static gint create_and_send_RECORD_message(GstRTSPsink* sink, GTimeVal *timeout, char *szSessionNumber)
{
GstRTSPMethod method;
GstRTSPMessage msg = { 0 };
const gchar *url_server_str_full = g_strdup_printf("rtsp://%s:%d/%s", sink->host, sink->port, sink->stream_name); //"rtsp://192.168.2.108:1935/live/1";
GstRTSPResult res;
method = GST_RTSP_RECORD;
res = gst_rtsp_message_init_request(&msg, method, url_server_str_full);
if (res < 0)
return res;
gst_rtsp_message_add_header(&msg, GST_RTSP_HDR_RANGE, "npt=0.000-"); // start live.
gst_rtsp_message_add_header(&msg, GST_RTSP_HDR_SESSION, szSessionNumber);
// Send our packet receive server answer and check some basic checks.
if ((res = sendReceiveAndCheck(sink->conn, timeout, &msg, sink->debug)) != GST_RTSP_OK) {
return res;
}
return res;
}
GstRTSPResult rtsp_session_describe(RTSP_client_session *session)
{
GstRTSPMessage *message;
GstRTSPResult res;
gchar *uri, *cseq;
gint nseq;
uri = g_strdup_printf("rtsp://%s:%s%s", session->url->hostname,
session->url->port, session->url->path);
GST_RTSP_CHECK(
gst_rtsp_message_new_request(&message,
GST_RTSP_DESCRIBE,
uri
),
no_describe_message
);
nseq = ++session->rtsp_seq;
cseq = g_strdup_printf("%d", nseq);
gst_rtsp_message_add_header(message,
GST_RTSP_HDR_CSEQ,
cseq
);
gst_rtsp_message_add_header(message,
GST_RTSP_HDR_ACCEPT,
"application/sdp"
);
res = rtsp_sesseion_send_message(session, message);
gst_rtsp_message_free(message);
if (res == GST_RTSP_OK)
{
session->wait_for.seq = nseq;
session->wait_for.method = GST_RTSP_DESCRIBE;
}
g_free(cseq);
g_free(uri);
return res;
no_describe_message:
g_free(uri);
return GST_RTSP_ENOMEM;
}
void
rtsp_handle_options_request(RTSP_Client *client, RTSP_Ps *state)
{
GstRTSPMethod options;
gchar *str;
rtsp_client_remove_dead_session(client);
options = GST_RTSP_DESCRIBE |
GST_RTSP_OPTIONS |
GST_RTSP_PAUSE |
GST_RTSP_PLAY |
GST_RTSP_SETUP |
GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN;
str = gst_rtsp_options_as_text(options);
gst_rtsp_message_init_response(
state->response,
GST_RTSP_STS_OK,
gst_rtsp_status_as_text(GST_RTSP_STS_OK),
state->request
);
gst_rtsp_message_add_header(state->response, GST_RTSP_HDR_PUBLIC, str);
g_free(str);
rtsp_client_send_response(client, NULL, state->response);
}
static gint create_and_send_SETUP_message(GstRTSPsink* sink, GTimeVal *timeout, char *szSessionNumber)
{
GstRTSPMethod method;
const gchar *url_server_str_full = g_strdup_printf("rtsp://%s:%d/%s", sink->host, sink->port, sink->stream_name); //"rtsp://192.168.2.108:1935/live/1";
GstRTSPResult res;
GstRTSPMessage msg = { 0 };
gint video_start_port = 5002;
gint video_end_port = video_start_port + 1;
gchar *transfer_foramt;
gchar *tmp;
method = GST_RTSP_SETUP;
tmp = g_strdup_printf("%s/streamid=0", url_server_str_full);
res = gst_rtsp_message_init_request(&msg, method, tmp);
if (res < 0)
return res;
transfer_foramt = g_strdup_printf("RTP/AVP/UDP;unicast;client_port=%d-%d;mode=record", video_start_port, video_end_port);
gst_rtsp_message_add_header(&msg, GST_RTSP_HDR_TRANSPORT, transfer_foramt);
gst_rtsp_message_add_header(&msg, GST_RTSP_HDR_SESSION, szSessionNumber); // TODO: Get the session id from the responce.
gst_rtsp_message_add_header(&msg, GST_RTSP_HDR_CONTENT_LENGTH, "0"); // TODO: Get the session id from the responce.
