Changeset 72fc2d9c9d87002fb29a508271759397157ad99f

Timestamp:

06/15/08 21:06:54 (2 months ago)

Author:

Rémi Denis-Courmont <rdenis@simphalempin.com>

git-committer:

Rémi Denis-Courmont <rdenis@simphalempin.com> 1213556814 +0300

git-parent:

[fa14c0922728246fe6353042ce4ae052fa4f75d8]

git-author:

Rémi Denis-Courmont <rdenis@simphalempin.com> 1213555728 +0300

Message:

RTP sout: rudimentary SRTP support

Files:

• modules/stream_out/Modules.am (modified) (2 diffs)
• modules/stream_out/rtp.c (modified) (10 diffs)

modules/stream_out/Modules.am

@@ -8 +8 @@
-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
@@ -28 +27 @@
-    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 =

modules/stream_out/rtp.c

@@ -3 +3 @@
- *****************************************************************************
- * Copyright (C) 2003-2004 the VideoLAN team
-
-
+
- *
- * Authors: Laurent Aimar <fenrir@via.ecp.fr>
@@ -41 +40 @@
-#include <vlc_charset.h>
-#include <vlc_strings.h>
+#include <srtp.h>
-
-#include "rtp.h"
@@ -131 +131 @@
-    "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",
@@ -193 +202 @@
-              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 );
@@ -205 +219 @@
-    "dst", "name", "port", "port-audio", "port-video", "*sdp", "ttl", "mux",
-    "sap", "description", "url", "email", "phone",
-
+ "key", "salt",
-    "mp4a-latm", NULL
-};
@@ -300 +314 @@
-
-    /* Packetizer specific fields */
+    int                 i_mtu;
+    srtp_session_t     *srtp;
-    pf_rtp_packetizer_t pf_packetize;
-    int          i_mtu;
-
-    /* Packets sinks */
@@ -903 +918 @@
-    }
-
-    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 );
@@ -1252 +1290 @@
-    if( id->listen_fd != NULL )
-        net_ListenClose( id->listen_fd );
+    if( id->srtp != NULL )
+        srtp_destroy( id->srtp );
-
-    vlc_mutex_destroy( &id->lock_sink );
@@ -1398 +1438 @@
-            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;
@@ -1409 +1472 @@
-        for( int i = 0; i < id->sinkc; i++ )
-        {
-
+
+
-
-            if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 )