Live555源代码解读（5-3）
void RTSPServer::RTSPClientSession::handleCmd_SETUP(
char const* cseq,
char const* urlPreSuffix,
char const* urlSuffix,
char const* fullRequestStr)
{
// Normally, "urlPreSuffix" should be the session (stream) name,
// and "urlSuffix" should be the subsession (track) name.
// However (being "liberal in what we accept"), we also handle
// 'aggregate' SETUP requests (i.e., without a track name),
// in the special case where we have only a single track. I.e.,
// in this case, we also handle:
// "urlPreSuffix" is empty and "urlSuffix" is the session (stream) name, or
// "urlPreSuffix" concatenated with "urlSuffix" (with "/" inbetween)
// is the session (stream) name.
char const* streamName = urlPreSuffix; // in the normal case
char const* trackId = urlSuffix; // in the normal case
char* concatenatedStreamName = NULL; // in the normal case
do {
// First, make sure the specified stream name exists:
fOurServerMediaSession = fOurServer.lookupServerMediaSession(streamName);
if (fOurServerMediaSession == NULL) {
// Check for the special case (noted above), before we up:
if (urlPreSuffix[0] == '\0') {
streamName = urlSuffix;
} else {
concatenatedStreamName = new char[strlen(urlPreSuffix)
+ strlen(urlSuffix) + 2]; // allow for the "/" and the trailing '\0'
sprintf(concatenatedStreamName, "%s/%s", urlPreSuffix,
urlSuffix);
streamName = concatenatedStreamName;
}
trackId = NULL;
// Check again:
fOurServerMediaSession = fOurServer.lookupServerMediaSession(streamName);
}
if (fOurServerMediaSession == NULL) {
handleCmd_notFound(cseq);
break;
}
fOurServerMediaSession->incrementReferenceCount();
//为一个流中所有的track都分配一个stream state
if (fStreamStates == NULL) {
// This is the first "SETUP" for this session. Set up our
// array of states for all of this session's subsessions (tracks):
ServerMediaSubsessionIterator iter(*fOurServerMediaSession);
for (fNumStreamStates = 0; iter.next() != NULL;++fNumStreamStates) {
} // begin by counting the number of subsessions (tracks)
fStreamStates = new struct streamState[fNumStreamStates];
iter.reset();
ServerMediaSubsession* subsession;
for (unsigned i = 0; i < fNumStreamStates; ++i) {
subsession = iter.next();
fStreamStates[i].subsession = subsession;
fStreamStates[i].streamToken = NULL; // for now; it may be changed by the "getStreamParameters()" call that comes later
}
}
//查找当前请求的track的信息
// Look up information for the specified subsession (track):
ServerMediaSubsession* subsession = NULL;
unsigned streamNum;
if (trackId != NULL && trackId[0] != '\0') { // normal case
for (streamNum = 0; streamNum < fNumStreamStates; ++streamNum) {
subsession = fStreamStates[streamNum].subsession;
if (subsession != NULL&& strcmp(trackId, subsession->trackId()) == 0)
break; //找到啦！
}
if (streamNum >= fNumStreamStates) {
// The specified track id doesn't exist, so this request fails:
handleCmd_notFound(cseq);
break;
}
} else {
// Weird case: there was no track id in the URL.
// This works only if we have only one subsession:
if (fNumStreamStates != 1) {
handleCmd_bad(cseq);
break;
}
streamNum = 0;
subsession = fStreamStates[streamNum].subsession;
}
// ASSERT: subsession != NULL
//分析RTSP请求字符串中的传输要求
// Look for a "Transport:" header in the request string, to extract client parameters:
StreamingMode streamingMode;
char* streamingModeString = NULL; // set when RAW_UDP streaming is specified
char* clientsDestinationAddressStr;
u_int8_t clientsDestinationTTL;
portNumBits clientRTPPortNum, clientRTCPPortNum;
unsigned char rtpChannelId, rtcpChannelId;
parseTransportHeader(fullRequestStr, streamingMode, streamingModeString,
clientsDestinationAddressStr, clientsDestinationTTL,
clientRTPPortNum, clientRTCPPortNum, rtpChannelId,
rtcpChannelId);
if (streamingMode == RTP_TCP && rtpChannelId == 0xFF
|| streamingMode != RTP_TCP&&
fClientOutputSocket != fClientInputSocket) {
// An anomolous situation, caused by a buggy client. Either:
// 1/ TCP streaming was requested, but with no "interleaving=" fields. (QuickTime Player sometimes does this.), or
// 2/ TCP streaming was not requested, but we're doing RTSP-over-HTTP tunneling (which implies TCP streaming).
