live555学习笔记－RTSP服务运作

RTSP服务运作

基础基本搞明白了，那么RTSP,RTP等这些协议又是如何利用这些基础机制运作的呢？
首先来看RTSP.

RTSP首先需建立TCP侦听socket。可见于此函数：

DynamicRTSPServer*DynamicRTSPServer::createNew(UsageEnvironment&env,PortourPort,
UserAuthenticationDatabase*authDatabase,
unsignedreclamationTestSeconds){
intourSocket=setUpOurSocket(env,ourPort);//建立TCPsocket
if(ourSocket==-1)
returnNULL;


returnnewDynamicRTSPServer(env,ourSocket,ourPort,authDatabase,
reclamationTestSeconds);
}

要帧听客户端的连接，就需要利用任务调度机制了，所以需添加一个socket handler。可见于此函数：

RTSPServer::RTSPServer(UsageEnvironment&env,
intourSocket,
PortourPort,
UserAuthenticationDatabase*authDatabase,
unsignedreclamationTestSeconds):
Medium(env),
fRTSPServerSocket(ourSocket),
fRTSPServerPort(ourPort),
fHTTPServerSocket(-1),
fHTTPServerPort(0),
fClientSessionsForHTTPTunneling(NULL),
fAuthDB(authDatabase),
fReclamationTestSeconds(reclamationTestSeconds),
fServerMediaSessions(HashTable::create(STRING_HASH_KEYS))
{
#ifdefUSE_SIGNALS
//IgnoretheSIGPIPEsignal,sothatclientsonthesamehostthatarekilled
//don'talsokillus:
signal(SIGPIPE,SIG_IGN);
#endif


//Arrangetohandleconnectionsfromothers:
env.taskScheduler().turnOnBackgroundReadHandling(
fRTSPServerSocket,
(TaskScheduler::BackgroundHandlerProc*)&incomingConnectionHandlerRTSP,
this);
}

当收到客户的连接时需保存下代表客户端的新socket，以后用这个socket与这个客户通讯。每个客户将来会对应一个rtp会话，而且各客户的RTSP请求只控制自己的rtp会话，那么最好建立一个会话类，代表各客户的rtsp会话。于是类RTSPServer::RTSPClientSession产生，它保存的代表客户的socket。下为RTSPClientSession的创建过程

 

voidRTSPServer::incomingConnectionHandler(intserverSocket)
{
structsockaddr_inclientAddr;
SOCKLEN_TclientAddrLen=sizeofclientAddr;

//接受连接
intclientSocket=accept(serverSocket,
(structsockaddr*)&clientAddr,
&clientAddrLen);

if(clientSocket<0){
interr=envir().getErrno();
if(err!=EWOULDBLOCK){
envir().setResultErrMsg("accept()failed:");
}
return;
}

//设置socket的参数
makeSocketNonBlocking(clientSocket);
increaseSendBufferTo(envir(),clientSocket,50*1024);

#ifdefDEBUG
envir()<<"accept()edconnectionfrom"<<our_inet_ntoa(clientAddr.sin_addr)<<"\n";
#endif

//产生一个sessonid

//CreateanewobjectforthisRTSPsession.
//(Choosearandom32-bitintegerforthesessionid(itwillbeencodedasa8-digithexnumber).Wedon'tbothercheckingfor
//acollision;theprobabilityoftwoconcurrentsessionsgettingthesamesessionidisverylow.)
//(Wedo,however,avoidchoosingsessionid0,becausethathasaspecialuse(by"OnDemandServerMediaSubsession").)
unsignedsessionId;
do{
sessionId=(unsigned)our_random();
}while(sessionId==0);

//创建RTSPClientSession，注意传入的参数
(void)createNewClientSession(sessionId,clientSocket,clientAddr);
}

 

RTSPClientSession要提供什么功能呢？可以想象：需要监听客户端的rtsp请求并回应它，需要在DESCRIBE请求中返回所请求的流的信息，需要在SETUP请求中建立起RTP会话，需要在TEARDOWN请求中关闭RTP会话，等等...

