live555 android,Android之live555构建RTSP Server

=====================

myRTSPServer.cpp 1.建立任务调度 scheduler = BasicTaskScheduler::createNew(); env = BasicUsageEnvironment::createNew(*scheduler); 2.建立RTP/RTCP端口 unsigned short rtpPortNumAudio = getDestAudioPort(); unsigned short rtcpPortNumAudio = rtpPortNumAudio +1;   struct in_addr destinationAddress; destinationAddress.s_addr = our_inet_addr("192.168.10.100");  //不设置会导致组播，client端IP const Port rtpPortAudio(rtpPortNumAudio); const Port rtcpPortAudio(rtcpPortNumAudio); rtpGroupsockAudio = new Groupsock(*env, destinationAddress, rtpPortAudio, ttl);   rtcpGroupsockAudio = new Groupsock(*env, destinationAddress, rtcpPortAudio, ttl);  3.建立会话 CreateAudioSink(rtpGroupsockAudio);  audioRTCP = RTCPInstance::createNew(*env, rtcpGroupsockAudio, getBandwidthAudio(),  (const unsigned char*)getName(), audioSink, NULL /* we're a server */, isSSM);   4.创建RTSPServer rtspServer = RTSPServer::createNew(*env,8554); sms= ServerMediaSession::createNew(*env, "tanktest", "Audio Stream", "Session streamed by \"Tank\"",   isSSM); sms->addSubsession(PassiveServerMediaSubsession::createNew(*audioSink, audioRTCP)); rtspServer->addServerMediaSession(sms); char* url = rtspServer->rtspURL(sms); strcpy(remoteUrl,url); 注意： live555/liveMedia/RTSPServer.cpp void RTSPServer::RTSPClientSession ::handleCmd_SETUP(char const* cseq,  char const* urlPreSuffix, char const* urlSuffix,  char const* fullRequestStr); live555/liveMedia/PassiveServerMediaSubsession.cpp void PassiveServerMediaSubsession ::getStreamParameters(unsigned clientSessionId,      netAddressBits clientAddress,      Port const& /*clientRTPPort*/,      Port const& clientRTCPPort,      int /*tcpSocketNum*/,      unsigned char /*rtpChannelId*/,      unsigned char /*rtcpChannelId*/,      netAddressBits& destinationAddress,      u_int8_t& destinationTTL,      Boolean& isMulticast,      Port& serverRTPPort,      Port& serverRTCPPort,      void*& streamToken)；