable rate)", ts, isKey ? "key" : "i",
+                   tracks[track].packCount);
+    }
+    // Fill the new DTSC packet, buffer it.
+    DTSC::Packet nextPack;
+    nextPack.genericFill(newTs, offset, track, buffer, len, 0, isKey);
+    tracks[track].packCount++;
+    if (incomingPacketCallback){incomingPacketCallback(nextPack);}
+  }
+
+  void State::h265Packet(uint64_t ts, const uint64_t track, const char *buffer, const uint32_t len,
+                         bool isKey){
+    MEDIUM_MSG("H265: %llu@%llu, %lub%s", track, ts, len, isKey ? " (key)" : "");
+    // Ignore zero-length packets (e.g. only contained init data and nothing else)
+    if (!len){return;}
+
+    // Header data? Compare to init, set if needed, and throw away
+    uint8_t nalType = (buffer[4] & 0x7E) >> 1;
+    switch (nalType){
+    case 32: // VPS
+    case 33: // SPS
+    case 34: // PPS
+      tracks[track].hevcInfo.addUnit(buffer);
+      updateH265Init(track);
+      return;
+    default: // others, continue parsing
+      break;
+    }
+
+    double fps = tracks[track].fpsMeta;
+    uint32_t offset = 0;
+    uint64_t newTs = ts;
+    if (fps > 1){
+      // Assume a steady frame rate, clip the timestamp based on frame number.
+      uint64_t frameNo = (ts / (1000.0 / fps)) + 0.5;
+      while (frameNo < tracks[track].packCount){tracks[track].packCount--;}
+      // More than 32 frames behind? We probably skipped something, somewhere...
+      if ((frameNo - tracks[track].packCount) > 32){tracks[track].packCount = frameNo;}
+      // After some experimentation, we found that the time offset is the difference between the
+      // frame number and the packet counter, times the frame rate in ms
+      offset = (frameNo - tracks[track].packCount) * (1000.0 / fps);
+      //... and the timestamp is the packet counter times the frame rate in ms.
+      newTs = tracks[track].packCount * (1000.0 / fps);
+      VERYHIGH_MSG("Packing time %llu = %sframe %llu (%.2f FPS). Expected %llu -> +%llu/%lu", ts,
+                   isKey ? "key" : "i", frameNo, fps, tracks[track].packCount,
+                   (frameNo - tracks[track].packCount), offset);
+    }else{
+      // For non-steady frame rate, assume no offsets are used and the timestamp is already correct
+      VERYHIGH_MSG("Packing time %llu = %sframe %llu (variable rate)", ts, isKey ? "key" : "i",
+                   tracks[track].packCount);
+    }
+    // Fill the new DTSC packet, buffer it.
+    DTSC::Packet nextPack;
+    nextPack.genericFill(newTs, offset, track, buffer, len, 0, isKey);
+    tracks[track].packCount++;
+    if (incomingPacketCallback){incomingPacketCallback(nextPack);}
+  }
+
+  ///Returns the multiplier to use to get milliseconds from the RTP payload type for the given track
+  double getMultiplier(const DTSC::Track & Trk){
+    if (Trk.type == "video" || Trk.codec == "MP2" || Trk.codec == "MP3"){return 90.0;}
+    return ((double)Trk.rate / 1000.0);
+  }
+
+  /// Handles RTP packets generically, for both TCP and UDP-based connections.
+  /// In case of UDP, expects packets to be pre-sorted.
+  void State::handleIncomingRTP(const uint64_t track, const RTP::Packet &pkt){
+    DTSC::Track &Trk = myMeta->tracks[track];
+    if (!tracks[track].firstTime){tracks[track].firstTime = pkt.getTimeStamp() + 1;}
+    uint64_t millis = (pkt.getTimeStamp() - tracks[track].firstTime + 1) / getMultiplier(Trk);
+    char *pl = pkt.getPayload();
+    uint32_t plSize = pkt.getPayloadSize();
+    INSANE_MSG("Received RTP packet for track %llu, time %llu -> %llu", track, pkt.getTimeStamp(), millis);
+    if (Trk.codec == "ALAW" || Trk.codec == "opus" || Trk.codec == "PCM"){
+      DTSC::Packet nextPack;
+      nextPack.genericFill(millis, 0, track, pl, plSize, 0, false);
+      if (incomingPacketCallback){incomingPacketCallback(nextPack);}
+      return;
+    }
+    if (Trk.codec == "AAC"){
+      // assume AAC packets are single AU units
+      /// \todo Support other input than single AU units
+      unsigned int headLen =
+          (Bit::btohs(pl) >> 3) + 2; // in bits, so /8, plus two for the prepended size
+      DTSC::Packet nextPack;
+      uint16_t samples = aac::AudSpecConf::samples(Trk.init);
+      uint32_t sampleOffset = 0;
+      uint32_t offset = 0;
+      uint32_t auSize = 0;
+      for (uint32_t i = 2; i < headLen; i += 2){
+        auSize = Bit::btohs(pl + i) >> 3; // only the upper 13 bits
+        nextPack.genericFill((pkt.getTimeStamp() + sampleOffset - tracks[track].firstTime + 1) /
+                                 getMultiplier(Trk),
+                             0, track, pl + headLen + offset,
+                             std::min(auSize, plSize - headLen - offset), 0, false);
+        offset += auSize;
+        sampleOffset += samples;
+        if (incomingPacketCallback){incomingPacketCallback(nextPack);}
+      }
+      return;
+    }
+    if (Trk.codec == "MP2" || Trk.codec == "MP3"){
+      if (plSize < 5){
+        WARN_MSG("Empty packet ignored!");
+        return;
+      }
+      DTSC::Packet nextPack;
+      nextPack.genericFill(millis, 0, track, pl + 4, plSize - 4, 0, false);
+      if (incomingPacketCallback){incomingPacketCallback(nextPack);}
+      return;
+    }
+    if (Trk.codec == "MPEG2"){
+      if (plSize < 5){
+        WARN_MSG("Empty packet ignored!");
+        return;
+      }
+      ///\TODO Merge packets with same timestamp together
+      HIGH_MSG("Received MPEG2 packet: %s", RTP::MPEGVideoHeader(pl).toString().c_str());
+      DTSC::Packet nextPack;
+      nextPack.genericFill(millis, 0, track, pl + 4, plSize - 4, 0, false);
+      if (incomingPacketCallback){incomingPacketCallback(nextPack);}
+      return;
+    }
+    if (Trk.codec == "HEVC"){
+      if (plSize < 2){
+        WARN_MSG("Empty packet ignored!");
+        return;
+      }
+      uint8_t nalType = (pl[0] & 0x7E) >> 1;
+      if (nalType == 48){
+        ERROR_MSG("AP not supported yet");
+      }else if (nalType == 49){
+        DONTEVEN_MSG("H265 Fragmentation Unit");
+        static Util::ResizeablePointer fuaBuffer;
+
+        // No length yet? Check for start bit. Ignore rest.
+        if (!fuaBuffer.size() && (pl[2] & 0x80) == 0){
+          HIGH_MSG("Not start of a new FU - throwing away");
+          return;
+        }
+        if (fuaBuffer.size() &&
+            ((pl[2] & 0x80) || (tracks[track].rtpSeq != pkt.getSequence()))){
+          WARN_MSG("H265 FU packet incompleted: %lu", fuaBuffer.size());
+          Bit::htobl(fuaBuffer, fuaBuffer.size() - 4); // size-prepend
+          fuaBuffer[4] |= 0x80;                        // set error bit
+          h265Packet((pkt.getTimeStamp() - tracks[track].firstTime + 1) / 90, track, fuaBuffer,
+                     fuaBuffer.size(), h265::isKeyframe(fuaBuffer + 4, fuaBuffer.size() - 4));
+          fuaBuffer.size() = 0;
+          return;
+        }
+
+        unsigned long len = plSize - 3; // ignore the three FU bytes in front
+        if (!fuaBuffer.size()){len += 6;}// six extra bytes for the first packet
+        if (!fuaBuffer.allocate(fuaBuffer.size() + len)){return;}
+        if (!fuaBuffer.size()){
+          memcpy(fuaBuffer + 6, pl + 3, plSize - 3);
+          // reconstruct first byte
+          fuaBuffer[4] = ((pl[2] & 0x3F) << 1) | (pl[0] & 0x81);
+          fuaBuffer[5] = pl[1];
+        }else{
+          memcpy(fuaBuffer + fuaBuffer.size(), pl + 3, plSize - 3);
+        }
+        fuaBuffer.size() += len;
+
+        if (pl[2] & 0x40){// last packet
+          VERYHIGH_MSG("H265 FU packet type %s (%u) completed: %lu", h265::typeToStr((fuaBuffer[4] & 0x7E) >> 1), (uint8_t)((fuaBuffer[4] & 0x7E) >> 1),
+                     fuaBuffer.size());
+          Bit::htobl(fuaBuffer, fuaBuffer.size() - 4); // size-prepend
+          h265Packet((pkt.getTimeStamp() - tracks[track].firstTime + 1) / 90, track, fuaBuffer,
+                     fuaBuffer.size(), h265::isKeyframe(fuaBuffer + 4, fuaBuffer.size() - 4));
+          fuaBuffer.size() = 0;
+        }
+        return;
+      }else if (nalType == 50){
+        ERROR_MSG("PACI/TSCI not supported yet");
+      }else{
+        DONTEVEN_MSG("%s NAL unit (%u)", h265::typeToStr(nalType), nalType);
+        static Util::ResizeablePointer packBuffer;
+        if (!packBuffer.allocate(plSize + 4)){return;}
+        Bit::htobl(packBuffer, plSize); // size-prepend
+        memcpy(packBuffer + 4, pl, plSize);
+        h265Packet((pkt.getTimeStamp() - tracks[track].firstTime + 1) / 90, track, packBuffer,
+                   plSize + 4, h265::isKeyframe(packBuffer + 4, plSize));
+        return;
+      }
+      return;
+    }
+    if (Trk.codec == "H264"){
+      // Handles common H264 packets types, but not all.
+      // Generalizes and converts them all to a data format ready for DTSC, then calls h264Packet
+      // for that data.
+      // Prints a WARN-level message if packet type is unsupported.
+      /// \todo Support other H264 packets types?
+      if (!plSize){
+        WARN_MSG("Empty packet ignored!");
+        return;
+      }
+      if ((pl[0] & 0x1F) == 0){
+        WARN_MSG("H264 packet type null ignored");
+        return;
+      }
+      if ((pl[0] & 0x1F) < 24){
+        DONTEVEN_MSG("H264 single packet, type %u", (unsigned int)(pl[0] & 0x1F));
+        static Util::ResizeablePointer packBuffer;
+        if (!packBuffer.allocate(plSize + 4)){return;}
+        Bit::htobl(packBuffer, plSize); // size-prepend
+        memcpy(packBuffer + 4, pl, plSize);
+        h264Packet((pkt.getTimeStamp() - tracks[track].firstTime + 1) / 90, track, packBuffer,
+                   plSize + 4, h264::isKeyframe(packBuffer + 4, plSize));
+        return;
+      }
+      if ((pl[0] & 0x1F) == 24){
+        DONTEVEN_MSG("H264 STAP-A packet");
+        unsigned int len = 0;
+        unsigned int pos = 1;
+        while (pos + 1 < plSize){
+          unsigned int pLen = Bit::btohs(pl + pos);
+          INSANE_MSG("Packet of %ub and type %u", pLen, (unsigned int)(pl[pos + 2] & 0x1F));
+          pos += 2 + pLen;
+          len += 4 + pLen;
+        }
+        static Util::ResizeablePointer packBuffer;
+        if (!packBuffer.allocate(len)){return;}
+        pos = 1;
+        len = 0;
+        bool isKey = false;
+        while (pos + 1 < plSize){
+          unsigned int pLen = Bit::btohs(pl + pos);
+          isKey |= h264::isKeyframe(pl + pos + 2, pLen);
+          Bit::htobl(packBuffer + len, pLen); // size-prepend
+          memcpy(packBuffer + len + 4, pl + pos + 2, pLen);
+          len += 4 + pLen;
+          pos += 2 + pLen;
+        }
+        h264Packet((pkt.getTimeStamp() - tracks[track].firstTime + 1) / 90, track, packBuffer, len,
+                   isKey);
+        return;
+      }
+      if ((pl[0] & 0x1F) == 28){
+        DONTEVEN_MSG("H264 FU-A packet");
+        static Util::ResizeablePointer fuaBuffer;
+
+        // No length yet? Check for start bit. Ignore rest.
+        if (!fuaBuffer.size() && (pl[1] & 0x80) == 0){
+          HIGH_MSG("Not start of a new FU-A - throwing away");
+          return;
+        }
+        if (fuaBuffer.size() &&
+            ((pl[1] & 0x80) || (tracks[track].rtpSeq != pkt.getSequence()))){
+          WARN_MSG("Ending unfinished FU-A");
+          INSANE_MSG("H264 FU-A packet incompleted: %lu", fuaBuffer.size());
+          uint8_t nalType = (fuaBuffer[4] & 0x1F);
+          if (nalType == 7 || nalType == 8){
+            // attempt to detect multiple H264 packets, even though specs disallow it
+            h264MultiParse((pkt.getTimeStamp() - tracks[track].firstTime + 1) / 90, track,
+                           fuaBuffer, fuaBuffer.size());
+          }else{
+            Bit::htobl(fuaBuffer, fuaBuffer.size() - 4); // size-prepend
+            fuaBuffer[4] |= 0x80;                        // set error bit
+            h264Packet((pkt.getTimeStamp() - tracks[track].firstTime + 1) / 90, track, fuaBuffer,
+                       fuaBuffer.size(), h264::isKeyframe(fuaBuffer + 4, fuaBuffer.size() - 4));
+          }
+          fuaBuffer.size() = 0;
+          return;
+        }
+
+        unsigned long len = plSize - 2; // ignore the two FU-A bytes in front
+        if (!fuaBuffer.size()){len += 5;}// five extra bytes for the first packet
+        if (!fuaBuffer.allocate(fuaBuffer.size() + len)){return;}
+        if (!fuaBuffer.size()){
+          memcpy(fuaBuffer + 4, pl + 1, plSize - 1);
+          // reconstruct first byte
+          fuaBuffer[4] = (fuaBuffer[4] & 0x1F) | (pl[0] & 0xE0);
+        }else{
+          memcpy(fuaBuffer + fuaBuffer.size(), pl + 2, plSize - 2);
+        }
+        fuaBuffer.size() += len;
+
+        if (pl[1] & 0x40){// last packet
+          INSANE_MSG("H264 FU-A packet type %u completed: %lu", (unsigned int)(fuaBuffer[4] & 0x1F),
+                     fuaBuffer.size());
+          uint8_t nalType = (fuaBuffer[4] & 0x1F);
+          if (nalType == 7 || nalType == 8){
+            // attempt to detect multiple H264 packets, even though specs disallow it
+            h264MultiParse((pkt.getTimeStamp() - tracks[track].firstTime + 1) / 90, track,
+                           fuaBuffer, fuaBuffer.size());
+          }else{
+            Bit::htobl(fuaBuffer, fuaBuffer.size() - 4); // size-prepend
+            h264Packet((pkt.getTimeStamp() - tracks[track].firstTime + 1) / 90, track, fuaBuffer,
+                       fuaBuffer.size(), h264::isKeyframe(fuaBuffer + 4, fuaBuffer.size() - 4));
+          }
+          fuaBuffer.size() = 0;
+        }
+        return;
+      }
+      WARN_MSG("H264 packet type %u unsupported", (unsigned int)(pl[0] & 0x1F));
+      return;
+    }
+  }
+}
+
diff --git a/lib/sdp.h b/lib/sdp.h
new file mode 100644
index 00000000..8a4dcf5f
--- /dev/null
+++ b/lib/sdp.h
@@ -0,0 +1,66 @@
+#include "dtsc.h"
+#include "http_parser.h"
+#include "rtp.h"
+#include "socket.h"
+#include "h265.h"
+
+namespace SDP{
+
+  double getMultiplier(const DTSC::Track & Trk);
+
+  /// Structure used to keep track of selected tracks.
+  class Track{
+  public:
+    Socket::UDPConnection data;
+    Socket::UDPConnection rtcp;
+    RTP::Packet pack;
+    long long rtcpSent;
+    uint64_t firstTime;
+    int channel; /// Channel number, used in TCP sending
+    uint64_t packCount;
+    uint16_t rtpSeq;
+    std::map<uint16_t, RTP::Packet> packBuffer;
+    uint32_t cPort;
+    std::string transportString; /// Current transport string.
+    std::string control;
+    std::string fmtp; /// fmtp string, used by getParamString / getParamInt
+    std::string spsData;
+    std::string ppsData;
+    h265::initData hevcInfo;
+    uint64_t fpsTime;
+    double fpsMeta;
+    double fps;
+    Track();
+    std::string getParamString(const std::string &param) const;
+    uint64_t getParamInt(const std::string &param) const;
+    bool parseTransport(const std::string &transport, const std::string &host,
+                        const std::string &source, const DTSC::Track &trk);
+    std::string rtpInfo(const DTSC::Track &trk, const std::string &source, uint64_t currentTime);
+  };
+
+  class State{
+  public:
+    State(){
+      incomingPacketCallback = 0;
+      myMeta = 0;
+    }
+    DTSC::Meta *myMeta;
+    void (*incomingPacketCallback)(const DTSC::Packet &pkt);
+    std::map<uint32_t, Track> tracks; ///< List of selected tracks with SDP-specific session data.
+    void parseSDP(const std::string &sdp);
+    void updateH264Init(uint64_t trackNo);
+    void updateH265Init(uint64_t trackNo);
+    uint32_t getTrackNoForChannel(uint8_t chan);
+    uint32_t parseSetup(HTTP::Parser &H, const std::string &host,
+                        const std::string &source);
+    void handleIncomingRTP(const uint64_t track, const RTP::Packet &pkt);
+    void h264MultiParse(uint64_t ts, const uint64_t track, char *buffer, const uint32_t len);
+    void h264Packet(uint64_t ts, const uint64_t track, const char *buffer, const uint32_t len,
+                    bool isKey);
+    void h265Packet(uint64_t ts, const uint64_t track, const char *buffer, const uint32_t len,
+                    bool isKey);
+  };
+
+  std::string mediaDescription(const DTSC::Track &trk);
+}
+
diff --git a/src/analysers/analyser_rtsp.cpp b/src/analysers/analyser_rtsp.cpp
index 0f8b1767..bd8ac41b 100644
--- a/src/analysers/analyser_rtsp.cpp
+++ b/src/analysers/analyser_rtsp.cpp
@@ -1,197 +1,119 @@
-#include <cstdlib>
-#include <iostream>
-#include <fstream>
-#include <string>
-#include <string.h>
-#include <vector>
+#include "analyser_rtsp.h"
 
