Add support for SRTP

This commit is contained in:
Marvin W 2021-03-23 15:05:50 +01:00
parent 9fed5ea865
commit b393d41601
No known key found for this signature in database
GPG key ID: 072E9235DB996F2A
11 changed files with 1286 additions and 81 deletions

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@ -10,6 +10,7 @@ SOURCES
"src/cipher.vala"
"src/cipher_converter.vala"
"src/error.vala"
"src/random.vala"
CUSTOM_VAPIS
"${CMAKE_CURRENT_SOURCE_DIR}/vapi/gcrypt.vapi"
PACKAGES

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@ -0,0 +1,5 @@
namespace Crypto {
public static void randomize(uint8[] buffer) {
GCrypt.Random.randomize(buffer);
}
}

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@ -18,18 +18,20 @@ SOURCES
src/video_widget.vala
src/register_plugin.vala
CUSTOM_VAPIS
${CMAKE_BINARY_DIR}/exports/crypto-vala.vapi
${CMAKE_BINARY_DIR}/exports/xmpp-vala.vapi
${CMAKE_BINARY_DIR}/exports/dino.vapi
${CMAKE_BINARY_DIR}/exports/qlite.vapi
${CMAKE_CURRENT_SOURCE_DIR}/src/srtp.vapi
PACKAGES
${RTP_PACKAGES}
OPTIONS
--vapidir=${CMAKE_CURRENT_SOURCE_DIR}/vapi
)
add_definitions(${VALA_CFLAGS} -DG_LOG_DOMAIN="rtp")
add_library(rtp SHARED ${RTP_VALA_C})
target_link_libraries(rtp libdino ${RTP_PACKAGES})
add_definitions(${VALA_CFLAGS} -DG_LOG_DOMAIN="rtp" -I${CMAKE_CURRENT_SOURCE_DIR}/src)
add_library(rtp SHARED ${RTP_VALA_C} src/srtp.c)
target_link_libraries(rtp libdino crypto-vala ${RTP_PACKAGES})
set_target_properties(rtp PROPERTIES PREFIX "")
set_target_properties(rtp PROPERTIES LIBRARY_OUTPUT_DIRECTORY ${CMAKE_BINARY_DIR}/plugins/)

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@ -200,65 +200,23 @@ public class Dino.Plugins.Rtp.Module : JingleRtp.Module {
plugin.close_stream(rtp_stream);
}
// public uint32 get_session_id(string id) {
// return (uint32) id.split("-")[0].to_int();
// }
//
// public string create_feed(string media, bool incoming) {
// init();
// string id = random_uuid();
// if (media == "audio") {
// id = "0-" + id;
// } else {
// id = "1-" + id;
// }
// MediaDevice? device = plugin.get_preferred_device(media, incoming);
// Feed feed;
// if (incoming) {
// if (media == "audio") {
// feed = new IncomingAudioFeed(id, this, device);
// } else if (media == "video") {
// feed = new IncomingVideoFeed(id, this, device);
// } else {
// critical("Incoming feed of media '%s' not supported", media);
// return id;
// }
// } else {
// if (media == "audio") {
// string? matching_incoming_feed_id = null;
// foreach (Feed match in plugin.feeds.values) {
// if (match is IncomingAudioFeed) {
// matching_incoming_feed_id = match.id;
// }
// }
// feed = new OutgoingAudioFeed(id, this, device);
// } else if (media == "video") {
// feed = new OutgoingVideoFeed(id, this, device);
// } else {
// critical("Outgoing feed of media '%s' not supported", media);
// return id;
// }
// }
// plugin.add_feed(id, feed);
// return id;
// }
//
// public void connect_feed(string id, JingleRtp.PayloadType payload, Jingle.DatagramConnection connection) {
// if (!plugin.feeds.has_key(id)) {
// critical("Tried to connect feed with id %s, but no such feed found", id);
// return;
// }
// Feed feed = plugin.feeds[id];
// feed.connect(payload, connection);
// }
//
// public void destroy_feed(string id) {
// if (!plugin.feeds.has_key(id)) {
// critical("Tried to destroy feed with id %s, but no such feed found", id);
// return;
// }
// Feed feed = plugin.feeds[id];
// feed.destroy();
// plugin.feeds.remove(id);
// }
public override JingleRtp.Crypto? generate_local_crypto() {
uint8[] keyAndSalt = new uint8[30];
Crypto.randomize(keyAndSalt);
return JingleRtp.Crypto.create(JingleRtp.Crypto.AES_CM_128_HMAC_SHA1_80, keyAndSalt);
}
public override JingleRtp.Crypto? pick_remote_crypto(Gee.List<JingleRtp.Crypto> cryptos) {
foreach (JingleRtp.Crypto crypto in cryptos) {
if (crypto.is_valid) return crypto;
}
return null;
}
public override JingleRtp.Crypto? pick_local_crypto(JingleRtp.Crypto? remote) {
if (remote == null || !remote.is_valid) return null;
uint8[] keyAndSalt = new uint8[30];
Crypto.randomize(keyAndSalt);
return remote.rekey(keyAndSalt);
}
}

