[svn-commits] jrose: trunk r412759 - in /trunk: ./ channels/ channels/sip/include/ configs/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Apr 21 11:20:43 CDT 2014
Author: jrose
Date: Mon Apr 21 11:20:32 2014
New Revision: 412759
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=412759
Log:
chan_sip: Add sendrpid trust options
In r411189, some behavior was changed which made sendrpid behavior
act in a more trusting manner by sending full user data for peers
set with private caller presence in P-Asserted-Identity headers.
Since this changed long time expected behaviors, we decided to pull
that patch when that was pointed out by the community. Instead, this
patch provides a trust_id_outbound setting which will expose the data
per RFC-3325 if set to 'yes' and simply not send the PAI/RPID headers
at all if set to 'no'. By default trust_id_outbound will be set to
'legacy' which will preserve the behavior prior to these patches.
Extra special thanks to Walter Doekes for providing advice and
feedback.
(closes issue AST-1301)
(closes issue ASTERISK-19465)
Reported by: Krzysztof Chmielewski
Review: https://reviewboard.asterisk.org/r/3447/
........
Merged revisions 412744 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 412746 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 412747 from http://svn.asterisk.org/svn/asterisk/branches/12
Modified:
trunk/ (props changed)
trunk/CHANGES
trunk/channels/chan_sip.c
trunk/channels/sip/include/sip.h
trunk/configs/sip.conf.sample
Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-12-merged' - no diff available.
Modified: trunk/CHANGES
URL: http://svnview.digium.com/svn/asterisk/trunk/CHANGES?view=diff&rev=412759&r1=412758&r2=412759
==============================================================================
--- trunk/CHANGES (original)
+++ trunk/CHANGES Mon Apr 21 11:20:32 2014
@@ -24,6 +24,23 @@
URIs in that they must be stopped manually and will continue to occupy
a channel's ARI control queue until they are stopped. They also can not
be rewound or fastforwarded.
+
+chan_sip
+-----------
+ * SIP peers can now specify 'trust_id_outbound' which affects RPID/PAI
+ fields for prohibited callingpres information. Values are legacy, no, and
+ yes. By default, legacy is used.
+ trust_id_outbound=legacy: behavior remains the same as 1.8.26.1 - When
+ dealing with prohibited callingpres, RPID/PAI headers are created for both
+ sendrpid=pai and sendrpid=rpid are appended, but the data is anonymized.
+ When sendrpid=rpid, only the remote party's domain is anonymized.
+ trust_id_outbound=no: when dealing with prohibited callingpres, RPID/PAI
+ headers are not sent.
+ trust_id_outbound=yes: RPID/PAI headers are applied with the full
+ remote party information in tact even for prohibited callingpres
+ information. In the case of PAI, a Privacy: id header will be appended for
+ prohibited calling information to communicate that the private information
+ should not be relayed to untrusted parties.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 12.1.0 to Asterisk 12.2.0 ------------
@@ -1498,8 +1515,8 @@
a chan_pjsip configuration, but it is expected that configuration beyond
what the script provides will be needed.