-#include <mist/config.h>
-#include <mist/rtp.h>
-#include <mist/socket.h>
-#include <mist/http_parser.h>
-#include <sstream>
+AnalyserRTSP *classPointer = 0;
 
-namespace RtspRtp{
+void incomingPacket(const DTSC::Packet &pkt){
+  classPointer->incoming(pkt);
+}
 
-  int analyseRtspRtp(std::string rtspUrl){    
-    /*//parse hostname
-    std::string hostname = rtspUrl.substr(7);
-    hostname = hostname.substr(0,hostname.find('/'));
-    std::cout << hostname << std::endl;
-    HTTP::Parser HTTP_R, HTTP_S;//HTTP Receiver en HTTP Sender.
-    Socket::Connection conn(hostname,554,false);//setting rtsp connection
-    
-    bool optionsSent = false;
-    bool optionsRecvd = false;
-    bool descSent = false;
-    bool descRecvd = false;
-    bool setupComplete = false;
-    bool playSent = false;
-    int CSeq = 1;
-    while(conn.connected()){
-      if(!optionsSent){
-        HTTP_R.protocol="RTSP/1.0";
-        HTTP_R.method = "OPTIONS";
-        HTTP_R.url = rtspUrl;
-        HTTP_R.SetHeader("CSeq",CSeq);
-        CSeq++;
-        HTTP_R.SetHeader("User-Agent","mistANALyser");
-        HTTP_R.SendRequest(conn);
-        optionsSent = true;
-      }
-      
-      if (optionsSent&& !optionsRecvd && (conn.Received().size() || conn.spool() )){ 
-        if(HTTP_S.Read(conn)){
-          std::cout << "recv opts" << std::endl;
-          
-          std::cout << HTTP_S.BuildResponse(HTTP_S.method,HTTP_S.url);      
-          optionsRecvd = true;        
-        }
-      }
-      
+void AnalyserRTSP::init(Util::Config &conf){
+  Analyser::init(conf);
+}
 
