[asterisk-commits] twilson: trunk r268894 - in /trunk: ./ build_tools/ channels/ channels/sip/ c...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Jun 8 00:29:20 CDT 2010
Author: twilson
Date: Tue Jun 8 00:29:08 2010
New Revision: 268894
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=268894
Log:
Add SRTP support for Asterisk
After 5 years in mantis and over a year on reviewboard, SRTP support is finally
being comitted. This includes generic CHANNEL dialplan functions that work for
getting the status of whether a call has secure media or signaling as defined
by the underlying channel technology and for setting whether or not a new
channel being bridged to a calling channel should have secure signaling or
media. See doc/tex/secure-calls.tex for examples.
Original patch by mikma, updated for trunk and revised by me.
(closes issue #5413)
Reported by: mikma
Tested by: twilson, notthematrix, hemanshurpatel
Review: https://reviewboard.asterisk.org/r/191/
Added:
trunk/channels/sip/include/sdp_crypto.h (with props)
trunk/channels/sip/include/srtp.h (with props)
trunk/channels/sip/sdp_crypto.c (with props)
trunk/channels/sip/srtp.c (with props)
trunk/doc/tex/secure-calls.tex (with props)
trunk/include/asterisk/res_srtp.h (with props)
trunk/res/res_srtp.c (with props)
trunk/res/res_srtp.exports.in (with props)
Modified:
trunk/CHANGES
trunk/build_tools/menuselect-deps.in
trunk/channels/chan_iax2.c
trunk/channels/chan_sip.c
trunk/channels/sip/dialplan_functions.c
trunk/channels/sip/include/sip.h
trunk/configure
trunk/configure.ac
trunk/doc/tex/asterisk.tex
trunk/funcs/func_channel.c
trunk/include/asterisk/autoconfig.h.in
trunk/include/asterisk/frame.h
trunk/include/asterisk/global_datastores.h
trunk/include/asterisk/rtp_engine.h
trunk/main/asterisk.exports.in
trunk/main/channel.c
trunk/main/global_datastores.c
trunk/main/rtp_engine.c
trunk/makeopts.in
trunk/res/res_rtp_asterisk.c
Modified: trunk/CHANGES
URL: http://svnview.digium.com/svn/asterisk/trunk/CHANGES?view=diff&rev=268894&r1=268893&r2=268894
==============================================================================
--- trunk/CHANGES (original)
+++ trunk/CHANGES Tue Jun 8 00:29:08 2010
@@ -59,6 +59,10 @@
* When dialing SIP peers, a new component may be added to the end of the dialstring
to indicate that a specific remote IP address or host should be used when dialing
the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
+ * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
+ ability to selectively force bridged channels to also be encrypted is also
+ implemented. Branching in the dialplan can be done based on whether or not
+ a channel has secure media and/or signaling.
* Added directmediapermit/directmediadeny to limit which peers can send direct media
to each other
* Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
@@ -68,6 +72,10 @@
-----------
* Added rtsavesysname option into iax.conf to allow the systname to be saved
on realtime updates.
+ * Added the ability for chan_iax2 to inform the dialplan whether or not
+ encryption is being used. This interoperates with the SIP SRTP implementation
+ so that a secure SIP call can be bridged to a secure IAX call when the
+ dialplan requires bridged channels to be "secure".
MGCP Changes
------------
@@ -205,6 +213,11 @@
prefixing the name of the hash at assignment with the appropriate number of
underscores, just like variables.
* GROUP_MATCH_COUNT has been improved to allow regex matching on category
+ * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
+ whether or not channels that are bridged to the current channel will be
+ required to have secure signaling and/or media.
+ * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
+ the current channel has secure signaling and/or media.
* For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
"no_media_path" option.
Returns "0" if there is a B channel associated with the call.
Modified: trunk/build_tools/menuselect-deps.in
URL: http://svnview.digium.com/svn/asterisk/trunk/build_tools/menuselect-deps.in?view=diff&rev=268894&r1=268893&r2=268894
==============================================================================
--- trunk/build_tools/menuselect-deps.in (original)
+++ trunk/build_tools/menuselect-deps.in Tue Jun 8 00:29:08 2010
@@ -51,6 +51,7 @@
SPEEX_PREPROCESS=@PBX_SPEEX_PREPROCESS@
SQLITE3=@PBX_SQLITE3@
SQLITE=@PBX_SQLITE@
+SRTP=@PBX_SRTP@
SS7=@PBX_SS7@
OPENSSL=@PBX_OPENSSL@
SUPPSERV=@PBX_SUPPSERV@
Modified: trunk/channels/chan_iax2.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_iax2.c?view=diff&rev=268894&r1=268893&r2=268894
==============================================================================
--- trunk/channels/chan_iax2.c (original)
+++ trunk/channels/chan_iax2.c Tue Jun 8 00:29:08 2010
@@ -1171,6 +1171,7 @@
static int iax2_sendimage(struct ast_channel *c, struct ast_frame *img);
