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Merged revisions 284950 via svnmerge from

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https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r284950 | dvossel | 2010-09-03 12:29:02 -0500 (Fri, 03 Sep 2010) | 14 lines
  
  authenticate OPTIONS requests just like we would an INVITE
  
  OPTIONS requests should be treated the same as an INVITE
  This includes authentication.  This patch adds the ability for
  incoming out of dialog OPTION requests to be authenticated
  before providing a response indicating whether an extension
  is available or not.  The authentication routine works the
  exact same way as it does for incoming INVITEs.  This means
  that if a peer has 'insecure=invite' in their peer definition,
  the same will be true for the processing of the OPTIONS request.
  
  Review: https://reviewboard.asterisk.org/r/881/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
10-digiumphones
David Vossel 15 years ago
parent 01aef13e0c
commit d17eded2e9
4 changed files with 47 additions and 14 deletions
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3
CHANGES
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@ -83,6 +83,9 @@ SIP Changes
available in device configurations as well as in the dial plan.
* Addition of the 'subscribe_network_change' option for turning on and off
res_stun_monitor module support in chan_sip.
* Addition of the 'auth_options_requests' option for turning on and off
authentication for OPTIONS requests in chan_sip.
IAX2 Changes
-----------

49
channels/chan_sip.c
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@ -1512,7 +1512,7 @@ static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req);
static int handle_request_message(struct sip_pvt *p, struct sip_request *req);
static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, int seqno, const char *e);
static void handle_request_info(struct sip_pvt *p, struct sip_request *req);
static int handle_request_options(struct sip_pvt *p, struct sip_request *req);
static int handle_request_options(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const char *e);
static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, int debug, int seqno, struct ast_sockaddr *addr, int *nounlock);
static int handle_request_notify(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, int seqno, const char *e);
static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual *current, struct sip_request *req, int seqno, int *nounlock);
@ -7054,6 +7054,7 @@ struct sip_pvt *sip_alloc(ast_string_field callid, struct ast_sockaddr *addr,
char *sent_by, *branch;
const char *cseq = get_header(req, "Cseq");
unsigned int seqno;
/* get branch parameter from initial Request that started this dialog */
get_viabranch(ast_strdupa(get_header(req, "Via")), &sent_by, &branch);
/* only store the branch if it begins with the magic prefix "z9hG4bK", otherwise
@ -7068,7 +7069,8 @@ struct sip_pvt *sip_alloc(ast_string_field callid, struct ast_sockaddr *addr,
if (!ast_strlen_zero(cseq) && (sscanf(cseq, "%30u", &seqno) == 1)) {
p->init_icseq = seqno;
}
set_socket_transport(&p->socket, req->socket.type); /* Later in ast_sip_ouraddrfor we need this to choose the right ip and port for the specific transport */
/* Later in ast_sip_ouraddrfor we need this to choose the right ip and port for the specific transport */
set_socket_transport(&p->socket, req->socket.type);
} else {
set_socket_transport(&p->socket, SIP_TRANSPORT_UDP);
}
@ -20500,19 +20502,10 @@ static int handle_request_notify(struct sip_pvt *p, struct sip_request *req, str
/*! \brief Handle incoming OPTIONS request
An OPTIONS request should be answered like an INVITE from the same UA, including SDP
*/
static int handle_request_options(struct sip_pvt *p, struct sip_request *req)
static int handle_request_options(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const char *e)
{
int res;
/*! XXX get_destination assumes we're already authenticated. This means that a request from
a known device (peer) will end up in the wrong context if this is out-of-dialog.
However, we want to handle OPTIONS as light as possible, so we might want to have
a configuration option whether we care or not. Some devices use this for testing