836
plugins/rtp/src/srtp.c Normal file
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@ -0,0 +1,836 @@
/*
* Secure RTP with libgcrypt
* Copyright (C) 2007 Rémi Denis-Courmont
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser 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
*/
/* TODO:
* Useless stuff (because nothing depends on it):
* - non-nul key derivation rate
* - MKI payload
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include <stdint.h>
#include <stddef.h>
#include "srtp.h"
#include <stdbool.h>
#include <stdlib.h>
#include <assert.h>
#include <errno.h>
#include <gcrypt.h>
#ifdef _WIN32
# include <winsock2.h>
#else
# include <netinet/in.h>
#endif
#define debug( ... ) (void)0
typedef struct srtp_proto_t
{
gcry_cipher_hd_t cipher;
gcry_md_hd_t mac;
uint64_t window;
uint32_t salt[4];
} srtp_proto_t;
struct srtp_session_t
{
srtp_proto_t rtp;
srtp_proto_t rtcp;
unsigned flags;
unsigned kdr;
uint32_t rtcp_index;
uint32_t rtp_roc;
uint16_t rtp_seq;
uint16_t rtp_rcc;
uint8_t tag_len;
};
enum
{
SRTP_CRYPT,
SRTP_AUTH,
SRTP_SALT,
SRTCP_CRYPT,
SRTCP_AUTH,
SRTCP_SALT
};
static inline unsigned rcc_mode (const srtp_session_t *s)
{
return (s->flags >> 4) & 3;
}
static void proto_destroy (srtp_proto_t *p)
{
gcry_md_close (p->mac);
gcry_cipher_close (p->cipher);
}
/**
* Releases all resources associated with a Secure RTP session.
*/
void srtp_destroy (srtp_session_t *s)
{
assert (s != NULL);
proto_destroy (&s->rtcp);
proto_destroy (&s->rtp);
free (s);
}
static int proto_create (srtp_proto_t *p, int gcipher, int gmd)
{
if (gcry_cipher_open (&p->cipher, gcipher, GCRY_CIPHER_MODE_CTR, 0) == 0)
{
if (gcry_md_open (&p->mac, gmd, GCRY_MD_FLAG_HMAC) == 0)
return 0;
gcry_cipher_close (p->cipher);
}
return -1;
}
/**
* Allocates a Secure RTP one-way session.
* The same session cannot be used both ways because this would confuse
* internal cryptographic counters; it is however of course feasible to open
* multiple simultaneous sessions with the same master key.
*
* @param encr encryption algorithm number
* @param auth authentication algortihm number
* @param tag_len authentication tag byte length (NOT including RCC)
* @param flags OR'ed optional flags.
*
* @return NULL in case of error
*/
srtp_session_t *
srtp_create (int encr, int auth, unsigned tag_len, int prf, unsigned flags)
{
if ((flags & ~SRTP_FLAGS_MASK))
return NULL;
int cipher, md;
switch (encr)
{
case SRTP_ENCR_NULL:
cipher = GCRY_CIPHER_NONE;
break;
case SRTP_ENCR_AES_CM:
cipher = GCRY_CIPHER_AES;
break;
default:
return NULL;
}
switch (auth)
{
case SRTP_AUTH_NULL:
md = GCRY_MD_NONE;
break;
case SRTP_AUTH_HMAC_SHA1:
md = GCRY_MD_SHA1;
break;
default:
return NULL;
}
if (tag_len > gcry_md_get_algo_dlen (md))
return NULL;
if (prf != SRTP_PRF_AES_CM)
return NULL;
srtp_session_t *s = malloc (sizeof (*s));
if (s == NULL)
return NULL;
memset (s, 0, sizeof (*s));
s->flags = flags;
s->tag_len = tag_len;
s->rtp_rcc = 1; /* Default RCC rate */
if (rcc_mode (s))
{
if (tag_len < 4)
goto error;
}
if (proto_create (&s->rtp, cipher, md) == 0)
{
if (proto_create (&s->rtcp, cipher, md) == 0)
return s;
proto_destroy (&s->rtp);
}
error:
free (s);
return NULL;
}
/**
* Counter Mode encryption/decryption (ctr length = 16 bytes)
* with non-padded (truncated) text
*/
static int
do_ctr_crypt (gcry_cipher_hd_t hd, const void *ctr, uint8_t *data, size_t len)
{
const size_t ctrlen = 16;
div_t d = div (len, ctrlen);
if (gcry_cipher_setctr (hd, ctr, ctrlen)
|| gcry_cipher_encrypt (hd, data, d.quot * ctrlen, NULL, 0))
return -1;
if (d.rem)
{
/* Truncated last block */
uint8_t dummy[ctrlen];
data += d.quot * ctrlen;
memcpy (dummy, data, d.rem);
memset (dummy + d.rem, 0, ctrlen - d.rem);
if (gcry_cipher_encrypt (hd, dummy, ctrlen, data, ctrlen))
return -1;
memcpy (data, dummy, d.rem);
}
return 0;
}
/**
* AES-CM key derivation (saltlen = 14 bytes)
*/
static int
do_derive (gcry_cipher_hd_t prf, const void *salt,
const uint8_t *r, size_t rlen, uint8_t label,
void *out, size_t outlen)
{
uint8_t iv[16];
memcpy (iv, salt, 14);
iv[14] = iv[15] = 0;
assert (rlen < 14);
iv[13 - rlen] ^= label;
for (size_t i = 0; i < rlen; i++)
iv[sizeof (iv) - rlen + i] ^= r[i];
memset (out, 0, outlen);
return do_ctr_crypt (prf, iv, out, outlen);
}
/**
* Sets (or resets) the master key and master salt for a SRTP session.
* This must be done at least once before using srtp_send(), srtp_recv(),
* srtcp_send() or srtcp_recv(). Also, rekeying is required every
* 2^48 RTP packets or 2^31 RTCP packets (whichever comes first),
* otherwise the protocol security might be broken.
*
* @return 0 on success, in case of error:
* EINVAL invalid or unsupported key/salt sizes combination
*/
int
srtp_setkey (srtp_session_t *s, const void *key, size_t keylen,
const void *salt, size_t saltlen)
{
/* SRTP/SRTCP cipher/salt/MAC keys derivation */
gcry_cipher_hd_t prf;
uint8_t r[6], keybuf[20];
if (saltlen != 14)
return EINVAL;
if (gcry_cipher_open (&prf, GCRY_CIPHER_AES, GCRY_CIPHER_MODE_CTR, 0)
|| gcry_cipher_setkey (prf, key, keylen))
return EINVAL;
/* SRTP key derivation */
#if 0
if (s->kdr != 0)
{
uint64_t index = (((uint64_t)s->rtp_roc) << 16) | s->rtp_seq;
index /= s->kdr;
for (int i = sizeof (r) - 1; i >= 0; i--)
{
r[i] = index & 0xff;
index = index >> 8;
}
}
else
#endif
memset (r, 0, sizeof (r));
if (do_derive (prf, salt, r, 6, SRTP_CRYPT, keybuf, 16)
|| gcry_cipher_setkey (s->rtp.cipher, keybuf, 16)
|| do_derive (prf, salt, r, 6, SRTP_AUTH, keybuf, 20)
|| gcry_md_setkey (s->rtp.mac, keybuf, 20)
|| do_derive (prf, salt, r, 6, SRTP_SALT, s->rtp.salt, 14))
return -1;
/* SRTCP key derivation */
memcpy (r, &(uint32_t){ htonl (s->rtcp_index) }, 4);
if (do_derive (prf, salt, r, 4, SRTCP_CRYPT, keybuf, 16)
|| gcry_cipher_setkey (s->rtcp.cipher, keybuf, 16)
|| do_derive (prf, salt, r, 4, SRTCP_AUTH, keybuf, 20)
|| gcry_md_setkey (s->rtcp.mac, keybuf, 20)
|| do_derive (prf, salt, r, 4, SRTCP_SALT, s->rtcp.salt, 14))
return -1;
(void)gcry_cipher_close (prf);
return 0;
}
static int hexdigit (char c)
{
if ((c >= '0') && (c <= '9'))
return c - '0';
if ((c >= 'A') && (c <= 'F'))