static int iax2_sendtext(struct ast_channel *c, const char *text);
static int iax2_setoption(struct ast_channel *c, int option, void *data, int datalen);
+static int iax2_queryoption(struct ast_channel *c, int option, void *data, int *datalen);
static int iax2_transfer(struct ast_channel *c, const char *dest);
static int iax2_write(struct ast_channel *c, struct ast_frame *f);
static int send_trunk(struct iax2_trunk_peer *tpeer, struct timeval *now);
@@ -1218,6 +1219,7 @@
.write_video = iax2_write,
.indicate = iax2_indicate,
.setoption = iax2_setoption,
+ .queryoption = iax2_queryoption,
.bridge = iax2_bridge,
.transfer = iax2_transfer,
.fixup = iax2_fixup,
@@ -4903,6 +4905,11 @@
ast_log(LOG_WARNING, "No address associated with '%s'\n", pds.peer);
return -1;
}
+ if (ast_test_flag64(iaxs[callno], IAX_FORCE_ENCRYPT) && !cai.encmethods) {
+ ast_log(LOG_WARNING, "Encryption forced for call, but not enabled\n");
+ c->hangupcause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
+ return -1;
+ }
if (ast_strlen_zero(cai.secret) && ast_test_flag64(iaxs[callno], IAX_FORCE_ENCRYPT)) {
ast_log(LOG_WARNING, "Call terminated. No secret given and force encrypt enabled\n");
return -1;
@@ -5144,6 +5151,19 @@
case AST_OPTION_OPRMODE:
errno = EINVAL;
return -1;
+ case AST_OPTION_SECURE_SIGNALING:
+ case AST_OPTION_SECURE_MEDIA:
+ {
+ unsigned short callno = PTR_TO_CALLNO(c->tech_pvt);
+ ast_mutex_lock(&iaxsl[callno]);
+ if ((*(int *) data)) {
+ ast_set_flag64(iaxs[callno], IAX_FORCE_ENCRYPT);
+ } else {
+ ast_clear_flag64(iaxs[callno], IAX_FORCE_ENCRYPT);
+ }
+ ast_mutex_unlock(&iaxsl[callno]);
+ return 0;
+ }
default:
{
unsigned short callno = PTR_TO_CALLNO(c->tech_pvt);
@@ -5172,6 +5192,23 @@
ast_free(h);
return res;
}
+ }
+}
+
+static int iax2_queryoption(struct ast_channel *c, int option, void *data, int *datalen)
+{
+ switch (option) {
+ case AST_OPTION_SECURE_SIGNALING:
+ case AST_OPTION_SECURE_MEDIA:
+ {
+ unsigned short callno = PTR_TO_CALLNO(c->tech_pvt);
+ ast_mutex_lock(&iaxsl[callno]);
+ *((int *) data) = ast_test_flag64(iaxs[callno], IAX_FORCE_ENCRYPT) ? 1 : 0;
+ ast_mutex_unlock(&iaxsl[callno]);
+ return 0;
+ }
+ default:
+ return -1;
}
}
@@ -13609,6 +13646,8 @@
ast_copy_string(buf, pvt->addr.sin_addr.s_addr ? ast_inet_ntoa(pvt->addr.sin_addr) : "", buflen);
} else if (!strcasecmp(args, "peername")) {
ast_copy_string(buf, pvt->username, buflen);
+ } else if (!strcasecmp(args, "secure_signaling") || !strcasecmp(args, "secure_media")) {
+ snprintf(buf, buflen, "%s", IAX_CALLENCRYPTED(pvt) ? "1" : "");
} else {
res = -1;
}
Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=268894&r1=268893&r2=268894
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Tue Jun 8 00:29:08 2010
@@ -267,10 +267,13 @@
#include "sip/include/config_parser.h"
#include "sip/include/reqresp_parser.h"
#include "sip/include/sip_utils.h"
+#include "sip/include/srtp.h"
+#include "sip/include/sdp_crypto.h"
#include "asterisk/ccss.h"
#include "asterisk/xml.h"
#include "sip/include/dialog.h"
#include "sip/include/dialplan_functions.h"
+
/*** DOCUMENTATION
<application name="SIPDtmfMode" language="en_US">
@@ -1566,6 +1569,10 @@
static int handle_response_register(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, int seqno);
static void handle_response(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, int seqno);
+/*------ SRTP Support -------- */
+static int setup_srtp(struct sip_srtp **srtp);
+static int process_crypto(struct sip_pvt *p, struct ast_rtp_instance *rtp, struct sip_srtp **srtp, const char *a);
+
/*------ T38 Support --------- */
static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans);
static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan);
@@ -3910,7 +3917,16 @@
res = 0;
}
break;
+ case AST_OPTION_SECURE_SIGNALING:
+ p->req_secure_signaling = *(unsigned int *) data;
+ res = 0;
+ break;
+ case AST_OPTION_SECURE_MEDIA:
+ ast_set2_flag(&p->flags[1], *(unsigned int *) data, SIP_PAGE2_USE_SRTP);
+ res = 0;
+ break;
default:
+ ast_log(LOG_NOTICE, "Unknown option: %d\n", option);
break;
}
@@ -3960,6 +3976,14 @@
cp = (char *) data;
*cp = p->dsp ? 1 : 0;
ast_debug(1, "Reporting digit detection %sabled on %s\n", *cp ? "en" : "dis", chan->name);
+ break;
+ case AST_OPTION_SECURE_SIGNALING:
+ *((unsigned int *) data) = p->req_secure_signaling;
+ res = 0;
+ break;
+ case AST_OPTION_SECURE_MEDIA:
+ *((unsigned int *) data) = ast_test_flag(&p->flags[1], SIP_PAGE2_USE_SRTP) ? 1 : 0;
+ res = 0;
break;
case AST_OPTION_DEVICE_NAME:
if (p && p->outgoing_call) {
@@ -5004,6 +5028,35 @@
}
}
+ /* Check to see if we should try to force encryption */
+ if (p->req_secure_signaling && p->socket.type != SIP_TRANSPORT_TLS) {
+ ast_log(LOG_WARNING, "Encrypted signaling is required\n");
+ ast->hangupcause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
+ return -1;
+ }
+
+ if (ast_test_flag(&p->flags[1], SIP_PAGE2_USE_SRTP)) {
