[asterisk-commits] kmoore: branch group/pimp_my_sip r387097 - in /team/group/pimp_my_sip: channe...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed May 1 08:16:24 CDT 2013
Author: kmoore
Date: Wed May 1 08:16:20 2013
New Revision: 387097
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=387097
Log:
Add SDES SRTP support to res_sip
Pull large parts of the SRTP negotiation and key management code out of
chan_sip so that they can be accessed by res_sip and add SDES SRTP
support to res_sip including support for mid-call key changes. This
feature can be activated by setting media_encryption=sdes on the
endpoint (dtls will come later).
This patch also adds support for AVPF via the use_avpf endpoint
setting.
Review: https://reviewboard.asterisk.org/r/2468
(closes issue ASTERISK-21416)
Patch-By: Kinsey Moore <kmoore at digium.com>
Added:
team/group/pimp_my_sip/include/asterisk/sdp_srtp.h (with props)
team/group/pimp_my_sip/main/sdp_srtp.c (with props)
Removed:
team/group/pimp_my_sip/channels/sip/include/sdp_crypto.h
team/group/pimp_my_sip/channels/sip/include/srtp.h
team/group/pimp_my_sip/channels/sip/sdp_crypto.c
team/group/pimp_my_sip/channels/sip/srtp.c
Modified:
team/group/pimp_my_sip/channels/chan_sip.c
team/group/pimp_my_sip/channels/sip/include/sip.h
team/group/pimp_my_sip/configs/res_sip.conf.sample
team/group/pimp_my_sip/include/asterisk/res_sip.h
team/group/pimp_my_sip/include/asterisk/res_sip_session.h
team/group/pimp_my_sip/res/res_sip/sip_configuration.c
team/group/pimp_my_sip/res/res_sip_sdp_rtp.c
team/group/pimp_my_sip/res/res_sip_session.c
Modified: team/group/pimp_my_sip/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/group/pimp_my_sip/channels/chan_sip.c?view=diff&rev=387097&r1=387096&r2=387097
==============================================================================
--- team/group/pimp_my_sip/channels/chan_sip.c (original)
+++ team/group/pimp_my_sip/channels/chan_sip.c Wed May 1 08:16:20 2013
@@ -286,8 +286,7 @@
#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/sdp_srtp.h"
#include "asterisk/ccss.h"
#include "asterisk/xml.h"
#include "sip/include/dialog.h"
@@ -1487,8 +1486,7 @@
static void handle_response(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t 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);
+static int process_crypto(struct sip_pvt *p, struct ast_rtp_instance *rtp, struct ast_sdp_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);
@@ -5913,7 +5911,7 @@
}
/*! \brief Initialize DTLS-SRTP support on an RTP instance */
-static int dialog_initialize_dtls_srtp(const struct sip_pvt *dialog, struct ast_rtp_instance *rtp, struct sip_srtp **srtp)
+static int dialog_initialize_dtls_srtp(const struct sip_pvt *dialog, struct ast_rtp_instance *rtp, struct ast_sdp_srtp **srtp)
{
struct ast_rtp_engine_dtls *dtls;
@@ -5938,7 +5936,7 @@
return -1;
}
- if (!(*srtp = sip_srtp_alloc())) {
+ if (!(*srtp = ast_sdp_srtp_alloc())) {
ast_log(LOG_ERROR, "Failed to create required SRTP structure on RTP instance '%p'\n",
rtp);
return -1;
@@ -6413,17 +6411,17 @@
ast_clear_flag(&p->flags[0], SIP_REINVITE);
}
- if (p->rtp && !p->srtp && setup_srtp(&p->srtp) < 0) {
+ if (p->rtp && !p->srtp && !(p->srtp = ast_sdp_srtp_alloc())) {
ast_log(LOG_WARNING, "SRTP audio setup failed\n");
return -1;
}
- if (p->vrtp && !p->vsrtp && setup_srtp(&p->vsrtp) < 0) {
+ if (p->vrtp && !p->vsrtp && !(p->vsrtp = ast_sdp_srtp_alloc())) {
ast_log(LOG_WARNING, "SRTP video setup failed\n");
return -1;
}
- if (p->trtp && !p->tsrtp && setup_srtp(&p->tsrtp) < 0) {
+ if (p->trtp && !p->tsrtp && !(p->tsrtp = ast_sdp_srtp_alloc())) {
ast_log(LOG_WARNING, "SRTP text setup failed\n");
return -1;
}
@@ -6688,17 +6686,17 @@
destroy_msg_headers(p);
if (p->srtp) {
- sip_srtp_destroy(p->srtp);
+ ast_sdp_srtp_destroy(p->srtp);
p->srtp = NULL;
}
if (p->vsrtp) {
- sip_srtp_destroy(p->vsrtp);
+ ast_sdp_srtp_destroy(p->vsrtp);
p->vsrtp = NULL;
}
if (p->tsrtp) {
- sip_srtp_destroy(p->tsrtp);
