[asterisk-commits] dvossel: branch 1.6.0 r196454 - in /branches/1.6.0: ./ channels/ configs/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri May 22 17:51:16 CDT 2009
Author: dvossel
Date: Fri May 22 17:51:09 2009
New Revision: 196454
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=196454
Log:
Merged revisions 196416 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
........
r196416 | dvossel | 2009-05-22 16:09:45 -0500 (Fri, 22 May 2009) | 19 lines
SIP set outbound transport type from Registration
In sip.conf the transport option allows for the configuration of what transport types (udp, tcp, and tls) a peer will accept, but only the first type listed was used for outbound connections. This patch changes this. Now the default transport type is only used until the peer registers. When registration takes place the transport type is parsed out of the Contact header. If the Contact header's transport type is equal to one that the peer supports, the peer's default transport type for outbound connections is set to match the Contact header's type. If the Contact header's transport type is not present, then the peer's default transport type is set to match the one the peer registered with. When a peer unregisters or the registration expires, the default transport type for that peer is reset.
(closes issue #12282)
Reported by: rjain
Patches:
reg_patch_1.diff uploaded by dvossel (license 671)
Tested by: dvossel
(closes issue #14727)
Reported by: pj
Patches:
reg_patch_3.diff uploaded by dvossel (license 671)
Tested by: pj, dvossel
Review: https://reviewboard.asterisk.org/r/249/
........
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.asterisk.org/svn-view/asterisk/branches/1.6.0/channels/chan_sip.c?view=diff&rev=196454&r1=196453&r2=196454
==============================================================================
--- branches/1.6.0/channels/chan_sip.c (original)
+++ branches/1.6.0/channels/chan_sip.c Fri May 22 17:51:09 2009
@@ -1449,6 +1449,7 @@
ASTOBJ_COMPONENTS(struct sip_peer); /*!< name, refcount, objflags, object pointers */
/*!< peer->name is the unique name of this object */
struct sip_socket socket; /*!< Socket used for this peer */
+ enum sip_transport default_outbound_transport; /*!< Peer Registration may change the default outbound transport. */
unsigned int transports:3; /*!< Transports (enum sip_transport) that are acceptable for this peer */
char secret[80]; /*!< Password */
char md5secret[80]; /*!< Password in MD5 */
@@ -1481,7 +1482,7 @@
int lastmsgssent;
unsigned int sipoptions; /*!< Supported SIP options */
struct ast_flags flags[2]; /*!< SIP_ flags */
-
+
/*! Mailboxes that this peer cares about */
AST_LIST_HEAD_NOLOCK(, sip_mailbox) mailboxes;
@@ -1502,7 +1503,7 @@
struct ast_dnsmgr_entry *dnsmgr;/*!< DNS refresh manager for peer */
struct sockaddr_in addr; /*!< IP address of peer */
int maxcallbitrate; /*!< Maximum Bitrate for a video call */
-
+
/* Qualification */
struct sip_pvt *call; /*!< Call pointer */
int pokeexpire; /*!< When to expire poke (qualify= checking) */
@@ -2551,6 +2552,27 @@
return 0;
return sip_debug_test_addr(sip_real_dst(p));
}
+ /*! \brief Return int representing a bit field of transport types found in const char *transport */
+ static int get_transport_str2enum(const char *transport)
+ {
+ int res = 0;
+
+ if (ast_strlen_zero(transport)) {
+ return res;
+ }
+
+ if (!strcasecmp(transport, "udp")) {
+ res |= SIP_TRANSPORT_UDP;
+ }
+ if (!strcasecmp(transport, "tcp")) {
+ res |= SIP_TRANSPORT_TCP;
+ }
+ if (!strcasecmp(transport, "tls")) {
+ res |= SIP_TRANSPORT_TLS;
+ }
+
+ return res;
+}
static inline const char *get_transport_list(struct sip_peer *peer) {
switch (peer->transports) {
@@ -3338,7 +3360,7 @@
* \endverbatim
*/
static int parse_uri(char *uri, char *scheme,
- char **ret_name, char **pass, char **domain, char **port, char **options)
+ char **ret_name, char **pass, char **domain, char **port, char **options, char **transport)
{
char *name = NULL;
int error = 0;
@@ -3357,6 +3379,17 @@
error = -1;
}
}
+ if (transport) {
+ char *t, *type = "";
+ *transport = "";
+ if ((t = strstr(uri, "transport="))) {
+ strsep(&t, "=");
+ if ((type = strsep(&t, ";"))) {
+ *transport = type;
+ }
+ }
+ }
+
if (!domain) {
/* if we don't want to split around domain, keep everything as a name,
* so we need to do nothing here, except remember why.