+ if (ast_test_flag(&p->flags[0], SIP_REINVITE)) {
+ ast_debug(1, "Direct media not possible when using SRTP, ignoring canreinvite setting\n");
+ ast_clear_flag(&p->flags[0], SIP_REINVITE);
+ }
+
+ if (p->rtp && !p->srtp && setup_srtp(&p->srtp) < 0) {
+ ast_log(LOG_WARNING, "SRTP audio setup failed\n");
+ return -1;
+ }
+
+ if (p->vrtp && !p->vsrtp && setup_srtp(&p->vsrtp) < 0) {
+ ast_log(LOG_WARNING, "SRTP video setup failed\n");
+ return -1;
+ }
+
+ if (p->trtp && !p->vsrtp && setup_srtp(&p->tsrtp) < 0) {
+ ast_log(LOG_WARNING, "SRTP text setup failed\n");
+ return -1;
+ }
+ }
+
res = 0;
ast_set_flag(&p->flags[0], SIP_OUTGOING);
@@ -5209,6 +5262,21 @@
if (p->chanvars) {
ast_variables_destroy(p->chanvars);
p->chanvars = NULL;
+ }
+
+ if (p->srtp) {
+ sip_srtp_destroy(p->srtp);
+ p->srtp = NULL;
+ }
+
+ if (p->vsrtp) {
+ sip_srtp_destroy(p->vsrtp);
+ p->vsrtp = NULL;
+ }
+
+ if (p->tsrtp) {
+ sip_srtp_destroy(p->tsrtp);
+ p->tsrtp = NULL;
}
if (p->directmediaha) {
@@ -7671,6 +7739,10 @@
const char *codecs;
int codec;
+ /* SRTP */
+ int secure_audio = FALSE;
+ int secure_video = FALSE;
+
/* Others */
int sendonly = -1;
int vsendonly = -1;
@@ -7770,6 +7842,7 @@
int video = FALSE;
int image = FALSE;
int text = FALSE;
+ char protocol[5] = {0,};
int x;
numberofports = 1;
@@ -7780,8 +7853,14 @@
nextm = get_sdp_iterate(&next, req, "m");
/* Search for audio media definition */
- if ((sscanf(m, "audio %30u/%30u RTP/AVP %n", &x, &numberofports, &len) == 2 && len > 0) ||
- (sscanf(m, "audio %30u RTP/AVP %n", &x, &len) == 1 && len > 0)) {
+ if ((sscanf(m, "audio %30u/%30u RTP/%4s %n", &x, &numberofports, protocol, &len) == 3 && len > 0) ||
+ (sscanf(m, "audio %30u RTP/%4s %n", &x, protocol, &len) == 2 && len > 0)) {
+ if (!strcmp(protocol, "SAVP")) {
+ secure_audio = 1;
+ } else if (strcmp(protocol, "AVP")) {
+ ast_log(LOG_WARNING, "unknown SDP media protocol in offer: %s\n", protocol);
+ continue;
+ }
audio = TRUE;
p->offered_media[SDP_AUDIO].offered = TRUE;
numberofmediastreams++;
@@ -7801,8 +7880,14 @@
ast_rtp_codecs_payloads_set_m_type(&newaudiortp, NULL, codec);
}
/* Search for video media definition */
- } else if ((sscanf(m, "video %30u/%30u RTP/AVP %n", &x, &numberofports, &len) == 2 && len > 0) ||
- (sscanf(m, "video %30u RTP/AVP %n", &x, &len) == 1 && len >= 0)) {
+ } else if ((sscanf(m, "video %30u/%30u RTP/%4s %n", &x, &numberofports, protocol, &len) == 3 && len > 0) ||
+ (sscanf(m, "video %30u RTP/%4s %n", &x, protocol, &len) == 2 && len >= 0)) {
+ if (!strcmp(protocol, "SAVP")) {
+ secure_video = 1;
+ } else if (strcmp(protocol, "AVP")) {
+ ast_log(LOG_WARNING, "unknown SDP media protocol in offer: %s\n", protocol);
+ continue;
+ }
video = TRUE;
p->novideo = FALSE;
p->offered_media[SDP_VIDEO].offered = TRUE;
@@ -7864,8 +7949,6 @@
if (numberofports > 1)
ast_log(LOG_WARNING, "SDP offered %d ports for media, not supported by Asterisk. Will try anyway...\n", numberofports);
-
-
/* Media stream specific parameters */
while ((type = get_sdp_line(&iterator, next - 1, req, &value)) != '\0') {
int processed = FALSE;
@@ -7899,12 +7982,16 @@
if (audio) {
if (process_sdp_a_sendonly(value, &sendonly))
processed = TRUE;
+ else if (process_crypto(p, p->rtp, &p->srtp, value))
+ processed = TRUE;
else if (process_sdp_a_audio(value, p, &newaudiortp, &last_rtpmap_codec))
processed = TRUE;
}
/* Video specific scanning */
else if (video) {
if (process_sdp_a_sendonly(value, &vsendonly))
+ processed = TRUE;
+ else if (process_crypto(p, p->vrtp, &p->vsrtp, value))
processed = TRUE;
else if (process_sdp_a_video(value, p, &newvideortp, &last_rtpmap_codec))
processed = TRUE;
@@ -7913,6 +8000,8 @@
else if (text) {
if (process_sdp_a_text(value, p, &newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec))
processed = TRUE;
+ else if (process_crypto(p, p->trtp, &p->tsrtp, value))
+ processed = TRUE;
}
/* Image (T.38 FAX) specific scanning */
else if (image) {
@@ -7937,19 +8026,47 @@
return -1;
}
- if (portno == -1 && vportno == -1 && udptlportno == -1 && tportno == -1)
+ if (portno == -1 && vportno == -1 && udptlportno == -1 && tportno == -1) {
/* No acceptable offer found in SDP - we have no ports */
/* Do not change RTP or VRTP if this is a re-invite */
+ ast_log(LOG_WARNING, "Failing due to no acceptable offer found\n");
return -2;
-
- if (numberofmediastreams > 3)
+ }
+
+ if (numberofmediastreams > 3) {
/* We have too many fax, audio and/or video and/or text media streams, fail this offer */
+ ast_log(LOG_WARNING, "Faling due to too many media streams\n");
return -3;
+ }
+
+ if (secure_audio && !(p->srtp && (ast_test_flag(p->srtp, SRTP_CRYPTO_OFFER_OK)))) {
+ ast_log(LOG_WARNING, "Can't provide secure audio requested in SDP offer\n");
+ return -4;
+ }
+
+ if (!secure_audio && p->srtp) {
+ ast_log(LOG_WARNING, "We are requesting SRTP, but they responded without it!\n");
+ return -4;
+ }
+
