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;
}
/**
* gst_rtsp_message_new_request:
* @msg: a location for the new #GstRTSPMessage
* @method: the request method to use
* @uri: the uri of the request
*
* Create a new #GstRTSPMessage with @method and @uri and store the result
* request message in @msg.
*
* Returns: a #GstRTSPResult. Free with gst_rtsp_message_free().
*/
GstRTSPResult
gst_rtsp_message_new_request (GstRTSPMessage ** msg, GstRTSPMethod method,
const gchar * uri)
{
GstRTSPMessage *newmsg;
g_return_val_if_fail (msg != NULL, GST_RTSP_EINVAL);
g_return_val_if_fail (uri != NULL, GST_RTSP_EINVAL);
newmsg = g_new0 (GstRTSPMessage, 1);
*msg = newmsg;
return gst_rtsp_message_init_request (newmsg, method, uri);
}
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 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;
}
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;
}
