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Merged revisions 336936 via svnmerge from

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https://origsvn.digium.com/svn/asterisk/branches/10

........
  r336936 | irroot | 2011-09-20 18:51:59 +0200 (Tue, 20 Sep 2011) | 14 lines
  
  
  Allow Setting Auth Tag Bit length Based on invite or config option
  
  Update the SIP SRTP API to allow use of 32 or 80 bit taglen.
  Curently only 80 bit is supported.
  
  The outgoing invite will use the taglen of the incoming invite preventing
  one-way audio.
  
  (Closes issue ASTERISK-17895)
  
  Review: https://reviewboard.asterisk.org/r/1173/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336937 65c4cc65-6c06-0410-ace0-fbb531ad65f3
certified/11.2
Gregory Nietsky 14 years ago
parent 14d3f891e0
commit 8493c46308
7 changed files with 36 additions and 13 deletions
Show Stats Download Patch File Download Diff File

1
CHANGES
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@ -213,6 +213,7 @@ res_fax
SIP Changes
-----------
* Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
* Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
Queue changes
-------------

25
channels/chan_sip.c
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@ -11127,14 +11127,25 @@ static void get_our_media_address(struct sip_pvt *p, int needvideo, int needtext
}
}
static void get_crypto_attrib(struct sip_srtp *srtp, const char **a_crypto)
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 (srtp->crypto && (sdp_crypto_offer(srtp->crypto) >= 0)) {
/* 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);
}
@ -11302,7 +11313,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
/* 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->vsrtp, &v_a_crypto);
get_crypto_attrib(p, p->vsrtp, &v_a_crypto);
ast_str_append(&m_video, 0, "m=video %d RTP/%s", ast_sockaddr_port(&vdest),
v_a_crypto ? "SAVP" : "AVP");
@ -11319,7 +11330,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
if (needtext) {
if (sipdebug_text)
ast_verbose("Lets set up the text sdp\n");
get_crypto_attrib(p->tsrtp, &t_a_crypto);
get_crypto_attrib(p, p->tsrtp, &t_a_crypto);
ast_str_append(&m_text, 0, "m=text %d RTP/%s", ast_sockaddr_port(&tdest),
t_a_crypto ? "SAVP" : "AVP");
if (debug) { /* XXX should I use tdest below ? */
@ -11332,7 +11343,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
/* 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->srtp, &a_crypto);
get_crypto_attrib(p, p->srtp, &a_crypto);
ast_str_append(&m_audio, 0, "m=audio %d RTP/%s", ast_sockaddr_port(&dest),
a_crypto ? "SAVP" : "AVP");
@ -27701,6 +27712,8 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, str
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, "encryption_taglen")) {
ast_set2_flag(&peer->flags[2], !strcasecmp(v->value, "32"), SIP_PAGE3_SRTP_TAG_32);
} else if (!strcasecmp(v->name, "snom_aoc_enabled")) {
ast_set2_flag(&peer->flags[2], ast_true(v->value), SIP_PAGE3_SNOM_AOC);
}
@ -29647,7 +29660,7 @@ static int process_crypto(struct sip_pvt *p, struct ast_rtp_instance *rtp, struc
return FALSE;
}
if (sdp_crypto_process((*srtp)->crypto, a, rtp) < 0) {
if (sdp_crypto_process((*srtp)->crypto, a, rtp, *srtp) < 0) {
return FALSE;
}

6
channels/sip/include/sdp_crypto.h
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@ -31,6 +31,7 @@
#include <asterisk/rtp_engine.h>
struct sdp_crypto;
struct sip_srtp;
/*! \brief Initialize an return an sdp_crypto struct
*
@ -51,11 +52,12 @@ void sdp_crypto_destroy(struct sdp_crypto *crypto);
* \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
* \param srtp SRTP structure
*
* \retval 0 success
* \retval nonzero failure
*/
int sdp_crypto_process(struct sdp_crypto *p, const char *attr, struct ast_rtp_instance *rtp);
int sdp_crypto_process(struct sdp_crypto *p, const char *attr, struct ast_rtp_instance *rtp, struct sip_srtp *srtp);
/*! \brief Generate an SRTP a=crypto offer
@ -68,7 +70,7 @@ int sdp_crypto_process(struct sdp_crypto *p, const char *attr, struct ast_rtp_in
* \retval 0 success
* \retval nonzero failure
*/
int sdp_crypto_offer(struct sdp_crypto *p);
int sdp_crypto_offer(struct sdp_crypto *p, int taglen);
/*! \brief Return the a_crypto value of the sdp_crypto struct