return c - 'A' + 0xA;
if ((c >= 'a') && (c <= 'f'))
return c - 'a' + 0xa;
return -1;
}
static ssize_t hexstring (const char *in, uint8_t *out, size_t outlen)
{
size_t inlen = strlen (in);
if ((inlen > (2 * outlen)) || (inlen & 1))
return -1;
for (size_t i = 0; i < inlen; i += 2)
{
int a = hexdigit (in[i]), b = hexdigit (in[i + 1]);
if ((a == -1) || (b == -1))
return -1;
out[i / 2] = (a << 4) | b;
}
return inlen / 2;
}
/**
* Sets (or resets) the master key and master salt for a SRTP session
* from hexadecimal strings. See also srtp_setkey().
*
* @return 0 on success, in case of error:
* EINVAL invalid or unsupported key/salt sizes combination
*/
int
srtp_setkeystring (srtp_session_t *s, const char *key, const char *salt)
{
uint8_t bkey[16]; /* TODO/NOTE: hard-coded for AES */
uint8_t bsalt[14]; /* TODO/NOTE: hard-coded for the PRF-AES-CM */
ssize_t bkeylen = hexstring (key, bkey, sizeof (bkey));
ssize_t bsaltlen = hexstring (salt, bsalt, sizeof (bsalt));
if ((bkeylen == -1) || (bsaltlen == -1))
return EINVAL;
return srtp_setkey (s, bkey, bkeylen, bsalt, bsaltlen) ? EINVAL : 0;
}
/**
* Sets Roll-over-Counter Carry (RCC) rate for the SRTP session. If not
* specified (through this function), the default rate of ONE is assumed
* (i.e. every RTP packets will carry the RoC). RCC rate is ignored if none
* of the RCC mode has been selected.
*
* The RCC mode is selected through one of these flags for srtp_create():
* SRTP_RCC_MODE1: integrity protection only for RoC carrying packets
* SRTP_RCC_MODE2: integrity protection for all packets
* SRTP_RCC_MODE3: no integrity protection
*
* RCC mode 3 is insecure. Compared to plain RTP, it provides confidentiality
* (through encryption) but is much more prone to DoS. It can only be used if
* anti-spoofing protection is provided by lower network layers (e.g. IPsec,
* or trusted routers and proper source address filtering).
*
* If RCC rate is 1, RCC mode 1 and 2 are functionally identical.
*
* @param rate RoC Carry rate (MUST NOT be zero)
*/
void srtp_setrcc_rate (srtp_session_t *s, uint16_t rate)
{
assert (rate != 0);
s->rtp_rcc = rate;
}
/** AES-CM for RTP (salt = 14 bytes + 2 nul bytes) */
static int
rtp_crypt (gcry_cipher_hd_t hd, uint32_t ssrc, uint32_t roc, uint16_t seq,
const uint32_t *salt, uint8_t *data, size_t len)
{
/* Determines cryptographic counter (IV) */
uint32_t counter[4];
counter[0] = salt[0];
counter[1] = salt[1] ^ ssrc;
counter[2] = salt[2] ^ htonl (roc);
counter[3] = salt[3] ^ htonl (seq << 16);
/* Encryption */
return do_ctr_crypt (hd, counter, data, len);
}
/** Determines SRTP Roll-Over-Counter (in host-byte order) */
static uint32_t
srtp_compute_roc (const srtp_session_t *s, uint16_t seq)
{
uint32_t roc = s->rtp_roc;
if (((seq - s->rtp_seq) & 0xffff) < 0x8000)
{
/* Sequence is ahead, good */
if (seq < s->rtp_seq)
roc++; /* Sequence number wrap */
}
else
{
/* Sequence is late, bad */
if (seq > s->rtp_seq)
roc--; /* Wrap back */
}
return roc;
}
/** Returns RTP sequence (in host-byte order) */
static inline uint16_t rtp_seq (const uint8_t *buf)
{
return (buf[2] << 8) | buf[3];
}
/** Message Authentication and Integrity for RTP */
static const uint8_t *
rtp_digest (gcry_md_hd_t md, const uint8_t *data, size_t len,
uint32_t roc)
{
gcry_md_reset (md);
gcry_md_write (md, data, len);
gcry_md_write (md, &(uint32_t){ htonl (roc) }, 4);
return gcry_md_read (md, 0);
}
/**
* Encrypts/decrypts a RTP packet and updates SRTP context
* (CTR block cypher mode of operation has identical encryption and
* decryption function).
*
* @param buf RTP packet to be en-/decrypted
* @param len RTP packet length
*
* @return 0 on success, in case of error:
* EINVAL malformatted RTP packet
* EACCES replayed packet or out-of-window or sync lost
*/
static int srtp_crypt (srtp_session_t *s, uint8_t *buf, size_t len)
{
assert (s != NULL);
assert (len >= 12u);
if ((buf[0] >> 6) != 2)
return EINVAL;
/* Computes encryption offset */
uint16_t offset = 12;
offset += (buf[0] & 0xf) * 4; // skips CSRC
if (buf[0] & 0x10)
{
uint16_t extlen;
offset += 4;
if (len < offset)
return EINVAL;
memcpy (&extlen, buf + offset - 2, 2);
offset += htons (extlen); // skips RTP extension header
}
if (len < offset)
return EINVAL;
/* Determines RTP 48-bits counter and SSRC */
uint16_t seq = rtp_seq (buf);
uint32_t roc = srtp_compute_roc (s, seq), ssrc;
memcpy (&ssrc, buf + 8, 4);
/* Updates ROC and sequence (it's safe now) */
int16_t diff = seq - s->rtp_seq;
if (diff > 0)
{
/* Sequence in the future, good */
s->rtp.window = s->rtp.window << diff;
s->rtp.window |= UINT64_C(1);
s->rtp_seq = seq, s->rtp_roc = roc;
}
else
{
/* Sequence in the past/present, bad */
diff = -diff;
if ((diff >= 64) || ((s->rtp.window >> diff) & 1))
return EACCES; /* Replay attack */
s->rtp.window |= UINT64_C(1) << diff;
}
/* Encrypt/Decrypt */
if (s->flags & SRTP_UNENCRYPTED)
return 0;
if (rtp_crypt (s->rtp.cipher, ssrc, roc, seq, s->rtp.salt,
buf + offset, len - offset))
return EINVAL;
return 0;
}
/**
* Turns a RTP packet into a SRTP packet: encrypt it, then computes
* the authentication tag and appends it.
* Note that you can encrypt packet in disorder.
*
* @param buf RTP packet to be encrypted/digested
* @param lenp pointer to the RTP packet length on entry,
* set to the SRTP length on exit (undefined on non-ENOSPC error)
* @param bufsize size (bytes) of the packet buffer
*
* @return 0 on success, in case of error:
* EINVAL malformatted RTP packet or internal error
* ENOSPC bufsize is too small to add authentication tag
* (<lenp> will hold the required byte size)
* EACCES packet would trigger a replay error on receiver
*/
int
srtp_send (srtp_session_t *s, uint8_t *buf, size_t *lenp, size_t bufsize)
{
size_t len = *lenp;
size_t tag_len;
size_t roc_len = 0;
/* Compute required buffer size */
if (len < 12u)
return EINVAL;
if (!(s->flags & SRTP_UNAUTHENTICATED))
{
tag_len = s->tag_len;
if (rcc_mode (s))
{
assert (tag_len >= 4);
assert (s->rtp_rcc != 0);
if ((rtp_seq (buf) % s->rtp_rcc) == 0)
{
roc_len = 4;
if (rcc_mode (s) == 3)
tag_len = 0; /* RCC mode 3 -> no auth*/