RTSPClientSession要侦听客户端的请求，就需把自己的socket handler加入计划任务。证据如下：

RTSPServer::RTSPClientSession::RTSPClientSession(
RTSPServer&ourServer,
unsignedsessionId,
intclientSocket,
structsockaddr_inclientAddr):
fOurServer(ourServer),
fOurSessionId(sessionId),
fOurServerMediaSession(NULL),
fClientInputSocket(clientSocket),
fClientOutputSocket(clientSocket),
fClientAddr(clientAddr),
fSessionCookie(NULL),
fLivenessCheckTask(NULL),
fIsMulticast(False),
fSessionIsActive(True),
fStreamAfterSETUP(False),
fTCPStreamIdCount(0),
fNumStreamStates(0),
fStreamStates(NULL),
fRecursionCount(0)
{
//Arrangetohandleincomingrequests:
resetRequestBuffer();
envir().taskScheduler().turnOnBackgroundReadHandling(fClientInputSocket,
(TaskScheduler::BackgroundHandlerProc*)&incomingRequestHandler,
this);
noteLiveness();
}

下面重点讲一下下RTSPClientSession响应DESCRIBE请求的过程：

voidRTSPServer::RTSPClientSession::handleCmd_DESCRIBE(
charconst*cseq,
charconst*urlPreSuffix,
charconst*urlSuffix,
charconst*fullRequestStr)
{
char*sdpDescription=NULL;
char*rtspURL=NULL;
do{
//整理一下下RTSP地址
charurlTotalSuffix[RTSP_PARAM_STRING_MAX];
if(strlen(urlPreSuffix)+strlen(urlSuffix)+2
>sizeofurlTotalSuffix){
handleCmd_bad(cseq);
break;
}
urlTotalSuffix[0]='\0';
if(urlPreSuffix[0]!='\0'){
strcat(urlTotalSuffix,urlPreSuffix);
strcat(urlTotalSuffix,"/");
}
strcat(urlTotalSuffix,urlSuffix);


//验证帐户和密码
if(!authenticationOK("DESCRIBE",cseq,urlTotalSuffix,fullRequestStr))
break;


//Weshouldreallycheckthattherequestcontainsan"Accept:"#####
//for"application/sdp",becausethat'swhatwe'resendingback#####


//Beginbylookingupthe"ServerMediaSession"objectforthespecified"urlTotalSuffix":
//跟据流的名字查找ServerMediaSession，如果找不到，会创建一个。每个ServerMediaSession中至少要包含一个
//ServerMediaSubsession。一个ServerMediaSession对应一个媒体，可以认为是Server上的一个文件，或一个实时获取设备。其包含的每个ServerMediaSubSession代表媒体中的一个Track。所以一个ServerMediaSession对应一个媒体，如果客户请求的媒体名相同，就使用已存在的ServerMediaSession，如果不同，就创建一个新的。一个流对应一个StreamState，StreamState与ServerMediaSubsession相关，但代表的是动态的，而ServerMediaSubsession代表静态的。
ServerMediaSession*session=fOurServer.lookupServerMediaSession(urlTotalSuffix);
if(session==NULL){
handleCmd_notFound(cseq);
break;
}


//Then,assembleaSDPdescriptionforthissession:
//获取SDP字符串，在函数内会依次获取每个ServerMediaSubSession的字符串然连接起来。
sdpDescription=session->generateSDPDescription();
if(sdpDescription==NULL){
//Thisusuallymeansthatafilenamethatwasspecifiedfora
//"ServerMediaSubsession"doesnotexist.
snprintf((char*)fResponseBuffer,sizeoffResponseBuffer,
"RTSP/1.0404FileNotFound,OrInIncorrectFormat\r\n"
"CSeq:%s\r\n"
"%s\r\n",cseq,dateHeader());
break;
}
unsignedsdpDescriptionSize=strlen(sdpDescription);