+ ast_sdp_srtp_destroy(p->tsrtp);
p->tsrtp = NULL;
}
@@ -10151,7 +10149,7 @@
secure_audio = 1;
if (p->srtp) {
- ast_set_flag(p->srtp, SRTP_CRYPTO_OFFER_OK);
+ ast_set_flag(p->srtp, AST_SRTP_CRYPTO_OFFER_OK);
}
} else if (!strcmp(protocol, "RTP/SAVP") || !strcmp(protocol, "RTP/SAVPF")) {
secure_audio = 1;
@@ -10232,8 +10230,8 @@
} else if (!strcmp(protocol, "UDP/TLS/RTP/SAVP") || !strcmp(protocol, "UDP/TLS/RTP/SAVPF")) {
secure_video = 1;
- if (p->vsrtp || (p->vsrtp = sip_srtp_alloc())) {
- ast_set_flag(p->vsrtp, SRTP_CRYPTO_OFFER_OK);
+ if (p->vsrtp || (p->vsrtp = ast_sdp_srtp_alloc())) {
+ ast_set_flag(p->vsrtp, AST_SRTP_CRYPTO_OFFER_OK);
}
} else if (!strcmp(protocol, "RTP/SAVP") || !strcmp(protocol, "RTP/SAVPF")) {
secure_video = 1;
@@ -10513,7 +10511,7 @@
goto process_sdp_cleanup;
}
- if (secure_audio && !(p->srtp && (ast_test_flag(p->srtp, SRTP_CRYPTO_OFFER_OK)))) {
+ if (secure_audio && !(p->srtp && (ast_test_flag(p->srtp, AST_SRTP_CRYPTO_OFFER_OK)))) {
ast_log(LOG_WARNING, "Can't provide secure audio requested in SDP offer\n");
res = -1;
goto process_sdp_cleanup;
@@ -10525,7 +10523,7 @@
goto process_sdp_cleanup;
}
- if (secure_video && !(p->vsrtp && (ast_test_flag(p->vsrtp, SRTP_CRYPTO_OFFER_OK)))) {
+ if (secure_video && !(p->vsrtp && (ast_test_flag(p->vsrtp, AST_SRTP_CRYPTO_OFFER_OK)))) {
ast_log(LOG_WARNING, "Can't provide secure video requested in SDP offer\n");
res = -1;
goto process_sdp_cleanup;
@@ -12992,52 +12990,14 @@
}
}
-static void get_crypto_attrib(struct sip_pvt *p, struct sip_srtp *srtp, const char **a_crypto)
-{
- int taglen = 80;
-
- /* Set encryption properties */
- if (srtp) {
- if (!srtp->crypto) {
- srtp->crypto = sdp_crypto_setup();
- }
-
- if (p->dtls_cfg.enabled) {
- /* If DTLS-SRTP is enabled the key details will be pulled from TLS */
- return;
- }
-
- /* set the key length based on INVITE or settings */
- if (ast_test_flag(srtp, SRTP_CRYPTO_TAG_80)) {
- taglen = 80;
- } else if (ast_test_flag(&p->flags[2], SIP_PAGE3_SRTP_TAG_32) ||
- ast_test_flag(srtp, SRTP_CRYPTO_TAG_32)) {
- taglen = 32;
- }
-
- if (srtp->crypto && (sdp_crypto_offer(srtp->crypto, taglen) >= 0)) {
- *a_crypto = sdp_crypto_attrib(srtp->crypto);
- }
-
- if (!*a_crypto) {
- ast_log(LOG_WARNING, "No SRTP key management enabled\n");
- }
- }
-}
-
-static char *get_sdp_rtp_profile(const struct sip_pvt *p, unsigned int secure, struct ast_rtp_instance *instance)
-{
- struct ast_rtp_engine_dtls *dtls;
-
- if ((dtls = ast_rtp_instance_get_dtls(instance)) && dtls->active(instance)) {
- return ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF) ? "UDP/TLS/RTP/SAVPF" : "UDP/TLS/RTP/SAVP";
- } else {
- if (ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
- return secure ? "RTP/SAVPF" : "RTP/AVPF";
- } else {
- return secure ? "RTP/SAVP" : "RTP/AVP";
- }
- }
+static char *crypto_get_attrib(struct ast_sdp_srtp *srtp, int dtls_enabled, int default_taglen_32)
+{
+ char *a_crypto;
+ char *orig_crypto = ast_strdupa(ast_sdp_srtp_get_attrib(srtp, dtls_enabled, default_taglen_32));
+ if (ast_asprintf(&a_crypto, "a=crypto:%s\r\n", orig_crypto) == -1) {
+ return NULL;
+ }
+ return a_crypto;
}
/*! \brief Add Session Description Protocol message
@@ -13078,9 +13038,9 @@
struct ast_str *a_video = ast_str_create(256); /* Attributes for video */
struct ast_str *a_text = ast_str_create(256); /* 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;
+ RAII_VAR(char *, a_crypto, NULL, ast_free);
+ RAII_VAR(char *, v_a_crypto, NULL, ast_free);
+ RAII_VAR(char *, t_a_crypto, NULL, ast_free);
int x;
struct ast_format tmp_fmt;
@@ -13198,9 +13158,11 @@
/* 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) {
- get_crypto_attrib(p, p->vsrtp, &v_a_crypto);
+ v_a_crypto = crypto_get_attrib(p->vsrtp, p->dtls_cfg.enabled,
+ ast_test_flag(&p->flags[2], SIP_PAGE3_SRTP_TAG_32));
ast_str_append(&m_video, 0, "m=video %d %s", ast_sockaddr_port(&vdest),
- get_sdp_rtp_profile(p, v_a_crypto ? 1 : 0, p->vrtp));