else
tag_len -= 4; /* RCC mode 1 or 2 -> auth*/
}
else
{
if (rcc_mode (s) & 1)
tag_len = 0; /* RCC mode 1 or 3 -> no auth */
}
}
*lenp = len + roc_len + tag_len;
}
else
tag_len = 0;
if (bufsize < *lenp)
return ENOSPC;
/* Encrypt payload */
int val = srtp_crypt (s, buf, len);
if (val)
return val;
/* Authenticate payload */
if (!(s->flags & SRTP_UNAUTHENTICATED))
{
uint32_t roc = srtp_compute_roc (s, rtp_seq (buf));
const uint8_t *tag = rtp_digest (s->rtp.mac, buf, len, roc);
if (roc_len)
{
memcpy (buf + len, &(uint32_t){ htonl (s->rtp_roc) }, 4);
len += 4;
}
memcpy (buf + len, tag, tag_len);
#if 0
printf ("Sent : 0x");
for (unsigned i = 0; i < tag_len; i++)
printf ("%02x", tag[i]);
puts ("");
#endif
}
return 0;
}
/**
* Turns a SRTP packet into a RTP packet: authenticates the packet,
* then decrypts it.
*
* @param buf RTP packet to be digested/decrypted
* @param lenp pointer to the SRTP packet length on entry,
* set to the RTP length on exit (undefined in case of error)
*
* @return 0 on success, in case of error:
* EINVAL malformatted SRTP packet
* EACCES authentication failed (spoofed packet or out-of-sync)
*/
int
srtp_recv (srtp_session_t *s, uint8_t *buf, size_t *lenp)
{
size_t len = *lenp;
if (len < 12u)
return EINVAL;
if (!(s->flags & SRTP_UNAUTHENTICATED))
{
size_t tag_len = s->tag_len, roc_len = 0;
if (rcc_mode (s))
{
if ((rtp_seq (buf) % s->rtp_rcc) == 0)
{
roc_len = 4;
if (rcc_mode (s) == 3)
tag_len = 0;
else
tag_len -= 4;
}
else
{
if (rcc_mode (s) & 1)
tag_len = 0; // RCC mode 1 or 3: no auth
}
}
if (len < (12u + roc_len + tag_len))
return EINVAL;
len -= roc_len + tag_len;
uint32_t roc = srtp_compute_roc (s, rtp_seq (buf)), rcc;
if (roc_len)
{
assert (roc_len == 4);
memcpy (&rcc, buf + len, 4);
rcc = ntohl (rcc);
}
else
rcc = roc;
const uint8_t *tag = rtp_digest (s->rtp.mac, buf, len, rcc);
#if 0
printf ("Computed: 0x");
for (unsigned i = 0; i < tag_len; i++)
printf ("%02x", tag[i]);
printf ("\nReceived: 0x");
for (unsigned i = 0; i < tag_len; i++)
printf ("%02x", buf[len + roc_len + i]);
puts ("");
#endif
if (memcmp (buf + len + roc_len, tag, tag_len))
return EACCES;
if (roc_len)
{
/* Authenticated packet carried a Roll-Over-Counter */
s->rtp_roc += rcc - roc;
assert (srtp_compute_roc (s, rtp_seq (buf)) == rcc);
}
*lenp = len;
}
return srtp_crypt (s, buf, len);
}
/** AES-CM for RTCP (salt = 14 bytes + 2 nul bytes) */
static int
rtcp_crypt (gcry_cipher_hd_t hd, uint32_t ssrc, uint32_t index,
const uint32_t *salt, uint8_t *data, size_t len)
{
return rtp_crypt (hd, ssrc, index >> 16, index & 0xffff, salt, data, len);
}
/** Message Authentication and Integrity for RTCP */
static const uint8_t *
rtcp_digest (gcry_md_hd_t md, const void *data, size_t len)
{
gcry_md_reset (md);
gcry_md_write (md, data, len);
return gcry_md_read (md, 0);
}
/**
* Encrypts/decrypts a RTCP packet and updates SRTCP context
* (CTR block cypher mode of operation has identical encryption and
* decryption function).
*
* @param buf RTCP packet to be en-/decrypted
* @param len RTCP packet length
*
* @return 0 on success, in case of error:
* EINVAL malformatted RTCP packet
*/
static int srtcp_crypt (srtp_session_t *s, uint8_t *buf, size_t len)
{
assert (s != NULL);
/* 8-bytes unencrypted header, and 4-bytes unencrypted footer */
if ((len < 12) || ((buf[0] >> 6) != 2))
return EINVAL;
uint32_t index;
memcpy (&index, buf + len, 4);
index = ntohl (index);
if (((index >> 31) != 0) != ((s->flags & SRTCP_UNENCRYPTED) == 0))
return EINVAL; // E-bit mismatch
index &= ~(1 << 31); // clear E-bit for counter
/* Updates SRTCP index (safe here) */
int32_t diff = index - s->rtcp_index;
if (diff > 0)
{
/* Packet in the future, good */
s->rtcp.window = s->rtcp.window << diff;
s->rtcp.window |= UINT64_C(1);
s->rtcp_index = index;
}
else
{
/* Packet in the past/present, bad */
diff = -diff;
if ((diff >= 64) || ((s->rtcp.window >> diff) & 1))
return EACCES; // replay attack!
s->rtp.window |= UINT64_C(1) << diff;
}
/* Crypts SRTCP */
if (s->flags & SRTCP_UNENCRYPTED)
return 0;
uint32_t ssrc;
memcpy (&ssrc, buf + 4, 4);
if (rtcp_crypt (s->rtcp.cipher, ssrc, index, s->rtp.salt,
buf + 8, len - 8))
return EINVAL;
return 0;
}
/**
* Turns a RTCP packet into a SRTCP packet: encrypt it, then computes
* the authentication tag and appends it.
*
* @param buf RTCP packet to be encrypted/digested
* @param lenp pointer to the RTCP packet length on entry,
* set to the SRTCP length on exit (undefined in case of error)
* @param bufsize size (bytes) of the packet buffer
*
* @return 0 on success, in case of error:
* EINVAL malformatted RTCP packet or internal error
* ENOSPC bufsize is too small (to add index and authentication tag)
*/
int
srtcp_send (srtp_session_t *s, uint8_t *buf, size_t *lenp, size_t bufsize)
{
size_t len = *lenp;
if (bufsize < (len + 4 + s->tag_len))
return ENOSPC;
uint32_t index = ++s->rtcp_index;
if (index >> 31)
s->rtcp_index = index = 0; /* 31-bit wrap */
if ((s->flags & SRTCP_UNENCRYPTED) == 0)
index |= 0x80000000; /* Set Encrypted bit */
memcpy (buf + len, &(uint32_t){ htonl (index) }, 4);
int val = srtcp_crypt (s, buf, len);
if (val)
return val;
len += 4; /* Digests SRTCP index too */
const uint8_t *tag = rtcp_digest (s->rtcp.mac, buf, len);
memcpy (buf + len, tag, s->tag_len);
*lenp = len + s->tag_len;
return 0;
}
/**
* Turns a SRTCP packet into a RTCP packet: authenticates the packet,
* then decrypts it.
*
* @param buf RTCP packet to be digested/decrypted
* @param lenp pointer to the SRTCP packet length on entry,
* set to the RTCP length on exit (undefined in case of error)
*
* @return 0 on success, in case of error:
* EINVAL malformatted SRTCP packet
* EACCES authentication failed (spoofed packet or out-of-sync)
*/
int
srtcp_recv (srtp_session_t *s, uint8_t *buf, size_t *lenp)
{
size_t len = *lenp;
if (len < (4u + s->tag_len))
return EINVAL;
len -= s->tag_len;
const uint8_t *tag = rtcp_digest (s->rtcp.mac, buf, len);
if (memcmp (buf + len, tag, s->tag_len))
return EACCES;
len -= 4; /* Remove SRTCP index before decryption */
*lenp = len;
return srtcp_crypt (s, buf, len);
}