//Also,generateourRTSPURL,forthe"Content-Base:"header
//(whichisnecessarytoensurethatthecorrectURLgetsusedin
//subsequent"SETUP"requests).
rtspURL=fOurServer.rtspURL(session,fClientInputSocket);


//形成响应DESCRIBE请求的RTSP字符串。
snprintf((char*)fResponseBuffer,sizeoffResponseBuffer,
"RTSP/1.0200OK\r\nCSeq:%s\r\n"
"%s"
"Content-Base:%s/\r\n"
"Content-Type:application/sdp\r\n"
"Content-Length:%d\r\n\r\n"
"%s",cseq,dateHeader(),rtspURL,sdpDescriptionSize,
sdpDescription);
}while(0);


delete[]sdpDescription;
delete[]rtspURL;


//返回后会被立即发送（没有把socketwrite操作放入计划任务中）。
}

fOurServer.lookupServerMediaSession(urlTotalSuffix)中会在找不到同名ServerMediaSession时新建一个，代表一个RTP流的ServerMediaSession们是被RTSPServer管理的，而不是被RTSPClientSession拥有。为什么呢？因为ServerMediaSession代表的是一个静态的流，也就是可以从它里面获取一个流的各种信息，但不能获取传输状态。不同客户可能连接到同一个流，所以ServerMediaSession应被RTSPServer所拥有。创建一个ServerMediaSession过程值得一观：

staticServerMediaSession*createNewSMS(UsageEnvironment&env,charconst*fileName,FILE*/*fid*/)
{
//Usethefilenameextensiontodeterminethetypeof"ServerMediaSession":
charconst*extension=strrchr(fileName,'.');
if(extension==NULL)
returnNULL;