+ ast_sdp_get_rtp_profile(v_a_crypto ? 1 : 0, p->vrtp,
+ ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)));
/* Build max bitrate string */
if (p->maxcallbitrate)
@@ -13223,9 +13185,11 @@
if (needtext) {
if (sipdebug_text)
ast_verbose("Lets set up the text sdp\n");
- get_crypto_attrib(p, p->tsrtp, &t_a_crypto);
+ t_a_crypto = crypto_get_attrib(p->tsrtp, p->dtls_cfg.enabled,
+ ast_test_flag(&p->flags[2], SIP_PAGE3_SRTP_TAG_32));
ast_str_append(&m_text, 0, "m=text %d %s", ast_sockaddr_port(&tdest),
- get_sdp_rtp_profile(p, t_a_crypto ? 1 : 0, p->trtp));
+ ast_sdp_get_rtp_profile(t_a_crypto ? 1 : 0, p->trtp,
+ ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)));
if (debug) { /* XXX should I use tdest below ? */
ast_verbose("Text is at %s\n", ast_sockaddr_stringify(&taddr));
}
@@ -13244,9 +13208,11 @@
/* We break with the "recommendation" and send our IP, in order that our
peer doesn't have to ast_gethostbyname() us */
- get_crypto_attrib(p, p->srtp, &a_crypto);
+ a_crypto = crypto_get_attrib(p->srtp, p->dtls_cfg.enabled,
+ ast_test_flag(&p->flags[2], SIP_PAGE3_SRTP_TAG_32));
ast_str_append(&m_audio, 0, "m=audio %d %s", ast_sockaddr_port(&dest),
- get_sdp_rtp_profile(p, a_crypto ? 1 : 0, p->rtp));
+ ast_sdp_get_rtp_profile(a_crypto ? 1 : 0, p->rtp,
+ ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)));
/* Now, start adding audio codecs. These are added in this order:
- First what was requested by the calling channel
@@ -26017,7 +25983,7 @@
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) || (p->t38.state == T38_REJECTED)) {
/* If this is not a re-invite or something to ignore - it's critical */
- if (p->srtp && !ast_test_flag(p->srtp, SRTP_CRYPTO_OFFER_OK)) {
+ if (p->srtp && !ast_test_flag(p->srtp, AST_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 {
@@ -33305,22 +33271,7 @@
} while (0));
}
-/* SRTP */
-static int setup_srtp(struct sip_srtp **srtp)
-{
- if (!ast_rtp_engine_srtp_is_registered()) {
- ast_debug(1, "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)
+static int process_crypto(struct sip_pvt *p, struct ast_rtp_instance *rtp, struct ast_sdp_srtp **srtp, const char *a)
{
struct ast_rtp_engine_dtls *dtls;
@@ -33333,26 +33284,27 @@
if (strncasecmp(a, "crypto:", 7)) {
return FALSE;
}
+ /* skip "crypto:" */
+ a += strlen("crypto:");
+
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) {
+ if (!(*srtp = ast_sdp_srtp_alloc())) {
return FALSE;
}
}
- if (!(*srtp)->crypto && !((*srtp)->crypto = sdp_crypto_setup())) {
+ if (!(*srtp)->crypto && !((*srtp)->crypto = ast_sdp_crypto_alloc())) {
return FALSE;
}
- if (sdp_crypto_process((*srtp)->crypto, a, rtp, *srtp) < 0) {
+ if (ast_sdp_crypto_process(rtp, *srtp, a) < 0) {
return FALSE;
}
-
- ast_set_flag(*srtp, SRTP_CRYPTO_OFFER_OK);
if ((dtls = ast_rtp_instance_get_dtls(rtp))) {
dtls->stop(rtp);
Modified: team/group/pimp_my_sip/channels/sip/include/sip.h
URL: http://svnview.digium.com/svn/asterisk/team/group/pimp_my_sip/channels/sip/include/sip.h?view=diff&rev=387097&r1=387096&r2=387097
==============================================================================
--- team/group/pimp_my_sip/channels/sip/include/sip.h (original)
+++ team/group/pimp_my_sip/channels/sip/include/sip.h Wed May 1 08:16:20 2013
@@ -1189,9 +1189,9 @@
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 */
+ struct ast_sdp_srtp *srtp; /*!< Structure to hold Secure RTP session data for audio */
+ struct ast_sdp_srtp *vsrtp; /*!< Structure to hold Secure RTP session data for video */
+ struct ast_sdp_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) */
Modified: team/group/pimp_my_sip/configs/res_sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/group/pimp_my_sip/configs/res_sip.conf.sample?view=diff&rev=387097&r1=387096&r2=387097
==============================================================================
--- team/group/pimp_my_sip/configs/res_sip.conf.sample (original)
