[vlc-devel] commit: RTP sout: rudimentary SRTP support ( Rémi Denis-Courmont )
git version control
git at videolan.org
Sun Jun 15 21:05:00 CEST 2008
vlc | branch: master | Rémi Denis-Courmont <rdenis at simphalempin.com> | Sun Jun 15 21:48:48 2008 +0300| [72fc2d9c9d87002fb29a508271759397157ad99f]
RTP sout: rudimentary SRTP support
> http://git.videolan.org/gitweb.cgi/vlc.git/?a=commit;h=72fc2d9c9d87002fb29a508271759397157ad99f
---
modules/stream_out/Modules.am | 9 ++++-
modules/stream_out/rtp.c | 78 +++++++++++++++++++++++++++++++++++++----
2 files changed, 79 insertions(+), 8 deletions(-)
diff --git a/modules/stream_out/Modules.am b/modules/stream_out/Modules.am
index 961f7e8..f01cc1c 100644
--- a/modules/stream_out/Modules.am
+++ b/modules/stream_out/Modules.am
@@ -7,7 +7,6 @@ SOURCES_stream_out_duplicate = duplicate.c
SOURCES_stream_out_es = es.c
SOURCES_stream_out_display = display.c
SOURCES_stream_out_gather = gather.c
-SOURCES_stream_out_rtp = rtp.h rtp.c rtpfmt.c rtcp.c rtsp.c
SOURCES_stream_out_switcher = switcher.c
SOURCES_stream_out_bridge = bridge.c
SOURCES_stream_out_mosaic_bridge = mosaic_bridge.c
@@ -27,3 +26,11 @@ libvlc_LTLIBRARIES += \
libstream_out_mosaic_bridge_plugin.la \
libstream_out_autodel_plugin.la \
$(NULL)
+
+# RTP plugin
+libstream_out_rtp_plugin_la_SOURCES = \
+ rtp.c rtp.h rtpfmt.c rtcp.c rtsp.c
+libstream_out_rtp_plugin_la_CFLAGS = $(AM_CFLAGS) -I$(top_srcdir)/libs/srtp
+libstream_out_rtp_plugin_la_LIBADD = $(AM_LIBADD) \
+ $(top_builddir)/libs/srtp/libvlc_srtp.la
+libstream_out_rtp_plugin_la_DEPENDENCIES =
diff --git a/modules/stream_out/rtp.c b/modules/stream_out/rtp.c
index fe9a8f1..c74f659 100644
--- a/modules/stream_out/rtp.c
+++ b/modules/stream_out/rtp.c
@@ -2,8 +2,7 @@
* rtp.c: rtp stream output module
*****************************************************************************
* Copyright (C) 2003-2004 the VideoLAN team
- * Copyright © 2007 Rémi Denis-Courmont
- * $Id$
+ * Copyright © 2007-2008 Rémi Denis-Courmont
*
* Authors: Laurent Aimar <fenrir at via.ecp.fr>
*
@@ -40,6 +39,7 @@
#include <vlc_network.h>
#include <vlc_charset.h>
#include <vlc_strings.h>
+#include <srtp.h>
#include "rtp.h"
@@ -130,6 +130,15 @@
#define PROTO_LONGTEXT N_( \
"This selects which transport protocol to use for RTP." )
+#define SRTP_KEY_TEXT N_("SRTP key (hexadecimal)")
+#define SRTP_KEY_LONGTEXT N_( \
+ "RTP packets will be integrity-protected and ciphered "\
+ "with this Secure RTP master shared secret key.")
+
+#define SRTP_SALT_TEXT N_("SRTP salt (hexadecimal)")
+#define SRTP_SALT_LONGTEXT N_( \
+ "Secure RTP requires a (non-secret) master salt value.")