-      
-      if(optionsRecvd && !descSent){
-        HTTP_S.Clean();
-        HTTP_R.protocol="RTSP/1.0";
-        HTTP_R.method = "DESCRIBE";
-        HTTP_R.url = rtspUrl;
-        HTTP_R.SetHeader("CSeq",CSeq);
-        CSeq++;
-        HTTP_R.SetHeader("User-Agent","mistANALyser");
-        HTTP_R.SendRequest(conn);
-        descSent = true;
-
-      }
-      
-      std::vector<std::string> trackIds;
-      
-      if (descSent&&!descRecvd && (conn.Received().size() || conn.spool() )){ 
-
-        if(HTTP_S.Read(conn)){
-          std::cout << "recv desc2" << std::endl;
-          std::cout << HTTP_S.BuildResponse(HTTP_S.method,HTTP_S.url);
-          size_t pos = HTTP_S.body.find("m=");
-          do{
-          //finding all track IDs
-             pos = HTTP_S.body.find("a=control:",pos);
-             if(pos !=std::string::npos){
-              trackIds.push_back(HTTP_S.body.substr(pos+10,HTTP_S.body.find("\r\n",pos)-pos-10 ) );//setting track IDs;
-              pos++;
-             }
-          }while(pos != std::string::npos);
-          //we have all the tracks
-
-          descRecvd = true;        
-        }
-      }
-
-      
-      unsigned int setupsSent = 0;
-      unsigned int setupsRecvd = 0;
-      Socket::UDPConnection connectors[trackIds.size()];
-      unsigned int setports[trackIds.size()];
-      uint32_t bport = 10000;
-      std::string sessionID = "";
-      
-      std::stringstream setup;
-      
-      if(descRecvd && !setupComplete){
-        //time to setup.
-        for(std::vector<std::string>::iterator it = trackIds.begin();it!=trackIds.end();it++){
-          std::cout << "setup " << setupsSent<< std::endl;        
-          while(!connectors[setupsSent].SetConnection( bport,false) ){
-            bport +=2;//finding an available port
-          }
-          std::cout << "setup" << bport<< std::endl;
-          setports[setupsSent] = bport;
-          bport +=2;
-          if(setupsSent == setupsRecvd){
-            //send only one setup
-            HTTP_S.Clean();
-            HTTP_R.protocol="RTSP/1.0";
-            HTTP_R.method = "SETUP";
-            HTTP_R.url = rtspUrl+ '/' + *(it);
-            setup << "RTP/AVP/UDP;unicast;client_port="<< setports[setupsSent] <<"-" <<setports[setupsSent]+1 ;
-            HTTP_R.SetHeader("Transport",setup.str() );
-            std:: cout << setup.str()<<std::endl;
-            setup.str(std::string());
-            setup.clear();
-            HTTP_R.SetHeader("CSeq",CSeq);
-            CSeq++;
-            if(sessionID != ""){
-             HTTP_R.SetHeader("Session",sessionID);
-            }
-            HTTP_R.SetHeader("User-Agent","mistANALyser");  
-            HTTP_R.SendRequest(conn);          
-            setupsSent ++;   
-          }
-                   
-
-          
-          while(setupsSent == setupsRecvd+1){
-            //lets Assume we assume we always receive a response          
-            if ( (conn.Received().size() || conn.spool() )){ 
-              if(HTTP_S.Read(conn)){
-                std::cout << "recv setup" << std::endl;
-                std::cout << HTTP_S.BuildResponse(HTTP_S.method,HTTP_S.url);      
-                optionsRecvd = true;    
-                sessionID = HTTP_S.GetHeader("Session");
-                setupsRecvd++;    
-              }
-            }
-          }     
-          //set up all parameters, and then after the for loop we have to listen to setups and all. sent if both are equal, and recv if one is sent          
-  
-        }
-        setupComplete = true;
-      }
-      
-      if(setupComplete && !playSent){
-      //time to play
-        HTTP_S.Clean();
-        HTTP_R.protocol="RTSP/1.0";
-        HTTP_R.method = "PLAY";
-        HTTP_R.url = rtspUrl;
-      
-        HTTP_R.SetHeader("CSeq",CSeq);
-        CSeq++;
-        HTTP_R.SetHeader("User-Agent","mistANALyser");
-        HTTP_R.SetHeader("Session",sessionID);
-        HTTP_R.SendRequest(conn);
-        playSent = true;
-        std::cout << "sent play" << std::endl;
-        char buffer[2000];
-        while(!connectors[0].iread((void*)buffer,2000)) {
-          std::cout << "buffer";
-        }
-        std::cout <<"buffer is not empty" << std::endl;
-        
-      }
-      
-      //streams set up
-      //time to read some packets
-
-
-      
-      
-      
-      if(descRecvd){
-        conn.close();
-      }
+void AnalyserRTSP::incoming(const DTSC::Packet &pkt){
+  char *dataPtr;
+  uint32_t dataSize;
+  pkt.getString("data", dataPtr, dataSize);
+  DETAIL_MED("Received %ub %sfor track %lu (%s) @ %llums", dataSize, pkt.getFlag("keyframe")?"keyframe ":"", pkt.getTrackId(),
+             myMeta.tracks[pkt.getTrackId()].getIdentifier().c_str(), pkt.getTime());
+  if (detail >= 8){
+    for (uint32_t i = 0; i < dataSize; ++i){
+      std::cout << std::hex << std::setw(2) << std::setfill('0') << (int)dataPtr[i] << " ";
+      if (i % 32 == 31){std::cout << std::endl;}
     }
-    conn.close();*/
-    return 0; 
+    std::cout << std::endl;
   }
-
-
 }
 
-
-
-int main(int argc, char ** argv){
-  Util::Config conf = Util::Config(argv[0]);
-  conf.addOption("url",JSON::fromString("{\"arg\":\"string\",\"short\":\"u\",\"long\":\"url\",\"help\":\"URL To get.\", \"default\":\"rtsp://localhost/s1k\"}"));
-  conf.parseArgs(argc, argv);
-  return RtspRtp::analyseRtspRtp(conf.getString("url"));
+AnalyserRTSP::AnalyserRTSP(Util::Config &conf) : Analyser(conf){
+  myConn = Socket::Connection(1, 0);
+  sdpState.myMeta = &myMeta;
+  sdpState.incomingPacketCallback = incomingPacket;
+  classPointer = this;
 }
+
+bool AnalyserRTSP::isOpen(){
+  return myConn;
+}
+
+bool AnalyserRTSP::parsePacket(){
+  do{
+    // No new data? Sleep and retry, if connection still open
+    if (!myConn.Received().size() || !myConn.Received().available(1)){
+      if (!myConn.spool() && isOpen()){Util::sleep(500);}
+      continue;
+    }
+    if (myConn.Received().copy(1) != "$"){
+      // not a TCP RTP packet, read RTSP commands
+      if (HTTP.Read(myConn)){
+        if (HTTP.hasHeader("Content-Type") && HTTP.GetHeader("Content-Type") == "application/sdp"){
+          sdpState.parseSDP(HTTP.body);
+          HTTP.Clean();
+          return true;
+        }
+        if (HTTP.hasHeader("Transport")){
+          uint32_t trackNo = sdpState.parseSetup(HTTP, "", "");
+          if (trackNo){
+            DETAIL_MED("Parsed transport for track: %lu", trackNo);
+          }else{
+            DETAIL_MED("Could not parse transport string!");
+          }
+          HTTP.Clean();
+          return true;
+        }
+
+        std::cout << HTTP.BuildRequest() << std::endl;
+
+        HTTP.Clean();
+        return true;
+      }else{
+        if (!myConn.spool() && isOpen()){Util::sleep(500);}
+      }
+      continue;
+    }
+    if (!myConn.Received().available(4)){
+      if (!myConn.spool() && isOpen()){Util::sleep(500);}
+      continue;
+    }// a TCP RTP packet, but not complete yet
+
+    // We have a TCP packet! Read it...
+    // Format: 1 byte '$', 1 byte channel, 2 bytes len, len bytes binary data
+    std::string tcpHead = myConn.Received().copy(4);
+    uint16_t len = ntohs(*(short *)(tcpHead.data() + 2));
+    if (!myConn.Received().available(len + 4)){
+      if (!myConn.spool() && isOpen()){Util::sleep(500);}
+      continue;
+    }// a TCP RTP packet, but not complete yet
+    // remove whole packet from buffer, including 4 byte header
+    std::string tcpPacket = myConn.Received().remove(len + 4);
+    RTP::Packet pkt(tcpPacket.data() + 4, len);
+    uint8_t chan = tcpHead.data()[1];
+    uint32_t trackNo = sdpState.getTrackNoForChannel(chan);
+    DETAIL_HI("Received %ub RTP packet #%u on channel %u, time %llu", len,
+              (unsigned int)pkt.getSequence(), chan, pkt.getTimeStamp());
+    if (!trackNo && (chan % 2) != 1){
+      DETAIL_MED("Received packet for unknown track number on channel %u", chan);
+    }
+    if (trackNo){
+      sdpState.tracks[trackNo].rtpSeq = pkt.getSequence();
+    }
+
+    if (detail >= 10){
+      char *pl = pkt.getPayload();
+      uint32_t payLen = pkt.getPayloadSize();
+      for (uint32_t i = 0; i < payLen; ++i){
+        std::cout << std::hex << std::setw(2) << std::setfill('0') << (int)pl[i] << " ";
+        if (i % 32 == 31){std::cout << std::endl;}
+      }
+      std::cout << std::endl;
+    }
+
+    sdpState.handleIncomingRTP(trackNo, pkt);
+
+    return true;
+
+  }while (isOpen());
+
+  // if needed, parse TCP packets, and cancel if it is not safe (yet) to read HTTP/RTSP packets
+}
+
diff --git a/src/analysers/analyser_rtsp.h b/src/analysers/analyser_rtsp.h
index 8d3ff74d..4194d7fe 100644
--- a/src/analysers/analyser_rtsp.h
+++ b/src/analysers/analyser_rtsp.h
@@ -1,27 +1,23 @@
-#include <mist/config.h>
+#pragma once
+
 #include "analyser.h"
-#include <cstdlib>
-#include <iomanip>
-#include <vector>
-#include <sstream>
-#include <mist/socket.h>
-#include <mist/rtp.h>
+#include <mist/h264.h>
 #include <mist/http_parser.h>
+#include <mist/rtp.h>
+#include <mist/sdp.h>
 
-class rtpAnalyser : public analysers 
-{
-  Socket::Connection conn;
-  HTTP::Parser HTTP_R, HTTP_S;//HTTP Receiver en HTTP Sender.
-  int step;
-  std::vector<std::string> tracks;
-  std::vector<Socket::UDPConnection> connections;
-  unsigned int trackIt;
+class AnalyserRTSP : public Analyser{
+public:
+  AnalyserRTSP(Util::Config &conf);
+  static void init(Util::Config &cfg);
+  bool parsePacket();
+  void incoming(const DTSC::Packet &pkt);
+  bool isOpen();
 
-  public:
-    rtpAnalyser(Util::Config config);
-    bool packetReady();
-    void PreProcessing();
-    //int Analyse();
-    int doAnalyse();
-    void doValidate();
+private:
+  Socket::Connection myConn;
+  HTTP::Parser HTTP;
+  DTSC::Meta myMeta;
+  SDP::State sdpState;
 };
+
diff --git a/src/output/output_rtsp.cpp b/src/output/output_rtsp.cpp
index be1297ea..667dade4 100644
--- a/src/output/output_rtsp.cpp
+++ b/src/output/output_rtsp.cpp
@@ -1,19 +1,26 @@
-#include <mist/defines.h>
+#include "output_rtsp.h"
+#include <mist/adts.h>
 #include <mist/auth.h>
-#include <mist/encode.h>
-#include <mist/stream.h>
 #include <mist/bitfields.h>
 #include <mist/bitstream.h>
-#include <mist/adts.h>
+#include <mist/defines.h>
+#include <mist/encode.h>
+#include <mist/stream.h>
 #include <mist/triggers.h>
-#include "output_rtsp.h"
 #include <sys/stat.h>
 
-namespace Mist {
+namespace Mist{
 
-  Socket::Connection * mainConn = 0;
+  Socket::Connection *mainConn = 0;
+  OutRTSP *classPointer = 0;
 
-  OutRTSP::OutRTSP(Socket::Connection & myConn) : Output(myConn){
+  /// Helper function for passing packets into the OutRTSP class
+  void insertPacket(const DTSC::Packet &pkt){classPointer->incomingPacket(pkt);}
+
+  /// Takes incoming packets and buffers them.
+  void OutRTSP::incomingPacket(const DTSC::Packet &pkt){bufferLivePacket(pkt);}
+
+  OutRTSP::OutRTSP(Socket::Connection &myConn) : Output(myConn){
     connectedAt = Util::epoch() + 2208988800ll;
     pausepoint = 0;
     setBlocking(false);
@@ -22,51 +29,57 @@ namespace Mist {
     expectTCP = false;
     lastTimeSync = 0;
     mainConn = &myConn;
+    classPointer = this;
+    sdpState.incomingPacketCallback = insertPacket;
+    sdpState.myMeta = &myMeta;
   }
-  
+
   /// Function used to send RTP packets over UDP
   ///\param socket A UDP Connection pointer, sent as a void*, to keep portability.
   ///\param data The RTP Packet that needs to be sent
   ///\param len The size of data
   ///\param channel Not used here, but is kept for compatibility with sendTCP
-  void sendUDP(void * socket, char * data, unsigned int len, unsigned int channel) {
-    ((Socket::UDPConnection *) socket)->SendNow(data, len);
+  void sendUDP(void *socket, char *data, unsigned int len, unsigned int channel){
+    ((Socket::UDPConnection *)socket)->SendNow(data, len);
     if (mainConn){mainConn->addUp(len);}
   }
 
-
   /// Function used to send RTP packets over TCP
   ///\param socket A TCP Connection pointer, sent as a void*, to keep portability.
   ///\param data The RTP Packet that needs to be sent
   ///\param len The size of data
   ///\param channel Used to distinguish different data streams when sending RTP over TCP
-  void sendTCP(void * socket, char * data, unsigned int len, unsigned int channel) {
-    //1 byte '$', 1 byte channel, 2 bytes length
+  void sendTCP(void *socket, char *data, unsigned int len, unsigned int channel){
+    // 1 byte '$', 1 byte channel, 2 bytes length
     char buf[] = "$$$$";
     buf[1] = channel;
-    ((short *) buf)[1] = htons(len);
-    ((Socket::Connection *) socket)->SendNow(buf, 4);
-    ((Socket::Connection *) socket)->SendNow(data, len);
+    ((short *)buf)[1] = htons(len);
+    ((Socket::Connection *)socket)->SendNow(buf, 4);
+    ((Socket::Connection *)socket)->SendNow(data, len);
   }
 