// In either case, we assume TCP streaming, and set the RTP and RTCP channel ids to proper values:
streamingMode = RTP_TCP;
rtpChannelId = fTCPStreamIdCount;
rtcpChannelId = fTCPStreamIdCount + 1;
}
fTCPStreamIdCount += 2;
Port clientRTPPort(clientRTPPortNum);
Port clientRTCPPort(clientRTCPPortNum);
// Next, check whether a "Range:" header is present in the request.
// This isn't legal, but some clients do this to combine "SETUP" and "PLAY":
double rangeStart = 0.0, rangeEnd = 0.0;
fStreamAfterSETUP = parseRangeHeader(fullRequestStr, rangeStart,
rangeEnd) || parsePlayNowHeader(fullRequestStr);
// Then, get server parameters from the 'subsession':
int tcpSocketNum = streamingMode == RTP_TCP ? fClientOutputSocket : -1;
netAddressBits destinationAddress = 0;
u_int8_t destinationTTL = 255;
#ifdef RTSP_ALLOW_CLIENT_DESTINATION_SETTING
if (clientsDestinationAddressStr != NULL) {
// Use the client-provided "destination" address.
// Note: This potentially allows the server to be used in denial-of-service
// attacks, so don't enable this code unless you're sure that clients are
// trusted.
destinationAddress = our_inet_addr(clientsDestinationAddressStr);
}
// Also use the client-provided TTL.
destinationTTL = clientsDestinationTTL;
#endif
delete[] clientsDestinationAddressStr;
Port serverRTPPort(0);
Port serverRTCPPort(0);
// Make sure that we transmit on the same interface that's used by
// the client (in case we're a multi-homed server):
struct sockaddr_in sourceAddr;
SOCKLEN_T namelen = sizeof sourceAddr;
getsockname(fClientInputSocket, (struct sockaddr*) &sourceAddr,&namelen);
netAddressBits origSendingInterfaceAddr = SendingInterfaceAddr;
netAddressBits origReceivingInterfaceAddr = ReceivingInterfaceAddr;
// NOTE: The following might not work properly, so we ifdef it out for now:
#ifdef HACK_FOR_MULTIHOMED_SERVERS
ReceivingInterfaceAddr = SendingInterfaceAddr = sourceAddr.sin_addr.s_addr;
#endif
//获取rtp连接信息，在其中已建立起了server端的rtp和rtcp socket，返回
//fStreamStates[streamNum].streamToken表示数据流已经建立起来了
subsession->getStreamParameters(fOurSessionId,
fClientAddr.sin_addr.s_addr, clientRTPPort, clientRTCPPort,
tcpSocketNum, rtpChannelId, rtcpChannelId, destinationAddress,
destinationTTL, fIsMulticast, serverRTPPort, serverRTCPPort,
fStreamStates[streamNum].streamToken);
SendingInterfaceAddr = origSendingInterfaceAddr;
ReceivingInterfaceAddr = origReceivingInterfaceAddr;
//形成RTSP回应字符串
struct in_addr destinationAddr;
destinationAddr.s_addr = destinationAddress;
char* destAddrStr = strDup(our_inet_ntoa(destinationAddr));
char* sourceAddrStr = strDup(our_inet_ntoa(sourceAddr.sin_addr));
if (fIsMulticast) {
switch (streamingMode) {
case RTP_UDP:
snprintf(
(char*) fResponseBuffer,
sizeof fResponseBuffer,
"RTSP/1.0 200 OK\r\n"
"CSeq: %s\r\n"
"%s"
"Transport: RTP/AVP;multicast;destination=%s;source=%s;port=%d-%d;ttl=%d\r\n"
"Session: %08X\r\n\r\n", cseq, dateHeader(),
destAddrStr, sourceAddrStr, ntohs(serverRTPPort.num()),
ntohs(serverRTCPPort.num()), destinationTTL,
fOurSessionId);
break;
case RTP_TCP:
// multicast streams can't be sent via TCP
handleCmd_unsupportedTransport(cseq);
break;
case RAW_UDP:
snprintf(
(char*) fResponseBuffer,
sizeof fResponseBuffer,
"RTSP/1.0 200 OK\r\n"
"CSeq: %s\r\n"
"%s"
"Transport: %s;multicast;destination=%s;source=%s;port=%d;ttl=%d\r\n"
"Session: %08X\r\n\r\n", cseq, dateHeader(),
streamingModeString, destAddrStr, sourceAddrStr,
ntohs(serverRTPPort.num()), destinationTTL,
fOurSessionId);
break;
}
} else {
switch (streamingMode) {
case RTP_UDP: {
snprintf(
(char*) fResponseBuffer,
sizeof fResponseBuffer,
"RTSP/1.0 200 OK\r\n"
"CSeq: %s\r\n"
"%s"
"Transport: RTP/AVP;unicast;destination=%s;source=%s;client_port=%d-%d;server_port=%d-%d\r\n"
"Session: %08X\r\n\r\n", cseq, dateHeader(),
destAddrStr, sourceAddrStr, ntohs(clientRTPPort.num()),
ntohs(clientRTCPPort.num()), ntohs(serverRTPPort.num()),
ntohs(serverRTCPPort.num()), fOurSessionId);
break;
}
case RTP_TCP: {
