gstreamer/gst/rtsp-server/rtsp-sdp.c

524 lines
14 KiB
C

/* GStreamer
* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:rtsp-sdp
* @short_description: Make SDP messages
* @see_also: #GstRTSPMedia
*
* Last reviewed on 2013-07-11 (1.0.0)
*/
#include <string.h>
#include <gst/sdp/gstmikey.h>
#include "rtsp-sdp.h"
#define AES_128_KEY_LEN 16
#define AES_256_KEY_LEN 32
#define HMAC_32_KEY_LEN 4
#define HMAC_80_KEY_LEN 10
static gboolean
get_info_from_tags (GstPad * pad, GstEvent ** event, gpointer user_data)
{
GstSDPMedia *media = (GstSDPMedia *) user_data;
if (GST_EVENT_TYPE (*event) == GST_EVENT_TAG) {
GstTagList *tags;
guint bitrate = 0;
gst_event_parse_tag (*event, &tags);
if (gst_tag_list_get_scope (tags) != GST_TAG_SCOPE_STREAM)
return TRUE;
if (!gst_tag_list_get_uint (tags, GST_TAG_MAXIMUM_BITRATE,
&bitrate) || bitrate == 0)
if (!gst_tag_list_get_uint (tags, GST_TAG_BITRATE, &bitrate) ||
bitrate == 0)
return TRUE;
/* set bandwidth (kbits/s) */
gst_sdp_media_add_bandwidth (media, GST_SDP_BWTYPE_AS, bitrate / 1000);
return FALSE;
}
return TRUE;
}
static void
update_sdp_from_tags (GstRTSPStream * stream, GstSDPMedia * stream_media)
{
GstPad *src_pad;
src_pad = gst_rtsp_stream_get_srcpad (stream);
gst_pad_sticky_events_foreach (src_pad, get_info_from_tags, stream_media);
gst_object_unref (src_pad);
}
static guint8
enc_key_length_from_cipher_name (const gchar * cipher)
{
if (g_strcmp0 (cipher, "aes-128-icm") == 0)
return AES_128_KEY_LEN;
else if (g_strcmp0 (cipher, "aes-256-icm") == 0)
return AES_256_KEY_LEN;
else {
GST_ERROR ("encryption algorithm '%s' not supported", cipher);
return 0;
}
}
static guint8
auth_key_length_from_auth_name (const gchar * auth)
{
if (g_strcmp0 (auth, "hmac-sha1-32") == 0)
return HMAC_32_KEY_LEN;
else if (g_strcmp0 (auth, "hmac-sha1-80") == 0)
return HMAC_80_KEY_LEN;
else {
GST_ERROR ("authentication algorithm '%s' not supported", auth);
return 0;
}
}
static void
make_media (GstSDPMessage * sdp, GstSDPInfo * info, GstRTSPMedia * media,
GstRTSPStream * stream, GstStructure * s, GstRTSPProfile profile)
{
GstSDPMedia *smedia;
const gchar *caps_str, *caps_enc, *caps_params;
gchar *tmp;
gint caps_pt, caps_rate;
guint n_fields, j;
gboolean first;
GString *fmtp;
GstRTSPLowerTrans ltrans;
GSocketFamily family;
const gchar *addrtype, *proto;
gchar *address;
guint ttl;
GstClockTime rtx_time;
gst_sdp_media_new (&smedia);
/* get media type and payload for the m= line */
caps_str = gst_structure_get_string (s, "media");
gst_sdp_media_set_media (smedia, caps_str);
gst_structure_get_int (s, "payload", &caps_pt);
tmp = g_strdup_printf ("%d", caps_pt);
gst_sdp_media_add_format (smedia, tmp);
g_free (tmp);
gst_sdp_media_set_port_info (smedia, 0, 1);
switch (profile) {
case GST_RTSP_PROFILE_AVP:
proto = "RTP/AVP";
break;
case GST_RTSP_PROFILE_AVPF:
proto = "RTP/AVPF";
break;
case GST_RTSP_PROFILE_SAVP:
proto = "RTP/SAVP";
break;
case GST_RTSP_PROFILE_SAVPF:
proto = "RTP/SAVPF";
break;
default:
proto = "udp";
break;
}
gst_sdp_media_set_proto (smedia, proto);
if (info->is_ipv6) {
addrtype = "IP6";
family = G_SOCKET_FAMILY_IPV6;
} else {
addrtype = "IP4";
family = G_SOCKET_FAMILY_IPV4;
}
ltrans = gst_rtsp_stream_get_protocols (stream);
if (ltrans == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
GstRTSPAddress *addr;
addr = gst_rtsp_stream_get_multicast_address (stream, family);
if (addr == NULL)
goto no_multicast;
address = g_strdup (addr->address);
ttl = addr->ttl;
gst_rtsp_address_free (addr);
} else {
ttl = 16;
if (info->is_ipv6)
address = g_strdup ("::");
else
address = g_strdup ("0.0.0.0");
}
/* for the c= line */
gst_sdp_media_add_connection (smedia, "IN", addrtype, address, ttl, 1);
g_free (address);
/* get clock-rate, media type and params for the rtpmap attribute */
gst_structure_get_int (s, "clock-rate", &caps_rate);
caps_enc = gst_structure_get_string (s, "encoding-name");
caps_params = gst_structure_get_string (s, "encoding-params");