82
plugins/rtp/src/srtp.h Normal file
View file

@ -0,0 +1,82 @@
/*
* Secure RTP with libgcrypt
* Copyright (C) 2007 Rémi Denis-Courmont
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public License
* as published by the Free Software Foundation; either version 2.1
* 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 Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser 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
****************************************************************************/
#ifndef LIBVLC_SRTP_H
# define LIBVLC_SRTP_H 1
#include <stdint.h>
typedef struct srtp_session_t srtp_session_t;
enum
{
SRTP_UNENCRYPTED=0x1, //< do not encrypt SRTP packets
SRTCP_UNENCRYPTED=0x2, //< do not encrypt SRTCP packets
SRTP_UNAUTHENTICATED=0x4, //< authenticate only SRTCP packets
SRTP_RCC_MODE1=0x10, //< use Roll-over-Counter Carry mode 1
SRTP_RCC_MODE2=0x20, //< use Roll-over-Counter Carry mode 2
SRTP_RCC_MODE3=0x30, //< use Roll-over-Counter Carry mode 3 (insecure)
SRTP_FLAGS_MASK=0x37 //< mask for valid flags
};
/** SRTP encryption algorithms (ciphers); same values as MIKEY */
enum
{
SRTP_ENCR_NULL=0, //< no encryption
SRTP_ENCR_AES_CM=1, //< AES counter mode
SRTP_ENCR_AES_F8=2, //< AES F8 mode (not implemented)
};
/** SRTP authenticaton algorithms; same values as MIKEY */
enum
{
SRTP_AUTH_NULL=0, //< no authentication code
SRTP_AUTH_HMAC_SHA1=1, //< HMAC-SHA1
};
/** SRTP pseudo random function; same values as MIKEY */
enum
{
SRTP_PRF_AES_CM=0, //< AES counter mode
};
# ifdef __cplusplus
extern "C" {
# endif
srtp_session_t *srtp_create (int encr, int auth, unsigned tag_len, int prf,
unsigned flags);
void srtp_destroy (srtp_session_t *s);
int srtp_setkey (srtp_session_t *s, const void *key, size_t keylen,
const void *salt, size_t saltlen);
int srtp_setkeystring (srtp_session_t *s, const char *key, const char *salt);
void srtp_setrcc_rate (srtp_session_t *s, uint16_t rate);
int srtp_send (srtp_session_t *s, uint8_t *buf, size_t *lenp, size_t maxsize);
int srtp_recv (srtp_session_t *s, uint8_t *buf, size_t *lenp);
int srtcp_send (srtp_session_t *s, uint8_t *buf, size_t *lenp, size_t maxsiz);
int srtcp_recv (srtp_session_t *s, uint8_t *buf, size_t *lenp);
# ifdef __cplusplus
}
# endif
#endif