snprintf(
(char*) fResponseBuffer,
sizeof fResponseBuffer,
"RTSP/1.0 200 OK\r\n"
"CSeq: %s\r\n"
"%s"
"Transport: RTP/AVP/TCP;unicast;destination=%s;source=%s;interleaved=%d-%d\r\n"
"Session: %08X\r\n\r\n", cseq, dateHeader(),
destAddrStr, sourceAddrStr, rtpChannelId, rtcpChannelId,
fOurSessionId);
break;
}
case RAW_UDP: {
snprintf(
(char*) fResponseBuffer,
sizeof fResponseBuffer,
"RTSP/1.0 200 OK\r\n"
"CSeq: %s\r\n"
"%s"
"Transport: %s;unicast;destination=%s;source=%s;client_port=%d;server_port=%d\r\n"
"Session: %08X\r\n\r\n", cseq, dateHeader(),
streamingModeString, destAddrStr, sourceAddrStr,
ntohs(clientRTPPort.num()), ntohs(serverRTPPort.num()),
fOurSessionId);
break;
}
}
}
delete[] destAddrStr;
delete[] sourceAddrStr;
delete[] streamingModeString;
} while (0);
delete[] concatenatedStreamName;
//返回后，回应字符串会被立即发送
}
有个问题，如果这个streamToken使用的是已存在的（还记得ReuseFirstSource吗），为它再次调用startStream()时，究竟会做什么呢？你猜啊！呵呵我猜吧：应该是把这个客户端的地址和rtp/rtcp端口传给rtp server的groupSocket，rtp server自然就开始向这个客户端发送数据了。是不是这样尼？看代码吧：
void RTSPServer::RTSPClientSession::handleCmd_SETUP(
char const* cseq,
char const* urlPreSuffix,
char const* urlSuffix,
char const* fullRequestStr)
{
// Normally, "urlPreSuffix" should be the session (stream) name,
// and "urlSuffix" should be the subsession (track) name.
// However (being "liberal in what we accept"), we also handle
// 'aggregate' SETUP requests (i.e., without a track name),
// in the special case where we have only a single track. I.e.,
// in this case, we also handle:
// "urlPreSuffix" is empty and "urlSuffix" is the session (stream) name, or
// "urlPreSuffix" concatenated with "urlSuffix" (with "/" inbetween)
// is the session (stream) name.
char const* streamName = urlPreSuffix; // in the normal case
char const* trackId = urlSuffix; // in the normal case
char* concatenatedStreamName = NULL; // in the normal case
do {
// First, make sure the specified stream name exists:
fOurServerMediaSession = fOurServer.lookupServerMediaSession(streamName);
if (fOurServerMediaSession == NULL) {
// Check for the special case (noted above), before we up:
if (urlPreSuffix[0] == '\0') {
streamName = urlSuffix;
} else {
concatenatedStreamName = new char[strlen(urlPreSuffix)
+ strlen(urlSuffix) + 2]; // allow for the "/" and the trailing '\0'
sprintf(concatenatedStreamName, "%s/%s", urlPreSuffix,
urlSuffix);
streamName = concatenatedStreamName;
}
trackId = NULL;
// Check again:
fOurServerMediaSession = fOurServer.lookupServerMediaSession(streamName);
}
if (fOurServerMediaSession == NULL) {
handleCmd_notFound(cseq);
break;
}
fOurServerMediaSession->incrementReferenceCount();
//为一个流中所有的track都分配一个stream state
if (fStreamStates == NULL) {
// This is the first "SETUP" for this session. Set up our
// array of states for all of this session's subsessions (tracks):
ServerMediaSubsessionIterator iter(*fOurServerMediaSession);
for (fNumStreamStates = 0; iter.next() != NULL;++fNumStreamStates) {
} // begin by counting the number of subsessions (tracks)
fStreamStates = new struct streamState[fNumStreamStates];
iter.reset();
ServerMediaSubsession* subsession;
for (unsigned i = 0; i < fNumStreamStates; ++i) {
subsession = iter.next();
fStreamStates[i].subsession = subsession;
fStreamStates[i].streamToken = NULL; // for now; it may be changed by the "getStreamParameters()" call that comes later
}
}
//查找当前请求的track的信息
// Look up information for the specified subsession (track):
ServerMediaSubsession* subsession = NULL;
unsigned streamNum;
if (trackId != NULL && trackId[0] != '\0') { // normal case
for (streamNum = 0; streamNum < fNumStreamStates; ++streamNum) {
subsession = fStreamStates[streamNum].subsession;
if (subsession != NULL&& strcmp(trackId, subsession->trackId()) == 0)
break; //找到啦！
}
if (streamNum >= fNumStreamStates) {
// The specified track id doesn't exist, so this request fails:
handleCmd_notFound(cseq);
break;
}
} else {
// Weird case: there was no track id in the URL.
