[svn-commits] tilghman: branch 1.6.0 r135127 - in /branches/1.6.0: ./ channels/ configs/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Aug 1 11:44:33 CDT 2008
Author: tilghman
Date: Fri Aug 1 11:44:32 2008
New Revision: 135127
URL: http://svn.digium.com/view/asterisk?view=rev&rev=135127
Log:
Merged revisions 135126 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
........
r135126 | tilghman | 2008-08-01 11:39:51 -0500 (Fri, 01 Aug 2008) | 9 lines
SIP should use the transport type set in the Moved Temporarily for the next
invite.
(closes issue #11843)
Reported by: pestermann
Patches:
20080723__issue11843_302_ignores_transport_16branch.diff uploaded by bbryant (license 36)
20080723__issue11843_302_ignores_transport_trunk.diff uploaded by bbryant (license 36)
Tested by: pabelanger
........
Modified:
branches/1.6.0/ (props changed)
branches/1.6.0/channels/chan_sip.c
branches/1.6.0/configs/sip.conf.sample
Propchange: branches/1.6.0/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.
Modified: branches/1.6.0/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.6.0/channels/chan_sip.c?view=diff&rev=135127&r1=135126&r2=135127
==============================================================================
--- branches/1.6.0/channels/chan_sip.c (original)
+++ branches/1.6.0/channels/chan_sip.c Fri Aug 1 11:44:32 2008
@@ -1821,6 +1821,37 @@
static int sip_refer_allocate(struct sip_pvt *p);
static void ast_quiet_chan(struct ast_channel *chan);
static int attempt_transfer(struct sip_dual *transferer, struct sip_dual *target);
+/*!
+ * \brief generic function for determining if a correct transport is being
+ * used to contact a peer
+ *
+ * this is done as a macro so that the "tmpl" var can be passed either a
+ * sip_request or a sip_peer
+ */
+#define check_request_transport(peer, tmpl) ({ \
+ int ret = 0; \
+ if (peer->socket.type == tmpl->socket.type) \
+ ; \
+ else if (!(peer->transports & tmpl->socket.type)) {\
+ ast_log(LOG_ERROR, \
+ "'%s' is not a valid transport for '%s'. we only use '%s'! ending call.\n", \
+ get_transport(tmpl->socket.type), peer->name, get_transport_list(peer) \
+ ); \
+ ret = 1; \
+ } else if (peer->socket.type & SIP_TRANSPORT_TLS) { \
+ ast_log(LOG_WARNING, \
+ "peer '%s' HAS NOT USED (OR SWITCHED TO) TLS in favor of '%s' (but this was allowed in sip.conf)!\n", \
+ peer->name, get_transport(tmpl->socket.type) \
+ ); \
+ } else { \
+ ast_debug(1, \
+ "peer '%s' has contacted us over %s even though we prefer %s.\n", \
+ peer->name, get_transport(tmpl->socket.type), get_transport(peer->socket.type) \
+ ); \
+ }\
+ (ret); \
+})
+
/*--- Device monitoring and Device/extension state/event handling */
static int cb_extensionstate(char *context, char* exten, int state, void *data);
@@ -3956,6 +3987,11 @@
*/
static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
{
+ /* this checks that the dialog is contacting the peer on a valid
+ * transport type based on the peers transport configuration,
+ * otherwise, this function bails out */
+ if (dialog->socket.type && check_request_transport(peer, dialog))
+ return -1;
copy_socket_data(&dialog->socket, &peer->socket);
if ((peer->addr.sin_addr.s_addr || peer->defaddr.sin_addr.s_addr) &&
@@ -4102,10 +4138,6 @@
int res = create_addr_from_peer(dialog, peer);
unref_peer(peer);
return res;
- } else {
- /* Setup default parameters for this dialog's socket. Currently we only support regular UDP SIP as the default */
- dialog->socket.type = SIP_TRANSPORT_UDP;
- dialog->socket.port = bindaddr.sin_port;
}
ast_string_field_set(dialog, tohost, peername);
@@ -4124,7 +4156,12 @@
*/
hostn = peername;
- portno = port ? atoi(port) : (dialog->socket.type & SIP_TRANSPORT_TLS) ? STANDARD_TLS_PORT : STANDARD_SIP_PORT;
+ if (!dialog->socket.type)
+ dialog->socket.type = SIP_TRANSPORT_UDP;
+ if (ast_strlen_zero(port) || sscanf(port, "%u", &portno) != 1) {
+ portno = dialog->socket.type & SIP_TRANSPORT_TLS ?