-  void OutRTSP::init(Util::Config * cfg){
+  void OutRTSP::init(Util::Config *cfg){
     Output::init(cfg);
     capa["name"] = "RTSP";
-    capa["desc"] = "Provides Real Time Streaming Protocol output, supporting both UDP and TCP transports.";
+    capa["desc"] =
+        "Provides Real Time Streaming Protocol output, supporting both UDP and TCP transports.";
     capa["deps"] = "";
     capa["url_rel"] = "/$";
     capa["codecs"][0u][0u].append("H264");
+    capa["codecs"][0u][0u].append("HEVC");
+    capa["codecs"][0u][0u].append("MPEG2");
     capa["codecs"][0u][1u].append("AAC");
     capa["codecs"][0u][1u].append("MP3");
     capa["codecs"][0u][1u].append("AC3");
     capa["codecs"][0u][1u].append("ALAW");
     capa["codecs"][0u][1u].append("PCM");
     capa["codecs"][0u][1u].append("opus");
-    
+    capa["codecs"][0u][1u].append("MP2");
+
     capa["methods"][0u]["handler"] = "rtsp";
     capa["methods"][0u]["type"] = "rtsp";
     capa["methods"][0u]["priority"] = 2ll;
-    
+
     capa["optional"]["maxsend"]["name"] = "Max RTP packet size";
     capa["optional"]["maxsend"]["help"] = "Maximum size of RTP packets in bytes";
     capa["optional"]["maxsend"]["default"] = (long long)RTP::MAX_SEND;
@@ -79,92 +92,81 @@ namespace Mist {
   }
 
   void OutRTSP::sendNext(){
-    char * dataPointer = 0;
+    char *dataPointer = 0;
     unsigned int dataLen = 0;
     thisPacket.getString("data", dataPointer, dataLen);
     uint32_t tid = thisPacket.getTrackId();
     uint64_t timestamp = thisPacket.getTime();
-    
-    //if we're past the pausing point, seek to it, and pause immediately
+
+    // if we're past the pausing point, seek to it, and pause immediately
     if (pausepoint && timestamp > pausepoint){
       pausepoint = 0;
       stop();
       return;
     }
 
-
     if (myMeta.live && lastTimeSync + 666 < timestamp){
       lastTimeSync = timestamp;
       updateMeta();
-      DTSC::Track & mainTrk = myMeta.tracks[getMainSelectedTrack()];
+      DTSC::Track &mainTrk = myMeta.tracks[getMainSelectedTrack()];
       // The extra 2000ms here is for the metadata sync delay.
       // It can be removed once we get rid of that.
-      if (timestamp + 2000 + needsLookAhead < mainTrk.keys.rbegin()->getTime() && mainTrk.lastms - mainTrk.keys.rbegin()->getTime() > needsLookAhead){
-        INFO_MSG("Skipping forward %llums (%llu ms LA)", mainTrk.keys.rbegin()->getTime() - thisPacket.getTime(), needsLookAhead);
+      if (timestamp + 2000 + needsLookAhead < mainTrk.keys.rbegin()->getTime() &&
+          mainTrk.lastms - mainTrk.keys.rbegin()->getTime() > needsLookAhead){
+        INFO_MSG("Skipping forward %llums (%llu ms LA)",
+                 mainTrk.keys.rbegin()->getTime() - thisPacket.getTime(), needsLookAhead);
         seek(mainTrk.keys.rbegin()->getTime());
         return;
       }
     }
-    
-    void * socket = 0;
+
+    void *socket = 0;
     void (*callBack)(void *, char *, unsigned int, unsigned int) = 0;
-    
-    if (tracks[tid].channel == -1){//UDP connection
-      socket = &tracks[tid].data;
+
+    if (sdpState.tracks[tid].channel == -1){// UDP connection
+      socket = &sdpState.tracks[tid].data;
       callBack = sendUDP;
-      if (Util::epoch()/5 != tracks[tid].rtcpSent){
-        tracks[tid].rtcpSent = Util::epoch()/5;
-        tracks[tid].pack.sendRTCP(connectedAt, &tracks[tid].rtcp, tid, myMeta, sendUDP);
+      if (Util::epoch() / 5 != sdpState.tracks[tid].rtcpSent){
+        sdpState.tracks[tid].rtcpSent = Util::epoch() / 5;
+        sdpState.tracks[tid].pack.sendRTCP(connectedAt, &sdpState.tracks[tid].rtcp, tid, myMeta,
+                                           sendUDP);
       }
     }else{
       socket = &myConn;
       callBack = sendTCP;
     }
-    
-    if(myMeta.tracks[tid].codec == "H264"){
-      long long offset = thisPacket.getInt("offset");
-      tracks[tid].pack.setTimestamp(90 * (timestamp + offset));
-      unsigned long sent = 0;
-      while (sent < dataLen) {
-        unsigned long nalSize = ntohl(*((unsigned long *)(dataPointer + sent)));
-        tracks[tid].pack.sendH264(socket, callBack, dataPointer + sent + 4, nalSize, tracks[tid].channel);
-        sent += nalSize + 4;
-      }
-      return;
-    }
 
-    //Default packager
-    tracks[tid].pack.setTimestamp(timestamp * ((double) myMeta.tracks[tid].rate / 1000.0));
-    tracks[tid].pack.sendData(socket, callBack, dataPointer, dataLen, tracks[tid].channel,myMeta.tracks[tid].codec);
-    
+    uint64_t offset = thisPacket.getInt("offset");
+    sdpState.tracks[tid].pack.setTimestamp((timestamp+offset) * SDP::getMultiplier(myMeta.tracks[tid]));
+    sdpState.tracks[tid].pack.sendData(socket, callBack, dataPointer, dataLen,
+                                       sdpState.tracks[tid].channel, myMeta.tracks[tid].codec);
   }
 
   /// This request handler also checks for UDP packets
   void OutRTSP::requestHandler(){
-    if (!expectTCP){
-      handleUDP();
-    }
+    if (!expectTCP){handleUDP();}
     Output::requestHandler();
   }
 
   void OutRTSP::onRequest(){
     RTP::MAX_SEND = config->getInteger("maxsend");
-    //if needed, parse TCP packets, and cancel if it is not safe (yet) to read HTTP/RTSP packets
+    // if needed, parse TCP packets, and cancel if it is not safe (yet) to read HTTP/RTSP packets
     while ((!expectTCP || handleTCP()) && HTTP_R.Read(myConn)){
-      //cancel broken URLs
+      // cancel broken URLs
       if (HTTP_R.url.size() < 8){
-        WARN_MSG("Invalid data found in RTSP input around ~%llub - disconnecting!", myConn.dataDown());
+        WARN_MSG("Invalid data found in RTSP input around ~%llub - disconnecting!",
+                 myConn.dataDown());
         myConn.close();
         break;
       }
       HTTP_S.Clean();
       HTTP_S.protocol = "RTSP/1.0";
-      
-      //set the streamname and session
+
+      // set the streamname and session
       if (!source.size()){
         std::string source = HTTP_R.url.substr(7);
-        unsigned int loc = std::min(source.find(':'),source.find('/'));
-        source = source.substr(0,loc);
+        unsigned int loc = std::min(source.find(':'), source.find('/'));
+        source = source.substr(0, loc);
       }
       size_t found = HTTP_R.url.find('/', 7);
       if (!streamName.size()){
@@ -172,30 +174,29 @@ namespace Mist {
         Util::sanitizeName(streamName);
       }
       if (streamName.size()){
-        HTTP_S.SetHeader("Session", Secure::md5(HTTP_S.GetHeader("User-Agent") + getConnectedHost()) + "_" + streamName);
+        HTTP_S.SetHeader("Session",
+                         Secure::md5(HTTP_S.GetHeader("User-Agent") + getConnectedHost()) + "_" +
+                             streamName);
       }
-      
-      //set the date
+
+      // set the date
       time_t timer;
       time(&timer);
-      struct tm * timeNow = gmtime(&timer);
+      struct tm *timeNow = gmtime(&timer);
       char dString[42];
       strftime(dString, 42, "%a, %d %h %Y, %X GMT", timeNow);
       HTTP_S.SetHeader("Date", dString);
-      
-      //set the sequence number to match the received sequence number
-      if (HTTP_R.hasHeader("CSeq")){
-        HTTP_S.SetHeader("CSeq", HTTP_R.GetHeader("CSeq"));
-      }
-      if (HTTP_R.hasHeader("Cseq")){
-        HTTP_S.SetHeader("CSeq", HTTP_R.GetHeader("Cseq"));
-      }
-      
+
+      // set the sequence number to match the received sequence number
+      if (HTTP_R.hasHeader("CSeq")){HTTP_S.SetHeader("CSeq", HTTP_R.GetHeader("CSeq"));}
+      if (HTTP_R.hasHeader("Cseq")){HTTP_S.SetHeader("CSeq", HTTP_R.GetHeader("Cseq"));}
+
       INFO_MSG("Handling %s", HTTP_R.method.c_str());
 