// This works only if we have only one subsession:
if (fNumStreamStates != 1) {
handleCmd_bad(cseq);
break;
}
streamNum = 0;
subsession = fStreamStates[streamNum].subsession;
}
// ASSERT: subsession != NULL
//分析RTSP请求字符串中的传输要求
// Look for a "Transport:" header in the request string, to extract client parameters:
StreamingMode streamingMode;
char* streamingModeString = NULL; // set when RAW_UDP streaming is specified
char* clientsDestinationAddressStr;
u_int8_t clientsDestinationTTL;
portNumBits clientRTPPortNum, clientRTCPPortNum;
unsigned char rtpChannelId, rtcpChannelId;
parseTransportHeader(fullRequestStr, streamingMode, streamingModeString,
clientsDestinationAddressStr, clientsDestinationTTL,
clientRTPPortNum, clientRTCPPortNum, rtpChannelId,
rtcpChannelId);
if (streamingMode == RTP_TCP && rtpChannelId == 0xFF
|| streamingMode != RTP_TCP&&
fClientOutputSocket != fClientInputSocket) {
// An anomolous situation, caused by a buggy client. Either:
// 1/ TCP streaming was requested, but with no "interleaving=" fields. (QuickTime Player sometimes does this.), or
// 2/ TCP streaming was not requested, but we're doing RTSP-over-HTTP tunneling (which implies TCP streaming).
// In either case, we assume TCP streaming, and set the RTP and RTCP channel ids to proper values:
streamingMode = RTP_TCP;
rtpChannelId = fTCPStreamIdCount;
rtcpChannelId = fTCPStreamIdCount + 1;
}
fTCPStreamIdCount += 2;
Port clientRTPPort(clientRTPPortNum);
Port clientRTCPPort(clientRTCPPortNum);
// Next, check whether a "Range:" header is present in the request.
// This isn't legal, but some clients do this to combine "SETUP" and "PLAY":
double rangeStart = 0.0, rangeEnd = 0.0;
fStreamAfterSETUP = parseRangeHeader(fullRequestStr, rangeStart,
rangeEnd) || parsePlayNowHeader(fullRequestStr);
// Then, get server parameters from the 'subsession':
int tcpSocketNum = streamingMode == RTP_TCP ? fClientOutputSocket : -1;
netAddressBits destinationAddress = 0;
u_int8_t destinationTTL = 255;
#ifdef RTSP_ALLOW_CLIENT_DESTINATION_SETTING
if (clientsDestinationAddressStr != NULL) {
// Use the client-provided "destination" address.
// Note: This potentially allows the server to be used in denial-of-service
// attacks, so don't enable this code unless you're sure that clients are
// trusted.
destinationAddress = our_inet_addr(clientsDestinationAddressStr);
}
// Also use the client-provided TTL.