capabilities, which means that we need to match device to answer with proper
capabilities (including SDP).
\todo Fix handle_request_options device handling with optional authentication
(this needs to be fixed in 1.4 as well)
*/
struct sip_peer *authpeer = NULL; /* Matching Peer */
if (p->lastinvite) {
/* if this is a request in an active dialog, just confirm that the dialog exists. */
@ -20520,6 +20513,29 @@ static int handle_request_options(struct sip_pvt *p, struct sip_request *req)
return 0;
}
if (sip_cfg.auth_options_requests) {
/* Do authentication if this OPTIONS request began the dialog */
copy_request(&p->initreq, req);
set_pvt_allowed_methods(p, req);
res = check_user_full(p, req, SIP_OPTIONS, e, XMIT_UNRELIABLE, addr, &authpeer);
if (res == AUTH_CHALLENGE_SENT) {
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return 0;
}
if (res < 0) { /* Something failed in authentication */
if (res == AUTH_FAKE_AUTH) {
ast_log(LOG_NOTICE, "Sending fake auth rejection for device %s\n", get_header(req, "From"));
transmit_fake_auth_response(p, SIP_OPTIONS, req, XMIT_UNRELIABLE);
} else {
ast_log(LOG_NOTICE, "Failed to authenticate device %s\n", get_header(req, "From"));
transmit_response(p, "403 Forbidden", req);
}
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return 0;
}
}
/* must go through authentication before getting here */
res = (get_destination(p, req, NULL) == SIP_GET_DEST_EXTEN_FOUND ? 0 : -1);
build_contact(p);
@ -23546,7 +23562,7 @@ static int handle_incoming(struct sip_pvt *p, struct sip_request *req, struct as
/* Handle various incoming SIP methods in requests */
switch (p->method) {
case SIP_OPTIONS:
res = handle_request_options(p, req);
res = handle_request_options(p, req, addr, e);
break;
case SIP_INVITE:
res = handle_request_invite(p, req, debug, seqno, addr, recount, e, nounlock);
@ -26390,6 +26406,7 @@ static int reload_config(enum channelreloadreason reason)
sip_cfg.notifyhold = FALSE; /*!< Keep track of hold status for a peer */
sip_cfg.directrtpsetup = FALSE; /* Experimental feature, disabled by default */
sip_cfg.alwaysauthreject = DEFAULT_ALWAYSAUTHREJECT;
sip_cfg.auth_options_requests = 1;
sip_cfg.allowsubscribe = FALSE;
sip_cfg.disallowed_methods = SIP_UNKNOWN;
sip_cfg.contact_ha = NULL; /* Reset the contact ACL */
@ -26630,6 +26647,10 @@ static int reload_config(enum channelreloadreason reason)
}
} else if (!strcasecmp(v->name, "alwaysauthreject")) {
sip_cfg.alwaysauthreject = ast_true(v->value);
} else if (!strcasecmp(v->name, "auth_options_requests")) {
if (ast_false(v->value)) {
sip_cfg.auth_options_requests = 0;
}
} else if (!strcasecmp(v->name, "mohinterpret")) {
ast_copy_string(default_mohinterpret, v->value, sizeof(default_mohinterpret));
} else if (!strcasecmp(v->name, "mohsuggest")) {

1
channels/sip/include/sip.h
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@ -674,6 +674,7 @@ struct sip_settings {
int srvlookup; /*!< SRV Lookup on or off. Default is on */
int allowguest; /*!< allow unauthenticated peers to connect? */
int alwaysauthreject; /*!< Send 401 Unauthorized for all failing requests */
int auth_options_requests; /*!< Authenticate OPTIONS requests */
int compactheaders; /*!< send compact sip headers */
int allow_external_domains; /*!< Accept calls to external SIP domains? */
int callevents; /*!< Whether we send manager events or not */

8
configs/sip.conf.sample
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@ -370,6 +370,14 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; the ability of an attacker to scan for valid SIP usernames.
; This option is set to "yes" by default.
;auth_options_requests = no ; sip OPTIONS requests should be treated the exact same as
; an INVITE, this includes performing authentication. By default
; OPTIONS requests are authenticated, however this option allows
; OPTION requests to proceed unauthenticated in order to increase
; performance. This may be desirable if OPTIONS are only used to
; qualify the availabilty of the endpoint/extension. Disabling
; this option is not recommended.
;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
; order instead of RFC3551 packing order (this is required
; for Sipura and Grandstream ATAs, among others). This is

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