+++ team/group/pimp_my_sip/configs/res_sip.conf.sample Wed May 1 08:16:20 2013
@@ -22,3 +22,5 @@
;rtp_ipv6=yes ; Force IPv6 for RTP transport
;rtp_symmetric=yes ; Enable symmetric RTP support
;use_ptime=yes ; Whether to use the ptime value received from the endpoint or not
+;media_encryption=no ; Options for media encryption are no, and sdes
+;use_avpf=no ; Whether to force usage of AVPF transport for this endpoint
Modified: team/group/pimp_my_sip/include/asterisk/res_sip.h
URL: http://svnview.digium.com/svn/asterisk/team/group/pimp_my_sip/include/asterisk/res_sip.h?view=diff&rev=387097&r1=387096&r2=387097
==============================================================================
--- team/group/pimp_my_sip/include/asterisk/res_sip.h (original)
+++ team/group/pimp_my_sip/include/asterisk/res_sip.h Wed May 1 08:16:20 2013
@@ -251,6 +251,17 @@
* Subsequent session refreshes will be sent no matter the session direction
*/
AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_INCOMING,
+};
+
+enum ast_sip_session_media_encryption {
+ /*! Invalid media encryption configuration */
+ AST_SIP_MEDIA_TRANSPORT_INVALID = 0,
+ /*! Do not allow any encryption of session media */
+ AST_SIP_MEDIA_ENCRYPT_NONE,
+ /*! Offer SDES-encrypted session media */
+ AST_SIP_MEDIA_ENCRYPT_SDES,
+ /*! Offer encrypted session media with datagram TLS key exchange */
+ AST_SIP_MEDIA_ENCRYPT_DTLS,
};
/*!
@@ -336,6 +347,10 @@
unsigned int send_rpid;
/*! Should unsolicited MWI be aggregated into a single NOTIFY? */
unsigned int aggregate_mwi;
+ /*! Do we use media encryption? what type? */
+ enum ast_sip_session_media_encryption media_encryption;
+ /*! Do we use AVPF exclusively for this endpoint? */
+ unsigned int use_avpf;
};
/*!
Modified: team/group/pimp_my_sip/include/asterisk/res_sip_session.h
URL: http://svnview.digium.com/svn/asterisk/team/group/pimp_my_sip/include/asterisk/res_sip_session.h?view=diff&rev=387097&r1=387096&r2=387097
==============================================================================
--- team/group/pimp_my_sip/include/asterisk/res_sip_session.h (original)
+++ team/group/pimp_my_sip/include/asterisk/res_sip_session.h Wed May 1 08:16:20 2013
@@ -26,6 +26,8 @@
#include "asterisk/channel.h"
/* Needed for ast_sockaddr struct */
#include "asterisk/netsock.h"
+/* Neeed for ast_sdp_srtp struct */
+#include "asterisk/sdp_srtp.h"
/* Forward declarations */
struct ast_sip_endpoint;
@@ -54,6 +56,8 @@
struct ast_sockaddr direct_media_addr;
/*! \brief SDP handler that setup the RTP */
struct ast_sip_session_sdp_handler *handler;
+ /*! \brief Holds SRTP information */
+ struct ast_sdp_srtp *srtp;
/*! \brief Stream is on hold */
unsigned int held:1;
/*! \brief Stream type this session media handles */
Added: team/group/pimp_my_sip/include/asterisk/sdp_srtp.h
URL: http://svnview.digium.com/svn/asterisk/team/group/pimp_my_sip/include/asterisk/sdp_srtp.h?view=auto&rev=387097
==============================================================================
--- team/group/pimp_my_sip/include/asterisk/sdp_srtp.h (added)
+++ team/group/pimp_my_sip/include/asterisk/sdp_srtp.h Wed May 1 08:16:20 2013
@@ -1,0 +1,125 @@
+/*
+ * 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_srtp.h
+ *
+ * \brief SRTP and SDP Security descriptions
+ *
+ * Specified in RFC 4568
+ * Specified in RFC 3711
+ *
+ * \author Mikael Magnusson <mikma at users.sourceforge.net>
+ */
+
+#ifndef _SDP_SRTP_H
+#define _SDP_SRTP_H
+
+#include <asterisk/rtp_engine.h>
+
+struct ast_sdp_crypto;
+
+/*! \brief structure for secure RTP audio */
+struct ast_sdp_srtp {
+ unsigned int flags;
+ struct ast_sdp_crypto *crypto;
+};
+
+/* SRTP flags */
+#define AST_SRTP_CRYPTO_OFFER_OK (1 << 1)
+#define AST_SRTP_CRYPTO_TAG_32 (1 << 2)
+#define AST_SRTP_CRYPTO_TAG_80 (1 << 3)
+
+/*!
+ * \brief allocate a ast_sdp_srtp structure
+ * \retval a new malloc'd ast_sdp_srtp structure on success
+ * \retval NULL on failure
+*/
+struct ast_sdp_srtp *ast_sdp_srtp_alloc(void);
+
+/*!
