/// \file Connector_RTSP/main.cpp
/// Contains the main code for the RTSP Connector
#include <queue>
#include <cmath>
#include <ctime>
#include <cstdio>
#include <string>
#include <climits>
#include <cstdlib>
#include <cstring>
#include <unistd.h>
#include <getopt.h>
#include <iostream>
#include <sstream>
#include <sys/time.h>
#include <sys/wait.h>
#include <sys/types.h>
#include <sys/epoll.h>
#include "../util/socket.h"
#include "../util/flv_tag.h"
#include "../util/http_parser.h"
//JRTPLIB
#include "rtp.h"
/// Reads a single NALU from std::cin. Expected is H.264 Bytestream format.
/// Function was used as a way of debugging data. FLV does not contain all the metadata we need, so we had to try different approaches.
/// \return The Nalu data.
/// \todo Throw this function away when everything works, it is not needed.
std::string ReadNALU( ) {
static char Separator[3] = { (char)0x00, (char)0x00, (char)0x01 };
std::string Buffer;
std::string Result;
do {
Buffer += std::cin.get();
} while ( std::cin.good() && ( Buffer.find( Separator,0,3 ) == std::string::npos ) );
if( !std::cin.good() ) { return ""; }
Result = Buffer.substr(0, Buffer.find( Separator,0,3 ) );
while( *(Result.end() - 1) == (char)0x00 ) { Result.erase( Result.end() - 1 ); }
if( Result.size() == 0 ) { Result = ReadNALU( ); }
return Result;
}
/// The main function of the connector.
/// Used by server_setup.cpp in the bottom of the file, to start up the Connector.
/// This function contains the while loop the accepts connections, and sends them data.
/// \param conn A connection with the client.
int RTSP_Handler( Socket::Connection conn ) {
/// \todo Convert this to DTSC::DTMI, with an additional DTSC::Stream/
FLV::Tag tag;// Temporary tag buffer for incoming video data.
bool PlayVideo = false;
bool PlayAudio = true;
//JRTPlib Objects to handle the RTP connection, which runs "parallel" to RTSP.
jrtplib::RTPSession VideoSession;
jrtplib::RTPSessionParams VideoParams;
jrtplib::RTPUDPv6TransmissionParams VideoTransParams;
std::string PreviousRequest = "";
std::string streamname;
Socket::Connection ss(-1);
HTTP::Parser HTTP_R, HTTP_S;
//Some clients appear to expect a single request per connection. Don't know which ones.
bool PerRequest = false;
//The main loop of the function
while(conn.connected() && !FLV::Parse_Error) {
if( HTTP_R.Read(conn ) ) {
//send Debug info to stderr.
//send the appropriate responses on RTSP Commands.
fprintf( stderr, "REQUEST:\n%s\n", HTTP_R.BuildRequest().c_str() );
HTTP_S.protocol = "RTSP/1.0";
if( HTTP_R.method == "OPTIONS" ) {
//Always return the requested CSeq value.
HTTP_S.SetHeader( "CSeq", HTTP_R.GetHeader( "CSeq" ).c_str() );
//The minimal set of options required for RTSP, add new options here as well if we want to support these.
HTTP_S.SetHeader( "Public", "OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY" );
//End the HTTP body, IMPORTANT!! Connection hangs otherwise!!
HTTP_S.SetBody( "\r\n\r\n" );
fprintf( stderr, "RESPONSE:\n%s\n", HTTP_S.BuildResponse( "200", "OK" ).c_str() );
conn.write( HTTP_S.BuildResponse( "200", "OK" ) );
} else if ( HTTP_R.method == "DESCRIBE" ) {
///\todo Implement DESCRIBE option.
//Don't know if a 501 response is seen as valid. If it is, don't bother changing it.
if( HTTP_R.GetHeader( "Accept" ).find( "application/sdp" ) == std::string::npos ) {
fprintf( stderr, "RESPONSE:\n%s\n", HTTP_S.BuildResponse( "501", "Not Implemented" ).c_str() );
conn.write( HTTP_S.BuildResponse( "501", "Not Implemented" ) );
} else {
HTTP_S.SetHeader( "CSeq", HTTP_R.GetHeader( "CSeq" ).c_str() );
HTTP_S.SetHeader( "Content-Type", "application/sdp" );
/// \todo Retrieve presence of video and audio data, and process into response. Can now easily be done through DTSC::DTMI
/// \todo Retrieve Packetization mode ( is 0 for now ). I suppose this is the H264 packetization mode. Can maybe be retrieved from the docs on H64.
/// \todo Send a valid SDP file.
/// \todo Add audio to SDP file.
//This is just a dummy with data that was supposedly right for our teststream.
