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Add flag to select TLS in the transport. Enable TLS on the socketclient when we use a TLS uri.
179 lines
5.9 KiB
C
179 lines
5.9 KiB
C
/* GStreamer
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* Copyright (C) <2005,2006> Wim Taymans <wim@fluendo.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/*
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* Unless otherwise indicated, Source Code is licensed under MIT license.
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* See further explanation attached in License Statement (distributed in the file
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* LICENSE).
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy of
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* this software and associated documentation files (the "Software"), to deal in
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* the Software without restriction, including without limitation the rights to
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* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
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* of the Software, and to permit persons to whom the Software is furnished to do
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* so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in all
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* copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
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* SOFTWARE.
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*/
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#ifndef __GST_RTSP_TRANSPORT_H__
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#define __GST_RTSP_TRANSPORT_H__
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#include <gst/gstconfig.h>
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#include <gst/rtsp/gstrtspdefs.h>
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G_BEGIN_DECLS
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/**
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* GstRTSPTransMode:
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* @GST_RTSP_TRANS_UNKNOWN: invalid tansport mode
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* @GST_RTSP_TRANS_RTP: transfer RTP data
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* @GST_RTSP_TRANS_RDT: transfer RDT (RealMedia) data
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*
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* The transfer mode to use.
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*/
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typedef enum {
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GST_RTSP_TRANS_UNKNOWN = 0,
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GST_RTSP_TRANS_RTP = (1 << 0),
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GST_RTSP_TRANS_RDT = (1 << 1)
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} GstRTSPTransMode;
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/**
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* GstRTSPProfile:
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* @GST_RTSP_PROFILE_UNKNOWN: invalid profile
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* @GST_RTSP_PROFILE_AVP: the Audio/Visual profile
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* @GST_RTSP_PROFILE_SAVP: the secure Audio/Visual profile
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*
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* The transfer profile to use.
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*/
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typedef enum {
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GST_RTSP_PROFILE_UNKNOWN = 0,
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GST_RTSP_PROFILE_AVP = (1 << 0),
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GST_RTSP_PROFILE_SAVP = (1 << 1)
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} GstRTSPProfile;
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/**
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* GstRTSPLowerTrans:
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* @GST_RTSP_LOWER_TRANS_UNKNOWN: invalid transport flag
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* @GST_RTSP_LOWER_TRANS_UDP: stream data over UDP
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* @GST_RTSP_LOWER_TRANS_UDP_MCAST: stream data over UDP multicast
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* @GST_RTSP_LOWER_TRANS_TCP: stream data over TCP
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* @GST_RTSP_LOWER_TRANS_HTTP: stream data tunneled over HTTP.
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* @GST_RTSP_LOWER_TRANS_TLS: encrypt TCP and HTTP with TLS
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*
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* The different transport methods.
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*/
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typedef enum {
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GST_RTSP_LOWER_TRANS_UNKNOWN = 0,
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GST_RTSP_LOWER_TRANS_UDP = (1 << 0),
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GST_RTSP_LOWER_TRANS_UDP_MCAST = (1 << 1),
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GST_RTSP_LOWER_TRANS_TCP = (1 << 2),
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GST_RTSP_LOWER_TRANS_HTTP = (1 << 4),
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GST_RTSP_LOWER_TRANS_TLS = (1 << 5)
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} GstRTSPLowerTrans;
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#define GST_TYPE_RTSP_LOWER_TRANS (gst_rtsp_lower_trans_get_type())
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GType gst_rtsp_lower_trans_get_type (void);
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typedef struct _GstRTSPRange GstRTSPRange;
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typedef struct _GstRTSPTransport GstRTSPTransport;
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/**
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* GstRTSPRange:
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* @min: minimum value of the range
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* @max: maximum value of the range
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*
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* A type to specify a range.
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*/
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struct _GstRTSPRange {
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gint min;
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gint max;
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};
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/**
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* GstRTSPTransport:
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* @trans: the transport mode
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* @profile: the tansport profile
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* @lower_transport: the lower transport
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* @destination: the destination ip/hostname
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* @source: the source ip/hostname
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* @layers: the number of layers
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* @mode_play: if play mode was selected
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* @mode_record: if record mode was selected
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* @append: is append mode was selected
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* @interleaved: the interleave range
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* @ttl: the time to live for multicast UDP
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* @port: the port pair for multicast sessions
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* @client_port: the client port pair for receiving data
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* @server_port: the server port pair for receiving data
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* @ssrc: the ssrc that the sender/receiver will use
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*
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* A structure holding the RTSP transport values.
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*/
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struct _GstRTSPTransport {
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GstRTSPTransMode trans;
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GstRTSPProfile profile;
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GstRTSPLowerTrans lower_transport;
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gchar *destination;
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gchar *source;
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guint layers;
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gboolean mode_play;
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gboolean mode_record;
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gboolean append;
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GstRTSPRange interleaved;
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/* multicast specific */
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guint ttl;
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/* UDP specific */
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GstRTSPRange port;
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GstRTSPRange client_port;
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GstRTSPRange server_port;
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/* RTP specific */
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guint ssrc;
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/*< private >*/
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gpointer _gst_reserved[GST_PADDING];
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};
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GstRTSPResult gst_rtsp_transport_new (GstRTSPTransport **transport);
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GstRTSPResult gst_rtsp_transport_init (GstRTSPTransport *transport);
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GstRTSPResult gst_rtsp_transport_parse (const gchar *str, GstRTSPTransport *transport);
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gchar* gst_rtsp_transport_as_text (GstRTSPTransport *transport);
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GstRTSPResult gst_rtsp_transport_get_mime (GstRTSPTransMode trans, const gchar **mime);
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GstRTSPResult gst_rtsp_transport_get_manager (GstRTSPTransMode trans, const gchar **manager, guint option);
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GstRTSPResult gst_rtsp_transport_free (GstRTSPTransport *transport);
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G_END_DECLS
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#endif /* __GST_RTSP_TRANSPORT_H__ */
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