-
------------------------------------------------------------------------------
+>>>>>>> .merge-right.r412746
--- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
------------------------------------------------------------------------------
Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=412759&r1=412758&r2=412759
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Mon Apr 21 11:20:32 2014
@@ -12645,15 +12645,39 @@
}
lid_pres = ast_party_id_presentation(&connected_id);
- fromdomain = S_OR(p->fromdomain, ast_sockaddr_stringify_host_remote(&p->ourip));
+ if (((lid_pres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) &&
+ (ast_test_flag(&p->flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND) == SIP_PAGE2_TRUST_ID_OUTBOUND_NO)) {
+ /* If pres is not allowed and we don't trust the peer, we don't apply an RPID header */
+ return 0;
+ }
+
+ fromdomain = p->fromdomain;
+ if (!fromdomain ||
+ ((ast_test_flag(&p->flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND) == SIP_PAGE2_TRUST_ID_OUTBOUND_YES) &&
+ !strcmp("anonymous.invalid", fromdomain))) {
+ /* If the fromdomain is NULL or if it was set to anonymous.invalid due to privacy settings and we trust the peer,
+ * use the host IP address */
+ fromdomain = ast_sockaddr_stringify_host_remote(&p->ourip);
+ }
lid_num = ast_uri_encode(lid_num, tmp2, sizeof(tmp2), ast_uri_sip_user);
if (ast_test_flag(&p->flags[0], SIP_SENDRPID_PAI)) {
- if ((lid_pres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) {
- ast_str_set(&tmp, -1, "%s", anonymous_string);
+ if (ast_test_flag(&p->flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND) != SIP_PAGE2_TRUST_ID_OUTBOUND_LEGACY) {
+ /* trust_id_outbound = yes - Always give full information even if it's private, but append a privacy header
+ * When private data is included */
+ ast_str_set(&tmp, -1, "\"%s\" <sip:%s@%s>", lid_name, lid_num, fromdomain);
+ if ((lid_pres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) {
+ add_header(req, "Privacy", "id");
+ }
} else {
- ast_str_set(&tmp, -1, "\"%s\" <sip:%s@%s>", lid_name, lid_num, fromdomain);
+ /* trust_id_outbound = legacy - behave in a non RFC-3325 compliant manner and send anonymized data when
+ * when handling private data. */
+ if ((lid_pres & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED) {
+ ast_str_set(&tmp, -1, "\"%s\" <sip:%s@%s>", lid_name, lid_num, fromdomain);
+ } else {
+ ast_str_set(&tmp, -1, "%s", anonymous_string);
+ }
}
add_header(req, "P-Asserted-Identity", ast_str_buffer(tmp));
} else {
@@ -19641,6 +19665,18 @@
static const char *allowoverlap2str(int mode)
{
return map_x_s(allowoverlapstr, mode, "<error>");
+}
+
+static const struct _map_x_s trust_id_outboundstr[] = {
+ { SIP_PAGE2_TRUST_ID_OUTBOUND_LEGACY, "Legacy" },
+ { SIP_PAGE2_TRUST_ID_OUTBOUND_NO, "No" },
+ { SIP_PAGE2_TRUST_ID_OUTBOUND_YES, "Yes" },
+ { -1, NULL }, /* terminator */
+};
+
+static const char *trust_id_outbound2str(int mode)
+{
+ return map_x_s(trust_id_outboundstr, mode, "<error>");
}
/*! \brief Destroy disused contexts between reloads
@@ -20310,6 +20346,7 @@
ast_cli(fd, " Path : %s\n", ast_str_buffer(path));
ast_free(path);
}
+ ast_cli(fd, " TrustIDOutbnd: %s\n", trust_id_outbound2str(ast_test_flag(&peer->flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND)));
ast_cli(fd, " Subscriptions: %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)));
ast_cli(fd, " Overlap dial : %s\n", allowoverlap2str(ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWOVERLAP)));
if (peer->outboundproxy)
@@ -29863,6 +29900,19 @@
} else if (!strcasecmp(v->name, "rpid_immediate")) {
ast_set_flag(&mask[1], SIP_PAGE2_RPID_IMMEDIATE);
ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_RPID_IMMEDIATE);
+ } else if (!strcasecmp(v->name, "trust_id_outbound")) {
+ ast_set_flag(&mask[1], SIP_PAGE2_TRUST_ID_OUTBOUND);
+ ast_clear_flag(&flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND);
+ if (!strcasecmp(v->value, "legacy")) {
+ ast_set_flag(&flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND_LEGACY);
