From 32d0faac9cb7576520d326ddd15d9fc7a0b47252 Mon Sep 17 00:00:00 2001
From: Terry Wilson <twilson@digium.com>
Date: Mon, 21 Nov 2011 21:09:59 +0000
Subject: [PATCH] Default to nat=yes; warn when nat in general and peer differ
It is possible to enumerate SIP usernames when the general and user/peer
nat settings differ in whether to respond to the port a request is sent
from or the port listed for responses in the Via header. In 1.4 and 1.6.2,
this would mean if one setting was nat=yes or nat=route and the other was
either nat=no or nat=never. In 1.8 and 10, this would mean when one was
nat=force_rport and the other was nat=no.
In order to address this problem, it was decided to switch the default
behavior to nat=yes/force_rport as it is the most commonly used option
and to strongly discourage setting nat per-peer/user when at all possible.
For more discussion of the issue, please see:
http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html
(closes issue ASTERISK-18862)
Review: https://reviewboard.asterisk.org/r/1591/
........
Merged revisions 345776 from http://svn.asterisk.org/svn/asterisk/branches/1.4
........
Merged revisions 345800 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
........
Merged revisions 345828 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 345830 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345831 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
CHANGES | 5 +++++
channels/chan_sip.c | 23 ++++++++++++++++++-----
configs/sip.conf.sample | 15 ++++++++-------
3 files changed, 31 insertions(+), 12 deletions(-)
diff --git a/CHANGES b/CHANGES
index bfc73010d2..5dc86231e0 100644
--- a/CHANGES
+++ b/CHANGES
@@ -323,6 +323,11 @@ PBX Core
SIP Changes
-----------
+ * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
+ now defaults to force_rport. It is very important that phones requiring nat=no be
+ specifically set as such instead of relying on the default setting. If at all
+ possible, all devices should have nat settings configured in the general section as
+ opposed to configuring nat per-device.
* Added preferred_codec_only option in sip.conf. This feature limits the joint
codecs sent in response to an INVITE to the single most preferred codec.
* Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index d5401156ad..2eecb0a6a8 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -27222,12 +27222,11 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask
}
} else if (!strcasecmp(v->name, "nat")) {
ast_set_flag(&mask[0], SIP_NAT_FORCE_RPORT);
+ ast_set_flag(&flags[0], SIP_NAT_FORCE_RPORT); /* Default to "force_rport" */
if (!strcasecmp(v->value, "no")) {
ast_clear_flag(&flags[0], SIP_NAT_FORCE_RPORT);
- } else if (!strcasecmp(v->value, "force_rport")) {
- ast_set_flag(&flags[0], SIP_NAT_FORCE_RPORT);
} else if (!strcasecmp(v->value, "yes")) {
- ast_set_flag(&flags[0], SIP_NAT_FORCE_RPORT);
+ /* We've already defaulted to force_rport */
ast_set_flag(&mask[1], SIP_PAGE2_SYMMETRICRTP);
ast_set_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP);
} else if (!strcasecmp(v->value, "comedia")) {
@@ -28381,6 +28380,18 @@ static void sip_set_default_format_capabilities(struct ast_format_cap *cap)
ast_format_cap_add(cap, ast_format_set(&tmp_fmt, AST_FORMAT_H263, 0));
}
+static void display_nat_warning(const char *cat, int reason, struct ast_flags *flags) {
+ int global_nat, specific_nat;
+
+ if (reason == CHANNEL_MODULE_LOAD && (specific_nat = ast_test_flag(&flags[0], SIP_NAT_FORCE_RPORT)) != (global_nat = ast_test_flag(&global_flags[0], SIP_NAT_FORCE_RPORT))) {
+ ast_log(LOG_WARNING, "!!! PLEASE NOTE: Setting 'nat' for a peer/user that differs from the global setting can make\n");
+ ast_log(LOG_WARNING, "!!! the name of that peer/user discoverable by an attacker. Replies for non-existent peers/users\n");
+ ast_log(LOG_WARNING, "!!! will be sent to a different port than replies for an existing peer/user. If at all possible,\n");
+ ast_log(LOG_WARNING, "!!! use the global 'nat' setting and do not set 'nat' per peer/user.\n");
+ ast_log(LOG_WARNING, "!!! (config category='%s' global force_rport='%s' peer/user force_rport='%s')\n", cat, AST_CLI_YESNO(global_nat), AST_CLI_YESNO(specific_nat));
