416 lines
8.9 KiB
C

#include "Limelight-internal.h"
#include "Rtsp.h"
#define RTSP_MAX_RESP_SIZE 16384
static SOCKET sock = INVALID_SOCKET;
static IP_ADDRESS remoteAddr;
static int currentSeqNumber;
static char rtspTargetUrl[256];
static char sessionIdString[16];
static int hasSessionId;
static char responseBuffer[RTSP_MAX_RESP_SIZE];
/* GFE 2.1.1 */
#define RTSP_CLIENT_VERSION 10
#define RTSP_CLIENT_VERSION_S "10"
/* Create RTSP Option */
static POPTION_ITEM createOptionItem(char* option, char* content)
{
POPTION_ITEM item = malloc(sizeof(*item));
if (item == NULL) {
return NULL;
}
item->option = malloc(strlen(option) + 1);
if (item->option == NULL) {
free(item);
return NULL;
}
strcpy(item->option, option);
item->content = malloc(strlen(content) + 1);
if (item->content == NULL) {
free(item->option);
free(item);
return NULL;
}
strcpy(item->content, content);
item->next = NULL;
item->flags = FLAG_ALLOCATED_OPTION_FIELDS;
return item;
}
/* Add an option to the RTSP Message */
static int addOption(PRTSP_MESSAGE msg, char* option, char* content)
{
POPTION_ITEM item = createOptionItem(option, content);
if (item == NULL) {
return 0;
}
insertOption(&msg->options, item);
msg->flags |= FLAG_ALLOCATED_OPTION_ITEMS;
return 1;
}
/* Create an RTSP Request */
static int initializeRtspRequest(PRTSP_MESSAGE msg, char* command, char* target)
{
char sequenceNumberStr[16];
// FIXME: Hacked CSeq attribute due to RTSP parser bug
createRtspRequest(msg, NULL, 0, command, target, "RTSP/1.0",
0, NULL, NULL, 0);
sprintf(sequenceNumberStr, "%d", currentSeqNumber++);
if (!addOption(msg, "CSeq", sequenceNumberStr) ||
!addOption(msg, "X-GS-ClientVersion", RTSP_CLIENT_VERSION_S)) {
freeMessage(msg);
return 0;
}
return 1;
}
/* Returns 1 on success, 0 otherwise */
static int transactRtspMessage(PRTSP_MESSAGE request, PRTSP_MESSAGE response) {
SOCK_RET err;
int ret = 0;
int offset;
char* serializedMessage = NULL;
int messageLen;
sock = connectTcpSocket(remoteAddr, 48010);
if (sock == INVALID_SOCKET) {
return ret;
}
enableNoDelay(sock);
serializedMessage = serializeRtspMessage(request, &messageLen);
if (serializedMessage == NULL) {
closesocket(sock);
return ret;
}
// Send our message
err = send(sock, serializedMessage, messageLen, 0);
if (err == SOCKET_ERROR) {
goto Exit;
}
// Read the response until the server closes the connection
offset = 0;
for (;;) {
err = recv(sock, &responseBuffer[offset], RTSP_MAX_RESP_SIZE - offset, 0);
if (err <= 0) {
// Done reading
break;
}
offset += err;
// Warn if the RTSP message is too big
if (offset == RTSP_MAX_RESP_SIZE) {
Limelog("RTSP message too long\n");
goto Exit;
}
}
if (parseRtspMessage(response, responseBuffer, offset) == RTSP_ERROR_SUCCESS) {
// Successfully parsed response
ret = 1;
}
else {
Limelog("Failed to parse RTSP response\n");
}
Exit:
if (serializedMessage != NULL) {
free(serializedMessage);
}
closesocket(sock);
sock = INVALID_SOCKET;
return ret;
}
/* Terminate the RTSP Handshake process by closing the socket */
void terminateRtspHandshake(void) {
if (sock != INVALID_SOCKET) {
closesocket(sock);
sock = INVALID_SOCKET;
}
}
/* Send RTSP OPTIONS request */
static int requestOptions(PRTSP_MESSAGE response) {
RTSP_MESSAGE request;
int ret;
ret = initializeRtspRequest(&request, "OPTIONS", rtspTargetUrl);
if (ret != 0) {
ret = transactRtspMessage(&request, response);
freeMessage(&request);
}
return ret;
}
/* Send RTSP DESCRIBE request */
static int requestDescribe(PRTSP_MESSAGE response) {
RTSP_MESSAGE request;
int ret;
ret = initializeRtspRequest(&request, "DESCRIBE", rtspTargetUrl);
if (ret != 0) {
if (addOption(&request, "Accept",
"application/sdp") &&
addOption(&request, "If-Modified-Since",
"Thu, 01 Jan 1970 00:00:00 GMT")) {
ret = transactRtspMessage(&request, response);
}
else {
ret = 0;
}
freeMessage(&request);
}
return ret;
}
/* Send RTSP SETUP request */
static int setupStream(PRTSP_MESSAGE response, char* target) {
RTSP_MESSAGE request;
int ret;
ret = initializeRtspRequest(&request, "SETUP", target);
if (ret != 0) {
if (hasSessionId) {
if (!addOption(&request, "Session", sessionIdString)) {
ret = 0;
goto FreeMessage;
}
}
if (addOption(&request, "Transport", " ") &&
addOption(&request, "If-Modified-Since",
"Thu, 01 Jan 1970 00:00:00 GMT")) {