+ } else if (ast_true(v->value)) {
+ ast_set_flag(&flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND_YES);
+ } else if (ast_false(v->value)) {
+ ast_set_flag(&flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND_NO);
+ } else {
+ ast_log(LOG_WARNING, "Unknown trust_id_outbound mode '%s' on line %d, using legacy\n", v->value, v->lineno);
+ ast_set_flag(&flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND_LEGACY);
+ }
} else if (!strcasecmp(v->name, "g726nonstandard")) {
ast_set_flag(&mask[0], SIP_G726_NONSTANDARD);
ast_set2_flag(&flags[0], ast_true(v->value), SIP_G726_NONSTANDARD);
Modified: trunk/channels/sip/include/sip.h
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sip/include/sip.h?view=diff&rev=412759&r1=412758&r2=412759
==============================================================================
--- trunk/channels/sip/include/sip.h (original)
+++ trunk/channels/sip/include/sip.h Mon Apr 21 11:20:32 2014
@@ -361,13 +361,18 @@
#define SIP_PAGE2_HAVEPEERCONTEXT (1 << 28) /*< Are we associated with a configured peer context? */
#define SIP_PAGE2_USE_SRTP (1 << 29) /*!< DP: Whether we should offer (only) SRTP */
+#define SIP_PAGE2_TRUST_ID_OUTBOUND (3 << 30) /*!< DP: Do we trust the peer with private presence information? */
+#define SIP_PAGE2_TRUST_ID_OUTBOUND_LEGACY (0 << 30) /*!< Legacy, Do not provide private presence information, but include PAI/RPID when private */
+#define SIP_PAGE2_TRUST_ID_OUTBOUND_NO (1 << 30) /*!< No, Do not provide private presence information, do not include PAI/RPID when private */
+#define SIP_PAGE2_TRUST_ID_OUTBOUND_YES (2 << 30) /*!< Yes, provide private presence information in PAI/RPID headers */
+
#define SIP_PAGE2_FLAGS_TO_COPY \
(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_IGNORESDPVERSION | \
SIP_PAGE2_VIDEOSUPPORT | SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | \
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_USE_SRTP)
+ SIP_PAGE2_Q850_REASON | SIP_PAGE2_HAVEPEERCONTEXT | SIP_PAGE2_USE_SRTP | SIP_PAGE2_TRUST_ID_OUTBOUND)
#define SIP_PAGE3_SNOM_AOC (1 << 0) /*!< DPG: Allow snom aoc messages */
Modified: trunk/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/trunk/configs/sip.conf.sample?view=diff&rev=412759&r1=412758&r2=412759
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Mon Apr 21 11:20:32 2014
@@ -350,6 +350,17 @@
; transmit such UPDATE messages to it, then you must enable this option.
; Otherwise, we will have to wait until we can send a reinvite to
; transmit the information.
+;trust_id_outbound = no ; Controls whether or not we trust this peer with private identity
+ ; information (when the remote party has callingpres=prohib or equivalent).
+ ; no - RPID/PAI headers will not be included for private peer information
+ ; yes - RPID/PAI headers will include the private peer information. Privacy
+ ; requirements will be indicated in a Privacy header for sendrpid=pai
+ ; legacy - RPID/PAI will be included for private peer information. In the
+ ; case of sendrpid=pai, private data that would be included in them
+ ; will be anonymized. For sendrpid=rpid, private data may be included
+ ; but the remote party's domain will be anonymized. The way legacy
+ ; behaves may violate RFC-3325, but it follows historic behavior.
+ ; This option is set to 'legacy' by default
;prematuremedia=no ; Some ISDN links send empty media frames before
; the call is in ringing or progress state. The SIP
; channel will then send 183 indicating early media
@@ -1219,6 +1230,7 @@
; autoframing
; insecure
; trustrpid
+; trust_id_outbound
; progressinband
; promiscredir
; useclientcode
@@ -1431,7 +1443,8 @@
;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
;allow=g729 ; Pass-thru only unless g729 license obtained
;callingpres=allowed_passed_screen ; Set caller ID presentation
- ; See README.callingpres for more information
+ ; See function CALLERPRES documentation for possible
+ ; values.
;[xlite1]
; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
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