// Send our packet receive server answer and check some basic checks.
if ((res = sendReceiveAndCheck(sink->conn, timeout, &msg, sink->debug)) != GST_RTSP_OK) {
return res;
}
GstRTSPTransport *transport;
res = gst_rtsp_transport_new(&transport);
res = extractTransportFromMessage(&msg, transport);
g_print("Got server port %d", transport->server_port);
sink->server_rtp_port = transport->server_port.min;
if (res != GST_RTSP_OK)
return -ERR_PARSING;
return GST_RTSP_OK;
}
/**
* gst_rtsp_message_init_response:
* @msg: a #GstRTSPMessage
* @code: the status code
* @reason: the status reason or #NULL
* @request: the request that triggered the response or #NULL
*
* Initialize @msg with @code and @reason.
*
* When @reason is #NULL, the default reason for @code will be used.
*
* When @request is not #NULL, the relevant headers will be copied to the new
* response message.
*
* Returns: a #GstRTSPResult.
*/
GstRTSPResult
gst_rtsp_message_init_response (GstRTSPMessage * msg, GstRTSPStatusCode code,
const gchar * reason, const GstRTSPMessage * request)
{
g_return_val_if_fail (msg != NULL, GST_RTSP_EINVAL);
gst_rtsp_message_unset (msg);
if (reason == NULL)
reason = gst_rtsp_status_as_text (code);
msg->type = GST_RTSP_MESSAGE_RESPONSE;
msg->type_data.response.code = code;
msg->type_data.response.reason = g_strdup (reason);
msg->type_data.response.version = GST_RTSP_VERSION_1_0;
msg->hdr_fields = g_array_new (FALSE, FALSE, sizeof (RTSPKeyValue));
if (request) {
gchar *header;
/* copy CSEQ */
if (gst_rtsp_message_get_header (request, GST_RTSP_HDR_CSEQ, &header,
0) == GST_RTSP_OK) {
gst_rtsp_message_add_header (msg, GST_RTSP_HDR_CSEQ, header);
}
/* copy session id */
if (gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &header,
0) == GST_RTSP_OK) {
char *pos;
header = g_strdup (header);
if ((pos = strchr (header, ';'))) {
*pos = '\0';
}
g_strchomp (header);
gst_rtsp_message_add_header (msg, GST_RTSP_HDR_SESSION, header);
g_free (header);
}
/* FIXME copy more headers? */
}
return GST_RTSP_OK;
}
static gint create_and_send_OPTION_message(GstRTSPsink* sink, GTimeVal *timeout) {
GstRTSPResult res;
const gchar *url_server_str = g_strdup_printf("rtsp://%s", sink->host); //"rtsp://192.168.2.108"; // TODO: get ip and port from parameters.
const gchar *url_server_ip_str = sink->host;// "192.168.2.108";
//GstRTSPConnection *conn = sink->conn ;
int port = sink->port;
GstRTSPUrl * url;
GstRTSPMessage msg = { 0 };
// set parameters
res = gst_rtsp_url_parse((const guint8*)url_server_str, &url);
res = gst_rtsp_url_set_port(url, port);
// create connection
res = gst_rtsp_connection_create(url, &sink->conn);
res = gst_rtsp_connection_connect(sink->conn, timeout);
if (res != GST_RTSP_OK)
goto beach;
GstRTSPMethod method = GST_RTSP_OPTIONS;
res = gst_rtsp_message_init_request(&msg, method, url_server_str);
if (res < 0)
return res;
/* set user-agent */
if (sink->user_agent)
gst_rtsp_message_add_header(&msg, GST_RTSP_HDR_USER_AGENT, sink->user_agent);
// Send our packet receive server answer and check some basic checks.
if ((res = sendReceiveAndCheck(sink->conn, timeout, &msg, sink->debug)) != GST_RTSP_OK) {
return res;
}
// check if server supports RECORD.