+ if (secure_video && !(p->vsrtp && (ast_test_flag(p->vsrtp, SRTP_CRYPTO_OFFER_OK)))) {
+ ast_log(LOG_WARNING, "Can't provide secure video requested in SDP offer\n");
+ return -4;
+ }
+
+ if (!p->novideo && !secure_video && p->vsrtp) {
+ ast_log(LOG_WARNING, "We are requesting SRTP, but they responded without it!\n");
+ return -4;
+ }
+
+ if (!(secure_audio || secure_video) && ast_test_flag(&p->flags[1], SIP_PAGE2_USE_SRTP)) {
+ ast_log(LOG_WARNING, "Matched device setup to use SRTP, but request was not!\n");
+ return -4;
+ }
if (udptlportno == -1) {
change_t38_state(p, T38_DISABLED);
}
-
/* Now gather all of the codecs that we are asked for: */
ast_rtp_codecs_payload_formats(&newaudiortp, &peercapability, &peernoncodeccapability);
@@ -9843,6 +9960,23 @@
}
}
+static void get_crypto_attrib(struct sip_srtp *srtp, const char **a_crypto)
+{
+ /* Set encryption properties */
+ if (srtp) {
+ if (!srtp->crypto) {
+ srtp->crypto = sdp_crypto_setup();
+ }
+ if (srtp->crypto && (sdp_crypto_offer(srtp->crypto) >= 0)) {
+ *a_crypto = sdp_crypto_attrib(srtp->crypto);
+ }
+
+ if (!*a_crypto) {
+ ast_log(LOG_WARNING, "No SRTP key management enabled\n");
+ }
+ }
+}
+
/*! \brief Add Session Description Protocol message
If oldsdp is TRUE, then the SDP version number is not incremented. This mechanism
@@ -9879,6 +10013,9 @@
struct ast_str *a_video = ast_str_alloca(1024); /* Attributes for video */
struct ast_str *a_text = ast_str_alloca(1024); /* Attributes for text */
struct ast_str *a_modem = ast_str_alloca(1024); /* Attributes for modem */
+ const char *a_crypto = NULL;
+ const char *v_a_crypto = NULL;
+ const char *t_a_crypto = NULL;
format_t x;
format_t capability = 0;
@@ -9962,7 +10099,9 @@
/* Ok, we need video. Let's add what we need for video and set codecs.
Video is handled differently than audio since we can not transcode. */
if (needvideo) {
- ast_str_append(&m_video, 0, "m=video %d RTP/AVP", ntohs(vdest.sin_port));
+ get_crypto_attrib(p->vsrtp, &v_a_crypto);
+ ast_str_append(&m_video, 0, "m=video %d RTP/%s", ntohs(vdest.sin_port),
+ v_a_crypto ? "SAVP" : "AVP");
/* Build max bitrate string */
if (p->maxcallbitrate)
@@ -9976,7 +10115,9 @@
if (needtext) {
if (sipdebug_text)
ast_verbose("Lets set up the text sdp\n");
- ast_str_append(&m_text, 0, "m=text %d RTP/AVP", ntohs(tdest.sin_port));
+ get_crypto_attrib(p->tsrtp, &t_a_crypto);
+ ast_str_append(&m_text, 0, "m=text %d RTP/%s", ntohs(tdest.sin_port),
+ t_a_crypto ? "SAVP" : "AVP");
if (debug) /* XXX should I use tdest below ? */
ast_verbose("Text is at %s port %d\n", ast_inet_ntoa(p->ourip.sin_addr), ntohs(tsin.sin_port));
@@ -9987,7 +10128,9 @@
/* We break with the "recommendation" and send our IP, in order that our
peer doesn't have to ast_gethostbyname() us */
- ast_str_append(&m_audio, 0, "m=audio %d RTP/AVP", ntohs(dest.sin_port));
+ get_crypto_attrib(p->srtp, &a_crypto);
+ ast_str_append(&m_audio, 0, "m=audio %d RTP/%s", ntohs(dest.sin_port),
+ a_crypto ? "SAVP" : "AVP");
if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) == SIP_PAGE2_CALL_ONHOLD_ONEDIR)
hold = "a=recvonly\r\n";
@@ -10149,7 +10292,15 @@
len += m_text->used + a_text->used + strlen(hold);
if (add_t38)
len += m_modem->used + a_modem->used;
-
+ if (a_crypto) {
+ len += strlen(a_crypto);
+ }
+ if (v_a_crypto) {
+ len += strlen(v_a_crypto);
+ }
+ if (t_a_crypto) {
+ len += strlen(t_a_crypto);
+ }
add_header(resp, "Content-Type", "application/sdp");
add_header_contentLength(resp, len);
add_line(resp, version);
@@ -10163,6 +10314,9 @@
add_line(resp, m_audio->str);
add_line(resp, a_audio->str);
add_line(resp, hold);
+ if (a_crypto) {
+ add_line(resp, a_crypto);
+ }
} else if (p->offered_media[SDP_AUDIO].offered) {
snprintf(dummy_answer, sizeof(dummy_answer), "m=audio 0 RTP/AVP %s\r\n", p->offered_media[SDP_AUDIO].codecs);
add_line(resp, dummy_answer);
@@ -10171,6 +10325,9 @@
add_line(resp, m_video->str);
add_line(resp, a_video->str);
add_line(resp, hold); /* Repeat hold for the video stream */
+ if (v_a_crypto) {
+ add_line(resp, v_a_crypto);
+ }
} else if (p->offered_media[SDP_VIDEO].offered) {
snprintf(dummy_answer, sizeof(dummy_answer), "m=video 0 RTP/AVP %s\r\n", p->offered_media[SDP_VIDEO].codecs);
add_line(resp, dummy_answer);
@@ -10179,6 +10336,9 @@
add_line(resp, m_text->str);
add_line(resp, a_text->str);
add_line(resp, hold); /* Repeat hold for the text stream */
+ if (t_a_crypto) {
+ add_line(resp, t_a_crypto);
+ }
} else if (p->offered_media[SDP_TEXT].offered) {
snprintf(dummy_answer, sizeof(dummy_answer), "m=text 0 RTP/AVP %s\r\n", p->offered_media[SDP_TEXT].codecs);
add_line(resp, dummy_answer);
@@ -17482,6 +17642,8 @@
ast_copy_string(buf, peer->cid_num, len);
} else if (!strcasecmp(colname, "codecs")) {
ast_getformatname_multiple(buf, len -1, peer->capability);
+ } else if (!strcasecmp(colname, "encryption")) {
+ snprintf(buf, len, "%d", ast_test_flag(&peer->flags[1], SIP_PAGE2_USE_SRTP));