+ STANDARD_TLS_PORT : STANDARD_SIP_PORT;
+ }
if (global_srvlookup) {
char service[MAXHOSTNAMELEN];
int tportno;
@@ -4180,7 +4217,7 @@
struct sip_pvt *p = ast->tech_pvt; /* chan is locked, so the reference cannot go away */
struct varshead *headp;
struct ast_var_t *current;
- const char *referer = NULL; /* SIP referrer */
+ const char *referer = NULL; /* SIP referrer */
if ((ast->_state != AST_STATE_DOWN) && (ast->_state != AST_STATE_RESERVED)) {
ast_log(LOG_WARNING, "sip_call called on %s, neither down nor reserved\n", ast->name);
@@ -4211,9 +4248,8 @@
p->t38.state = T38_LOCAL_DIRECT;
ast_debug(1, "T38State change to %d on channel %s\n", p->t38.state, ast->name);
}
-
- }
-
+ }
+
res = 0;
ast_set_flag(&p->flags[0], SIP_OUTGOING);
@@ -10452,27 +10488,10 @@
break;
}
- if (peer->socket.type != req->socket.type ) {
- if (!(peer->transports & req->socket.type)) {
- ast_log(LOG_ERROR,
- "peer '%s' has contacted us over %s, but we only accept '%s' for this peer! ending call.\n",
- peer->name, get_transport(req->socket.type), get_transport_list(peer)
- );
-
- ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
- transmit_response_with_date(p, "403 Forbidden", req);
- res = AUTH_BAD_TRANSPORT;
- } else if (peer->socket.type & SIP_TRANSPORT_TLS) {
- ast_log(LOG_WARNING,
- "peer '%s' HAS STOPPED USING TLS in favor of '%s' (but this was allowed in sip.conf)!\n",
- peer->name, get_transport(req->socket.type)
- );
- } else {
- ast_log(LOG_DEBUG,
- "peer '%s' has contacted us over %s even though we prefer %s.\n",
- peer->name, get_transport(req->socket.type), get_transport(peer->socket.type)
- );
- }
+ if (check_request_transport(peer, req)) {
+ ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
+ transmit_response_with_date(p, "403 Forbidden", req);
+ res = AUTH_BAD_TRANSPORT;
}
}
}
@@ -14527,13 +14546,41 @@
static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req)
{
char tmp[SIPBUFSIZE];
- char *s, *e, *t;
+ char *s, *e, *t, *trans;
char *domain;
+ enum sip_transport transport = SIP_TRANSPORT_UDP;
ast_copy_string(tmp, get_header(req, "Contact"), sizeof(tmp));
if ((t = strchr(tmp, ',')))
*t = '\0';
- s = remove_uri_parameters(get_in_brackets(tmp));
+
+ s = get_in_brackets(tmp);
+ if ((trans = strcasestr(s, ";transport="))) do {
+ trans += 11;
+
+ if ((e = strchr(trans, ';')))
+ *e = '\0';
+
+ if (!strncasecmp(trans, "tcp", 3))
+ transport = SIP_TRANSPORT_TCP;
+ else if (!strncasecmp(trans, "tls", 3))
+ transport = SIP_TRANSPORT_TLS;
+ else {
+ if (strncasecmp(trans, "udp", 3))
+ ast_debug(1, "received contact with an invalid transport, '%s'\n", s);
+ transport = SIP_TRANSPORT_UDP;
+ }
+ } while(0);
+ s = remove_uri_parameters(s);
+
+ if (p->socket.ser) {
+ ao2_ref(p->socket.ser, -1);
+ p->socket.ser = NULL;
+ }
+
+ p->socket.fd = -1;
+ p->socket.type = transport;
+