+ * \brief free a ast_sdp_srtp structure
+ * \param srtp a ast_sdp_srtp structure
+*/
+void ast_sdp_srtp_destroy(struct ast_sdp_srtp *srtp);
+
+/*! \brief Initialize an return an ast_sdp_crypto struct
+ *
+ * \details
+ * This function allocates a new ast_sdp_crypto struct and initializes its values
+ *
+ * \retval NULL on failure
+ * \retval a pointer to a new ast_sdp_crypto structure
+ */
+struct ast_sdp_crypto *ast_sdp_crypto_alloc(void);
+
+/*! \brief Destroy a previously allocated ast_sdp_crypto struct */
+void ast_sdp_crypto_destroy(struct ast_sdp_crypto *crypto);
+
+/*! \brief Parse the a=crypto line from SDP and set appropriate values on the
+ * ast_sdp_crypto struct.
+ *
+ * The attribute line should already have "a=crypto:" removed.
+ *
+ * \param p A valid ast_sdp_crypto struct
+ * \param attr the a:crypto line from SDP
+ * \param rtp The rtp instance associated with the SDP being parsed
+ * \param srtp SRTP structure
+ *
+ * \retval 0 success
+ * \retval nonzero failure
+ */
+int ast_sdp_crypto_process(struct ast_rtp_instance *rtp, struct ast_sdp_srtp *srtp, const char *attr);
+
+/*! \brief Generate an SRTP a=crypto offer
+ *
+ * \details
+ * The offer is stored on the ast_sdp_crypto struct in a_crypto
+ *
+ * \param p A valid ast_sdp_crypto struct
+ * \param taglen Length
+ *
+ * \retval 0 success
+ * \retval nonzero failure
+ */
+int ast_sdp_crypto_build_offer(struct ast_sdp_crypto *p, int taglen);
+
+
+/*! \brief Get the crypto attribute line for the srtp structure
+ *
+ * The attribute line does not contain the initial "a=crypto:" and does
+ * not terminate with "\r\n".
+ *
+ * \param srtp The ast_sdp_srtp structure for which to get an attribute line
+ * \param dtls_enabled Whether this connection is encrypted with datagram TLS
+ * \param default_taglen_32 Whether to default to a tag length of 32 instead of 80
+ *
+ * \retval An attribute line containing cryptographic information
+ * \retval NULL if the srtp structure does not require an attribute line containing crypto information
+ */
+const char *ast_sdp_srtp_get_attrib(struct ast_sdp_srtp *srtp, int dtls_enabled, int default_taglen_32);
+
+/*! \brief Get the RTP profile in use by a media session
+ *
+ * \param sdes_active Whether the media session is using SDES-SRTP
+ * \param instance The RTP instance associated with this media session
+ * \param using_avpf Whether the media session is using early feedback (AVPF)
+ *
+ * \retval A non-allocated string describing the profile in use (does not need to be freed)
+ */
+char *ast_sdp_get_rtp_profile(unsigned int sdes_active, struct ast_rtp_instance *instance, unsigned int using_avpf);
+#endif /* _SDP_CRYPTO_H */
Propchange: team/group/pimp_my_sip/include/asterisk/sdp_srtp.h
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: team/group/pimp_my_sip/include/asterisk/sdp_srtp.h
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: team/group/pimp_my_sip/include/asterisk/sdp_srtp.h
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: team/group/pimp_my_sip/main/sdp_srtp.c
URL: http://svnview.digium.com/svn/asterisk/team/group/pimp_my_sip/main/sdp_srtp.c?view=auto&rev=387097
==============================================================================
--- team/group/pimp_my_sip/main/sdp_srtp.c (added)
+++ team/group/pimp_my_sip/main/sdp_srtp.c Wed May 1 08:16:20 2013
@@ -1,0 +1,382 @@
+/*
+ * 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 ast_sdp_crypto.c
+ *
+ * \brief SRTP and SDP Security descriptions
+ *
+ * Specified in RFC 3711
+ * Specified in RFC 4568
+ *
+ * \author Mikael Magnusson <mikma at users.sourceforge.net>
+ */
+
+/*** MODULEINFO
+ <support_level>core</support_level>
+ ***/
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/options.h"
+#include "asterisk/utils.h"
+#include "asterisk/sdp_srtp.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 ast_sdp_srtp *ast_sdp_srtp_alloc(void)
+{
+ if (!ast_rtp_engine_srtp_is_registered()) {
+ ast_debug(1, "No SRTP module loaded, can't setup SRTP session.\n");
+ return NULL;
+ }
+
+ return ast_calloc(1, sizeof(struct ast_sdp_srtp));
+}
+
+void ast_sdp_srtp_destroy(struct ast_sdp_srtp *srtp)
+{
+ if (srtp->crypto) {
+ ast_sdp_crypto_destroy(srtp->crypto);