} else if (!strncasecmp(colname, "chanvar[", 8)) {
char *chanvar=colname + 8;
struct ast_variable *v;
@@ -21015,8 +21177,13 @@
transmit_response_with_t38_sdp(p, "200 OK", req, (reinvite ? XMIT_RELIABLE : (req->ignore ? XMIT_UNRELIABLE : XMIT_CRITICAL)));
} else if (p->t38.state == T38_DISABLED) {
/* If this is not a re-invite or something to ignore - it's critical */
- ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
- transmit_response_with_sdp(p, "200 OK", req, (reinvite ? XMIT_RELIABLE : (req->ignore ? XMIT_UNRELIABLE : XMIT_CRITICAL)), p->session_modify == TRUE ? FALSE : TRUE, FALSE);
+ if (p->srtp && !ast_test_flag(p->srtp, SRTP_CRYPTO_OFFER_OK)) {
+ ast_log(LOG_WARNING, "Target does not support required crypto\n");
+ transmit_response_reliable(p, "488 Not Acceptable Here (crypto)", req);
+ } else {
+ ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
+ transmit_response_with_sdp(p, "200 OK", req, (reinvite ? XMIT_RELIABLE : (req->ignore ? XMIT_UNRELIABLE : XMIT_CRITICAL)), p->session_modify == TRUE ? FALSE : TRUE, FALSE);
+ }
}
p->invitestate = INV_TERMINATED;
@@ -25374,6 +25541,8 @@
ast_string_field_set(peer, unsolicited_mailbox, v->value);
} else if (!strcasecmp(v->name, "use_q850_reason")) {
ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_Q850_REASON);
+ } else if (!strcasecmp(v->name, "encryption")) {
+ ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_USE_SRTP);
} else if (!strcasecmp(v->name, "snom_aoc_enabled")) {
ast_set2_flag(&peer->flags[2], ast_true(v->value), SIP_PAGE3_SNOM_AOC);
}
@@ -26666,6 +26835,10 @@
res = AST_RTP_GLUE_RESULT_FORBID;
}
+ if (p->srtp) {
+ res = AST_RTP_GLUE_RESULT_FORBID;
+ }
+
sip_pvt_unlock(p);
return res;
@@ -27084,6 +27257,50 @@
} while (0));
}
+/* SRTP */
+static int setup_srtp(struct sip_srtp **srtp)
+{
+ if (!ast_rtp_engine_srtp_is_registered()) {
+ ast_log(LOG_ERROR, "No SRTP module loaded, can't setup SRTP session.\n");
+ return -1;
+ }
+
+ if (!(*srtp = sip_srtp_alloc())) { /* Allocate SRTP data structure */
+ return -1;
+ }
+
+ return 0;
+}
+
+static int process_crypto(struct sip_pvt *p, struct ast_rtp_instance *rtp, struct sip_srtp **srtp, const char *a)
+{
+ if (strncasecmp(a, "crypto:", 7)) {
+ return FALSE;
+ }
+ if (!*srtp) {
+ if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
+ ast_log(LOG_WARNING, "Ignoring unexpected crypto attribute in SDP answer\n");
+ return FALSE;
+ }
+
+ if (setup_srtp(srtp) < 0) {
+ return FALSE;
+ }
+ }
+
+ if (!(*srtp)->crypto && !((*srtp)->crypto = sdp_crypto_setup())) {
+ return FALSE;
+ }
+
+ if (sdp_crypto_process((*srtp)->crypto, a, rtp) < 0) {
+ return FALSE;
+ }
+
+ ast_set_flag(*srtp, SRTP_CRYPTO_OFFER_OK);
+
+ return TRUE;
+}
+
/*! \brief Reload module */
static int sip_do_reload(enum channelreloadreason reason)
{
Modified: trunk/channels/sip/dialplan_functions.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sip/dialplan_functions.c?view=diff&rev=268894&r1=268893&r2=268894
==============================================================================
--- trunk/channels/sip/dialplan_functions.c (original)
+++ trunk/channels/sip/dialplan_functions.c Tue Jun 8 00:29:08 2010
@@ -214,6 +214,10 @@
ast_log(LOG_WARNING, "Unrecognized argument '%s' to %s\n", preparse, funcname);
return -1;
}
+ } else if (!strcasecmp(args.param, "secure_signaling")) {
+ snprintf(buf, buflen, "%s", p->socket.type == SIP_TRANSPORT_TLS ? "1" : "");
+ } else if (!strcasecmp(args.param, "secure_media")) {
+ snprintf(buf, buflen, "%s", p->srtp ? "1" : "");
} else {
res = -1;
}
Added: trunk/channels/sip/include/sdp_crypto.h
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sip/include/sdp_crypto.h?view=auto&rev=268894
==============================================================================
--- trunk/channels/sip/include/sdp_crypto.h (added)
+++ trunk/channels/sip/include/sdp_crypto.h Tue Jun 8 00:29:08 2010
@@ -1,0 +1,82 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2006 - 2007, Mikael Magnusson
+ *
+ * Mikael Magnusson <mikma at users.sourceforge.net>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file sdp_crypto.h
+ *
+ * \brief SDP Security descriptions
+ *
+ * Specified in RFC 4568
+ *
+ * \author Mikael Magnusson <mikma at users.sourceforge.net>
+ */
+
+#ifndef _SDP_CRYPTO_H
+#define _SDP_CRYPTO_H
+
+#include <asterisk/rtp_engine.h>
+
+struct sdp_crypto;
+
+/*! \brief Initialize an return an sdp_crypto struct
+ *
+ * \details
+ * This function allocates a new sdp_crypto struct and initializes its values
+ *
+ * \retval NULL on failure
+ * \retval a pointer to a new sdp_crypto structure
+ */
+struct sdp_crypto *sdp_crypto_setup(void);
+
+/*! \brief Destroy a previously allocated sdp_crypto struct */
+void sdp_crypto_destroy(struct sdp_crypto *crypto);
+
+/*! \brief Parse the a=crypto line from SDP and set appropriate values on the
+ * sdp_crypto struct.