103
plugins/rtp/src/srtp.vapi Normal file
View file

@ -0,0 +1,103 @@
[Compact]
[CCode (cname = "srtp_session_t", free_function = "srtp_destroy", cheader_filename="srtp.h")]
public class Dino.Plugins.Rtp.SrtpSession {
[CCode (cname = "srtp_create")]
public SrtpSession(SrtpEncryption encr, SrtpAuthentication auth, uint tag_len, SrtpPrf prf, SrtpFlags flags);
[CCode (cname = "srtp_setkey")]
public int setkey(uint8[] key, uint8[] salt);
[CCode (cname = "srtp_setkeystring")]
public int setkeystring(string key, string salt);
[CCode (cname = "srtp_setrcc_rate")]
public void setrcc_rate(uint16 rate);
[CCode (cname = "srtp_send")]
private int rtp_send([CCode (array_length = false)] uint8[] buf, ref size_t len, size_t maxsize);
[CCode (cname = "srtcp_send")]
private int rtcp_send([CCode (array_length = false)] uint8[] buf, ref size_t len, size_t maxsize);
[CCode (cname = "srtp_recv")]
private int rtp_recv([CCode (array_length = false)] uint8[] buf, ref size_t len);
[CCode (cname = "srtcp_recv")]
private int rtcp_recv([CCode (array_length = false)] uint8[] buf, ref size_t len);
public uint8[] encrypt_rtp(uint8[] input, uint tag_len = 10) throws GLib.Error {
uint8[] buf = new uint8[input.length+tag_len];
GLib.Memory.copy(buf, input, input.length);
size_t buf_use = input.length;
int res = rtp_send(buf, ref buf_use, buf.length);
if (res != 0) {
throw new GLib.Error(-1, res, "RTP encrypt failed");
}
uint8[] ret = new uint8[buf_use];
GLib.Memory.copy(ret, buf, buf_use);
return ret;
}
public uint8[] encrypt_rtcp(uint8[] input, uint tag_len = 10) throws GLib.Error {
uint8[] buf = new uint8[input.length+tag_len+4];
GLib.Memory.copy(buf, input, input.length);
size_t buf_use = input.length;
int res = rtcp_send(buf, ref buf_use, buf.length);
if (res != 0) {
throw new GLib.Error(-1, res, "RTCP encrypt failed");
}
uint8[] ret = new uint8[buf_use];
GLib.Memory.copy(ret, buf, buf_use);
return ret;
}
public uint8[] decrypt_rtp(uint8[] input) throws GLib.Error {
uint8[] buf = new uint8[input.length];
GLib.Memory.copy(buf, input, input.length);
size_t buf_use = input.length;
int res = rtp_recv(buf, ref buf_use);
if (res != 0) {
throw new GLib.Error(-1, res, "RTP decrypt failed");
}
uint8[] ret = new uint8[buf_use];
GLib.Memory.copy(ret, buf, buf_use);
return ret;
}
public uint8[] decrypt_rtcp(uint8[] input) throws GLib.Error {
uint8[] buf = new uint8[input.length];
GLib.Memory.copy(buf, input, input.length);
size_t buf_use = input.length;
int res = rtcp_recv(buf, ref buf_use);
if (res != 0) {
throw new GLib.Error(-1, res, "RTCP decrypt failed");
}
uint8[] ret = new uint8[buf_use];
GLib.Memory.copy(ret, buf, buf_use);
return ret;
}
}
[Flags]
[CCode (cname = "unsigned", cprefix = "", cheader_filename="srtp.h", has_type_id = false)]
public enum Dino.Plugins.Rtp.SrtpFlags {
SRTP_UNENCRYPTED,
SRTCP_UNENCRYPTED,
SRTP_UNAUTHENTICATED,
SRTP_RCC_MODE1,
SRTP_RCC_MODE2,
SRTP_RCC_MODE3
}
[CCode (cname = "int", cprefix = "SRTP_ENCR_", cheader_filename="srtp.h", has_type_id = false)]
public enum Dino.Plugins.Rtp.SrtpEncryption {
NULL,
AES_CM,
AES_F8
}
[CCode (cname = "int", cprefix = "SRTP_AUTH_", cheader_filename="srtp.h", has_type_id = false)]
public enum Dino.Plugins.Rtp.SrtpAuthentication {
NULL,
HMAC_SHA1
}
[CCode (cname = "int", cprefix = "SRTP_PRF_", cheader_filename="srtp.h", has_type_id = false)]
public enum Dino.Plugins.Rtp.SrtpPrf {
AES_CM
}

View file

@ -53,6 +53,9 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
private Gst.Pad send_rtp_sink_pad;
private Gst.Pad send_rtp_src_pad;
private SrtpSession? local_crypto_session;
private SrtpSession? remote_crypto_session;
public Stream(Plugin plugin, Xmpp.Xep.Jingle.Content content) {
base(content);
this.plugin = plugin;
@ -144,6 +147,20 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
plugin.unpause();
}
private void prepare_local_crypto() {
if (local_crypto != null && local_crypto_session == null) {
local_crypto_session = new SrtpSession(
local_crypto.crypto_suite == Xep.JingleRtp.Crypto.F8_128_HMAC_SHA1_80 ? SrtpEncryption.AES_F8 : SrtpEncryption.AES_CM,
SrtpAuthentication.HMAC_SHA1,
local_crypto.crypto_suite == Xep.JingleRtp.Crypto.AES_CM_128_HMAC_SHA1_32 ? 4 : 10,
SrtpPrf.AES_CM,
0
);
local_crypto_session.setkey(local_crypto.key, local_crypto.salt);
debug("Setting up encryption with key params %s", local_crypto.key_params);
}
}
private Gst.FlowReturn on_new_sample(Gst.App.Sink sink) {
if (sink == null) {
debug("Sink is null");
@ -153,9 +170,16 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
Gst.Buffer buffer = sample.get_buffer();
uint8[] data;
buffer.extract_dup(0, buffer.get_size(), out data);
prepare_local_crypto();
if (sink == send_rtp) {
if (local_crypto_session != null) {
data = local_crypto_session.encrypt_rtp(data, local_crypto.crypto_suite == Xep.JingleRtp.Crypto.AES_CM_128_HMAC_SHA1_32 ? 4 : 10);
}
on_send_rtp_data(new Bytes.take(data));
} else if (sink == send_rtcp) {
if (local_crypto_session != null) {
data = local_crypto_session.encrypt_rtcp(data, local_crypto.crypto_suite == Xep.JingleRtp.Crypto.AES_CM_128_HMAC_SHA1_32 ? 4 : 10);
}
on_send_rtcp_data(new Bytes.take(data));
} else {
warning("unknown sample");
@ -258,15 +282,47 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
recv_rtp_src_pad = null;
}
private void prepare_remote_crypto() {
if (remote_crypto != null && remote_crypto_session == null) {
remote_crypto_session = new SrtpSession(
remote_crypto.crypto_suite == Xep.JingleRtp.Crypto.F8_128_HMAC_SHA1_80 ? SrtpEncryption.AES_F8 : SrtpEncryption.AES_CM,
SrtpAuthentication.HMAC_SHA1,
remote_crypto.crypto_suite == Xep.JingleRtp.Crypto.AES_CM_128_HMAC_SHA1_32 ? 4 : 10,
SrtpPrf.AES_CM,
0
);
remote_crypto_session.setkey(remote_crypto.key, remote_crypto.salt);
debug("Setting up decryption with key params %s", remote_crypto.key_params);
}
}
public override void on_recv_rtp_data(Bytes bytes) {
prepare_remote_crypto();
uint8[] data = bytes.get_data();
if (remote_crypto_session != null) {
try {
data = remote_crypto_session.decrypt_rtp(data);
} catch (Error e) {
warning("%s (%d)", e.message, e.code);
}
}
if (push_recv_data) {
recv_rtp.push_buffer(new Gst.Buffer.wrapped_bytes(bytes));
recv_rtp.push_buffer(new Gst.Buffer.wrapped((owned) data));
}
}
public override void on_recv_rtcp_data(Bytes bytes) {
prepare_remote_crypto();
uint8[] data = bytes.get_data();
if (remote_crypto_session != null) {
try {
data = remote_crypto_session.decrypt_rtcp(data);
} catch (Error e) {
warning("%s (%d)", e.message, e.code);
}
}
if (push_recv_data) {
recv_rtcp.push_buffer(new Gst.Buffer.wrapped_bytes(bytes));
recv_rtcp.push_buffer(new Gst.Buffer.wrapped((owned) data));
}
}