+ }
+ srtp->crypto = NULL;
+ ast_free(srtp);
+}
+
+struct ast_sdp_crypto {
+ char *a_crypto;
+ unsigned char local_key[SRTP_MASTER_LEN];
+ char *tag;
+ char local_key64[SRTP_MASTER_LEN64];
+ unsigned char remote_key[SRTP_MASTER_LEN];
+};
+
+static int set_crypto_policy(struct ast_srtp_policy *policy, int suite_val, const unsigned char *master_key, unsigned long ssrc, int inbound);
+
+void ast_sdp_crypto_destroy(struct ast_sdp_crypto *crypto)
+{
+ ast_free(crypto->a_crypto);
+ crypto->a_crypto = NULL;
+ ast_free(crypto->tag);
+ crypto->tag = NULL;
+ ast_free(crypto);
+}
+
+struct ast_sdp_crypto *ast_sdp_crypto_alloc(void)
+{
+ struct ast_sdp_crypto *p;
+ int key_len;
+ unsigned char remote_key[SRTP_MASTER_LEN];
+
+ if (!ast_rtp_engine_srtp_is_registered()) {
+ return NULL;
+ }
+
+ if (!(p = ast_calloc(1, sizeof(*p)))) {
+ return NULL;
+ }
+
+ if (res_srtp->get_random(p->local_key, sizeof(p->local_key)) < 0) {
+ ast_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 crypto_activate(struct ast_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;
+ }
+
+ /* Add the SRTP policies */
+ if (ast_rtp_instance_add_srtp_policy(rtp, remote_policy, local_policy)) {
+ ast_log(LOG_WARNING, "Could not set SRTP policies\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 ast_sdp_crypto_process(struct ast_rtp_instance *rtp, struct ast_sdp_srtp *srtp, const char *attr)
+{
+ char *str = 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 key_len = 0;
+ int suite_val = 0;
+ unsigned char remote_key[SRTP_MASTER_LEN];
+ int taglen = 0;
+ struct ast_sdp_crypto *crypto = srtp->crypto;
+
+ if (!ast_rtp_engine_srtp_is_registered()) {
+ return -1;
+ }
+
+ str = ast_strdupa(attr);
+
+ 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 (!ast_strlen_zero(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;
+ ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_80);
+ taglen = 80;
+ } else if (!strcmp(suite, "AES_CM_128_HMAC_SHA1_32")) {
+ suite_val = AST_AES_CM_128_HMAC_SHA1_32;
+ ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_32);
+ taglen = 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;
+ }
+
+ found = 1;
+ break;
+ }
+ }
+
+ if (!found) {
+ ast_log(LOG_NOTICE, "SRTP crypto offer not acceptable\n");
+ return -1;
+ }
+
+ if ((key_len = ast_base64decode(remote_key, key_salt, sizeof(remote_key))) != SRTP_MASTER_LEN) {
+ ast_log(LOG_WARNING, "SRTP descriptions key %d != %d\n", key_len, SRTP_MASTER_LEN);
+ return -1;
+ }
+
+ if (!memcmp(crypto->remote_key, remote_key, sizeof(crypto->remote_key))) {
+ ast_debug(1, "SRTP remote key unchanged; maintaining current policy\n");
+ ast_set_flag(srtp, AST_SRTP_CRYPTO_OFFER_OK);
+ return 0;
+ }
+ memcpy(crypto->remote_key, remote_key, sizeof(crypto->remote_key));
+
+ if (crypto_activate(crypto, suite_val, remote_key, rtp) < 0) {
+ return -1;
+ }
+
+ if (!crypto->tag) {
+ ast_log(LOG_DEBUG, "Accepting crypto tag %s\n", tag);
+ crypto->tag = ast_strdup(tag);
+ if (!crypto->tag) {
+ ast_log(LOG_ERROR, "Could not allocate memory for tag\n");
+ return -1;
+ }
+ }
+
+ /* Finally, rebuild the crypto line */
+ if (ast_sdp_crypto_build_offer(crypto, taglen)) {
+ return -1;
+ }
+
+ ast_set_flag(srtp, AST_SRTP_CRYPTO_OFFER_OK);
+ return 0;
+}
+
+int ast_sdp_crypto_build_offer(struct ast_sdp_crypto *p, int taglen)
+{
+ /* Rebuild the crypto line */
+ if (p->a_crypto) {
+ ast_free(p->a_crypto);
+ }
+
+ if (ast_asprintf(&p->a_crypto, "%s AES_CM_128_HMAC_SHA1_%i inline:%s",
+ p->tag ? p->tag : "1", taglen, p->local_key64) == -1) {
+ ast_log(LOG_ERROR, "Could not allocate memory for crypto line\n");
+ return -1;
+ }
+
+ ast_log(LOG_DEBUG, "Crypto line: a=crypto:%s\n", p->a_crypto);
+
+ return 0;
+}
+
+const char *ast_sdp_srtp_get_attrib(struct ast_sdp_srtp *srtp, int dtls_enabled, int default_taglen_32)
+{
+ int taglen = default_taglen_32 ? 32 : 80;
+
+ if (!srtp) {
+ return NULL;
+ }
+
+ /* Set encryption properties */
+ if (!srtp->crypto) {
+ srtp->crypto = ast_sdp_crypto_alloc();
+ }
+
+ if (dtls_enabled) {
+ /* If DTLS-SRTP is enabled the key details will be pulled from TLS */
+ return NULL;
+ }
+
+ /* set the key length based on INVITE or settings */
+ if (ast_test_flag(srtp, AST_SRTP_CRYPTO_TAG_80)) {