if (caps_enc) {
if (caps_params)
tmp = g_strdup_printf ("%d %s/%d/%s", caps_pt, caps_enc, caps_rate,
caps_params);
else
tmp = g_strdup_printf ("%d %s/%d", caps_pt, caps_enc, caps_rate);
gst_sdp_media_add_attribute (smedia, "rtpmap", tmp);
g_free (tmp);
}
/* the config uri */
tmp = gst_rtsp_stream_get_control (stream);
gst_sdp_media_add_attribute (smedia, "control", tmp);
g_free (tmp);
/* check for srtp */
do {
GstBuffer *srtpkey;
const GValue *val;
const gchar *srtpcipher, *srtpauth, *srtcpcipher, *srtcpauth;
GstMIKEYMessage *msg;
GstMIKEYPayload *payload, *pkd;
GBytes *bytes;
GstMapInfo info;
const guint8 *data;
gsize size;
gchar *base64;
guint8 byte;
guint32 ssrc;
val = gst_structure_get_value (s, "srtp-key");
if (val == NULL)
break;
srtpkey = gst_value_get_buffer (val);
if (srtpkey == NULL)
break;
srtpcipher = gst_structure_get_string (s, "srtp-cipher");
srtpauth = gst_structure_get_string (s, "srtp-auth");
srtcpcipher = gst_structure_get_string (s, "srtcp-cipher");
srtcpauth = gst_structure_get_string (s, "srtcp-auth");
if (srtpcipher == NULL || srtpauth == NULL || srtcpcipher == NULL ||
srtcpauth == NULL)
break;
msg = gst_mikey_message_new ();
/* unencrypted MIKEY message, we send this over TLS so this is allowed */
gst_mikey_message_set_info (msg, GST_MIKEY_VERSION, GST_MIKEY_TYPE_PSK_INIT,
FALSE, GST_MIKEY_PRF_MIKEY_1, 0, GST_MIKEY_MAP_TYPE_SRTP);
/* add policy '0' for our SSRC */
gst_rtsp_stream_get_ssrc (stream, &ssrc);
gst_mikey_message_add_cs_srtp (msg, 0, ssrc, 0);
/* timestamp is now */
gst_mikey_message_add_t_now_ntp_utc (msg);
/* add some random data */
gst_mikey_message_add_rand_len (msg, 16);
/* the policy '0' is SRTP with the above discovered algorithms */
payload = gst_mikey_payload_new (GST_MIKEY_PT_SP);
gst_mikey_payload_sp_set (payload, 0, GST_MIKEY_SEC_PROTO_SRTP);
/* only AES-CM is supported */
byte = 1;
gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_ENC_ALG, 1,
&byte);
/* Encryption key length */
byte = enc_key_length_from_cipher_name (srtpcipher);
gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_ENC_KEY_LEN, 1,
&byte);
/* only HMAC-SHA1 */
gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_AUTH_ALG, 1,
&byte);
/* Authentication key length */
byte = auth_key_length_from_auth_name (srtpauth);
gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_AUTH_KEY_LEN, 1,
&byte);
/* we enable encryption on RTP and RTCP */
gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTP_ENC, 1,
&byte);
gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTCP_ENC, 1,
&byte);
/* we enable authentication on RTP and RTCP */
gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTP_AUTH, 1,
&byte);
gst_mikey_message_add_payload (msg, payload);
/* make unencrypted KEMAC */
payload = gst_mikey_payload_new (GST_MIKEY_PT_KEMAC);
gst_mikey_payload_kemac_set (payload, GST_MIKEY_ENC_NULL,
GST_MIKEY_MAC_NULL);
/* add the key in key data */
pkd = gst_mikey_payload_new (GST_MIKEY_PT_KEY_DATA);
gst_buffer_map (srtpkey, &info, GST_MAP_READ);
gst_mikey_payload_key_data_set_key (pkd, GST_MIKEY_KD_TEK, info.size,
info.data);
gst_buffer_unmap (srtpkey, &info);
/* add key data to KEMAC */
gst_mikey_payload_kemac_add_sub (payload, pkd);
gst_mikey_message_add_payload (msg, payload);
/* now serialize this to bytes */
bytes = gst_mikey_message_to_bytes (msg, NULL, NULL);
gst_mikey_message_unref (msg);
/* and make it into base64 */
data = g_bytes_get_data (bytes, &size);
base64 = g_base64_encode (data, size);
g_bytes_unref (bytes);
tmp = g_strdup_printf ("mikey %s", base64);
g_free (base64);
gst_sdp_media_add_attribute (smedia, "key-mgmt", tmp);
g_free (tmp);
} while (FALSE);
/* collect all other properties and add them to fmtp or attributes */
fmtp = g_string_new ("");
g_string_append_printf (fmtp, "%d ", caps_pt);
first = TRUE;
n_fields = gst_structure_n_fields (s);
for (j = 0; j < n_fields; j++) {
const gchar *fname, *fval;