View file

@ -17,7 +17,9 @@ public class Xmpp.Xep.JingleRtp.Parameters : Jingle.ContentParameters, Object {
public bool encryption_required { get; private set; default = false; }
public PayloadType? agreed_payload_type { get; private set; }
public Gee.List<PayloadType> payload_types = new ArrayList<PayloadType>(PayloadType.equals_func);
public Gee.List<Crypto> cryptos = new ArrayList<Crypto>();
public Gee.List<Crypto> remote_cryptos = new ArrayList<Crypto>();
public Crypto? local_crypto = null;
public Crypto? remote_crypto = null;
public weak Stream? stream { get; private set; }
@ -27,7 +29,7 @@ public class Xmpp.Xep.JingleRtp.Parameters : Jingle.ContentParameters, Object {
string media, Gee.List<PayloadType> payload_types,
string? ssrc = null, bool rtcp_mux = false,
string? bandwidth = null, string? bandwidth_type = null,
bool encryption_required = false, Gee.List<Crypto> cryptos = new ArrayList<Crypto>()
bool encryption_required = false, Crypto? local_crypto = null
) {
this.parent = parent;
this.media = media;
@ -37,7 +39,7 @@ public class Xmpp.Xep.JingleRtp.Parameters : Jingle.ContentParameters, Object {
this.bandwidth_type = bandwidth_type;
this.encryption_required = encryption_required;
this.payload_types = payload_types;
this.cryptos = cryptos;
this.local_crypto = local_crypto;
}
public Parameters.from_node(Module parent, StanzaNode node) throws Jingle.IqError {
@ -49,7 +51,7 @@ public class Xmpp.Xep.JingleRtp.Parameters : Jingle.ContentParameters, Object {
if (encryption != null) {
this.encryption_required = encryption.get_attribute_bool("required", this.encryption_required);
foreach (StanzaNode crypto in encryption.get_subnodes("crypto")) {
this.cryptos.add(Crypto.parse(crypto));
this.remote_cryptos.add(Crypto.parse(crypto));
}
}
foreach (StanzaNode payloadType in node.get_subnodes("payload-type")) {
@ -64,6 +66,15 @@ public class Xmpp.Xep.JingleRtp.Parameters : Jingle.ContentParameters, Object {
content.reject();
return;
}
remote_crypto = parent.pick_remote_crypto(remote_cryptos);
if (local_crypto == null && remote_crypto != null) {
local_crypto = parent.pick_local_crypto(remote_crypto);
}
if ((local_crypto == null || remote_crypto == null) && encryption_required) {
debug("no usable encryption, but encryption required");
content.reject();
return;
}
}
public void accept(XmppStream stream, Jingle.Session session, Jingle.Content content) {
@ -97,6 +108,15 @@ public class Xmpp.Xep.JingleRtp.Parameters : Jingle.ContentParameters, Object {
}
});
if (remote_crypto == null || local_crypto == null) {
if (encryption_required) {
warning("Encryption required but not provided in both directions");
return;
}
remote_crypto = null;
local_crypto = null;
}
this.stream = parent.create_stream(content);
rtp_datagram.datagram_received.connect(this.stream.on_recv_rtp_data);
rtcp_datagram.datagram_received.connect(this.stream.on_recv_rtcp_data);
@ -118,6 +138,20 @@ public class Xmpp.Xep.JingleRtp.Parameters : Jingle.ContentParameters, Object {
}
agreed_payload_type = preferred_payload_type;
Gee.List<StanzaNode> crypto_nodes = description_node.get_deep_subnodes("encryption", "crypto");
if (crypto_nodes.size == 0) {
warning("Counterpart didn't include any cryptos");
if (encryption_required) {
return;
}
} else {
Crypto preferred_crypto = Crypto.parse(crypto_nodes[0]);
if (local_crypto.crypto_suite != preferred_crypto.crypto_suite) {
warning("Counterpart's crypto suite doesn't match any of our sent ones");
}
remote_crypto = preferred_crypto;
}
accept(stream, session, content);
}
@ -137,6 +171,10 @@ public class Xmpp.Xep.JingleRtp.Parameters : Jingle.ContentParameters, Object {
ret.put_node(payload_type.to_xml());
}
}
if (local_crypto != null) {
ret.put_node(new StanzaNode.build("encryption", NS_URI)
.put_node(local_crypto.to_xml()));
}
return ret;
}
}