-      //handle the request
+      // handle the request
       if (HTTP_R.method == "OPTIONS"){
-        HTTP_S.SetHeader("Public", "SETUP, TEARDOWN, PLAY, PAUSE, DESCRIBE, GET_PARAMETER, ANNOUNCE, RECORD");
+        HTTP_S.SetHeader("Public",
+                         "SETUP, TEARDOWN, PLAY, PAUSE, DESCRIBE, GET_PARAMETER, ANNOUNCE, RECORD");
         HTTP_S.SendResponse("200", "OK", myConn);
         HTTP_R.Clean();
         continue;
@@ -210,27 +211,35 @@ namespace Mist {
         selectedTracks.clear();
         std::stringstream transportString;
         transportString << "v=0\r\n"
-        "o=- " << Util::getMS() << " 1 IN IP4 127.0.0.1\r\n"
-        "s=" << streamName << "\r\n"
-        "c=IN IP4 0.0.0.0\r\n"
-        "i=" << streamName << "\r\n"
-        "u=" << HTTP_R.url.substr(0, HTTP_R.url.rfind('/')) << "/" << streamName << "\r\n"
-        "t=0 0\r\n"
-        "a=tool:MistServer\r\n"
-        "a=type:broadcast\r\n"
-        "a=control:*\r\n";
+                           "o=- "
+                        << Util::getMS() << " 1 IN IP4 127.0.0.1\r\n"
+                                            "s="
+                        << streamName << "\r\n"
+                                         "c=IN IP4 0.0.0.0\r\n"
+                                         "i="
+                        << streamName << "\r\n"
+                                         "u="
+                        << HTTP_R.url.substr(0, HTTP_R.url.rfind('/')) << "/" << streamName
+                        << "\r\n"
+                           "t=0 0\r\n"
+                           "a=tool:MistServer\r\n"
+                           "a=type:broadcast\r\n"
+                           "a=control:*\r\n";
         if (myMeta.live){
           transportString << "a=range:npt=" << ((double)startTime()) / 1000.0 << "-\r\n";
         }else{
-          transportString << "a=range:npt=" << ((double)startTime()) / 1000.0 << "-" << ((double)endTime()) / 1000.0 << "\r\n";
+          transportString << "a=range:npt=" << ((double)startTime()) / 1000.0 << "-"
+                          << ((double)endTime()) / 1000.0 << "\r\n";
         }
-        
-        for (std::map<unsigned int, DTSC::Track>::iterator objIt = myMeta.tracks.begin(); objIt != myMeta.tracks.end(); ++objIt) {
-          transportString << tracks[objIt->first].mediaDescription(objIt->second);
+
+        for (std::map<unsigned int, DTSC::Track>::iterator objIt = myMeta.tracks.begin();
+             objIt != myMeta.tracks.end(); ++objIt){
+          transportString << SDP::mediaDescription(objIt->second);
         }
         transportString << "\r\n";
         HIGH_MSG("Reply: %s", transportString.str().c_str());
-        HTTP_S.SetHeader("Content-Base", HTTP_R.url.substr(0, HTTP_R.url.rfind('/')) + "/" + streamName);
+        HTTP_S.SetHeader("Content-Base",
+                         HTTP_R.url.substr(0, HTTP_R.url.rfind('/')) + "/" + streamName);
         HTTP_S.SetHeader("Content-Type", "application/sdp");
         HTTP_S.SetBody(transportString.str());
         HTTP_S.SendResponse("200", "OK", myConn);
@@ -238,95 +247,72 @@ namespace Mist {
         continue;
       }
       if (HTTP_R.method == "SETUP"){
-        size_t trackPos = HTTP_R.url.rfind("/track");
-        if (trackPos != std::string::npos){
-          unsigned int trId = atol(HTTP_R.url.substr(trackPos + 6).c_str());
-          if (myMeta.tracks.count(trId)){
-            if (tracks[trId].parseTransport(HTTP_R.GetHeader("Transport"), getConnectedHost(), source, myMeta.tracks[trId])){
-              selectedTracks.insert(trId);
-              HTTP_S.SetHeader("Expires", HTTP_S.GetHeader("Date"));
-              HTTP_S.SetHeader("Transport", tracks[trId].transportString);
-              HTTP_S.SetHeader("Cache-Control", "no-cache");
-              HTTP_S.SendResponse("200", "OK", myConn);
-            }else{
-              HTTP_S.SendResponse("404", "Track not available", myConn);
-            }
-            HTTP_R.Clean();
-            continue;
-          }
+        uint32_t trackNo = sdpState.parseSetup(HTTP_R, getConnectedHost(), source);
+        HTTP_S.SetHeader("Expires", HTTP_S.GetHeader("Date"));
+        HTTP_S.SetHeader("Cache-Control", "no-cache");
+        if (trackNo){
+          selectedTracks.insert(trackNo);
+          SDP::Track &sdpTrack = sdpState.tracks[trackNo];
+          if (sdpTrack.channel != -1){expectTCP = true;}
+          HTTP_S.SetHeader("Transport", sdpTrack.transportString);
+          HTTP_S.SendResponse("200", "OK", myConn);
+          INFO_MSG("Setup completed for track %lu (%s): %s", trackNo,
+                   myMeta.tracks[trackNo].codec.c_str(), sdpTrack.transportString.c_str());
+        }else{
+          HTTP_S.SendResponse("404", "Track not known or allowed", myConn);
+          FAIL_MSG("Could not handle setup for %s", HTTP_R.url.c_str());
         }
-        //might be push setup - check known control points
-        if (pushing && tracks.size()){
-          bool setupHandled = false;
-          for (std::map<int, RTPTrack>::iterator it = tracks.begin(); it != tracks.end(); ++it){
-            if (it->second.control.size() && (HTTP_R.url.find(it->second.control) != std::string::npos || HTTP_R.GetVar("pass").find(it->second.control) != std::string::npos)){
-              if (it->second.parseTransport(HTTP_R.GetHeader("Transport"), getConnectedHost(), source, myMeta.tracks[it->first])){
-                if (it->second.channel != -1){
-                  expectTCP = true;
-                }
-                HTTP_S.SetHeader("Expires", HTTP_S.GetHeader("Date"));
-                HTTP_S.SetHeader("Transport", it->second.transportString);
-                HTTP_S.SetHeader("Cache-Control", "no-cache");
-                HTTP_S.SendResponse("200", "OK", myConn);
-                INFO_MSG("Setup completed for track %llu (%s): %s", it->first, myMeta.tracks[it->first].codec.c_str(), it->second.transportString.c_str());
-              }else{
-                HTTP_S.SendResponse("404", "Track not known or allowed", myConn);
-              }
-              setupHandled = true;
-              HTTP_R.Clean();
-              break;
-            }
-          }
-          if (!setupHandled){
-            HTTP_S.SendResponse("404", "Track not known", myConn);
-            HTTP_R.Clean();
-          }
-          continue;
-        }
-        FAIL_MSG("Could not handle setup: pushing=%s, trackSize=%u", pushing?"true":"false", tracks.size());
+        HTTP_R.Clean();
+        continue;
       }
       if (HTTP_R.method == "PLAY"){
         initialSeek();
         std::string range = HTTP_R.GetHeader("Range");
         if (range != ""){
-          range = range.substr(range.find("npt=")+4);
-          if (!range.empty()) {
+          range = range.substr(range.find("npt=") + 4);
+          if (!range.empty()){
             range = range.substr(0, range.find('-'));
-            uint64_t targetPos = 1000*atof(range.c_str());
+            uint64_t targetPos = 1000 * atof(range.c_str());
             if (targetPos || myMeta.vod){seek(targetPos);}
           }
         }
         std::stringstream rangeStr;
         if (myMeta.live){
-          rangeStr << "npt=" << currentTime()/1000 << "." << std::setw(3) << std::setfill('0') << currentTime()%1000 << "-";
+          rangeStr << "npt=" << currentTime() / 1000 << "." << std::setw(3) << std::setfill('0')
+                   << currentTime() % 1000 << "-";
         }else{
-          rangeStr << "npt=" << currentTime()/1000 << "." << std::setw(3) << std::setfill('0') << currentTime()%1000 << "-" << std::setw(1) << endTime()/1000 << "." << std::setw(3) << std::setfill('0') << endTime()%1000;
+          rangeStr << "npt=" << currentTime() / 1000 << "." << std::setw(3) << std::setfill('0')
+                   << currentTime() % 1000 << "-" << std::setw(1) << endTime() / 1000 << "."
+                   << std::setw(3) << std::setfill('0') << endTime() % 1000;
         }
         HTTP_S.SetHeader("Range", rangeStr.str());
         std::stringstream infoString;
         if (selectedTracks.size()){
-          for (std::set<unsigned long>::iterator it = selectedTracks.begin(); it != selectedTracks.end(); ++it){
+          for (std::set<unsigned long>::iterator it = selectedTracks.begin();
+               it != selectedTracks.end(); ++it){
             if (!infoString.str().empty()){infoString << ",";}
-            infoString << tracks[*it].rtpInfo(myMeta.tracks[*it], source+"/"+streamName, currentTime());
+            infoString << sdpState.tracks[*it].rtpInfo(myMeta.tracks[*it],
+                                                       source + "/" + streamName, currentTime());
           }
         }
         HTTP_S.SetHeader("RTP-Info", infoString.str());
         HTTP_S.SendResponse("200", "OK", myConn);
-        parseData = true; 
+        parseData = true;
         HTTP_R.Clean();
         continue;
       }
       if (HTTP_R.method == "PAUSE"){
         HTTP_S.SendResponse("200", "OK", myConn);
         std::string range = HTTP_R.GetHeader("Range");
-        if (!range.empty()){
-          range = range.substr(range.find("npt=")+4);
-        }
+        if (!range.empty()){range = range.substr(range.find("npt=") + 4);}
         if (range.empty()){
           stop();
         }else{
-          pausepoint = 1000 * (int) atof(range.c_str());
-          if (pausepoint > currentTime()){pausepoint = 0; stop();}
+          pausepoint = 1000 * (int)atof(range.c_str());
+          if (pausepoint > currentTime()){
+            pausepoint = 0;
+            stop();
+          }
         }
         HTTP_R.Clean();
         continue;
@@ -343,7 +329,7 @@ namespace Mist {
           return;
         }
         INFO_MSG("Pushing to stream %s", streamName.c_str());
-        parseSDP(HTTP_R.body);
+        sdpState.parseSDP(HTTP_R.body);
         HTTP_S.SendResponse("200", "OK", myConn);
         HTTP_R.Clean();
         continue;
@@ -360,519 +346,84 @@ namespace Mist {
 
   /// Disconnects the user
   bool OutRTSP::onFinish(){
-    if (myConn){
-      myConn.close();
-    }
+    if (myConn){myConn.close();}
     return false;
   }
 
   /// Attempts to parse TCP RTP packets at the beginning of the header.
   /// Returns whether it is safe to attempt to read HTTP/RTSP packets (true) or not (false).
   bool OutRTSP::handleTCP(){
-    if (!myConn.Received().size() || !myConn.Received().available(1)){return false;}//no data
-    if (myConn.Received().copy(1) != "$"){return true;}//not a TCP RTP packet
-    if (!myConn.Received().available(4)){return false;}//a TCP RTP packet, but not complete yet
-    //We have a TCP packet! Read it...
-    //Format: 1 byte '$', 1 byte channel, 2 bytes len, len bytes binary data
+    if (!myConn.Received().size() || !myConn.Received().available(1)){return false;}// no data
+    if (myConn.Received().copy(1) != "$"){return true;}// not a TCP RTP packet
+    if (!myConn.Received().available(4)){return false;}// a TCP RTP packet, but not complete yet
+    // We have a TCP packet! Read it...
+    // Format: 1 byte '$', 1 byte channel, 2 bytes len, len bytes binary data
     std::string tcpHead = myConn.Received().copy(4);
-    uint16_t len = ntohs(*(short*)(tcpHead.data()+2));
-    if (!myConn.Received().available(len+4)){return false;}//a TCP RTP packet, but not complete yet
-    //remove whole packet from buffer, including 4 byte header
-    std::string tcpPacket = myConn.Received().remove(len+4);
-    for (std::map<int, RTPTrack>::iterator it = tracks.begin(); it != tracks.end(); ++it){
-      if (tcpHead.data()[1] == it->second.channel){
-        RTP::Packet pkt(tcpPacket.data()+4, len);
-        it->second.rtpSeq = pkt.getSequence();
-        handleIncomingRTP(it->first, pkt);
-        break;
-      }
+    uint16_t len = ntohs(*(short *)(tcpHead.data() + 2));
+    if (!myConn.Received().available(len + 4)){
+      return false;
+    }// a TCP RTP packet, but not complete yet
+    // remove whole packet from buffer, including 4 byte header
+    std::string tcpPacket = myConn.Received().remove(len + 4);
+    uint32_t trackNo = sdpState.getTrackNoForChannel(tcpHead.data()[1]);
+    if (trackNo && isPushing()){
+      RTP::Packet pkt(tcpPacket.data() + 4, len);
+      sdpState.tracks[trackNo].rtpSeq = pkt.getSequence();
+      sdpState.handleIncomingRTP(trackNo, pkt);
     }
-    //attempt to read more packets
+    // attempt to read more packets
     return handleTCP();
   }
 