+ taglen = 80;
+ } else if (ast_test_flag(srtp, AST_SRTP_CRYPTO_TAG_32)) {
+ taglen = 32;
+ }
+
+ if (srtp->crypto && (ast_sdp_crypto_build_offer(srtp->crypto, taglen) >= 0)) {
+ return srtp->crypto->a_crypto;
+ }
+
+ ast_log(LOG_WARNING, "No SRTP key management enabled\n");
+ return NULL;
+}
+
+char *ast_sdp_get_rtp_profile(unsigned int sdes_active, struct ast_rtp_instance *instance, unsigned int using_avpf)
+{
+ struct ast_rtp_engine_dtls *dtls;
+
+ if ((dtls = ast_rtp_instance_get_dtls(instance)) && dtls->active(instance)) {
+ return using_avpf ? "UDP/TLS/RTP/SAVPF" : "UDP/TLS/RTP/SAVP";
+ } else {
+ if (using_avpf) {
+ return sdes_active ? "RTP/SAVPF" : "RTP/AVPF";
+ } else {
+ return sdes_active ? "RTP/SAVP" : "RTP/AVP";
+ }
+ }
+}
+
Propchange: team/group/pimp_my_sip/main/sdp_srtp.c
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: team/group/pimp_my_sip/main/sdp_srtp.c
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: team/group/pimp_my_sip/main/sdp_srtp.c
------------------------------------------------------------------------------
svn:mime-type = text/plain
Modified: team/group/pimp_my_sip/res/res_sip/sip_configuration.c
URL: http://svnview.digium.com/svn/asterisk/team/group/pimp_my_sip/res/res_sip/sip_configuration.c?view=diff&rev=387097&r1=387096&r2=387097
==============================================================================
--- team/group/pimp_my_sip/res/res_sip/sip_configuration.c (original)
+++ team/group/pimp_my_sip/res/res_sip/sip_configuration.c Wed May 1 08:16:20 2013
@@ -274,6 +274,23 @@
struct ast_sip_endpoint *endpoint = obj;
endpoint->id.tag = ast_strdup(var->value);
return endpoint->id.tag ? 0 : -1;
+}
+
+static int media_encryption_handler(const struct aco_option *opt, struct ast_variable *var, void *obj)
+{
+ struct ast_sip_endpoint *endpoint = obj;
+
+ if (!strcasecmp("no", var->value)) {
+ endpoint->media_encryption = AST_SIP_MEDIA_ENCRYPT_NONE;
+ } else if (!strcasecmp("sdes", var->value)) {
+ endpoint->media_encryption = AST_SIP_MEDIA_ENCRYPT_SDES;
+ /*} else if (!strcasecmp("dtls", var->value)) {
+ endpoint->media_encryption = AST_SIP_MEDIA_ENCRYPT_DTLS;*/
+ } else {
+ return -1;
+ }
+
+ return 0;
}
static void *sip_nat_hook_alloc(const char *name)
@@ -351,6 +368,8 @@
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "send_rpid", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, send_rpid));
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "mailboxes", "", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, mailboxes));
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "aggregate_mwi", "yes", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, aggregate_mwi));
+ ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "media_encryption", "no", media_encryption_handler, NULL, 0, 0);
+ ast_sorcery_object_field_register(sip_sorcery, "endpoint", "use_avpf", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, use_avpf));
if (ast_sip_initialize_sorcery_transport(sip_sorcery)) {
ast_log(LOG_ERROR, "Failed to register SIP transport support with sorcery\n");
Modified: team/group/pimp_my_sip/res/res_sip_sdp_rtp.c
URL: http://svnview.digium.com/svn/asterisk/team/group/pimp_my_sip/res/res_sip_sdp_rtp.c?view=diff&rev=387097&r1=387096&r2=387097
==============================================================================
--- team/group/pimp_my_sip/res/res_sip_sdp_rtp.c (original)
+++ team/group/pimp_my_sip/res/res_sip_sdp_rtp.c Wed May 1 08:16:20 2013
@@ -47,6 +47,7 @@
#include "asterisk/causes.h"
#include "asterisk/sched.h"
#include "asterisk/acl.h"
+#include "asterisk/sdp_srtp.h"
#include "asterisk/res_sip.h"
#include "asterisk/res_sip_session.h"
@@ -454,6 +455,93 @@
session_media->rtp, pref);
}
+/*! \brief figure out media transport encryption type from the media transport string */
+static enum ast_sip_session_media_encryption get_media_encryption_type(pj_str_t transport)
+{
+ RAII_VAR(char *, transport_str, ast_strndup(transport.ptr, transport.slen), ast_free);
+ if (strstr(transport_str, "UDP/TLS")) {
+ return AST_SIP_MEDIA_ENCRYPT_DTLS;
+ } else if (strstr(transport_str, "SAVP")) {
+ return AST_SIP_MEDIA_ENCRYPT_SDES;
+ } else {
+ return AST_SIP_MEDIA_ENCRYPT_NONE;
+ }
+}
+
+/*!