View file

@ -20,6 +20,9 @@ public abstract class Module : XmppStreamModule {
public abstract async Gee.List<PayloadType> get_supported_payloads(string media);
public abstract async PayloadType? pick_payload_type(string media, Gee.List<PayloadType> payloads);
public abstract Crypto? generate_local_crypto();
public abstract Crypto? pick_remote_crypto(Gee.List<Crypto> cryptos);
public abstract Crypto? pick_local_crypto(Crypto? remote);
public abstract Stream create_stream(Jingle.Content content);
public abstract void close_stream(Stream stream);
@ -36,6 +39,7 @@ public abstract class Module : XmppStreamModule {
// Create audio content
Parameters audio_content_parameters = new Parameters(this, "audio", yield get_supported_payloads("audio"));
audio_content_parameters.local_crypto = generate_local_crypto();
Jingle.Transport? audio_transport = yield jingle_module.select_transport(stream, content_type.required_transport_type, content_type.required_components, receiver_full_jid, Set.empty());
if (audio_transport == null) {
throw new Jingle.Error.NO_SHARED_PROTOCOLS("No suitable audio transports");
@ -52,6 +56,7 @@ public abstract class Module : XmppStreamModule {
if (video) {
// Create video content
Parameters video_content_parameters = new Parameters(this, "video", yield get_supported_payloads("video"));
video_content_parameters.local_crypto = generate_local_crypto();
Jingle.Transport? video_transport = yield stream.get_module(Jingle.Module.IDENTITY).select_transport(stream, content_type.required_transport_type, content_type.required_components, receiver_full_jid, Set.empty());
if (video_transport == null) {
throw new Jingle.Error.NO_SHARED_PROTOCOLS("No suitable video transports");
@ -92,6 +97,7 @@ public abstract class Module : XmppStreamModule {
if (content == null) {
// Content for video does not yet exist -> create it
Parameters video_content_parameters = new Parameters(this, "video", yield get_supported_payloads("video"));
video_content_parameters.local_crypto = generate_local_crypto();
Jingle.Transport? video_transport = yield stream.get_module(Jingle.Module.IDENTITY).select_transport(stream, content_type.required_transport_type, content_type.required_components, receiver_full_jid, Set.empty());
if (video_transport == null) {
throw new Jingle.Error.NO_SHARED_PROTOCOLS("No suitable video transports");
@ -148,26 +154,130 @@ public abstract class Module : XmppStreamModule {
}
public class Crypto {
public string cryptoSuite { get; private set; }
public string keyParams { get; private set; }
public string? sessionParams { get; private set; }
public string? tag { get; private set; }
public const string AES_CM_128_HMAC_SHA1_80 = "AES_CM_128_HMAC_SHA1_80";
public const string AES_CM_128_HMAC_SHA1_32 = "AES_CM_128_HMAC_SHA1_32";
public const string F8_128_HMAC_SHA1_80 = "F8_128_HMAC_SHA1_80";
public string crypto_suite { get; private set; }
public string key_params { get; private set; }
public string? session_params { get; private set; }
public string tag { get; private set; }
public uint8[] key_and_salt { owned get {
if (!key_params.has_prefix("inline:")) return null;
int endIndex = key_params.index_of("|");
if (endIndex < 0) endIndex = key_params.length;
string sub = key_params.substring(7, endIndex - 7);
return Base64.decode(sub);
}}
public string? lifetime { owned get {
if (!key_params.has_prefix("inline:")) return null;
int firstIndex = key_params.index_of("|");
if (firstIndex < 0) return null;
int endIndex = key_params.index_of("|", firstIndex + 1);
if (endIndex < 0) {
if (key_params.index_of(":", firstIndex) > 0) return null; // Is MKI
endIndex = key_params.length;
}
return key_params.substring(firstIndex + 1, endIndex);
}}
public int mki { get {
if (!key_params.has_prefix("inline:")) return -1;
int firstIndex = key_params.index_of("|");
if (firstIndex < 0) return -1;
int splitIndex = key_params.index_of(":", firstIndex);
if (splitIndex < 0) return -1;
int secondIndex = key_params.index_of("|", firstIndex + 1);
if (secondIndex < 0) {
return int.parse(key_params.substring(firstIndex + 1, splitIndex));
} else if (splitIndex > secondIndex) {
return int.parse(key_params.substring(secondIndex + 1, splitIndex));
}
return -1;
}}
public int mki_length { get {
if (!key_params.has_prefix("inline:")) return -1;
int firstIndex = key_params.index_of("|");
if (firstIndex < 0) return -1;
int splitIndex = key_params.index_of(":", firstIndex);
if (splitIndex < 0) return -1;
int secondIndex = key_params.index_of("|", firstIndex + 1);
if (secondIndex < 0 || splitIndex > secondIndex) {
return int.parse(key_params.substring(splitIndex + 1, key_params.length));
}
return -1;
}}
public bool is_valid { get {
switch(crypto_suite) {
case AES_CM_128_HMAC_SHA1_80:
case AES_CM_128_HMAC_SHA1_32:
case F8_128_HMAC_SHA1_80:
return key_and_salt.length == 30;
}
return false;
}}
public uint8[] key { owned get {
uint8[] key_and_salt = key_and_salt;
switch(crypto_suite) {
case AES_CM_128_HMAC_SHA1_80:
case AES_CM_128_HMAC_SHA1_32:
case F8_128_HMAC_SHA1_80:
if (key_and_salt.length >= 16) return key_and_salt[0:16];
break;
}
return null;
}}
public uint8[] salt { owned get {
uint8[] keyAndSalt = key_and_salt;
switch(crypto_suite) {
case AES_CM_128_HMAC_SHA1_80:
case AES_CM_128_HMAC_SHA1_32:
case F8_128_HMAC_SHA1_80:
if (keyAndSalt.length >= 30) return keyAndSalt[16:30];
break;
}
return null;
}}
public static Crypto create(string crypto_suite, uint8[] key_and_salt, string? session_params = null, string tag = "1") {
Crypto crypto = new Crypto();
crypto.crypto_suite = crypto_suite;
crypto.key_params = "inline:" + Base64.encode(key_and_salt);
crypto.session_params = session_params;
crypto.tag = tag;
return crypto;
}
public Crypto rekey(uint8[] key_and_salt) {
Crypto crypto = new Crypto();
crypto.crypto_suite = crypto_suite;
crypto.key_params = "inline:" + Base64.encode(key_and_salt);
crypto.session_params = session_params;
crypto.tag = tag;
return crypto;
}
public static Crypto parse(StanzaNode node) {
Crypto crypto = new Crypto();
crypto.cryptoSuite = node.get_attribute("crypto-suite");
crypto.keyParams = node.get_attribute("key-params");
crypto.sessionParams = node.get_attribute("session-params");
crypto.crypto_suite = node.get_attribute("crypto-suite");
crypto.key_params = node.get_attribute("key-params");
crypto.session_params = node.get_attribute("session-params");
crypto.tag = node.get_attribute("tag");
return crypto;
}
public StanzaNode to_xml() {
StanzaNode node = new StanzaNode.build("crypto", NS_URI)
.put_attribute("crypto-suite", cryptoSuite)
.put_attribute("key-params", keyParams);
if (sessionParams != null) node.put_attribute("session-params", sessionParams);
if (tag != null) node.put_attribute("tag", tag);
.put_attribute("crypto-suite", crypto_suite)
.put_attribute("key-params", key_params)
.put_attribute("tag", tag);
if (session_params != null) node.put_attribute("session-params", session_params);
return node;
}
}

View file

@ -17,6 +17,20 @@ public abstract class Xmpp.Xep.JingleRtp.Stream : Object {
}
return null;
}}
public JingleRtp.Crypto? local_crypto { get {
var content_params = content.content_params;
if (content_params is Parameters) {
return ((Parameters)content_params).local_crypto;
}
return null;
}}
public JingleRtp.Crypto? remote_crypto { get {
var content_params = content.content_params;
if (content_params is Parameters) {
return ((Parameters)content_params).remote_crypto;
}
return null;
}}
public bool sending { get {
return content.session.senders_include_us(content.senders);
}}