+ }
+}
+
/*! \brief Re-read SIP.conf config file
\note This function reloads all config data, except for
active peers (with registrations). They will only
@@ -28608,8 +28619,9 @@ static int reload_config(enum channelreloadreason reason)
ast_copy_string(default_mohinterpret, DEFAULT_MOHINTERPRET, sizeof(default_mohinterpret));
ast_copy_string(default_mohsuggest, DEFAULT_MOHSUGGEST, sizeof(default_mohsuggest));
ast_copy_string(default_vmexten, DEFAULT_VMEXTEN, sizeof(default_vmexten));
- ast_set_flag(&global_flags[0], SIP_DTMF_RFC2833); /*!< Default DTMF setting: RFC2833 */
- ast_set_flag(&global_flags[0], SIP_DIRECT_MEDIA); /*!< Allow re-invites */
+ ast_set_flag(&global_flags[0], SIP_DTMF_RFC2833); /*!< Default DTMF setting: RFC2833 */
+ ast_set_flag(&global_flags[0], SIP_DIRECT_MEDIA); /*!< Allow re-invites */
+ ast_set_flag(&global_flags[0], SIP_NAT_FORCE_RPORT); /*!< Default to nat=force_rport */
ast_copy_string(default_engine, DEFAULT_ENGINE, sizeof(default_engine));
ast_copy_string(default_parkinglot, DEFAULT_PARKINGLOT, sizeof(default_parkinglot));
@@ -29394,6 +29406,7 @@ static int reload_config(enum channelreloadreason reason)
}
peer = build_peer(cat, ast_variable_browse(cfg, cat), NULL, 0, 0);
if (peer) {
+ display_nat_warning(cat, reason, &peer->flags[0]);
ao2_t_link(peers, peer, "link peer into peers table");
if ((peer->type & SIP_TYPE_PEER) && !ast_sockaddr_isnull(&peer->addr)) {
ao2_t_link(peers_by_ip, peer, "link peer into peers_by_ip table");
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 3c77a88be9..cb492161e5 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -824,6 +824,14 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; for their media streams is not actual port number that will be used on the nearer
; side of the NAT.
;
+; IT IS IMPORTANT TO NOTE that if the nat setting in the general section differs from
+; the nat setting in a peer definition, then the peer username will be discoverable
+; by outside parties as Asterisk will respond to different ports for defined and
+; undefined peers. For this reason it is recommended to ONLY DEFINE NAT SETTINGS IN THE
+; GENERAL SECTION. Specifically, if nat=force_rport in one section and nat=no in the
+; other, then valid users with settings differing from those in the general section will
+; be discoverable.
+;
; In addition to these settings, Asterisk *always* uses 'symmetric RTP' mode as defined by
; RFC 4961; Asterisk will always send RTP packets from the same port number it expects
; to receive them on.
@@ -1212,12 +1220,10 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
type=friend
[natted-phone](!,basic-options) ; another template inheriting basic-options
- nat=yes
directmedia=no
host=dynamic
[public-phone](!,basic-options) ; another template inheriting basic-options
- nat=no
directmedia=yes
[my-codecs](!) ; a template for my preferred codecs
@@ -1257,7 +1263,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;description=Courtesy Phone ; Description of the peer. Shown when doing 'sip show peers'.
;host=192.168.0.23 ; we have a static but private IP address
; No registration allowed
-;nat=no ; there is not NAT between phone and Asterisk
;directmedia=yes ; allow RTP voice traffic to bypass Asterisk
;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
@@ -1287,7 +1292,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;regexten=1234 ; When they register, create extension 1234
;callerid="Jane Smith" <5678>
;host=dynamic ; This device needs to register
-;nat=yes ; X-Lite is behind a NAT router
;directmedia=no ; Typically set to NO if behind NAT
;disallow=all
;allow=gsm ; GSM consumes far less bandwidth than ulaw
@@ -1361,9 +1365,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;type=friend
;secret=blah
;qualify=200 ; Qualify peer is no more than 200ms away
-;nat=yes ; This phone may be natted
- ; Send SIP and RTP to the IP address that packet is
- ; received from instead of trusting SIP headers
;host=dynamic ; This device registers with us
;directmedia=no ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go directly from
--
GitLab