Esempio n. 1
0
static __inline__ gint
__rtp_pkt_send_rot(RTP_session *session, RTP_transport *transport, 
	RTP_packet *packet, gsize packet_size)
{
	GstRTSPMessage *msg;

	gst_rtsp_message_new_data(&msg, transport->rtp_ch);
	gst_rtsp_message_set_body(msg, (const guint8*)packet, packet_size);
	rtp_watch_write_message((RTP_watch*)transport->passby, msg);
	gst_rtsp_message_free(msg);

	return 0;
}
Esempio n. 2
0
static __inline__ gint
__rtp_pkt_send_ror(RTP_session *session, RTP_transport *transport, 
	RTP_packet *packet, gsize packet_size)
{
	GstRTSPMessage *msg;
	extern gint rtsp_client_send_message(RTSP_Client *client, GstRTSPMessage *msg);

	gst_rtsp_message_new_data(&msg, transport->rtp_ch);
	gst_rtsp_message_set_body(msg, (const guint8*)packet, packet_size);
	rtsp_client_send_message((RTSP_Client*)transport->passby, msg);
	gst_rtsp_message_free(msg);

	return 0;
}
Esempio n. 3
0
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;
}