   /// Reads and handles RTP packets over UDP, if needed
   void OutRTSP::handleUDP(){
-    for (std::map<int, RTPTrack>::iterator it = tracks.begin(); it != tracks.end(); ++it){
-      Socket::UDPConnection & s = it->second.data;
+    if (!isPushing()){return;}
+    for (std::map<uint32_t, SDP::Track>::iterator it = sdpState.tracks.begin();
+         it != sdpState.tracks.end(); ++it){
+      Socket::UDPConnection &s = it->second.data;
       while (s.Receive()){
         if (s.getDestPort() != it->second.cPort){
-          //wrong sending port, ignore packet
+          // wrong sending port, ignore packet
           continue;
         }
-        lastRecv = Util::epoch();//prevent disconnect of idle TCP connection when using UDP
+        lastRecv = Util::epoch(); // prevent disconnect of idle TCP connection when using UDP
         myConn.addDown(s.data_len);
         RTP::Packet pack(s.data, s.data_len);
         if (!it->second.rtpSeq){it->second.rtpSeq = pack.getSequence();}
-        //packet is very early - assume dropped after 10 packets
+        // packet is very early - assume dropped after 10 packets
         while ((int16_t)(((uint16_t)it->second.rtpSeq) - ((uint16_t)pack.getSequence())) < -10){
           WARN_MSG("Giving up on packet %u", it->second.rtpSeq);
           ++(it->second.rtpSeq);
-          //send any buffered packets we may have
+          // send any buffered packets we may have
           while (it->second.packBuffer.count(it->second.rtpSeq)){
-            handleIncomingRTP(it->first, pack);
+            sdpState.handleIncomingRTP(it->first, pack);
             ++(it->second.rtpSeq);
           }
         }
-        //send any buffered packets we may have
+        // send any buffered packets we may have
         while (it->second.packBuffer.count(it->second.rtpSeq)){
-          handleIncomingRTP(it->first, pack);
+          sdpState.handleIncomingRTP(it->first, pack);
           ++(it->second.rtpSeq);
         }
-        //packet is slightly early - buffer it
+        // packet is slightly early - buffer it
         if (((int16_t)(((uint16_t)it->second.rtpSeq) - ((uint16_t)pack.getSequence())) < 0)){
           INFO_MSG("Buffering early packet #%u->%u", it->second.rtpSeq, pack.getSequence());
           it->second.packBuffer[pack.getSequence()] = pack;
         }
-        //packet is late
+        // packet is late
         if ((int16_t)(((uint16_t)it->second.rtpSeq) - ((uint16_t)pack.getSequence())) > 0){
-          //negative difference?
-          WARN_MSG("Dropped a packet that arrived too late! (%d packets difference)", (int16_t)(((uint16_t)it->second.rtpSeq) - ((uint16_t)pack.getSequence())));
+          // negative difference?
+          WARN_MSG("Dropped a packet that arrived too late! (%d packets difference)",
+                   (int16_t)(((uint16_t)it->second.rtpSeq) - ((uint16_t)pack.getSequence())));
           return;
         }
-        //packet is in order
+        // packet is in order
         if (it->second.rtpSeq == pack.getSequence()){
-          handleIncomingRTP(it->first, pack);
+          sdpState.handleIncomingRTP(it->first, pack);
           ++(it->second.rtpSeq);
         }
       }
     }
   }
-
-  /// Handles a single H264 packet, checking if others are appended at the end in Annex B format.
-  /// If so, splits them up and calls h264Packet for each. If not, calls it only once for the whole payload.
-  void OutRTSP::h264MultiParse(uint64_t ts, const uint64_t track, char * buffer, const uint32_t len){
-    uint32_t lastStart = 0;
-    for (uint32_t i = 0; i < len-4; ++i){
-      //search for start code
-      if (buffer[i] == 0 && buffer[i+1] == 0 && buffer[i+2] == 0 && buffer[i+3] == 1){
-        //if found, handle a packet from the last start code up to this start code
-        Bit::htobl(buffer+lastStart, (i-lastStart-1)-4);//size-prepend
-        h264Packet(ts, track, buffer+lastStart, (i-lastStart-1), h264::isKeyframe(buffer+lastStart+4, i-lastStart-5));
-        lastStart = i;
-      }
-    }
-    //Last packet (might be first, if no start codes found)
-    Bit::htobl(buffer+lastStart, (len-lastStart)-4);//size-prepend
-    h264Packet(ts, track, buffer+lastStart, (len-lastStart), h264::isKeyframe(buffer+lastStart+4, len-lastStart-4));
-  }
-
-  void OutRTSP::h264Packet(uint64_t ts, const uint64_t track, const char * buffer, const uint32_t len, bool isKey){
-    //Ignore zero-length packets (e.g. only contained init data and nothing else)
-    if (!len){return;}
-    
-    //Header data? Compare to init, set if needed, and throw away
-    uint8_t nalType = (buffer[4] & 0x1F);
-    switch (nalType){
-      case 7: //SPS
-        if (tracks[track].spsData.size() != len-4 || memcmp(buffer+4, tracks[track].spsData.data(), len-4) != 0){
-          INFO_MSG("Updated SPS from RTP data");
-          tracks[track].spsData.assign(buffer+4, len-4);
-          updateH264Init(track);
-        }
-        return;
-      case 8: //PPS
-        if (tracks[track].ppsData.size() != len-4 || memcmp(buffer+4, tracks[track].ppsData.data(), len-4) != 0){
-          INFO_MSG("Updated PPS from RTP data");
-          tracks[track].ppsData.assign(buffer+4, len-4);
-          updateH264Init(track);
-        }
-        return;
-      default://others, continue parsing
-        break;
-    }
-
-    double fps = tracks[track].fpsMeta;
-    uint32_t offset = 0;
-    uint64_t newTs = ts;
-    if (fps > 1){
-      //Assume a steady frame rate, clip the timestamp based on frame number.
-      uint64_t frameNo = (ts / (1000.0/fps))+0.5;
-      while (frameNo < tracks[track].packCount){
-        tracks[track].packCount--;
-      }
-      //More than 32 frames behind? We probably skipped something, somewhere...
-      if ((frameNo-tracks[track].packCount) > 32){
-        tracks[track].packCount = frameNo;
-      }
-      //After some experimentation, we found that the time offset is the difference between the frame number and the packet counter, times the frame rate in ms
-      offset = (frameNo-tracks[track].packCount) * (1000.0/fps);
-      //... and the timestamp is the packet counter times the frame rate in ms.
-      newTs = tracks[track].packCount * (1000.0/fps);
-      VERYHIGH_MSG("Packing time %llu = %sframe %llu (%.2f FPS). Expected %llu -> +%llu/%lu", ts, isKey?"key":"i", frameNo, fps, tracks[track].packCount, (frameNo-tracks[track].packCount), offset);
-    }else{
-      //For non-steady frame rate, assume no offsets are used and the timestamp is already correct
-      VERYHIGH_MSG("Packing time %llu = %sframe %llu (variable rate)", ts, isKey?"key":"i", tracks[track].packCount);
-    }
-    //Fill the new DTSC packet, buffer it.
-    DTSC::Packet nextPack;
-    nextPack.genericFill(newTs, offset, track, buffer, len, 0, isKey);
-    tracks[track].packCount++;
-    bufferLivePacket(nextPack);
-  }
-
-  /// Handles RTP packets generically, for both TCP and UDP-based connections.
-  /// In case of UDP, expects packets to be pre-sorted.
-  void OutRTSP::handleIncomingRTP(const uint64_t track, const RTP::Packet & pkt){
-    if (!tracks[track].firstTime){
-      tracks[track].firstTime = pkt.getTimeStamp() + 1;
-    }
-    if (myMeta.tracks[track].codec == "ALAW" || myMeta.tracks[track].codec == "opus" || myMeta.tracks[track].codec == "MP3" || myMeta.tracks[track].codec == "PCM"){
-      char * pl = pkt.getPayload();
-      DTSC::Packet nextPack;
-      nextPack.genericFill((pkt.getTimeStamp() - tracks[track].firstTime + 1) / ((double)myMeta.tracks[track].rate / 1000.0), 0, track, pl, pkt.getPayloadSize(), 0, false);
-      bufferLivePacket(nextPack);
-      return;
-    }
-    if (myMeta.tracks[track].codec == "AAC"){
-      //assume AAC packets are single AU units
-      /// \todo Support other input than single AU units
-      char * pl = pkt.getPayload();
-      unsigned int headLen = (Bit::btohs(pl) >> 3) + 2;//in bits, so /8, plus two for the prepended size
-      DTSC::Packet nextPack;
-      uint16_t samples = aac::AudSpecConf::samples(myMeta.tracks[track].init);
-      uint32_t sampleOffset = 0;
-      uint32_t offset = 0;
-      uint32_t auSize = 0;
-      for (uint32_t i = 2; i < headLen; i += 2){
-        auSize = Bit::btohs(pl+i) >> 3;//only the upper 13 bits
-        nextPack.genericFill((pkt.getTimeStamp() + sampleOffset - tracks[track].firstTime + 1) / ((double)myMeta.tracks[track].rate / 1000.0), 0, track, pl+headLen+offset, std::min(auSize, pkt.getPayloadSize() - headLen - offset), 0, false);
-        offset += auSize;
-        sampleOffset += samples;
-        bufferLivePacket(nextPack);
-      }
-      return;
-    }
-    if (myMeta.tracks[track].codec == "H264"){
-      //Handles common H264 packets types, but not all.
-      //Generalizes and converts them all to a data format ready for DTSC, then calls h264Packet for that data.
-      //Prints a WARN-level message if packet type is unsupported.
-      /// \todo Support other H264 packets types?
-      char * pl = pkt.getPayload();
-      if (!pkt.getPayloadSize()){
-        WARN_MSG("Empty packet ignored!");
-        return;
-      }
-      if ((pl[0] & 0x1F) == 0){
-        WARN_MSG("H264 packet type null ignored");
-        return;
-      }
-      if ((pl[0] & 0x1F) < 24){
-        DONTEVEN_MSG("H264 single packet, type %u", (unsigned int)(pl[0] & 0x1F));
-        static char * packBuffer = 0;
-        static unsigned long packBufferSize = 0;
-        unsigned long len = pkt.getPayloadSize();
-        if (packBufferSize < len+4){
-          char * tmp = (char*)realloc(packBuffer, len+4);
-          if (tmp){
-            packBuffer = tmp;
-            packBufferSize = len+4;
-          }else{
-            free(packBuffer);
-            packBufferSize = 0;
-            packBuffer = 0;
-            FAIL_MSG("Failed to allocate memory for H264 packet");
-            return;
-          }
-        }
-        Bit::htobl(packBuffer, len);//size-prepend
-        memcpy(packBuffer+4, pl, len);
-        h264Packet((pkt.getTimeStamp() - tracks[track].firstTime + 1) / 90, track, packBuffer, len+4, h264::isKeyframe(packBuffer+4, len));
-        return;
-      }
-      if ((pl[0] & 0x1F) == 24){
-        DONTEVEN_MSG("H264 STAP-A packet");
-        unsigned int len = 0;
-        unsigned int pos = 1;
-        while (pos + 1 < pkt.getPayloadSize()){
-          unsigned int pLen = Bit::btohs(pl+pos);
-          INSANE_MSG("Packet of %ub and type %u", pLen, (unsigned int)(pl[pos+2] & 0x1F));
-          pos += 2+pLen;
-          len += 4+pLen;
-        }
-        static char * packBuffer = 0;
-        static unsigned long packBufferSize = 0;
-        if (packBufferSize < len){
-          char * tmp = (char*)realloc(packBuffer, len);
-          if (tmp){
-            packBuffer = tmp;
-            packBufferSize = len;
-          }else{
-            free(packBuffer);
-            packBufferSize = 0;
-            packBuffer = 0;
-            FAIL_MSG("Failed to allocate memory for H264 STAP-A packet");
-            return;
-          }
-        }
-        pos = 1;