+ *
+ * \param p A valid sdp_crypto struct
+ * \param attr the a:crypto line from SDP
+ * \param rtp The rtp instance associated with the SDP being parsed
+ *
+ * \retval 0 success
+ * \retval nonzero failure
+ */
+int sdp_crypto_process(struct sdp_crypto *p, const char *attr, struct ast_rtp_instance *rtp);
+
+
+/*! \brief Generate an SRTP a=crypto offer
+ *
+ * \details
+ * The offer is stored on the sdp_crypto struct in a_crypto
+ *
+ * \param A valid sdp_crypto struct
+ *
+ * \retval 0 success
+ * \retval nonzero failure
+ */
+int sdp_crypto_offer(struct sdp_crypto *p);
+
+
+/*! \brief Return the a_crypto value of the sdp_crypto struct
+ *
+ * \param p An sdp_crypto struct that has had sdp_crypto_offer called
+ *
+ * \retval The value of the a_crypto for p
+ */
+const char *sdp_crypto_attrib(struct sdp_crypto *p);
+
+#endif /* _SDP_CRYPTO_H */
Propchange: trunk/channels/sip/include/sdp_crypto.h
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: trunk/channels/sip/include/sdp_crypto.h
------------------------------------------------------------------------------
svn:keywords = Author ID Date Revision
Propchange: trunk/channels/sip/include/sdp_crypto.h
------------------------------------------------------------------------------
svn:mime-type = text/plain
Modified: trunk/channels/sip/include/sip.h
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sip/include/sip.h?view=diff&rev=268894&r1=268893&r2=268894
==============================================================================
--- trunk/channels/sip/include/sip.h (original)
+++ trunk/channels/sip/include/sip.h Tue Jun 8 00:29:08 2010
@@ -307,10 +307,8 @@
#define SIP_PAGE2_RTAUTOCLEAR (1 << 1) /*!< GP: Should we clean memory from peers after expiry? */
#define SIP_PAGE2_RPID_UPDATE (1 << 2)
#define SIP_PAGE2_Q850_REASON (1 << 3) /*!< DP: Get/send cause code via Reason header */
-
#define SIP_PAGE2_SYMMETRICRTP (1 << 4) /*!< GDP: Whether symmetric RTP is enabled or not */
#define SIP_PAGE2_STATECHANGEQUEUE (1 << 5) /*!< D: Unsent state pending change exists */
-
#define SIP_PAGE2_CONNECTLINEUPDATE_PEND (1 << 6)
#define SIP_PAGE2_RPID_IMMEDIATE (1 << 7)
#define SIP_PAGE2_RPORT_PRESENT (1 << 8) /*!< Was rport received in the Via header? */
@@ -345,6 +343,7 @@
#define SIP_PAGE2_UDPTL_DESTINATION (1 << 25) /*!< DP: Use source IP of RTP as destination if NAT is enabled */
#define SIP_PAGE2_VIDEOSUPPORT_ALWAYS (1 << 26) /*!< DP: Always set up video, even if endpoints don't support it */
#define SIP_PAGE2_HAVEPEERCONTEXT (1 << 27) /*< Are we associated with a configured peer context? */
+#define SIP_PAGE2_USE_SRTP (1 << 28) /*!< DP: Whether we should offer (only) SRTP */
#define SIP_PAGE2_FLAGS_TO_COPY \
(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_IGNORESDPVERSION | \
@@ -352,7 +351,7 @@
SIP_PAGE2_BUGGY_MWI | SIP_PAGE2_TEXTSUPPORT | SIP_PAGE2_FAX_DETECT | \
SIP_PAGE2_UDPTL_DESTINATION | SIP_PAGE2_VIDEOSUPPORT_ALWAYS | SIP_PAGE2_PREFERRED_CODEC | \
SIP_PAGE2_RPID_IMMEDIATE | SIP_PAGE2_RPID_UPDATE | SIP_PAGE2_SYMMETRICRTP |\
- SIP_PAGE2_Q850_REASON | SIP_PAGE2_HAVEPEERCONTEXT)
+ SIP_PAGE2_Q850_REASON | SIP_PAGE2_HAVEPEERCONTEXT | SIP_PAGE2_USE_SRTP)
#define SIP_PAGE3_SNOM_AOC (1 << 0) /*!< DPG: Allow snom aoc messages */
@@ -965,6 +964,7 @@
* or respect the other endpoint's request for frame sizes (on)
* for incoming calls
*/
+ unsigned short req_secure_signaling:1;/*!< Whether we are required to have secure signaling or not */
char tag[11]; /*!< Our tag for this session */
int timer_t1; /*!< SIP timer T1, ms rtt */
int timer_b; /*!< SIP timer B, ms */
@@ -1048,6 +1048,9 @@
AST_LIST_HEAD_NOLOCK(request_queue, sip_request) request_queue; /*!< Requests that arrived but could not be processed immediately */
struct sip_invite_param *options; /*!< Options for INVITE */
struct sip_st_dlg *stimer; /*!< SIP Session-Timers */
+ struct sip_srtp *srtp; /*!< Structure to hold Secure RTP session data for audio */
+ struct sip_srtp *vsrtp; /*!< Structure to hold Secure RTP session data for video */
+ struct sip_srtp *tsrtp; /*!< Structure to hold Secure RTP session data for text */