+
static const char *const ppsz_protos[] = {
"dccp", "sctp", "tcp", "udp", "udplite",
};
@@ -192,6 +201,11 @@ vlc_module_begin();
add_bool( SOUT_CFG_PREFIX "rtcp-mux", false, NULL,
RTCP_MUX_TEXT, RTCP_MUX_LONGTEXT, false );
+ add_string( SOUT_CFG_PREFIX "key", "", NULL,
+ SRTP_KEY_TEXT, SRTP_KEY_LONGTEXT, false );
+ add_string( SOUT_CFG_PREFIX "salt", "", NULL,
+ SRTP_SALT_TEXT, SRTP_SALT_LONGTEXT, false );
+
add_bool( SOUT_CFG_PREFIX "mp4a-latm", 0, NULL, RFC3016_TEXT,
RFC3016_LONGTEXT, false );
@@ -204,7 +218,7 @@ vlc_module_end();
static const char *const ppsz_sout_options[] = {
"dst", "name", "port", "port-audio", "port-video", "*sdp", "ttl", "mux",
"sap", "description", "url", "email", "phone",
- "proto", "rtcp-mux",
+ "proto", "rtcp-mux", "key", "salt",
"mp4a-latm", NULL
};
@@ -299,8 +313,9 @@ struct sout_stream_id_t
int i_bitrate;
/* Packetizer specific fields */
+ int i_mtu;
+ srtp_session_t *srtp;
pf_rtp_packetizer_t pf_packetize;
- int i_mtu;
/* Packets sinks */
vlc_mutex_t lock_sink;
@@ -902,13 +917,36 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt )
id->i_bitrate = 0;
}
- id->pf_packetize = NULL;
id->i_mtu = config_GetInt( p_stream, "mtu" );
if( id->i_mtu <= 12 + 16 )
id->i_mtu = 576 - 20 - 8; /* pessimistic */
-
msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu );
+ id->srtp = NULL;
+ id->pf_packetize = NULL;
+
+ char *key = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key");
+ if (key)
+ {
+ id->srtp = srtp_create (SRTP_ENCR_AES_CM, SRTP_AUTH_HMAC_SHA1, 10,
+ SRTP_PRF_AES_CM, SRTP_RCC_MODE1);
+ if (id->srtp == NULL)
+ {
+ free (key);
+ goto error;
+ }
+
+ char *salt = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt");
+ errno = srtp_setkeystring (id->srtp, key, salt ? salt : "");
+ free (salt);
+ free (key);
+ if (errno)
+ {
+ msg_Err (p_stream, "bad SRTP key/salt combination (%m)");
+ goto error;
+ }
+ }
+
vlc_mutex_init( &id->lock_sink );
id->sinkc = 0;
id->sinkv = NULL;
@@ -1251,6 +1289,8 @@ static int Del( sout_stream_t *p_stream, sout_stream_id_t *id )
rtp_del_sink( id, id->sinkv[0].rtp_fd ); /* sink for explicit dst= */
if( id->listen_fd != NULL )
net_ListenClose( id->listen_fd );
+ if( id->srtp != NULL )
+ srtp_destroy( id->srtp );
vlc_mutex_destroy( &id->lock_sink );
@@ -1397,6 +1437,29 @@ static void ThreadSend( vlc_object_t *p_this )
if( out == NULL )
continue; /* Forced wakeup */
+ if( id->srtp )
+ { /* FIXME: this is awfully inefficient */
+ size_t len = out->i_buffer;
+ int val = srtp_send( id->srtp, out->p_buffer, &len,
+ out->i_buffer );
+ if( val == ENOSPC )
+ {
+ out = block_Realloc( out, 0, len );
+ if( out == NULL )
+ continue;
+ val = srtp_send( id->srtp, out->p_buffer, &len,
+ out->i_buffer );
+ }
+ if( val )
+ {
+ errno = val;
+ msg_Dbg( id, "SRTP sending error: %m" );
+ block_Release( out );
+ continue;
+ }
+ out->i_buffer = len;
+ }
+
mtime_t i_date = out->i_dts + i_caching;
ssize_t len = out->i_buffer;
@@ -1408,7 +1471,8 @@ static void ThreadSend( vlc_object_t *p_this )
for( int i = 0; i < id->sinkc; i++ )
{
- SendRTCP( id->sinkv[i].rtcp, out );
+ if( !id->srtp ) /* FIXME: SRTCP support */
+ SendRTCP( id->sinkv[i].rtcp, out );
if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 )
continue;
More information about the vlc-devel
mailing list