-        len = 0;
-        bool isKey = false;
-        while (pos + 1 < pkt.getPayloadSize()){
-          unsigned int pLen = Bit::btohs(pl+pos);
-          isKey |= h264::isKeyframe(pl+pos+2, pLen);
-          Bit::htobl(packBuffer+len, pLen);//size-prepend
-          memcpy(packBuffer+len+4, pl+pos+2, pLen);
-          len += 4+pLen;
-          pos += 2+pLen;
-        }
-        h264Packet((pkt.getTimeStamp() - tracks[track].firstTime + 1) / 90, track, packBuffer, len, isKey);
-        return;
-      }
-      if ((pl[0] & 0x1F) == 28){
-        DONTEVEN_MSG("H264 FU-A packet");
-        static char * fuaBuffer = 0;
-        static unsigned long fuaBufferSize = 0;
-        static unsigned long fuaCurrLen = 0;
-
-        //No length yet? Check for start bit. Ignore rest.
-        if (!fuaCurrLen && (pkt.getPayload()[1] & 0x80) == 0){
-          HIGH_MSG("Not start of a new FU-A - throwing away");
-          return;
-        }
-        if (fuaCurrLen && ((pkt.getPayload()[1] & 0x80) || (tracks[track].rtpSeq != pkt.getSequence()))){
-          WARN_MSG("Ending unfinished FU-A");
-          INSANE_MSG("H264 FU-A packet incompleted: %lu", fuaCurrLen);
-          uint8_t nalType = (fuaBuffer[4] & 0x1F);
-          if (nalType == 7 || nalType == 8){
-            //attempt to detect multiple H264 packets, even though specs disallow it
-            h264MultiParse((pkt.getTimeStamp() - tracks[track].firstTime + 1) / 90, track, fuaBuffer, fuaCurrLen);
-          }else{
-            Bit::htobl(fuaBuffer, fuaCurrLen-4);//size-prepend
-            fuaBuffer[4] |= 0x80;//set error bit
-            h264Packet((pkt.getTimeStamp() - tracks[track].firstTime + 1) / 90, track, fuaBuffer, fuaCurrLen, h264::isKeyframe(fuaBuffer+4, fuaCurrLen-4));
-          }
-          fuaCurrLen = 0;
-          return;
-        }
-
-        unsigned long len = pkt.getPayloadSize() - 2;//ignore the two FU-A bytes in front
-        if (!fuaCurrLen){len += 5;}//five extra bytes for the first packet
-
-
-        if (fuaBufferSize < fuaCurrLen + len){
-          char * tmp = (char*)realloc(fuaBuffer, fuaCurrLen + len);
-          if (tmp){
-            fuaBuffer = tmp;
-            fuaBufferSize = fuaCurrLen + len;
-          }else{
-            free(fuaBuffer);
-            fuaBufferSize = 0;
-            fuaBuffer = 0;
-            FAIL_MSG("Failed to allocate memory for H264 FU-A packet");
-            return;
-          }
-        }
-  
-        if (fuaCurrLen == 0){
-          memcpy(fuaBuffer+4, pkt.getPayload()+1, pkt.getPayloadSize()-1);
-          //reconstruct first byte
-          fuaBuffer[4] = (fuaBuffer[4] & 0x1F) | (pkt.getPayload()[0] & 0xE0);
-        }else{
-          memcpy(fuaBuffer+fuaCurrLen, pkt.getPayload()+2, pkt.getPayloadSize()-2);
-        }
-        fuaCurrLen += len;
-
-        if (pkt.getPayload()[1] & 0x40){//last packet
-          INSANE_MSG("H264 FU-A packet type %u completed: %lu", (unsigned int)(fuaBuffer[4] & 0x1F), fuaCurrLen);
-          uint8_t nalType = (fuaBuffer[4] & 0x1F);
-          if (nalType == 7 || nalType == 8){
-            //attempt to detect multiple H264 packets, even though specs disallow it
-            h264MultiParse((pkt.getTimeStamp() - tracks[track].firstTime + 1) / 90, track, fuaBuffer, fuaCurrLen);
-          }else{
-            Bit::htobl(fuaBuffer, fuaCurrLen-4);//size-prepend
-            h264Packet((pkt.getTimeStamp() - tracks[track].firstTime + 1) / 90, track, fuaBuffer, fuaCurrLen, h264::isKeyframe(fuaBuffer+4, fuaCurrLen-4));
-          }
-          fuaCurrLen = 0;
-        }
-        return;
-      }
-      WARN_MSG("H264 packet type %u unsupported", (unsigned int)(pl[0] & 0x1F));
-      return;
-    }
-  }
-
-  /// Calculates H264 track metadata from sps and pps data stored in tracks[trackNo]
-  void OutRTSP::updateH264Init(uint64_t trackNo){
-    DTSC::Track & Trk = myMeta.tracks[trackNo];
-    RTPTrack & RTrk = tracks[trackNo];
-    h264::sequenceParameterSet sps(RTrk.spsData.data(), RTrk.spsData.size());
-    h264::SPSMeta hMeta = sps.getCharacteristics();
-    MP4::AVCC avccBox;
-    avccBox.setVersion(1);
-    avccBox.setProfile(RTrk.spsData[1]);
-    avccBox.setCompatibleProfiles(RTrk.spsData[2]);
-    avccBox.setLevel(RTrk.spsData[3]);
-    avccBox.setSPSNumber(1);
-    avccBox.setSPS(RTrk.spsData);
-    avccBox.setPPSNumber(1);
-    avccBox.setPPS(RTrk.ppsData);
-    RTrk.fpsMeta = hMeta.fps;
-    Trk.width = hMeta.width;
-    Trk.height = hMeta.height;
-    Trk.fpks = hMeta.fps * 1000;
-    Trk.init = std::string(avccBox.payload(), avccBox.payloadSize());
-  }
-
-  void OutRTSP::parseSDP(const std::string & sdp){
-    std::stringstream ss(sdp);
-    std::string to;
-    uint64_t trackNo = 0;
-    bool nope = true; //true if we have no valid track to fill
-    DTSC::Track * thisTrack = 0;
-    while(std::getline(ss,to,'\n')){
-      if (!to.empty() && *to.rbegin() == '\r'){to.erase(to.size()-1, 1);}
-
-      // All tracks start with a media line
-      if (to.substr(0,2) == "m="){
-        nope = true;
-        ++trackNo;
-        thisTrack = &(myMeta.tracks[trackNo]);
-        std::stringstream words(to.substr(2));
-        std::string item;
-        if (getline(words, item, ' ') && (item == "audio" || item == "video")){
-            thisTrack->type = item;
-            thisTrack->trackID = trackNo;
-        }else{
-          WARN_MSG("Media type not supported: %s", item.c_str());
-          continue;
-        }
-        getline(words, item, ' ');
-        if (!getline(words, item, ' ') || item != "RTP/AVP"){
-          WARN_MSG("Media transport not supported: %s", item.c_str());
-          continue;
-        }
-        if (getline(words, item, ' ')){
-          uint64_t avp_type = JSON::Value(item).asInt();
-          switch (avp_type){
-            case 8: //PCM A-law
-              INFO_MSG("PCM A-law payload type");
-              nope = false;
-              thisTrack->codec = "ALAW";
-              thisTrack->rate = 8000;
-              thisTrack->channels = 1;
-              INFO_MSG("Incoming track %s", thisTrack->getIdentifier().c_str());
-              break;
-            case 10: //PCM Stereo, 44.1kHz
-              INFO_MSG("Linear PCM stereo 44.1kHz payload type");
-              nope = false;
-              thisTrack->codec = "PCM";
-              thisTrack->size = 16;
-              thisTrack->rate = 44100;
-              thisTrack->channels = 2;
-              INFO_MSG("Incoming track %s", thisTrack->getIdentifier().c_str());
-              break;
-            case 11: //PCM Mono, 44.1kHz
-              INFO_MSG("Linear PCM mono 44.1kHz payload type");
-              nope = false;
-              thisTrack->codec = "PCM";
-              thisTrack->rate = 44100;
-              thisTrack->size = 16;
-              thisTrack->channels = 1;
-              INFO_MSG("Incoming track %s", thisTrack->getIdentifier().c_str());
-              break;
-            default:
-              //dynamic type
-              if (avp_type >= 96 && avp_type <= 127){
-                INFO_MSG("Dynamic payload type (%llu) detected", avp_type);
-                nope = false;
-                continue;
-              }else{
-                FAIL_MSG("Payload type %llu not supported!", avp_type);
-                continue;
-              }
-          }
-        }
-        continue;
-      }
-      if (nope){continue;}//ignore lines if we have no valid track
-      // RTP mapping
-      if (to.substr(0, 8) == "a=rtpmap"){
-        std::string mediaType = to.substr(to.find(' ', 8)+1);
-        std::string trCodec = mediaType.substr(0, mediaType.find('/'));
-        //convert to fullcaps
-        for(unsigned int i=0;i<trCodec.size();++i){
-          if(trCodec[i]<=122 && trCodec[i]>=97){trCodec[i]-=32;}
-        }
-        if (thisTrack->type == "audio"){
-          std::string extraInfo = mediaType.substr(mediaType.find('/')+1);
-          if (extraInfo.find('/') != std::string::npos){
-            size_t lastSlash = extraInfo.find('/');
-            thisTrack->rate = atoll(extraInfo.substr(0, lastSlash).c_str());
-            thisTrack->channels = atoll(extraInfo.substr(lastSlash+1).c_str());
-          }else{
-            thisTrack->rate = atoll(extraInfo.c_str());
-            thisTrack->channels = 1;
-          }
-        }
-        if (trCodec == "H264"){thisTrack->codec = "H264";}
-        if (trCodec == "OPUS"){
-          thisTrack->codec = "opus";
-          thisTrack->init = std::string("OpusHead\001\002\170\000\200\273\000\000\000\000\000", 19);
-        }
-        if (trCodec == "PCMA"){thisTrack->codec = "ALAW";}
-        if (trCodec == "L8"){
-          thisTrack->codec = "PCM";
-          thisTrack->size = 8;
-        }
-        if (trCodec == "L16"){
-          thisTrack->codec = "PCM";
-          thisTrack->size = 16;
-        }
-        if (trCodec == "L20"){
-          thisTrack->codec = "PCM";
-          thisTrack->size = 20;
-        }
-        if (trCodec == "L24"){
-          thisTrack->codec = "PCM";
-          thisTrack->size = 24;
-        }
-        if (trCodec == "MPEG4-GENERIC"){thisTrack->codec = "AAC";}
-        INFO_MSG("Incoming track %s", thisTrack->getIdentifier().c_str());
-        continue;
-      }
-      if (to.substr(0, 10) == "a=control:"){
-        tracks[trackNo].control = to.substr(10);
-        continue;
-      }
-      if (to.substr(0, 7) == "a=fmtp:"){
-        tracks[trackNo].fmtp = to.substr(7);
-        if (thisTrack->codec == "AAC"){
-          if (tracks[trackNo].getParamString("mode") != "AAC-hbr"){
-            //a=fmtp:97 profile-level-id=1;mode=AAC-hbr;sizelength=13;indexlength=3;indexdeltalength=3; config=120856E500
-            FAIL_MSG("AAC transport mode not supported: %s", tracks[trackNo].getParamString("mode").c_str());
-            nope = true;
-            myMeta.tracks.erase(trackNo);
-            tracks.erase(trackNo);
-            continue;
-          }
-          thisTrack->init = Encodings::Hex::decode(tracks[trackNo].getParamString("config"));
-          //myMeta.tracks[trackNo].rate = aac::AudSpecConf::rate(myMeta.tracks[trackNo].init);
-
-        }
-        if (thisTrack->codec == "H264"){
-          //a=fmtp:96 packetization-mode=1; sprop-parameter-sets=Z0LAHtkA2D3m//AUABqxAAADAAEAAAMAMg8WLkg=,aMuDyyA=; profile-level-id=42C01E
-          std::string sprop = tracks[trackNo].getParamString("sprop-parameter-sets");
-          size_t comma = sprop.find(',');
-          tracks[trackNo].spsData = Encodings::Base64::decode(sprop.substr(0,comma));
-          tracks[trackNo].ppsData = Encodings::Base64::decode(sprop.substr(comma+1));
-          updateH264Init(trackNo);
-        }
-        continue;
-      }
-      // We ignore bandwidth lines
-      if (to.substr(0,2) == "b="){
-        continue;
-      }
-      //we ignore everything before the first media line.
-      if (!trackNo){
-        continue;
-      }
-      //at this point, the data is definitely for a track
-      INFO_MSG("Unhandled SDP line for track %llu: %s", trackNo, to.c_str());
-    }
-
-  }
-
 }
 