int red; /*!< T.140 RTP Redundancy */
int hangupcause; /*!< Storage of hangupcause copied from our owner before we disconnect from the AST channel (only used at hangup) */
Added: trunk/channels/sip/include/srtp.h
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sip/include/srtp.h?view=auto&rev=268894
==============================================================================
--- trunk/channels/sip/include/srtp.h (added)
+++ trunk/channels/sip/include/srtp.h Tue Jun 8 00:29:08 2010
@@ -1,0 +1,57 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2006 - 2007, Mikael Magnusson
+ *
+ * Mikael Magnusson <mikma at users.sourceforge.net>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file sip_srtp.h
+ *
+ * \brief SIP Secure RTP (SRTP)
+ *
+ * Specified in RFC 3711
+ *
+ * \author Mikael Magnusson <mikma at users.sourceforge.net>
+ */
+
+#ifndef _SIP_SRTP_H
+#define _SIP_SRTP_H
+
+#include "sdp_crypto.h"
+
+/* SRTP flags */
+#define SRTP_ENCR_OPTIONAL (1 << 1) /* SRTP encryption optional */
+#define SRTP_CRYPTO_ENABLE (1 << 2)
+#define SRTP_CRYPTO_OFFER_OK (1 << 3)
+
+/*! \brief structure for secure RTP audio */
+struct sip_srtp {
+ unsigned int flags;
+ struct sdp_crypto *crypto;
+};
+
+/*!
+ * \brief allocate a sip_srtp structure
+ * \retval a new malloc'd sip_srtp structure on success
+ * \retval NULL on failure
+*/
+struct sip_srtp *sip_srtp_alloc(void);
+
+/*!
+ * \brief free a sip_srtp structure
+ * \param srtp a sip_srtp structure
+*/
+void sip_srtp_destroy(struct sip_srtp *srtp);
+
+#endif /* _SIP_SRTP_H */
Propchange: trunk/channels/sip/include/srtp.h
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: trunk/channels/sip/include/srtp.h
------------------------------------------------------------------------------
svn:keywords = Author ID Date Revision
Propchange: trunk/channels/sip/include/srtp.h
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: trunk/channels/sip/sdp_crypto.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sip/sdp_crypto.c?view=auto&rev=268894
==============================================================================
--- trunk/channels/sip/sdp_crypto.c (added)
+++ trunk/channels/sip/sdp_crypto.c Tue Jun 8 00:29:08 2010
@@ -1,0 +1,310 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2006 - 2007, Mikael Magnusson
+ *
+ * Mikael Magnusson <mikma at users.sourceforge.net>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file sdp_crypto.c
+ *
+ * \brief SDP Security descriptions
+ *
+ * Specified in RFC 4568
+ *
+ * \author Mikael Magnusson <mikma at users.sourceforge.net>
+ */
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/options.h"
+#include "asterisk/utils.h"
+#include "include/sdp_crypto.h"
+
+#define SRTP_MASTER_LEN 30
+#define SRTP_MASTERKEY_LEN 16
+#define SRTP_MASTERSALT_LEN ((SRTP_MASTER_LEN) - (SRTP_MASTERKEY_LEN))
+#define SRTP_MASTER_LEN64 (((SRTP_MASTER_LEN) * 8 + 5) / 6 + 1)
+
+extern struct ast_srtp_res *res_srtp;
+extern struct ast_srtp_policy_res *res_srtp_policy;
+
+struct sdp_crypto {
+ char *a_crypto;
+ unsigned char local_key[SRTP_MASTER_LEN];
+ char local_key64[SRTP_MASTER_LEN64];
+};
+
+static int set_crypto_policy(struct ast_srtp_policy *policy, int suite_val, const unsigned char *master_key, unsigned long ssrc, int inbound);
+
+static struct sdp_crypto *sdp_crypto_alloc(void)
+{
+ struct sdp_crypto *crypto;
+
+ return crypto = ast_calloc(1, sizeof(*crypto));
+}
+
+void sdp_crypto_destroy(struct sdp_crypto *crypto)
+{
+ ast_free(crypto->a_crypto);
+ crypto->a_crypto = NULL;
+ ast_free(crypto);
+}
+
+struct sdp_crypto *sdp_crypto_setup(void)
+{
+ struct sdp_crypto *p;
+ int key_len;
+ unsigned char remote_key[SRTP_MASTER_LEN];
+
+ if (!ast_rtp_engine_srtp_is_registered()) {
+ return NULL;
+ }
+
+ if (!(p = sdp_crypto_alloc())) {
+ return NULL;
+ }
+
+ if (res_srtp->get_random(p->local_key, sizeof(p->local_key)) < 0) {
+ sdp_crypto_destroy(p);
+ return NULL;
+ }
+
+ ast_base64encode(p->local_key64, p->local_key, SRTP_MASTER_LEN, sizeof(p->local_key64));
+
+ key_len = ast_base64decode(remote_key, p->local_key64, sizeof(remote_key));
+
+ if (key_len != SRTP_MASTER_LEN) {