ret = transactRtspMessage(&request, response);
}
else {
ret = 0;
}
FreeMessage:
freeMessage(&request);
}
return ret;
}
/* Send RTSP PLAY request*/
static int playStream(PRTSP_MESSAGE response, char* target) {
RTSP_MESSAGE request;
int ret;
ret = initializeRtspRequest(&request, "PLAY", target);
if (ret != 0) {
if (addOption(&request, "Session", sessionIdString)) {
ret = transactRtspMessage(&request, response);
}
else {
ret = 0;
}
freeMessage(&request);
}
return ret;
}
/* Send RTSP ANNOUNCE message */
static int sendVideoAnnounce(PRTSP_MESSAGE response, PSTREAM_CONFIGURATION streamConfig) {
RTSP_MESSAGE request;
int ret;
int payloadLength;
char payloadLengthStr[16];
struct in_addr sdpAddr;
ret = initializeRtspRequest(&request, "ANNOUNCE", "streamid=video");
if (ret != 0) {
ret = 0;
if (!addOption(&request, "Session", sessionIdString) ||
!addOption(&request, "Content-type", "application/sdp")) {
goto FreeMessage;
}
memcpy(&sdpAddr, &remoteAddr, sizeof(remoteAddr));
request.payload = getSdpPayloadForStreamConfig(streamConfig, sdpAddr, &payloadLength);
if (request.payload == NULL) {
goto FreeMessage;
}
request.flags |= FLAG_ALLOCATED_PAYLOAD;
request.payloadLength = payloadLength;
sprintf(payloadLengthStr, "%d", payloadLength);
if (!addOption(&request, "Content-length", payloadLengthStr)) {
goto FreeMessage;
}
ret = transactRtspMessage(&request, response);
FreeMessage:
freeMessage(&request);
}
return ret;
}
/* Perform RTSP Handshake with the streaming server machine as part of the connection process */
int performRtspHandshake(IP_ADDRESS addr, PSTREAM_CONFIGURATION streamConfigPtr) {
struct in_addr inaddr;
// Initialize global state
remoteAddr = addr;
memcpy(&inaddr, &addr, sizeof(addr));
sprintf(rtspTargetUrl, "rtsp://%s", inet_ntoa(inaddr));
currentSeqNumber = 1;
hasSessionId = 0;
{
RTSP_MESSAGE response;
if (!requestOptions(&response)) {
Limelog("RTSP OPTIONS request failed\n");
return -1;
}
if (response.message.response.statusCode != 200) {
Limelog("RTSP OPTIONS request failed: %d\n",
response.message.response.statusCode);
return -1;
}
freeMessage(&response);
}
{
RTSP_MESSAGE response;
if (!requestDescribe(&response)) {
Limelog("RTSP DESCRIBE request failed\n");
return -1;
}
if (response.message.response.statusCode != 200) {
Limelog("RTSP DESCRIBE request failed: %d\n",
response.message.response.statusCode);
return -1;
}
freeMessage(&response);
}
{
RTSP_MESSAGE response;
char* sessionId;
if (!setupStream(&response, "streamid=audio")) {
Limelog("RTSP SETUP streamid=audio request failed\n");
return -1;
}
if (response.message.response.statusCode != 200) {
Limelog("RTSP SETUP streamid=audio request failed: %d\n",
response.message.response.statusCode);
return -1;
}
sessionId = getOptionContent(response.options, "Session");
if (sessionId == NULL) {
Limelog("RTSP SETUP streamid=audio is missing session attribute");
return -1;
}
strcpy(sessionIdString, sessionId);
hasSessionId = 1;
freeMessage(&response);
}
{
RTSP_MESSAGE response;
if (!setupStream(&response, "streamid=video")) {
Limelog("RTSP SETUP streamid=video request failed\n");
return -1;
}
if (response.message.response.statusCode != 200) {
Limelog("RTSP SETUP streamid=video request failed: %d\n",
response.message.response.statusCode);
return -1;
}
freeMessage(&response);
}
{
RTSP_MESSAGE response;
if (!sendVideoAnnounce(&response, streamConfigPtr)) {
Limelog("RTSP ANNOUNCE request failed\n");
return -1;
}
if (response.message.response.statusCode != 200) {
Limelog("RTSP ANNOUNCE request failed: %d\n",
response.message.response.statusCode);
return -1;
}
freeMessage(&response);
}
{
RTSP_MESSAGE response;
if (!playStream(&response, "streamid=video")) {
Limelog("RTSP PLAY streamid=video request failed\n");
return -1;
}
if (response.message.response.statusCode != 200) {
Limelog("RTSP PLAY streamid=video failed: %d\n",
response.message.response.statusCode);
return -1;
}
freeMessage(&response);
}
{
RTSP_MESSAGE response;
if (!playStream(&response, "streamid=audio")) {
Limelog("RTSP PLAY streamid=audio request failed\n");
return -1;
}
if (response.message.response.statusCode != 200) {
Limelog("RTSP PLAY streamid=audio failed: %d\n",
response.message.response.statusCode);
return -1;
}
freeMessage(&response);
}
return 0;
}