fname = gst_structure_nth_field_name (s, j);
/* filter out standard properties */
if (!strcmp (fname, "media"))
continue;
if (!strcmp (fname, "payload"))
continue;
if (!strcmp (fname, "clock-rate"))
continue;
if (!strcmp (fname, "encoding-name"))
continue;
if (!strcmp (fname, "encoding-params"))
continue;
if (!strcmp (fname, "ssrc"))
continue;
if (!strcmp (fname, "timestamp-offset"))
continue;
if (!strcmp (fname, "seqnum-offset"))
continue;
if (g_str_has_prefix (fname, "srtp-"))
continue;
if (g_str_has_prefix (fname, "srtcp-"))
continue;
/* handled later */
if (g_str_has_prefix (fname, "x-gst-rtsp-server-rtx-time"))
continue;
if (!strcmp (fname, "a-framesize")) {
/* a-framesize attribute */
if ((fval = gst_structure_get_string (s, fname))) {
tmp = g_strdup_printf ("%d %s", caps_pt, fval);
gst_sdp_media_add_attribute (smedia, fname + 2, tmp);
g_free (tmp);
}
continue;
}
if (g_str_has_prefix (fname, "a-")) {
/* attribute */
if ((fval = gst_structure_get_string (s, fname)))
gst_sdp_media_add_attribute (smedia, fname + 2, fval);
continue;
}
if (g_str_has_prefix (fname, "x-")) {
/* attribute */
if ((fval = gst_structure_get_string (s, fname)))
gst_sdp_media_add_attribute (smedia, fname, fval);
continue;
}
if ((fval = gst_structure_get_string (s, fname))) {
g_string_append_printf (fmtp, "%s%s=%s", first ? "" : ";", fname, fval);
first = FALSE;
}
}
if (!first) {
tmp = g_string_free (fmtp, FALSE);
gst_sdp_media_add_attribute (smedia, "fmtp", tmp);
g_free (tmp);
} else {
g_string_free (fmtp, TRUE);
}
update_sdp_from_tags (stream, smedia);
if ((profile == GST_RTSP_PROFILE_AVPF || profile == GST_RTSP_PROFILE_SAVPF)
&& (rtx_time = gst_rtsp_stream_get_retransmission_time (stream))) {
/* ssrc multiplexed retransmit functionality */
guint rtx_pt = gst_rtsp_stream_get_retransmission_pt (stream);
if (rtx_pt == 0) {
g_warning ("failed to find an available dynamic payload type. "
"Not adding retransmission");
} else {
gchar *tmp;
tmp = g_strdup_printf ("%d", rtx_pt);
gst_sdp_media_add_format (smedia, tmp);
g_free (tmp);
tmp = g_strdup_printf ("%d rtx/%d", rtx_pt, caps_rate);
gst_sdp_media_add_attribute (smedia, "rtpmap", tmp);
g_free (tmp);
tmp =
g_strdup_printf ("%d apt=%d;rtx-time=%" G_GINT64_FORMAT, rtx_pt,
caps_pt, GST_TIME_AS_MSECONDS (rtx_time));
gst_sdp_media_add_attribute (smedia, "fmtp", tmp);
g_free (tmp);
}
}
gst_sdp_message_add_media (sdp, smedia);
gst_sdp_media_free (smedia);
return;
/* ERRORS */
no_multicast:
{
gst_sdp_media_free (smedia);
g_warning ("ignoring stream %d without multicast address",
gst_rtsp_stream_get_index (stream));
return;
}
}
/**
* gst_rtsp_sdp_from_media:
* @sdp: a #GstSDPMessage
* @info: (transfer none): a #GstSDPInfo
* @media: (transfer none): a #GstRTSPMedia
*
* Add @media specific info to @sdp. @info is used to configure the connection
* information in the SDP.
*
* Returns: TRUE on success.
*/
gboolean
gst_rtsp_sdp_from_media (GstSDPMessage * sdp, GstSDPInfo * info,
GstRTSPMedia * media)
{
guint i, n_streams;
gchar *rangestr;
n_streams = gst_rtsp_media_n_streams (media);
rangestr = gst_rtsp_media_get_range_string (media, FALSE, GST_RTSP_RANGE_NPT);
if (rangestr == NULL)
goto not_prepared;
gst_sdp_message_add_attribute (sdp, "range", rangestr);
g_free (rangestr);
for (i = 0; i < n_streams; i++) {
GstRTSPStream *stream;
GstCaps *caps;
GstStructure *s;
GstRTSPProfile profiles;
guint mask;
stream = gst_rtsp_media_get_stream (media, i);
caps = gst_rtsp_stream_get_caps (stream);
if (caps == NULL) {
g_warning ("ignoring stream %d without media type", i);
continue;
}
s = gst_caps_get_structure (caps, 0);
if (s == NULL) {
gst_caps_unref (caps);
g_warning ("ignoring stream %d without media type", i);
continue;
}
/* make a new media for each profile */
profiles = gst_rtsp_stream_get_profiles (stream);
mask = 1;
while (profiles >= mask) {
GstRTSPProfile prof = profiles & mask;
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;
}
}