[asterisk-commits] tilghman: trunk r135126 - in /trunk: channels/chan_sip.c configs/sip.conf.sample

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Fri Aug 1 11:39:52 CDT 2008


Author: tilghman
Date: Fri Aug  1 11:39:51 2008
New Revision: 135126

URL: http://svn.digium.com/view/asterisk?view=rev&rev=135126
Log:
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:
    trunk/channels/chan_sip.c
    trunk/configs/sip.conf.sample

Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=135126&r1=135125&r2=135126
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Fri Aug  1 11:39:51 2008
@@ -1950,6 +1950,37 @@
 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);
@@ -2103,7 +2134,7 @@
 static const struct sockaddr_in *sip_real_dst(const struct sip_pvt *p);
 static void build_via(struct sip_pvt *p);
 static int create_addr_from_peer(struct sip_pvt *r, struct sip_peer *peer);
-static int create_addr(struct sip_pvt *dialog, const char *opeer, struct sockaddr_in *sin);
+static int create_addr(struct sip_pvt *dialog, const char *opeer, struct sockaddr_in *sin, int newdialog);
 static char *generate_random_string(char *buf, size_t size);
 static void build_callid_pvt(struct sip_pvt *pvt);
 static void build_callid_registry(struct sip_registry *reg, struct in_addr ourip, const char *fromdomain);
@@ -4135,6 +4166,11 @@
  */
 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) &&
@@ -4260,10 +4296,11 @@
 	
 	return 0;
 }
+
 /*! \brief create address structure from device name
  *      Or, if peer not found, find it in the global DNS 
  *      returns TRUE (-1) on failure, FALSE on success */
-static int create_addr(struct sip_pvt *dialog, const char *opeer, struct sockaddr_in *sin)
+static int create_addr(struct sip_pvt *dialog, const char *opeer, struct sockaddr_in *sin, int newdialog)
 {
 	struct hostent *hp;
 	struct ast_hostent ahp;
@@ -4284,13 +4321,12 @@
 	peer = find_peer(peername, NULL, TRUE, TRUE);
 
 	if (peer) {
-		int res = create_addr_from_peer(dialog, peer);
+		int res;
+		if (newdialog)
+			dialog->socket.type = 0;
+		res = create_addr_from_peer(dialog, peer);
 		unref_peer(peer, "create_addr: unref peer from find_peer hashtab lookup");
 		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);
@@ -4306,7 +4342,10 @@
 	if (sin) {
 		memcpy(&dialog->sa.sin_addr, &sin->sin_addr, sizeof(dialog->sa.sin_addr));
 		if (!sin->sin_port) {
-			portno = port ? atoi(port) : (dialog->socket.type & SIP_TRANSPORT_TLS) ? STANDARD_TLS_PORT : STANDARD_SIP_PORT;
+			if (ast_strlen_zero(port) || sscanf(port, "%u", &portno) != 1) {
+				portno = dialog->socket.type & SIP_TRANSPORT_TLS ?
+					STANDARD_TLS_PORT : STANDARD_SIP_PORT;
+			}
 		} else {
 			portno = ntohs(sin->sin_port);
 		}
@@ -4339,6 +4378,8 @@
 		memcpy(&dialog->sa.sin_addr, hp->h_addr, sizeof(dialog->sa.sin_addr));
 	}
 
+	if (!dialog->socket.type)
+		dialog->socket.type = SIP_TRANSPORT_UDP;
 	dialog->sa.sin_port = htons(portno);
 	dialog->recv = dialog->sa;
 	return 0;
@@ -4375,7 +4416,7 @@
 	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);
@@ -4406,9 +4447,8 @@
 			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);
 
@@ -9442,7 +9482,7 @@
 		return -1;
 	}
 
-	if (create_addr(p, channame, NULL)) {
+	if (create_addr(p, channame, NULL, 0)) {
 		/* Maybe they're not registered, etc. */
 		dialog_unlink_all(p, TRUE, TRUE);
 		dialog_unref(p, "unref dialog inside for loop" );
@@ -9654,7 +9694,7 @@
 			r->us.sin_port = htons(r->portno);
 
 		/* Find address to hostname */
-		if (create_addr(p, r->hostname, &r->us)) {
+		if (create_addr(p, r->hostname, &r->us, 0)) {
 			/* we have what we hope is a temporary network error,
 			 * probably DNS.  We need to reschedule a registration try */
 			dialog_unlink_all(p, TRUE, TRUE);
@@ -10852,27 +10892,10 @@
 					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;
 				}
 			} 
 		}
@@ -14423,7 +14446,7 @@
 			return CLI_FAILURE;
 		}
 
-		if (create_addr(p, a->argv[i], NULL)) {
+		if (create_addr(p, a->argv[i], NULL, 0)) {
 			/* Maybe they're not registered, etc. */
 			dialog_unlink_all(p, TRUE, TRUE);
 			dialog_unref(p, "unref dialog inside for loop" );
@@ -14996,13 +15019,41 @@
 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;
@@ -15011,9 +15062,9 @@
 		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) {
@@ -17095,7 +17146,7 @@
 			if ((peerorhost = strchr(uri, ':'))) {
 				*peerorhost++ = '\0';
 			}
-			create_addr(p, peerorhost, NULL);
+			create_addr(p, peerorhost, NULL, 0);
 			ast_string_field_set(p, theirtag, NULL);
 			for (pkt = p->packets; pkt; pkt = pkt->next) {
 				if (pkt->seqno == p->icseq && pkt->method == SIP_INVITE) {
@@ -20068,6 +20119,8 @@
  	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.
@@ -20119,29 +20172,48 @@
 		*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) 
 		ext = extension (user part of URI)
 		dnid = destination of the call (applies to the To: header)
 	*/
-	if (create_addr(p, host, NULL)) {
+	if (create_addr(p, host, NULL, 1)) {
 		*cause = AST_CAUSE_UNREGISTERED;
 		ast_debug(3, "Cant create SIP call - target device not registred\n");
 		dialog_unlink_all(p, TRUE, TRUE);

Modified: trunk/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/trunk/configs/sip.conf.sample?view=diff&rev=135126&r1=135125&r2=135126
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Fri Aug  1 11:39:51 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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