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Missing authorization, otherwise working fairly good

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This commit is contained in:
Erik Zandvliet 2011-09-13 13:35:21 +02:00
parent dd2222d748
commit 1c141f9ab8
2 changed files with 13 additions and 11 deletions
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22
Connector_RTSP/main.cpp
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@ -46,7 +46,7 @@ int RTSP_Handler( Socket::Connection conn ) {
bool PlayAudio = true; bool PlayAudio = true;
jrtplib::RTPSession VideoSession; jrtplib::RTPSession VideoSession;
jrtplib::RTPSessionParams VideoParams; jrtplib::RTPSessionParams VideoParams;
jrtplib::RTPUDPv4TransmissionParams VideoTransParams; jrtplib::RTPUDPv6TransmissionParams VideoTransParams;
std::string PreviousRequest = ""; std::string PreviousRequest = "";
Socket::Connection ss(-1); Socket::Connection ss(-1);
HTTP::Parser HTTP_R, HTTP_S; HTTP::Parser HTTP_R, HTTP_S;
@ -54,9 +54,6 @@ int RTSP_Handler( Socket::Connection conn ) {
while(conn.connected() && !FLV::Parse_Error) { while(conn.connected() && !FLV::Parse_Error) {
if( HTTP_R.Read(conn ) ) { if( HTTP_R.Read(conn ) ) {
fprintf( stderr, "REQUEST:\n%s\n", HTTP_R.BuildRequest().c_str() ); fprintf( stderr, "REQUEST:\n%s\n", HTTP_R.BuildRequest().c_str() );
// if( HTTP_R.GetHeader( "User-Agent" ).find( "RealMedia Player Version" ) != std::string::npos) {
// PerRequest = true;
// }
HTTP_S.protocol = "RTSP/1.0"; HTTP_S.protocol = "RTSP/1.0";
if( HTTP_R.method == "OPTIONS" ) { if( HTTP_R.method == "OPTIONS" ) {
HTTP_S.SetHeader( "CSeq", HTTP_R.GetHeader( "CSeq" ).c_str() ); HTTP_S.SetHeader( "CSeq", HTTP_R.GetHeader( "CSeq" ).c_str() );
@ -72,8 +69,8 @@ int RTSP_Handler( Socket::Connection conn ) {
HTTP_S.SetHeader( "CSeq", HTTP_R.GetHeader( "CSeq" ).c_str() ); HTTP_S.SetHeader( "CSeq", HTTP_R.GetHeader( "CSeq" ).c_str() );
HTTP_S.SetHeader( "Content-Type", "application/sdp" ); HTTP_S.SetHeader( "Content-Type", "application/sdp" );
/// \todo Retrieve presence of video and audio data, and process into response /// \todo Retrieve presence of video and audio data, and process into response
/// \todo Retrieve Packetization mode ( is 1 for now ). Where can I retrieve this? /// \todo Retrieve Packetization mode ( is 0 for now ). Where can I retrieve this?
HTTP_S.SetBody( "v=0\r\no=- 0 0 IN IP4 ddvtech.com\r\ns=Fifa Test\r\nc=IN IP4 127.0.0.1\r\nt=0 0\r\na=recvonly\r\nm=video 0 RTP/AVP 98\r\na=control:rtsp://localhost/fifa/video\r\na=rtpmap:98 H264/90000\r\na=fmtp:98 packetization-mode=1\r\nm=audio 0 RTP/AVP 96\r\na=control:rtsp://localhost/fifa/audio\r\na=rtpmap:96 mpeg4-generic/16000/2\r\n\r\n"); HTTP_S.SetBody( "v=0\r\no=- 0 0 IN IP4 ddvtech.com\r\ns=Fifa Test\r\nc=IN IP4 127.0.0.1\r\nt=0 0\r\na=recvonly\r\nm=video 0 RTP/AVP 98\r\na=control:rtsp://localhost/fifa/video\r\na=rtpmap:98 H264/90000\r\na=fmtp:98 packetization-mode=0\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");
