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RTSP: Choose random port, extra debugging messages

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This commit is contained in:
Lekensteyn 2012-03-14 18:23:27 +01:00
parent b068d53e2a
commit ad31299def
1 changed files with 18 additions and 9 deletions
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27
Connector_RTSP/main.cpp
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@ -12,6 +12,7 @@
#include <unistd.h> #include <unistd.h>
#include <getopt.h> #include <getopt.h>
#include <iostream> #include <iostream>
#include <sstream>
#include <sys/time.h> #include <sys/time.h>
#include <sys/wait.h> #include <sys/wait.h>
#include <sys/types.h> #include <sys/types.h>
@ -116,26 +117,34 @@ int RTSP_Handler( Socket::Connection conn ) {
} else { } else {
HTTP_S.SetHeader( "CSeq", HTTP_R.GetHeader( "CSeq" ).c_str() ); HTTP_S.SetHeader( "CSeq", HTTP_R.GetHeader( "CSeq" ).c_str() );
HTTP_S.SetHeader( "Session", time(NULL) ); HTTP_S.SetHeader( "Session", time(NULL) );
/// \todo "Random" generation of server_ports
/// \todo Add support for audio /// \todo Add support for audio
// if( HTTP_R.url.find( "audio" ) != std::string::npos ) { // if( HTTP_R.url.find( "audio" ) != std::string::npos ) {
// HTTP_S.SetHeader( "Transport", HTTP_R.GetHeader( "Transport" ) + ";server_port=50002-50003" ); // HTTP_S.SetHeader( "Transport", HTTP_R.GetHeader( "Transport" ) + ";server_port=50002-50003" );
// } else { // } else {
//send video data
HTTP_S.SetHeader( "Transport", HTTP_R.GetHeader( "Transport" ) + ";server_port=50000-50001" );
//Stub data for testing purposes. This should now be extracted somehow from DTSC::DTMI //Stub data for testing purposes. This should now be extracted somehow from DTSC::DTMI
VideoParams.SetOwnTimestampUnit( ( 1.0 / 29.917 ) * 90000.0 ); VideoParams.SetOwnTimestampUnit( ( 1.0 / 29.917 ) * 90000.0 );
VideoParams.SetMaximumPacketSize( 10000 ); VideoParams.SetMaximumPacketSize( 10000 );
//pick the right port here
VideoTransParams.SetPortbase( 50000 );
//create a JRTPlib session //create a JRTPlib session
int VideoStatus = VideoSession.Create( VideoParams, &VideoTransParams, jrtplib::RTPTransmitter::IPv6UDPProto ); 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 ) { if( VideoStatus < 0 ) {
std::cerr << jrtplib::RTPGetErrorString( VideoStatus ) << std::endl; std::cerr << "Video session could not be created: " << jrtplib::RTPGetErrorString( VideoStatus ) << std::endl;
exit( -1 ); exit( -1 );
} else { } else {
std::cerr << "Created video session\n"; 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 /// \todo Connect with clients other than localhost
uint8_t localip[32]; uint8_t localip[32];
@ -147,7 +156,7 @@ int RTSP_Handler( Socket::Connection conn ) {
//add the destination address to the VideoSession //add the destination address to the VideoSession
VideoStatus = VideoSession.AddDestination(addr); VideoStatus = VideoSession.AddDestination(addr);
if (VideoStatus < 0) { if (VideoStatus < 0) {
std::cerr << jrtplib::RTPGetErrorString(VideoStatus) << std::endl; std::cerr << "Destination could not be set: " << jrtplib::RTPGetErrorString(VideoStatus) << std::endl;
exit(-1); exit(-1);
} else { } else {
std::cerr << "Destination Set\n"; std::cerr << "Destination Set\n";