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4f40781fff
based off RFC 4588 and the server-rtpaux example in -good
513 lines
14 KiB
C
513 lines
14 KiB
C
/* GStreamer
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* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.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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* SECTION:rtsp-sdp
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* @short_description: Make SDP messages
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* @see_also: #GstRTSPMedia
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*
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* Last reviewed on 2013-07-11 (1.0.0)
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*/
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#include <string.h>
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#include <gst/sdp/gstmikey.h>
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#include "rtsp-sdp.h"
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#define AES_128_KEY_LEN 16
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#define AES_256_KEY_LEN 32
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#define HMAC_32_KEY_LEN 4
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#define HMAC_80_KEY_LEN 10
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static gboolean
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get_info_from_tags (GstPad * pad, GstEvent ** event, gpointer user_data)
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{
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GstSDPMedia *media = (GstSDPMedia *) user_data;
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if (GST_EVENT_TYPE (*event) == GST_EVENT_TAG) {
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GstTagList *tags;
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guint bitrate = 0;
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gst_event_parse_tag (*event, &tags);
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if (gst_tag_list_get_scope (tags) != GST_TAG_SCOPE_STREAM)
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return TRUE;
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if (!gst_tag_list_get_uint (tags, GST_TAG_MAXIMUM_BITRATE,
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&bitrate) || bitrate == 0)
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if (!gst_tag_list_get_uint (tags, GST_TAG_BITRATE, &bitrate) ||
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bitrate == 0)
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return TRUE;
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/* set bandwidth (kbits/s) */
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gst_sdp_media_add_bandwidth (media, GST_SDP_BWTYPE_AS, bitrate / 1000);
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return FALSE;
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}
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return TRUE;
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}
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static void
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update_sdp_from_tags (GstRTSPStream * stream, GstSDPMedia * stream_media)
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{
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GstPad *src_pad;
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src_pad = gst_rtsp_stream_get_srcpad (stream);
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gst_pad_sticky_events_foreach (src_pad, get_info_from_tags, stream_media);
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gst_object_unref (src_pad);
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}
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static guint8
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enc_key_length_from_cipher_name (const gchar * cipher)
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{
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if (g_strcmp0 (cipher, "aes-128-icm") == 0)
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return AES_128_KEY_LEN;
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else if (g_strcmp0 (cipher, "aes-256-icm") == 0)
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return AES_256_KEY_LEN;
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else {
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GST_ERROR ("encryption algorithm '%s' not supported", cipher);
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return 0;
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}
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}
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static guint8
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auth_key_length_from_auth_name (const gchar * auth)
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{
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if (g_strcmp0 (auth, "hmac-sha1-32") == 0)
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return HMAC_32_KEY_LEN;
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else if (g_strcmp0 (auth, "hmac-sha1-80") == 0)
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return HMAC_80_KEY_LEN;
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else {
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GST_ERROR ("authentication algorithm '%s' not supported", auth);
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return 0;
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}
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}
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static void
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make_media (GstSDPMessage * sdp, GstSDPInfo * info, GstRTSPMedia * media,
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GstRTSPStream * stream, GstStructure * s, GstRTSPProfile profile)
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{
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GstSDPMedia *smedia;
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const gchar *caps_str, *caps_enc, *caps_params;
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gchar *tmp;
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gint caps_pt, caps_rate;
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guint n_fields, j;
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gboolean first;
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GString *fmtp;
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GstRTSPLowerTrans ltrans;
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GSocketFamily family;
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const gchar *addrtype, *proto;
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gchar *address;
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guint ttl;
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GstClockTime rtx_time;
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gst_sdp_media_new (&smedia);
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/* get media type and payload for the m= line */
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caps_str = gst_structure_get_string (s, "media");
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gst_sdp_media_set_media (smedia, caps_str);
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gst_structure_get_int (s, "payload", &caps_pt);
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tmp = g_strdup_printf ("%d", caps_pt);
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gst_sdp_media_add_format (smedia, tmp);
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g_free (tmp);
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gst_sdp_media_set_port_info (smedia, 0, 1);
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switch (profile) {
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case GST_RTSP_PROFILE_AVP:
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proto = "RTP/AVP";
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break;
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case GST_RTSP_PROFILE_AVPF:
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proto = "RTP/AVPF";
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break;
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case GST_RTSP_PROFILE_SAVP:
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proto = "RTP/SAVP";
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break;
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case GST_RTSP_PROFILE_SAVPF:
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proto = "RTP/SAVPF";
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break;
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default:
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proto = "udp";
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break;
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}
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gst_sdp_media_set_proto (smedia, proto);
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if (info->is_ipv6) {
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addrtype = "IP6";
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family = G_SOCKET_FAMILY_IPV6;
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} else {
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addrtype = "IP4";
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family = G_SOCKET_FAMILY_IPV4;
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}
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ltrans = gst_rtsp_stream_get_protocols (stream);
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if (ltrans == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
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GstRTSPAddress *addr;
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addr = gst_rtsp_stream_get_multicast_address (stream, family);
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if (addr == NULL)
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goto no_multicast;
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address = g_strdup (addr->address);
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ttl = addr->ttl;
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gst_rtsp_address_free (addr);
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} else {
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ttl = 16;
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if (info->is_ipv6)
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address = g_strdup ("::");
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else
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address = g_strdup ("0.0.0.0");
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}
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/* for the c= line */
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gst_sdp_media_add_connection (smedia, "IN", addrtype, address, ttl, 1);
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g_free (address);
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/* get clock-rate, media type and params for the rtpmap attribute */
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gst_structure_get_int (s, "clock-rate", &caps_rate);
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caps_enc = gst_structure_get_string (s, "encoding-name");
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caps_params = gst_structure_get_string (s, "encoding-params");
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if (caps_enc) {
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if (caps_params)
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tmp = g_strdup_printf ("%d %s/%d/%s", caps_pt, caps_enc, caps_rate,
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caps_params);
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else
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tmp = g_strdup_printf ("%d %s/%d", caps_pt, caps_enc, caps_rate);
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gst_sdp_media_add_attribute (smedia, "rtpmap", tmp);
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g_free (tmp);
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}
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/* the config uri */
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tmp = gst_rtsp_stream_get_control (stream);
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gst_sdp_media_add_attribute (smedia, "control", tmp);
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g_free (tmp);
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/* check for srtp */
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do {
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GstBuffer *srtpkey;
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const GValue *val;
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const gchar *srtpcipher, *srtpauth, *srtcpcipher, *srtcpauth;
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GstMIKEYMessage *msg;
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GstMIKEYPayload *payload, *pkd;
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GBytes *bytes;
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GstMapInfo info;
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const guint8 *data;
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gsize size;
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gchar *base64;
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guint8 byte;
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guint32 ssrc;
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val = gst_structure_get_value (s, "srtp-key");
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if (val == NULL)
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break;
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srtpkey = gst_value_get_buffer (val);
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if (srtpkey == NULL)
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break;
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srtpcipher = gst_structure_get_string (s, "srtp-cipher");
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srtpauth = gst_structure_get_string (s, "srtp-auth");
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srtcpcipher = gst_structure_get_string (s, "srtcp-cipher");
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srtcpauth = gst_structure_get_string (s, "srtcp-auth");
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if (srtpcipher == NULL || srtpauth == NULL || srtcpcipher == NULL ||
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srtcpauth == NULL)
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break;
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msg = gst_mikey_message_new ();
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/* unencrypted MIKEY message, we send this over TLS so this is allowed */
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gst_mikey_message_set_info (msg, GST_MIKEY_VERSION, GST_MIKEY_TYPE_PSK_INIT,
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FALSE, GST_MIKEY_PRF_MIKEY_1, 0, GST_MIKEY_MAP_TYPE_SRTP);
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/* add policy '0' for our SSRC */
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gst_rtsp_stream_get_ssrc (stream, &ssrc);
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gst_mikey_message_add_cs_srtp (msg, 0, ssrc, 0);
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/* timestamp is now */
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gst_mikey_message_add_t_now_ntp_utc (msg);
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/* add some random data */
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gst_mikey_message_add_rand_len (msg, 16);
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/* the policy '0' is SRTP with the above discovered algorithms */
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payload = gst_mikey_payload_new (GST_MIKEY_PT_SP);
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gst_mikey_payload_sp_set (payload, 0, GST_MIKEY_SEC_PROTO_SRTP);
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/* only AES-CM is supported */