3
channels/sip/include/sip.h
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@ -354,9 +354,10 @@
#define SIP_PAGE3_SNOM_AOC (1 << 0) /*!< DPG: Allow snom aoc messages */
#define SIP_PAGE3_SRTP_TAG_32 (1 << 1) /*!< DP: Use a 32bit auth tag in INVITE not 80bit */
#define SIP_PAGE3_FLAGS_TO_COPY \
(SIP_PAGE3_SNOM_AOC)
(SIP_PAGE3_SNOM_AOC | SIP_PAGE3_SRTP_TAG_32)
/*@}*/

2
channels/sip/include/srtp.h
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@ -34,6 +34,8 @@
#define SRTP_ENCR_OPTIONAL (1 << 1) /* SRTP encryption optional */
#define SRTP_CRYPTO_ENABLE (1 << 2)
#define SRTP_CRYPTO_OFFER_OK (1 << 3)
#define SRTP_CRYPTO_TAG_32 (1 << 4)
#define SRTP_CRYPTO_TAG_80 (1 << 5)
/*! \brief structure for secure RTP audio */
struct sip_srtp {

11
channels/sip/sdp_crypto.c
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@ -32,6 +32,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/options.h"
#include "asterisk/utils.h"
#include "include/sdp_crypto.h"
#include "include/srtp.h"
#define SRTP_MASTER_LEN 30
#define SRTP_MASTERKEY_LEN 16
@ -188,7 +189,7 @@ err:
return res;
}
int sdp_crypto_process(struct sdp_crypto *p, const char *attr, struct ast_rtp_instance *rtp)
int sdp_crypto_process(struct sdp_crypto *p, const char *attr, struct ast_rtp_instance *rtp, struct sip_srtp *srtp)
{
char *str = NULL;
char *tag = NULL;
@ -228,8 +229,10 @@ int sdp_crypto_process(struct sdp_crypto *p, const char *attr, struct ast_rtp_in
if (!strcmp(suite, "AES_CM_128_HMAC_SHA1_80")) {
suite_val = AST_AES_CM_128_HMAC_SHA1_80;
ast_set_flag(srtp, SRTP_CRYPTO_TAG_80);
} else if (!strcmp(suite, "AES_CM_128_HMAC_SHA1_32")) {
suite_val = AST_AES_CM_128_HMAC_SHA1_32;
ast_set_flag(srtp, SRTP_CRYPTO_TAG_32);
} else {
ast_log(LOG_WARNING, "Unsupported crypto suite: %s\n", suite);
return -1;
@ -283,16 +286,16 @@ int sdp_crypto_process(struct sdp_crypto *p, const char *attr, struct ast_rtp_in
return 0;
}
int sdp_crypto_offer(struct sdp_crypto *p)
int sdp_crypto_offer(struct sdp_crypto *p, int taglen)
{
char crypto_buf[128];
const char *crypto_suite = "AES_CM_128_HMAC_SHA1_80"; /* Crypto offer */
if (p->a_crypto) {
ast_free(p->a_crypto);
}
if (snprintf(crypto_buf, sizeof(crypto_buf), "a=crypto:1 %s inline:%s\r\n", crypto_suite, p->local_key64) < 1) {
if (snprintf(crypto_buf, sizeof(crypto_buf), "a=crypto:1 AES_CM_128_HMAC_SHA1_%i inline:%s\r\n",
taglen, p->local_key64) < 1) {
return -1;
}

1
configs/sip.conf.sample
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@ -904,6 +904,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;encryption=no ; Whether to offer SRTP encrypted media (and only SRTP encrypted media)
; on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if
; the peer does not support SRTP. Defaults to no.
;encryption_taglen=80 ; Set the auth tag length offered in the INVITE either 32/80 default 80
;----------------------------------------- REALTIME SUPPORT ------------------------
; For additional information on ARA, the Asterisk Realtime Architecture,

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