+ * \brief Checks whether the encryption offered in SDP is compatible with the endpoint's configuration
+ * \internal
+ *
+ * \param endpoint_encryption Media encryption configured for the endpoint
+ * \param stream pjmedia_sdp_media stream description
+ *
+ * \retval AST_SIP_MEDIA_TRANSPORT_INVALID on encryption mismatch
+ * \retval The encryption requested in the SDP
+ */
+static enum ast_sip_session_media_encryption check_endpoint_media_transport(
+ struct ast_sip_endpoint *endpoint,
+ const struct pjmedia_sdp_media *stream)
+{
+ enum ast_sip_session_media_encryption incoming_encryption;
+
+ if (endpoint->use_avpf) {
+ char transport_end = stream->desc.transport.ptr[stream->desc.transport.slen - 1];
+ if (transport_end != 'F') {
+ return AST_SIP_MEDIA_TRANSPORT_INVALID;
+ }
+ }
+
+ incoming_encryption = get_media_encryption_type(stream->desc.transport);
+ if (incoming_encryption == AST_SIP_MEDIA_ENCRYPT_DTLS) {
+ /* DTLS not yet supported */
+ return AST_SIP_MEDIA_TRANSPORT_INVALID;
+ }
+
+ if (incoming_encryption == endpoint->media_encryption) {
+ return incoming_encryption;
+ }
+
+ return AST_SIP_MEDIA_TRANSPORT_INVALID;
+}
+
+static int setup_sdes_srtp(struct ast_sip_session_media *session_media,
+ const struct pjmedia_sdp_media *stream)
+{
+ pjmedia_sdp_attr *attr;
+ RAII_VAR(char *, crypto_str, NULL, ast_free);
+
+ /* check the stream for the required crypto attribute */
+ attr = pjmedia_sdp_media_find_attr2(stream, "crypto", NULL);
+ if (!attr) {
+ return -1;
+ }
+
+ crypto_str = ast_strndup(attr->value.ptr, attr->value.slen);
+ if (!crypto_str) {
+ return -1;
+ }
+
+ if (!session_media->srtp) {
+ session_media->srtp = ast_sdp_srtp_alloc();
+ if (!session_media->srtp) {
+ return -1;
+ }
+ }
+
+ if (!session_media->srtp->crypto) {
+ session_media->srtp->crypto = ast_sdp_crypto_alloc();
+ if (!session_media->srtp->crypto) {
+ return -1;
+ }
+ }
+
+ if (ast_sdp_crypto_process(session_media->rtp, session_media->srtp, crypto_str)) {
+ return -1;
+ }
+
+ return 0;
+}
+
/*! \brief Function which negotiates an incoming media stream */
static int negotiate_incoming_sdp_stream(struct ast_sip_session *session, struct ast_sip_session_media *session_media,
const struct pjmedia_sdp_session *sdp, const struct pjmedia_sdp_media *stream)
@@ -461,12 +549,19 @@
char host[NI_MAXHOST];
RAII_VAR(struct ast_sockaddr *, addrs, NULL, ast_free_ptr);
enum ast_format_type media_type = stream_to_media_type(session_media->stream_type);
+ enum ast_sip_session_media_encryption incoming_encryption;
/* If no type formats have been configured reject this stream */
if (!ast_format_cap_has_type(session->endpoint->codecs, media_type)) {
return 0;
}
+ /* Ensure incoming transport is compatible with the endpoint's configuration */
+ incoming_encryption = check_endpoint_media_transport(session->endpoint, stream);
+ if (incoming_encryption == AST_SIP_MEDIA_TRANSPORT_INVALID) {
+ return -1;
+ }
+
ast_copy_pj_str(host, stream->conn ? &stream->conn->addr : &sdp->conn->addr, sizeof(host));
/* Ensure that the address provided is valid */
@@ -480,7 +575,40 @@
return -1;
}
+ if (incoming_encryption == AST_SIP_MEDIA_ENCRYPT_SDES
+ && setup_sdes_srtp(session_media, stream)) {
+ return -1;
+ }
+
return set_caps(session, session_media, stream);
+}
+
+static int add_crypto_to_stream(struct ast_sip_session *session,
+ struct ast_sip_session_media *session_media,
+ pj_pool_t *pool, pjmedia_sdp_media *media)
+{
+ pj_str_t stmp;
+ pjmedia_sdp_attr *attr;
+ const char *crypto_attribute;
+
+ if (!session_media->srtp && session->endpoint->media_encryption != AST_SIP_MEDIA_ENCRYPT_NONE) {
+ session_media->srtp = ast_sdp_srtp_alloc();
+ if (!session_media->srtp) {
+ return -1;
+ }
+ }
+
+ crypto_attribute = ast_sdp_srtp_get_attrib(session_media->srtp,
+ 0 /* DTLS can not be enabled for res_sip */,
+ 0 /* don't prefer 32byte tag length */);
+ if (!crypto_attribute) {
+ /* No crypto attribute to add */
[... 108 lines stripped ...]
More information about the asterisk-commits
mailing list