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byte = 1;
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gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_ENC_ALG, 1,
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&byte);
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/* Encryption key length */
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byte = enc_key_length_from_cipher_name (srtpcipher);
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gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_ENC_KEY_LEN, 1,
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&byte);
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/* only HMAC-SHA1 */
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gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_AUTH_ALG, 1,
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&byte);
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/* Authentication key length */
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byte = auth_key_length_from_auth_name (srtpauth);
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gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_AUTH_KEY_LEN, 1,
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&byte);
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/* we enable encryption on RTP and RTCP */
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gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTP_ENC, 1,
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&byte);
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gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTCP_ENC, 1,
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&byte);
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/* we enable authentication on RTP and RTCP */
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gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTP_AUTH, 1,
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&byte);
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gst_mikey_message_add_payload (msg, payload);
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/* make unencrypted KEMAC */
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payload = gst_mikey_payload_new (GST_MIKEY_PT_KEMAC);
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gst_mikey_payload_kemac_set (payload, GST_MIKEY_ENC_NULL,
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GST_MIKEY_MAC_NULL);
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/* add the key in key data */
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pkd = gst_mikey_payload_new (GST_MIKEY_PT_KEY_DATA);
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gst_buffer_map (srtpkey, &info, GST_MAP_READ);
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gst_mikey_payload_key_data_set_key (pkd, GST_MIKEY_KD_TEK, info.size,
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info.data);
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gst_buffer_unmap (srtpkey, &info);
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/* add key data to KEMAC */
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gst_mikey_payload_kemac_add_sub (payload, pkd);
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gst_mikey_message_add_payload (msg, payload);
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/* now serialize this to bytes */
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bytes = gst_mikey_message_to_bytes (msg, NULL, NULL);
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gst_mikey_message_unref (msg);
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/* and make it into base64 */
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data = g_bytes_get_data (bytes, &size);
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base64 = g_base64_encode (data, size);
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g_bytes_unref (bytes);
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tmp = g_strdup_printf ("mikey %s", base64);
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g_free (base64);
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gst_sdp_media_add_attribute (smedia, "key-mgmt", tmp);
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g_free (tmp);
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} while (FALSE);
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/* collect all other properties and add them to fmtp or attributes */
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fmtp = g_string_new ("");
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g_string_append_printf (fmtp, "%d ", caps_pt);
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first = TRUE;
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n_fields = gst_structure_n_fields (s);
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for (j = 0; j < n_fields; j++) {
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const gchar *fname, *fval;
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fname = gst_structure_nth_field_name (s, j);
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/* filter out standard properties */
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if (!strcmp (fname, "media"))
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continue;
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if (!strcmp (fname, "payload"))
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continue;
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if (!strcmp (fname, "clock-rate"))
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continue;
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if (!strcmp (fname, "encoding-name"))
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continue;
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if (!strcmp (fname, "encoding-params"))
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continue;
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if (!strcmp (fname, "ssrc"))
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continue;
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if (!strcmp (fname, "timestamp-offset"))
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continue;
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if (!strcmp (fname, "seqnum-offset"))
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continue;
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if (g_str_has_prefix (fname, "srtp-"))
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continue;
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if (g_str_has_prefix (fname, "srtcp-"))
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continue;
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/* handled later */
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if (g_str_has_prefix (fname, "x-gst-rtsp-server-rtx-time"))
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continue;
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if (g_str_has_prefix (fname, "a-")) {
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/* attribute */
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if ((fval = gst_structure_get_string (s, fname)))
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gst_sdp_media_add_attribute (smedia, fname + 2, fval);
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continue;
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}
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if (g_str_has_prefix (fname, "x-")) {
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/* attribute */
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if ((fval = gst_structure_get_string (s, fname)))
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gst_sdp_media_add_attribute (smedia, fname, fval);
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continue;
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}
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if ((fval = gst_structure_get_string (s, fname))) {
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g_string_append_printf (fmtp, "%s%s=%s", first ? "" : ";", fname, fval);