if (ast_test_flag(&p->flags[0], SIP_PROMISCREDIR)) {
if (!strncasecmp(s, "sip:", 4))
s += 4;
@@ -14542,9 +14589,9 @@
e = strchr(s, '/');
if (e)
*e = '\0';
- ast_debug(2, "Found promiscuous redirection to 'SIP/%s'\n", s);
+ ast_debug(2, "Found promiscuous redirection to 'SIP/::::%s@%s'\n", get_transport(transport), s);
if (p->owner)
- ast_string_field_build(p->owner, call_forward, "SIP/%s", s);
+ ast_string_field_build(p->owner, call_forward, "SIP/::::%s@%s", get_transport(transport), s);
} else {
e = strchr(tmp, '@');
if (e) {
@@ -19491,6 +19538,8 @@
char *secret = NULL;
char *md5secret = NULL;
char *authname = NULL;
+ char *trans = NULL;
+ enum sip_transport transport = 0;
int oldformat = format;
/* mask request with some set of allowed formats.
@@ -19540,22 +19589,41 @@
*host++ = '\0';
ext = tmp;
secret = strchr(ext, ':');
- if (secret) {
- *secret++ = '\0';
- md5secret = strchr(secret, ':');
- if (md5secret) {
- *md5secret++ = '\0';
- authname = strchr(md5secret, ':');
- if (authname)
- *authname++ = '\0';
- }
- }
- } else {
+ }
+ if (secret) {
+ *secret++ = '\0';
+ md5secret = strchr(secret, ':');
+ }
+ if (md5secret) {
+ *md5secret++ = '\0';
+ authname = strchr(md5secret, ':');
+ }
+ if (authname) {
+ *authname++ = '\0';
+ trans = strchr(authname, ':');
+ }
+ if (trans) {
+ *trans++ = '\0';
+ if (!strcasecmp(trans, "tcp"))
+ transport = SIP_TRANSPORT_TCP;
+ else if (!strcasecmp(trans, "tls"))
+ transport = SIP_TRANSPORT_TLS;
+ else {
+ if (strcasecmp(trans, "udp"))
+ ast_log(LOG_WARNING, "'%s' is not a valid transport option to Dial() for SIP calls, using udp by default.\n", trans);
+ transport = SIP_TRANSPORT_UDP;
+ }
+ }
+
+ if (!host) {
ext = strchr(tmp, '/');
if (ext)
*ext++ = '\0';
host = tmp;
}
+
+ p->socket.fd = -1;
+ p->socket.type = transport;
/* We now have
host = peer name, DNS host name or DNS domain (for SRV)
@@ -19574,7 +19642,7 @@
ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip);
build_via(p);
build_callid_pvt(p);
-
+
/* We have an extension to call, don't use the full contact here */
/* This to enable dialing registered peers with extension dialling,
like SIP/peername/extension
Modified: branches/1.6.0/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/branches/1.6.0/configs/sip.conf.sample?view=diff&rev=135127&r1=135126&r2=135127
==============================================================================
--- branches/1.6.0/configs/sip.conf.sample (original)
+++ branches/1.6.0/configs/sip.conf.sample Fri Aug 1 11:44:32 2008
@@ -7,7 +7,7 @@
; syntaxes for dialing SIP devices.
; SIP/devicename
; SIP/username at domain (SIP uri)
-; SIP/username[:password[:md5secret[:authname]]]@host[:port]
+; SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
; SIP/devicename/extension
;
;
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