if (isServerSupportStreamPush(&msg) != GST_RTSP_OK) {
return -ERR_CANNOT_PUSH_STREAM;
}
beach:
return GST_RTSP_OK;
}
static __inline__ void
nmp_rtsp_give_options_response(NmpMediaDevice *device,
NmpRtspState *state, GstRTSPStatusCode code)
{
GstRTSPMethod options;
gchar *str;
if (code == GST_RTSP_STS_OK)
{
options = GST_RTSP_OPTIONS | GST_RTSP_TEARDOWN;
str = gst_rtsp_options_as_text(options);
gst_rtsp_message_init_response(state->response, GST_RTSP_STS_OK,
gst_rtsp_status_as_text(GST_RTSP_STS_OK), state->request);
gst_rtsp_message_add_header(state->response, GST_RTSP_HDR_PUBLIC, str);
g_free(str);
nmp_rtsp_device_send_response(device, state->response);
}
else
{
nmp_rtsp_device_send_generic_response(device, code, state);
}
}
static void
test_client_sdp (const gchar * launch_line, guint * bandwidth_val)
{
GstRTSPClient *client;
GstRTSPMessage request = { 0, };
gchar *str;
/* simple DESCRIBE for an existing url */
client = setup_client (launch_line);
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
"rtsp://localhost/test") == GST_RTSP_OK);
str = g_strdup_printf ("%d", cseq);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
g_free (str);
gst_rtsp_client_set_send_func (client, test_response_sdp,
(gpointer) bandwidth_val, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
teardown_client (client);
}
static gint create_and_send_ANNOUNCE_message2(GstRTSPsink* sink, GTimeVal *timeout, char **szSessionNumber) {
const gchar *url_client_ip_str = "0.0.0.0";//"192.168.2.104";
const gchar *url_server_str_full = g_strdup_printf("rtsp://%s:%d/%s", sink->host, sink->port, sink->stream_name); //"rtsp://192.168.2.108:1935/live/1";
//conn = sink->conn;
GstRTSPMessage msg = { 0 };
GstSDPMessage *sdp;
GstRTSPMethod method;
GstRTSPResult res;
guint num_ports = 1;
guint rtp_port = 5006;
char *szPayloadType = g_strdup_printf("%d", sink->payload);
method = GST_RTSP_ANNOUNCE ;
res = gst_rtsp_message_init_request(&msg, method, url_server_str_full);
if (res < 0)
return res;
/* set user-agent */
if (sink->user_agent)
gst_rtsp_message_add_header(&msg, GST_RTSP_HDR_USER_AGENT, sink->user_agent);
gst_rtsp_message_add_header(&msg, GST_RTSP_HDR_CONTENT_TYPE, "application/sdp");
// allocate sdp messege buffer...
res = gst_sdp_message_new(&sdp);
//v=..
res = gst_sdp_message_set_version(sdp, "0");
//o=...
res = gst_sdp_message_set_origin(sdp, "-", "0", "0", "IN", "IP4", "0.0.0.0");
//s=..
if (sink->session_name)
res = gst_sdp_message_set_session_name(sdp, "Unnamed");
//i=..
if (sink->information)
res = gst_sdp_message_set_information(sdp, "N/A");
//c=...
res = gst_sdp_message_set_connection(sdp, "IN", "IP4", url_client_ip_str, 0, 0);
//a=...
res = gst_sdp_message_add_attribute(sdp, "recvonly", NULL);
// create media
GstSDPMedia *media;
res = gst_sdp_media_new(&media);
res = gst_sdp_media_init(media);
//m=...
res = gst_sdp_media_set_media(media, "video");
res = gst_sdp_media_set_port_info(media, rtp_port, num_ports);
res = gst_sdp_media_set_proto(media, "RTP/AVP");
res = gst_sdp_media_add_format(media, szPayloadType);
//a=...
char *rtpmap = g_strdup_printf("%s %s/%d", szPayloadType, sink->encoding_name, sink->clock_rate);
res = gst_sdp_media_add_attribute(media, "rtpmap", rtpmap);
res = gst_sdp_media_add_attribute(media, "fmtp", szPayloadType);
res = gst_sdp_media_add_attribute(media, "control", "streamid=0");
// insert media into sdp
res = gst_sdp_message_add_media(sdp, media);
gchar * sdp_str = gst_sdp_message_as_text(sdp);
int size = g_utf8_strlen(sdp_str, 500);
gst_sdp_message_free(sdp);
gst_sdp_media_free(media);
res = gst_rtsp_message_set_body(&msg, sdp_str, size);
sink->responce = &msg;
// Send our packet receive server answer and check some basic checks.
if ((res = sendReceiveAndCheck(sink->conn, timeout, &msg, sink->debug)) != GST_RTSP_OK) {
return res;
}
// get session number
*szSessionNumber = extractSessionNumberFromMessage(&msg);
return GST_RTSP_OK;
}