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first = FALSE;
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}
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}
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if (!first) {
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tmp = g_string_free (fmtp, FALSE);
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gst_sdp_media_add_attribute (smedia, "fmtp", tmp);
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g_free (tmp);
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} else {
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g_string_free (fmtp, TRUE);
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}
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update_sdp_from_tags (stream, smedia);
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if ((rtx_time = gst_rtsp_stream_get_retransmission_time (stream))) {
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/* ssrc multiplexed retransmit functionality */
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guint rtx_pt = gst_rtsp_stream_get_retransmission_pt (stream);
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if (rtx_pt == 0) {
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g_warning ("failed to find an available dynamic payload type. "
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"Not adding retransmission");
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} else {
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gchar *tmp;
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tmp = g_strdup_printf ("%d", rtx_pt);
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gst_sdp_media_add_format (smedia, tmp);
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g_free (tmp);
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tmp = g_strdup_printf ("%d rtx/%d", rtx_pt, caps_rate);
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gst_sdp_media_add_attribute (smedia, "rtpmap", tmp);
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g_free (tmp);
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tmp =
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g_strdup_printf ("%d apt=%d;rtx-time=%" G_GINT64_FORMAT, rtx_pt,
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caps_pt, GST_TIME_AS_MSECONDS (rtx_time));
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gst_sdp_media_add_attribute (smedia, "fmtp", tmp);
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g_free (tmp);
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}
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}
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gst_sdp_message_add_media (sdp, smedia);
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gst_sdp_media_free (smedia);
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return;
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/* ERRORS */
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no_multicast:
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{
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gst_sdp_media_free (smedia);
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g_warning ("ignoring stream %d without multicast address",
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gst_rtsp_stream_get_index (stream));
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return;
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}
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}
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/**
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* gst_rtsp_sdp_from_media:
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* @sdp: a #GstSDPMessage
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* @info: (transfer none): a #GstSDPInfo
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* @media: (transfer none): a #GstRTSPMedia
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*
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* Add @media specific info to @sdp. @info is used to configure the connection
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* information in the SDP.
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*
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* Returns: TRUE on success.
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*/
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gboolean
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gst_rtsp_sdp_from_media (GstSDPMessage * sdp, GstSDPInfo * info,
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GstRTSPMedia * media)
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{
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guint i, n_streams;
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gchar *rangestr;
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n_streams = gst_rtsp_media_n_streams (media);
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rangestr = gst_rtsp_media_get_range_string (media, FALSE, GST_RTSP_RANGE_NPT);
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if (rangestr == NULL)
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goto not_prepared;
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gst_sdp_message_add_attribute (sdp, "range", rangestr);
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g_free (rangestr);
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for (i = 0; i < n_streams; i++) {
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GstRTSPStream *stream;
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GstCaps *caps;
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GstStructure *s;
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GstRTSPProfile profiles;
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guint mask;
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stream = gst_rtsp_media_get_stream (media, i);
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caps = gst_rtsp_stream_get_caps (stream);
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if (caps == NULL) {
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g_warning ("ignoring stream %d without media type", i);
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continue;
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}
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s = gst_caps_get_structure (caps, 0);
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if (s == NULL) {
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gst_caps_unref (caps);
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g_warning ("ignoring stream %d without media type", i);
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continue;
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}
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/* make a new media for each profile */
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profiles = gst_rtsp_stream_get_profiles (stream);
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mask = 1;
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while (profiles >= mask) {
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GstRTSPProfile prof = profiles & mask;
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if (prof)
|
|
make_media (sdp, info, media, stream, s, prof);
|
|
|
|
mask <<= 1;
|
|
}
|
|
gst_caps_unref (caps);
|
|
}
|
|
|
|
{
|
|
GstNetTimeProvider *provider;
|
|
|
|
if ((provider =
|
|
gst_rtsp_media_get_time_provider (media, info->server_ip, 0))) {
|
|
GstClock *clock;
|
|
gchar *address, *str;
|
|
gint port;
|
|
|
|
g_object_get (provider, "clock", &clock, "address", &address, "port",
|
|
&port, NULL);
|
|
|
|
str = g_strdup_printf ("GstNetTimeProvider %s %s:%d %" G_GUINT64_FORMAT,
|
|
g_type_name (G_TYPE_FROM_INSTANCE (clock)), address, port,
|
|
gst_clock_get_time (clock));
|
|
|
|
gst_sdp_message_add_attribute (sdp, "x-gst-clock", str);
|
|
g_free (str);
|
|
gst_object_unref (clock);
|
|
g_free (address);
|
|
gst_object_unref (provider);
|
|
}
|
|
}
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
not_prepared:
|
|
{
|
|
GST_ERROR ("media %p is not prepared", media);
|
|
return FALSE;
|
|
}
|
|
}
|