+ ast_log(LOG_ERROR, "base64 encode/decode bad len %d != %d\n", key_len, SRTP_MASTER_LEN);
+ ast_free(p);
+ return NULL;
+ }
+
+ if (memcmp(remote_key, p->local_key, SRTP_MASTER_LEN)) {
+ ast_log(LOG_ERROR, "base64 encode/decode bad key\n");
+ ast_free(p);
+ return NULL;
+ }
+
+ ast_debug(1 , "local_key64 %s len %zu\n", p->local_key64, strlen(p->local_key64));
+
+ return p;
+}
+
+static int set_crypto_policy(struct ast_srtp_policy *policy, int suite_val, const unsigned char *master_key, unsigned long ssrc, int inbound)
+{
+ const unsigned char *master_salt = NULL;
+
+ if (!ast_rtp_engine_srtp_is_registered()) {
+ return -1;
+ }
+
+ master_salt = master_key + SRTP_MASTERKEY_LEN;
+ if (res_srtp_policy->set_master_key(policy, master_key, SRTP_MASTERKEY_LEN, master_salt, SRTP_MASTERSALT_LEN) < 0) {
+ return -1;
+ }
+
+ if (res_srtp_policy->set_suite(policy, suite_val)) {
+ ast_log(LOG_WARNING, "Could not set remote SRTP suite\n");
+ return -1;
+ }
+
+ res_srtp_policy->set_ssrc(policy, ssrc, inbound);
+
+ return 0;
+}
+
+static int sdp_crypto_activate(struct sdp_crypto *p, int suite_val, unsigned char *remote_key, struct ast_rtp_instance *rtp)
+{
+ struct ast_srtp_policy *local_policy = NULL;
+ struct ast_srtp_policy *remote_policy = NULL;
+ struct ast_rtp_instance_stats stats = {0,};
+ int res = -1;
+
+ if (!ast_rtp_engine_srtp_is_registered()) {
+ return -1;
+ }
+
+ if (!p) {
+ return -1;
+ }
+
+ if (!(local_policy = res_srtp_policy->alloc())) {
+ return -1;
+ }
+
+ if (!(remote_policy = res_srtp_policy->alloc())) {
+ goto err;
+ }
+
+ if (ast_rtp_instance_get_stats(rtp, &stats, AST_RTP_INSTANCE_STAT_LOCAL_SSRC)) {
+ goto err;
+ }
+
+ if (set_crypto_policy(local_policy, suite_val, p->local_key, stats.local_ssrc, 0) < 0) {
+ goto err;
+ }
+
+ if (set_crypto_policy(remote_policy, suite_val, remote_key, 0, 1) < 0) {
+ goto err;
+ }
+
+ /* FIXME MIKMA */
+ /* ^^^ I wish I knew what needed fixing... */
+ if (ast_rtp_instance_add_srtp_policy(rtp, local_policy)) {
+ ast_log(LOG_WARNING, "Could not set local SRTP policy\n");
+ goto err;
+ }
+
+ if (ast_rtp_instance_add_srtp_policy(rtp, remote_policy)) {
+ ast_log(LOG_WARNING, "Could not set remote SRTP policy\n");
+ goto err;
+ }
+
+ ast_debug(1 , "SRTP policy activated\n");
+ res = 0;
+
+err:
+ if (local_policy) {
+ res_srtp_policy->destroy(local_policy);
+ }
+
+ if (remote_policy) {
+ res_srtp_policy->destroy(remote_policy);
+ }
+
+ return res;
+}
+
+int sdp_crypto_process(struct sdp_crypto *p, const char *attr, struct ast_rtp_instance *rtp)
+{
+ char *str = NULL;
+ char *name = NULL;
+ char *tag = NULL;
+ char *suite = NULL;
+ char *key_params = NULL;
+ char *key_param = NULL;
+ char *session_params = NULL;
+ char *key_salt = NULL;
+ char *lifetime = NULL;
+ int found = 0;
+ int attr_len = strlen(attr);
+ int key_len = 0;
+ int suite_val = 0;
+ unsigned char remote_key[SRTP_MASTER_LEN];
+
+ if (!ast_rtp_engine_srtp_is_registered()) {
+ return -1;
+ }
+
+ str = ast_strdupa(attr);
+
+ name = strsep(&str, ":");
+ tag = strsep(&str, " ");
+ suite = strsep(&str, " ");
+ key_params = strsep(&str, " ");
+ session_params = strsep(&str, " ");
+
+ if (!tag || !suite) {
+ ast_log(LOG_WARNING, "Unrecognized a=%s", attr);
+ return -1;
+ }
+
+ if (session_params) {
+ ast_log(LOG_WARNING, "Unsupported crypto parameters: %s", session_params);
+ return -1;
+ }
+
+ if (!strcmp(suite, "AES_CM_128_HMAC_SHA1_80")) {
+ suite_val = AST_AES_CM_128_HMAC_SHA1_80;
+ } else if (!strcmp(suite, "AES_CM_128_HMAC_SHA1_32")) {
+ suite_val = AST_AES_CM_128_HMAC_SHA1_32;
+ } else {
+ ast_log(LOG_WARNING, "Unsupported crypto suite: %s\n", suite);
+ return -1;
+ }
+
+ while ((key_param = strsep(&key_params, ";"))) {
+ char *method = NULL;
+ char *info = NULL;
+
+ method = strsep(&key_param, ":");
+ info = strsep(&key_param, ";");
+
+ if (!strcmp(method, "inline")) {
+ key_salt = strsep(&info, "|");
+ lifetime = strsep(&info, "|");
+
+ if (lifetime) {
+ ast_log(LOG_NOTICE, "Crypto life time unsupported: %s\n", attr);
+ continue;
+ }
+
[... 1469 lines stripped ...]
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