destinationTTL = clientsDestinationTTL;
#endif
delete[] clientsDestinationAddressStr;
Port serverRTPPort(0);
Port serverRTCPPort(0);
// Make sure that we transmit on the same interface that's used by
// the client (in case we're a multi-homed server):
struct sockaddr_in sourceAddr;
SOCKLEN_T namelen = sizeof sourceAddr;
getsockname(fClientInputSocket, (struct sockaddr*) &sourceAddr,&namelen);
netAddressBits origSendingInterfaceAddr = SendingInterfaceAddr;
netAddressBits origReceivingInterfaceAddr = ReceivingInterfaceAddr;
// NOTE: The following might not work properly, so we ifdef it out for now:
#ifdef HACK_FOR_MULTIHOMED_SERVERS
ReceivingInterfaceAddr = SendingInterfaceAddr = sourceAddr.sin_addr.s_addr;
#endif
//获取rtp连接信息，在其中已建立起了server端的rtp和rtcp socket，返回
//fStreamStates[streamNum].streamToken表示数据流已经建立起来了
subsession->getStreamParameters(fOurSessionId,
fClientAddr.sin_addr.s_addr, clientRTPPort, clientRTCPPort,
tcpSocketNum, rtpChannelId, rtcpChannelId, destinationAddress,
destinationTTL, fIsMulticast, serverRTPPort, serverRTCPPort,
fStreamStates[streamNum].streamToken);
SendingInterfaceAddr = origSendingInterfaceAddr;
ReceivingInterfaceAddr = origReceivingInterfaceAddr;
//形成RTSP回应字符串
struct in_addr destinationAddr;
destinationAddr.s_addr = destinationAddress;
char* destAddrStr = strDup(our_inet_ntoa(destinationAddr));
char* sourceAddrStr = strDup(our_inet_ntoa(sourceAddr.sin_addr));
if (fIsMulticast) {
switch (streamingMode) {
case RTP_UDP:
snprintf(
(char*) fResponseBuffer,
sizeof fResponseBuffer,
"RTSP/1.0 200 OK\r\n"
"CSeq: %s\r\n"
"%s"
"Transport: RTP/AVP;multicast;destination=%s;source=%s;port=%d-%d;ttl=%d\r\n"
"Session: %08X\r\n\r\n", cseq, dateHeader(),
destAddrStr, sourceAddrStr, ntohs(serverRTPPort.num()),
ntohs(serverRTCPPort.num()), destinationTTL,
fOurSessionId);
break;
case RTP_TCP:
// multicast streams can't be sent via TCP
handleCmd_unsupportedTransport(cseq);
break;
case RAW_UDP:
snprintf(
(char*) fResponseBuffer,
sizeof fResponseBuffer,
"RTSP/1.0 200 OK\r\n"
"CSeq: %s\r\n"
"%s"
"Transport: %s;multicast;destination=%s;source=%s;port=%d;ttl=%d\r\n"
"Session: %08X\r\n\r\n", cseq, dateHeader(),
streamingModeString, destAddrStr, sourceAddrStr,
ntohs(serverRTPPort.num()), destinationTTL,
fOurSessionId);
break;
}
} else {
switch (streamingMode) {
case RTP_UDP: {
snprintf(
(char*) fResponseBuffer,
sizeof fResponseBuffer,
"RTSP/1.0 200 OK\r\n"
"CSeq: %s\r\n"
"%s"
"Transport: RTP/AVP;unicast;destination=%s;source=%s;client_port=%d-%d;server_port=%d-%d\r\n"
"Session: %08X\r\n\r\n", cseq, dateHeader(),
destAddrStr, sourceAddrStr, ntohs(clientRTPPort.num()),
ntohs(clientRTCPPort.num()), ntohs(serverRTPPort.num()),
ntohs(serverRTCPPort.num()), fOurSessionId);
break;
}
case RTP_TCP: {
snprintf(
(char*) fResponseBuffer,
sizeof fResponseBuffer,
"RTSP/1.0 200 OK\r\n"
"CSeq: %s\r\n"
"%s"
"Transport: RTP/AVP/TCP;unicast;destination=%s;source=%s;interleaved=%d-%d\r\n"
"Session: %08X\r\n\r\n", cseq, dateHeader(),
destAddrStr, sourceAddrStr, rtpChannelId, rtcpChannelId,
fOurSessionId);
break;
}
case RAW_UDP: {
snprintf(
(char*) fResponseBuffer,
sizeof fResponseBuffer,
"RTSP/1.0 200 OK\r\n"
"CSeq: %s\r\n"
"%s"
"Transport: %s;unicast;destination=%s;source=%s;client_port=%d;server_port=%d\r\n"
"Session: %08X\r\n\r\n", cseq, dateHeader(),
streamingModeString, destAddrStr, sourceAddrStr,
ntohs(clientRTPPort.num()), ntohs(serverRTPPort.num()),
fOurSessionId);
break;
}
}
}
delete[] destAddrStr;
delete[] sourceAddrStr;
delete[] streamingModeString;
} while (0);
delete[] concatenatedStreamName;
//返回后，回应字符串会被立即发送
}