//SDP Docs: http://tools.ietf.org/html/rfc4566
HTTP_S.SetBody( "v=0\r\n" //protocol version
"o=- 0 0 IN IP4 ddvtech.com\r\n" //originator and session identifier (5.2):
//username sess-id sess-version nettype addrtype unicast-addr
//"-": no concept of User IDs, nettype IN(ternet)
//IP4: following address is a FQDN for IPv4
"s=Fifa Test\r\n" //session name (5.3)
//"c" - destination is specified in SETUP per rfc2326 C.1.7, set null as recommended
"c=IN IP4 0.0.0.0\r\n" //connection information -- not required if included in all media
//nettype addrtype connection-address
"t=0 0\r\n" //time the session is active: start-time stop-time; "0 0"=permanent session
"a=recvonly\r\n"//zero or more session attribute lines
"m=video 0 RTP/AVP 98\r\n"//media name and transport address: media port proto fmt ...
"a=control:" + HTTP_R.url + "\r\n"//rfc2326 C.1.1, URL for aggregate control on session level
"a=rtpmap:98 H264/90000\r\n"//rfc2326 C.1.3, dynamic payload type; see also http://tools.ietf.org/html/rfc1890#section-5
"a=fmtp:98 packetization-mode=0"//codec-specific parameters
"\r\n\r\n");//m=audio 0 RTP/AAP 96\r\na=control:rtsp://localhost/fifa/audio\r\na=rtpmap:96 mpeg4-generic/16000/2\r\n\r\n");
//important information when supporting multiple streams http://tools.ietf.org/html/rfc2326#appendix-C.3
fprintf( stderr, "RESPONSE:\n%s\n", HTTP_S.BuildResponse( "200", "OK" ).c_str() );
conn.write( HTTP_S.BuildResponse( "200", "OK" ) );
}
} else if ( HTTP_R.method == "SETUP" ) {
bool setup_session = false;//whether a session should be setup or not
std::string temp = HTTP_R.GetHeader("Transport");
//Extract the random UTP pair for video data ( RTP/RTCP)
int ClientRTPLoc = temp.find( "client_port=" ) + 12;
int PortSpacer = temp.find( "-", ClientRTPLoc );
int RTPClientPort = atoi( temp.substr( ClientRTPLoc, ( PortSpacer - ClientRTPLoc ) ).c_str() );
if( HTTP_S.GetHeader( "Session" ) != "" ) {
//Return an error if a second client tries to connect with an already running stream.
fprintf( stderr, "RESPONSE:\n%s\n", HTTP_S.BuildResponse( "459", "Aggregate Operation Not Allowed" ).c_str() );
conn.write( HTTP_S.BuildResponse( "459", "Aggregate Operation Not Allowed" ) );
} else {
do{
if (!ss.connected()){
/// \todo Put stream name-to-file mapping in a separate util file or even class
streamname = std::string(HTTP_R.url.c_str());
unsigned int slash_pos = streamname.rfind('/');
if (slash_pos != std::string::npos) streamname.erase(0, slash_pos);
for (std::string::iterator i=streamname.begin(); i != streamname.end(); ++i){
if (*i == '?'){
streamname.erase(i, streamname.end());
break;
}
if (!isalpha(*i) && !isdigit(*i) && *i != '_'){
streamname.erase(i);
--i;
}else{
*i = tolower(*i);
}
}
streamname = "/tmp/shared_socket_" + streamname;
ss = Socket::Connection(streamname);
if (!ss.connected()){
streamname = "";
HTTP_R.BuildResponse("404", "Not Found");
break; //skip the session below
}
}
setup_session = true;
}while(0);
}
if (setup_session) {
HTTP_S.SetHeader( "CSeq", HTTP_R.GetHeader( "CSeq" ).c_str() );
HTTP_S.SetHeader( "Session", time(NULL) );
/// \todo Add support for audio
// if( HTTP_R.url.find( "audio" ) != std::string::npos ) {
// HTTP_S.SetHeader( "Transport", HTTP_R.GetHeader( "Transport" ) + ";server_port=50002-50003" );
// } else {
//Stub data for testing purposes. This should now be extracted somehow from DTSC::DTMI
VideoParams.SetOwnTimestampUnit( ( 1.0 / 29.917 ) * 90000.0 );
VideoParams.SetMaximumPacketSize( 10000 );
//create a JRTPlib session
int VideoStatus;
uint16_t pbase;
//after 20 retries, just give up, most ports are likely in use
int retries = 20;
do {
//pick the right port here in the range 5000 to 5000 + 2 * 500 = 6000
pbase = 5000 + 2 * (rand() % 500);
VideoTransParams.SetPortbase( pbase );
VideoStatus = VideoSession.Create( VideoParams, &VideoTransParams, jrtplib::RTPTransmitter::IPv6UDPProto );
} while(VideoStatus < 0 && --retries > 0);
if( VideoStatus < 0 ) {
std::cerr << "Video session could not be created: " << jrtplib::RTPGetErrorString( VideoStatus ) << std::endl;
exit( -1 );
} else {
std::cerr << "Created video session using ports " << pbase << " and " << (pbase+1) << "\n";
}
//send video data
std::stringstream transport;
transport << HTTP_R.GetHeader( "Transport" ) << ";server_port=" << pbase << "-" << (pbase+1);
HTTP_S.SetHeader( "Transport", transport.str() );
/// \todo Connect with clients other than localhost
uint8_t localip[32];
int status = inet_pton( AF_INET6, conn.getHost().c_str(), localip ) ;
//Debug info
std::cerr << "Status: " << status << "\n";
jrtplib::RTPIPv6Address addr(localip,RTPClientPort);
//add the destination address to the VideoSession
VideoStatus = VideoSession.AddDestination(addr);
if (VideoStatus < 0) {
std::cerr << "Destination could not be set: " << jrtplib::RTPGetErrorString(VideoStatus) << std::endl;
exit(-1);
} else {
std::cerr << "Destination Set\n";
}
//Stub data for testing purposes.