ServerMediaSession*sms=NULL;
BooleanconstreuseSource=False;
if(strcmp(extension,".aac")==0){
//AssumedtobeanAACAudio(ADTSformat)file:
NEW_SMS("AACAudio");
sms->addSubsession(
ADTSAudioFileServerMediaSubsession::createNew(env,fileName,
reuseSource));
}elseif(strcmp(extension,".amr")==0){
//AssumedtobeanAMRAudiofile:
NEW_SMS("AMRAudio");
sms->addSubsession(
AMRAudioFileServerMediaSubsession::createNew(env,fileName,
reuseSource));
}elseif(strcmp(extension,".ac3")==0){
//AssumedtobeanAC-3Audiofile:
NEW_SMS("AC-3Audio");
sms->addSubsession(
AC3AudioFileServerMediaSubsession::createNew(env,fileName,
reuseSource));
}elseif(strcmp(extension,".m4e")==0){
//AssumedtobeaMPEG-4VideoElementaryStreamfile:
NEW_SMS("MPEG-4Video");
sms->addSubsession(
MPEG4VideoFileServerMediaSubsession::createNew(env,fileName,
reuseSource));
}elseif(strcmp(extension,".264")==0){
//AssumedtobeaH.264VideoElementaryStreamfile:
NEW_SMS("H.264Video");
OutPacketBuffer::maxSize=100000;//allowforsomepossiblylargeH.264frames
sms->addSubsession(
H264VideoFileServerMediaSubsession::createNew(env,fileName,
reuseSource));
}elseif(strcmp(extension,".mp3")==0){
//AssumedtobeaMPEG-1or2Audiofile:
NEW_SMS("MPEG-1or2Audio");
//Tostreamusing'ADUs'ratherthanrawMP3frames,uncommentthefollowing:
//#defineSTREAM_USING_ADUS1
//ToalsoreorderADUsbeforestreaming,uncommentthefollowing:
//#defineINTERLEAVE_ADUS1
//(FormoreinformationaboutADUsandinterleaving,
//see<http://www.live555.com/rtp-mp3/>)
BooleanuseADUs=False;
Interleaving*interleaving=NULL;
#ifdefSTREAM_USING_ADUS
useADUs=True;
#ifdefINTERLEAVE_ADUS
unsignedcharinterleaveCycle[]={0,2,1,3};//orchooseyourown...
unsignedconstinterleaveCycleSize
=(sizeofinterleaveCycle)/(sizeof(unsignedchar));
interleaving=newInterleaving(interleaveCycleSize,interleaveCycle);
#endif
#endif
sms->addSubsession(
MP3AudioFileServerMediaSubsession::createNew(env,fileName,
reuseSource,useADUs,interleaving));
}elseif(strcmp(extension,".mpg")==0){
//AssumedtobeaMPEG-1or2ProgramStream(audio+video)file:
NEW_SMS("MPEG-1or2ProgramStream");
MPEG1or2FileServerDemux*demux=MPEG1or2FileServerDemux::createNew(env,
fileName,reuseSource);
sms->addSubsession(demux->newVideoServerMediaSubsession());
sms->addSubsession(demux->newAudioServerMediaSubsession());
}elseif(strcmp(extension,".ts")==0){
//AssumedtobeaMPEGTransportStreamfile:
//Useanindexfilenamethat'sthesameastheTSfilename,exceptwith".tsx":
unsignedindexFileNameLen=strlen(fileName)+2;//allowfortrailing"x\0"
char*indexFileName=newchar[indexFileNameLen];
sprintf(indexFileName,"%sx",fileName);
NEW_SMS("MPEGTransportStream");
sms->addSubsession(
MPEG2TransportFileServerMediaSubsession::createNew(env,
fileName,indexFileName,reuseSource));
delete[]indexFileName;
}elseif(strcmp(extension,".wav")==0){
//AssumedtobeaWAVAudiofile:
NEW_SMS("WAVAudioStream");
//Toconvert16-bitPCMdatato8-bitu-law,priortostreaming,
//changethefollowingtoTrue:
BooleanconvertToULaw=False;
sms->addSubsession(
WAVAudioFileServerMediaSubsession::createNew(env,fileName,
reuseSource,convertToULaw));
}elseif(strcmp(extension,".dv")==0){
//AssumedtobeaDVVideofile
//First,makesurethattheRTPSinks'bufferswillbelargeenoughtohandlethehugesizeofDVframes(asbigas288000).
OutPacketBuffer::maxSize=300000;


NEW_SMS("DVVideo");
sms->addSubsession(
DVVideoFileServerMediaSubsession::createNew(env,fileName,
reuseSource));
}elseif(strcmp(extension,".mkv")==0){
//AssumedtobeaMatroskafile
NEW_SMS("Matroskavideo+audio+(optional)subtitles");


//CreateaMatroskafileserverdemultiplexorforthespecifiedfile.(Weentertheeventlooptowaitforthistocomplete.)
newMatroskaDemuxWatchVariable=0;
MatroskaFileServerDemux::createNew(env,fileName,
onMatroskaDemuxCreation,NULL);
env.taskScheduler().doEventLoop(&newMatroskaDemuxWatchVariable);


ServerMediaSubsession*smss;
while((smss=demux->newServerMediaSubsession())!=NULL){
sms->addSubsession(smss);
}
}


returnsms;
}

 

可以看到NEW_SMS("AMR Audio")会创建新的ServerMediaSession，之后马上调用sms->addSubsession（）为这个ServerMediaSession添加一个 ServerMediaSubSession 。看起来ServerMediaSession应该可以添加多个ServerMediaSubSession，但这里并没有这样做。如果可以添加多个 ServerMediaSubsession 那么ServerMediaSession与流名字所指定与文件是没有关系的，也就是说它不会操作文件，而文件的操作是放在 ServerMediaSubsession中的。具体应改是在ServerMediaSubsession的sdpLines()函数中打开。

 

原文地址：http://blog.csdn.net/niu_gao/article/details/6911130

live555源代码（VC6）：http://download.csdn.net/detail/leixiaohua1020/6374387