diff --git a/src/output/output_rtsp.h b/src/output/output_rtsp.h
index dca61b0d..4a76bfe0 100644
--- a/src/output/output_rtsp.h
+++ b/src/output/output_rtsp.h
@@ -1,214 +1,36 @@
 #pragma once
 
 #include "output.h"
-#include <mist/socket.h>
-#include <mist/rtp.h>
-#include <mist/http_parser.h>
-#include <mist/encode.h>
 #include <mist/h264.h>
+#include <mist/http_parser.h>
+#include <mist/rtp.h>
+#include <mist/sdp.h>
+#include <mist/socket.h>
 
-namespace Mist {  
-  ///Structure used to keep track of selected tracks.
-  class RTPTrack {
-    public:
-      Socket::UDPConnection data;
-      Socket::UDPConnection rtcp;
-      RTP::Packet pack;
-      long long rtcpSent;
-      uint64_t firstTime;
-      int channel;/// Channel number, used in TCP sending
-      uint64_t packCount;
-      uint16_t rtpSeq;
-      std::map<uint16_t, RTP::Packet> packBuffer;
-      uint32_t cPort;
-      std::string transportString;
-      std::string control;
-      std::string fmtp;
-      std::string spsData;
-      std::string ppsData;
-      uint64_t fpsTime;
-      double fpsMeta;
-      double fps;
-      RTPTrack(){
-        rtcpSent = 0;
-        channel = -1;
-        firstTime = 0;
-        packCount = 0;
-        cPort = 0;
-        rtpSeq = 0;
-        fpsTime = 0;
-        fpsMeta = 0;
-        fps = 0;
-      }
-      std::string getParamString(const std::string & param) const{
-        if (!fmtp.size()){return "";}
-        size_t pos = fmtp.find(param);
-        if (pos == std::string::npos){return "";}
-        pos += param.size()+1;
-        size_t ePos = fmtp.find_first_of(" ;", pos);
-        return fmtp.substr(pos, ePos-pos);
-      }
-      uint64_t getParamInt(const std::string & param) const{
-        return atoll(getParamString(param).c_str());
-      }
-      std::string mediaDescription(const DTSC::Track & trk){
-        std::stringstream mediaDesc;
-        if (trk.codec == "H264") {
-          MP4::AVCC avccbox;
-          avccbox.setPayload(trk.init);
-          mediaDesc << "m=video 0 RTP/AVP 97\r\n"
-          "a=rtpmap:97 H264/90000\r\n"
-          "a=cliprect:0,0," << trk.height << "," << trk.width << "\r\n"
-          "a=framesize:97 " << trk.width << '-' << trk.height << "\r\n"
-          "a=fmtp:97 packetization-mode=1;profile-level-id="
-          << std::hex << std::setw(2) << std::setfill('0') << (int)trk.init.data()[1] << std::dec << "E0"
-          << std::hex << std::setw(2) << std::setfill('0') << (int)trk.init.data()[3] << std::dec << ";"
-          "sprop-parameter-sets="
-          << Encodings::Base64::encode(std::string(avccbox.getSPS(), avccbox.getSPSLen()))
-          << ","
-          << Encodings::Base64::encode(std::string(avccbox.getPPS(), avccbox.getPPSLen()))
-          << "\r\n"
-          "a=framerate:" << ((double)trk.fpks)/1000.0 << "\r\n"
-          "a=control:track" << trk.trackID << "\r\n";
-        } else if (trk.codec == "AAC") {
-          mediaDesc << "m=audio 0 RTP/AVP 96" << "\r\n"
-          "a=rtpmap:96 mpeg4-generic/" << trk.rate << "/" << trk.channels << "\r\n"
-          "a=fmtp:96 streamtype=5; profile-level-id=15; config=";
-          for (unsigned int i = 0; i < trk.init.size(); i++) {
-            mediaDesc << std::hex << std::setw(2) << std::setfill('0') << (int)trk.init[i] << std::dec;
-          }
-          //these values are described in RFC 3640
-          mediaDesc << "; mode=AAC-hbr; SizeLength=13; IndexLength=3; IndexDeltaLength=3;\r\n"
-          "a=control:track" << trk.trackID << "\r\n";
-        }else if (trk.codec == "MP3") {
-          mediaDesc << "m=" << trk.type << " 0 RTP/AVP 14" << "\r\n"
-          "a=rtpmap:14 MPA/90000/" << trk.channels << "\r\n"
-          "a=control:track" << trk.trackID << "\r\n";
-        }else if ( trk.codec == "AC3") {
-          mediaDesc << "m=audio 0 RTP/AVP 100" << "\r\n"
-          "a=rtpmap:100 AC3/" << trk.rate << "/" << trk.channels << "\r\n"
-          "a=control:track" << trk.trackID << "\r\n";
-        }else if ( trk.codec == "ALAW") {
-          if (trk.channels == 1 && trk.rate == 8000){
-            mediaDesc << "m=audio 0 RTP/AVP 8" << "\r\n";
-          }else{
-            mediaDesc << "m=audio 0 RTP/AVP 101" << "\r\n";
-            mediaDesc << "a=rtpmap:101 PCMA/" << trk.rate <<  "/" << trk.channels << "\r\n";
-          }
-          mediaDesc << "a=control:track" << trk.trackID << "\r\n";
-        }else if ( trk.codec == "PCM") {
-          if (trk.size == 16 && trk.channels == 2 && trk.rate == 44100){
-            mediaDesc << "m=audio 0 RTP/AVP 10" << "\r\n";
-          } else if (trk.size == 16 && trk.channels == 1 && trk.rate == 44100){
-            mediaDesc << "m=audio 0 RTP/AVP 11" << "\r\n";
-          }else{
-            mediaDesc << "m=audio 0 RTP/AVP 103" << "\r\n";
-            mediaDesc << "a=rtpmap:103 L" << trk.size << "/" << trk.rate <<  "/" << trk.channels << "\r\n";
-          }
-          mediaDesc << "a=control:track" << trk.trackID << "\r\n";
-        }else if ( trk.codec == "opus") {
-          mediaDesc << "m=audio 0 RTP/AVP 102" << "\r\n"
-          "a=rtpmap:102 opus/" << trk.rate << "/" << trk.channels << "\r\n"
-          "a=control:track" << trk.trackID << "\r\n";
-        }
-        return mediaDesc.str();
-      }
-      bool parseTransport(const std::string & transport, const std::string & host, const std::string & source, const DTSC::Track & trk){
-        unsigned int SSrc = rand();
-        if (trk.codec == "H264") {
-          pack = RTP::Packet(97, 1, 0, SSrc);
-        }else if(trk.codec == "AAC"){
-          pack = RTP::Packet(96, 1, 0, SSrc);
-        }else if(trk.codec == "AC3"){
-          pack = RTP::Packet(100, 1, 0, SSrc);
-        }else if(trk.codec == "MP3"){
-          pack = RTP::Packet(14, 1, 0, SSrc);
-        }else if(trk.codec == "ALAW"){
-          if (trk.channels == 1 && trk.rate == 8000){
-            pack = RTP::Packet(8, 1, 0, SSrc);
-          }else{
-            pack = RTP::Packet(101, 1, 0, SSrc);
-          }
-        }else if ( trk.codec == "PCM") {
-          if (trk.size == 16 && trk.channels == 2 && trk.rate == 44100){
-            pack = RTP::Packet(10, 1, 0, SSrc);
-          } else if (trk.size == 16 && trk.channels == 1 && trk.rate == 44100){
-            pack = RTP::Packet(11, 1, 0, SSrc);
-          }else{
-            pack = RTP::Packet(103, 1, 0, SSrc);
-          }
-        }else if(trk.codec == "opus"){
-          pack = RTP::Packet(102, 1, 0, SSrc);
-        }else{
-          ERROR_MSG("Unsupported codec %s for RTSP on track %u", trk.codec.c_str(), trk.trackID);
-          return false;
-        }
-        if (transport.find("TCP") != std::string::npos) {
-          std::string chanE =  transport.substr(transport.find("interleaved=") + 12, (transport.size() - transport.rfind('-') - 1)); //extract channel ID
-          channel = atol(chanE.c_str());
-          rtcpSent = 0;
-          transportString = transport;
-        } else {
-          channel = -1;
-          size_t port_loc = transport.rfind("client_port=") + 12;
-          cPort = atol(transport.substr(port_loc, transport.rfind('-') - port_loc).c_str());
-          uint32_t portA, portB;
-          //find available ports locally;
-          int sendbuff = 4*1024*1024;
-          data.SetDestination(host, cPort);
-          portA = data.bind(0);
-          setsockopt(data.getSock(), SOL_SOCKET, SO_SNDBUF, &sendbuff, sizeof(sendbuff));
-          rtcp.SetDestination(host, cPort + 1);
-          portB = rtcp.bind(0);
-          setsockopt(rtcp.getSock(), SOL_SOCKET, SO_SNDBUF, &sendbuff, sizeof(sendbuff));
-          std::stringstream tStr;
-          tStr << "RTP/AVP/UDP;unicast;client_port=" << cPort << '-' << cPort + 1 << ";";
-          if (source.size()){
-            tStr << "source=" << source << ";";
-          }
-          tStr << "server_port=" << portA << "-" << portB << ";ssrc=" << std::hex << SSrc << std::dec;
-          transportString = tStr.str();
-          INFO_MSG("Transport string: %s", transportString.c_str());
-        }
-        return true;
-      }
-      std::string rtpInfo(const DTSC::Track & trk, const std::string & source, uint64_t currentTime){
-        unsigned int timeMultiplier = 1;
-        timeMultiplier = ((double)trk.rate / 1000.0);
-        if (trk.codec == "H264") {
-          timeMultiplier = 90;
-        }
-        std::stringstream rInfo;
-        rInfo << "url=" << source << "/track" << trk.trackID << ";"; //get the current url, not localhost
-        rInfo << "sequence=" << pack.getSequence() << ";rtptime=" << currentTime * timeMultiplier;
-        return rInfo.str();
-      }
-  };
-  
-  class OutRTSP : public Output {
-    public:
-      OutRTSP(Socket::Connection & myConn);
-      static void init(Util::Config * cfg);
-      void sendNext();
-      void onRequest();
-      void requestHandler();
-      bool onFinish();
-    private:
-      void parseSDP(const std::string & sdp);
-      long long connectedAt;///< The timestamp the connection was made, as reference point for RTCP packets.
-      std::map<int, RTPTrack> tracks;///< List of selected tracks with RTSP-specific session data.
-      unsigned int pausepoint;///< Position to pause at, when reached
-      HTTP::Parser HTTP_R, HTTP_S;
-      std::string source;
-      uint64_t lastTimeSync;
-      bool expectTCP;
-      bool handleTCP();
-      void handleUDP();
-      void handleIncomingRTP(const uint64_t track, const RTP::Packet & pkt);
-      void h264MultiParse(uint64_t ts, const uint64_t track, char * buffer, const uint32_t len);
-      void h264Packet(uint64_t ts, const uint64_t track, const char * buffer, const uint32_t len, bool isKey);
-      void updateH264Init(uint64_t trackNo);
+namespace Mist{
+  class OutRTSP : public Output{
+  public:
+    OutRTSP(Socket::Connection &myConn);
+    static void init(Util::Config *cfg);
+    void sendNext();
+    void onRequest();
+    void requestHandler();
+    bool onFinish();
+    void incomingPacket(const DTSC::Packet &pkt);
+
+  private:
+    long long connectedAt;   ///< The timestamp the connection was made, as reference point for RTCP
+                             ///packets.
+    unsigned int pausepoint; ///< Position to pause at, when reached
+    SDP::State sdpState;
+    HTTP::Parser HTTP_R, HTTP_S;
+    std::string source;
+    uint64_t lastTimeSync;
+    bool expectTCP;
+    bool handleTCP();
+    void handleUDP();
   };
 }
 
 typedef Mist::OutRTSP mistOut;
+