fprintf( stderr, "RESPONSE:\n%s\n", HTTP_S.BuildResponse( "200", "OK" ).c_str() ); fprintf( stderr, "RESPONSE:\n%s\n", HTTP_S.BuildResponse( "200", "OK" ).c_str() );
conn.write( HTTP_S.BuildResponse( "200", "OK" ) ); conn.write( HTTP_S.BuildResponse( "200", "OK" ) );
} }
@ -97,7 +94,7 @@ int RTSP_Handler( Socket::Connection conn ) {
VideoParams.SetMaximumPacketSize( 10000 ); VideoParams.SetMaximumPacketSize( 10000 );
//pick the right port here //pick the right port here
VideoTransParams.SetPortbase( 50000 ); VideoTransParams.SetPortbase( 50000 );
int VideoStatus = VideoSession.Create( VideoParams, &VideoTransParams ); int VideoStatus = VideoSession.Create( VideoParams, &VideoTransParams, jrtplib::RTPTransmitter::IPv6UDPProto );
if( VideoStatus < 0 ) { if( VideoStatus < 0 ) {
std::cerr << jrtplib::RTPGetErrorString( VideoStatus ) << std::endl; std::cerr << jrtplib::RTPGetErrorString( VideoStatus ) << std::endl;
exit( -1 ); exit( -1 );
@ -105,8 +102,11 @@ int RTSP_Handler( Socket::Connection conn ) {
std::cerr << "Created video session\n"; std::cerr << "Created video session\n";
} }
/// \todo retrieve other client than localhost --> Socket::Connection has no support for this yet? /// \todo retrieve other client than localhost --> Socket::Connection has no support for this yet?
uint8_t localip[]={127,0,0,1};
jrtplib::RTPIPv4Address addr(localip,RTPClientPort); uint8_t localip[32];
int status = inet_pton( AF_INET6, conn.getHost().c_str(), localip ) ;
std::cerr << "Status: " << status << "\n";
jrtplib::RTPIPv6Address addr(localip,RTPClientPort);
VideoStatus = VideoSession.AddDestination(addr); VideoStatus = VideoSession.AddDestination(addr);
if (VideoStatus < 0) { if (VideoStatus < 0) {
@ -161,8 +161,8 @@ int RTSP_Handler( Socket::Connection conn ) {
VideoSession.BYEDestroy(delay,"Out of data",11); VideoSession.BYEDestroy(delay,"Out of data",11);
conn.close(); conn.close();
} else { } else {
VideoSession.SendPacket( VideoBuf.c_str(), VideoBuf.size() );//, 98, true, ( 1.0 / 29.917 ) ); VideoSession.SendPacket( VideoBuf.c_str(), VideoBuf.size(), 98, false, ( 1.0 / 29.917 ) * 90000 );
// jrtplib::RTPTime delay( 1.0 / 29.917 ); // jrtplib::RTPTime delay( ( 1.0 / 29.917 ) * 90000 );
// jrtplib::RTPTime::Wait( delay ); // jrtplib::RTPTime::Wait( delay );
} }
} }

2
Connector_RTSP/rtp.h
View file

@ -1,7 +1,9 @@
#include <jrtplib3/rtpsession.h> #include <jrtplib3/rtpsession.h>
#include <jrtplib3/rtppacket.h> #include <jrtplib3/rtppacket.h>
#include <jrtplib3/rtpudpv4transmitter.h> #include <jrtplib3/rtpudpv4transmitter.h>
#include <jrtplib3/rtpudpv6transmitter.h>
#include <jrtplib3/rtpipv4address.h> #include <jrtplib3/rtpipv4address.h>
#include <jrtplib3/rtpipv6address.h>
#include <jrtplib3/rtpsessionparams.h> #include <jrtplib3/rtpsessionparams.h>
#include <jrtplib3/rtperrors.h> #include <jrtplib3/rtperrors.h>