//Payload type should confirm with the SDP File. 98 == H264 / AVC
VideoSession.SetDefaultPayloadType(98);
VideoSession.SetDefaultMark(false);
//We have no idea if this timestamp has to correspond with the OwnTimeStampUnit() above.
VideoSession.SetDefaultTimestampIncrement( ( 1.0 / 29.917 ) * 90000 );
// }
HTTP_S.SetBody( "\r\n\r\n" );
fprintf( stderr, "RESPONSE:\n%s\n", HTTP_S.BuildResponse( "200", "OK" ).c_str() );
conn.write( HTTP_S.BuildResponse( "200", "OK" ) );
}
} else if( HTTP_R.method == "PLAY" ) {
if( HTTP_R.GetHeader( "Range" ).substr(0,4) != "npt=" ) {
//We do not support this, whatever it is. Not needed for minimal compliance.
fprintf( stderr, "RESPONSE:\n%s\n", HTTP_S.BuildResponse( "501", "Not Implemented" ).c_str() );
conn.write( HTTP_S.BuildResponse( "501", "Not Implemented" ) );
} else {
//Initializes for actual streaming over the SETUP connection.
HTTP_S.SetHeader( "CSeq", HTTP_R.GetHeader( "CSeq" ).c_str() );
HTTP_S.SetHeader( "Session", HTTP_R.GetHeader( "Session" ) );
HTTP_S.SetHeader( "Range", HTTP_R.GetHeader( "Range" ) );
HTTP_S.SetHeader( "RTP-Info", "url=" + HTTP_R.url + ";seq=0;rtptime=0" );
HTTP_S.SetBody( "\r\n\r\n" );
fprintf( stderr, "RESPONSE:\n%s\n", HTTP_S.BuildResponse( "200", "OK" ).c_str() );
conn.write( HTTP_S.BuildResponse( "200", "OK" ) );
//Used further down, to start streaming video.
//PlayAudio = true;
PlayVideo = true;
}
} else if( HTTP_R.method == "TEARDOWN" ) {
//If we were sending any stream data at this point, stop it, but keep the setup.
HTTP_S.SetHeader( "CSeq", HTTP_R.GetHeader( "CSeq" ).c_str() );
HTTP_S.SetBody( "\r\n\r\n" );
fprintf( stderr, "RESPONSE:\n%s\n", HTTP_S.BuildResponse( "200", "OK" ).c_str() );
conn.write( HTTP_S.BuildResponse( "200", "OK" ) );
//PlayAudio = false;
PlayVideo = false;
} else {
//We do not implement other commands ( yet )
fprintf( stderr, "RESPONSE:\n%s\n", HTTP_S.BuildResponse( "501", "Not Implemented" ).c_str() );
conn.write( HTTP_S.BuildResponse( "501", "Not Implemented" ) );
}
HTTP_R.Clean();
HTTP_S.Clean();
if( PerRequest ) {
conn.close();
}
}
if( PlayVideo ) {
bool no_data_ignore = false;
std::string VideoBuf;
ss.canRead();
switch (ss.ready()) {
case -1:
std::cerr << "Buffer socket is disconnected\n";
break;
case 0://not ready
no_data_ignore = true;
break;
default:
///\todo Make it work!
DTSC::Stream ds;
ss.spool();
if (ds.parsePacket(ss.Received())){
VideoBuf = ds.lastData();
}else{
std::cerr << "Failed to parse packet" << std::endl;
no_data_ignore = true;//perhaps corrupt?
}
break;
}
if(no_data_ignore){}else if( VideoBuf == "" ) {
//videobuffer is empty, no more data.
jrtplib::RTPTime delay = jrtplib::RTPTime(10.0);
VideoSession.BYEDestroy(delay,"Out of data",11);
conn.close();
std::cerr << "Buffer empty - closing connection" << std::endl;
} else {
//Send a single NALU (H264 block) here.
VideoSession.SendPacket( VideoBuf.c_str(), VideoBuf.size(), 98, false, ( 1.0 / 29.917 ) * 90000 );
//we can add delays here as follows:
//don't know if these are nescecary or not, but good for testing nonetheless
// jrtplib::RTPTime delay( ( 1.0 / 29.917 ) * 90000 );
// jrtplib::RTPTime::Wait( delay );
}
}
}
return 0;
}
//Set Default Port
#define DEFAULT_PORT 554
//Set the function that should be forked for each client
#define MAINHANDLER RTSP_Handler
//Set the section in the Config file, though we will not use this yet
#define CONFIGSECT RTSP
//Include the main functionality, as well as fork support and everything.
#include "../util/server_setup.cpp"