@@ -9922,18 +9955,32 @@
}
}
+static void set_peer_transport(struct sip_peer *peer, int transport)
+{
+ /* if the transport type changes, clear all socket data */
+ if (peer->socket.type != transport) {
+ peer->socket.type = transport;
+ peer->socket.fd = -1;
+ if (peer->socket.tcptls_session) {
+ ao2_ref(peer->socket.tcptls_session, -1);
+ peer->socket.tcptls_session = NULL;
+ }
+ }
+}
+
/*! \brief Expire registration of SIP peer */
static int expire_register(const void *data)
{
struct sip_peer *peer = (struct sip_peer *)data;
-
+
if (!peer) /* Hmmm. We have no peer. Weird. */
return 0;
memset(&peer->addr, 0, sizeof(peer->addr));
destroy_association(peer); /* remove registration data from storage */
-
+ set_peer_transport(peer, peer->default_outbound_transport);
+
manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Unregistered\r\nCause: Expired\r\n", peer->name);
register_peer_exten(peer, FALSE); /* Remove regexten */
peer->expire = -1;
@@ -10058,13 +10105,13 @@
/* We have only the part in <brackets> here so we just need to parse a SIP URI.*/
if (tcp) {
- if (parse_uri(contact, "sips:", &contact, NULL, &host, &pt, NULL)) {
- if (parse_uri(contact2, "sip:", &contact, NULL, &host, &pt, NULL))
+ if (parse_uri(contact, "sips:", &contact, NULL, &host, &pt, NULL, NULL)) {
+ if (parse_uri(contact2, "sip:", &contact, NULL, &host, &pt, NULL, NULL))
ast_log(LOG_NOTICE, "'%s' is not a valid SIP contact (missing sip:) trying to use anyway\n", contact);
}
port = !ast_strlen_zero(pt) ? atoi(pt) : STANDARD_TLS_PORT;
} else {
- if (parse_uri(contact, "sip:", &contact, NULL, &host, &pt, NULL))
+ if (parse_uri(contact, "sip:", &contact, NULL, &host, &pt, NULL, NULL))
ast_log(LOG_NOTICE, "'%s' is not a valid SIP contact (missing sip:) trying to use anyway\n", contact);
port = !ast_strlen_zero(pt) ? atoi(pt) : STANDARD_SIP_PORT;
}
@@ -10100,12 +10147,13 @@
/*! \brief Parse contact header and save registration (peer registration) */
static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, struct sip_peer *peer, struct sip_request *req)
{
- char contact[SIPBUFSIZE];
+ char contact[SIPBUFSIZE];
char data[SIPBUFSIZE];
const char *expires = get_header(req, "Expires");
int expiry = atoi(expires);
- char *curi, *host, *pt, *curi2;
+ char *curi, *host, *pt, *curi2, *transport;
int port;
+ int transport_type;
const char *useragent;
struct hostent *hp;
struct ast_hostent ahp;
@@ -10125,8 +10173,6 @@
}
}
- if (peer->socket.type == req->socket.type)
- copy_socket_data(&peer->socket, &req->socket);
copy_socket_data(&pvt->socket, &req->socket);
/* Look for brackets */
@@ -10148,10 +10194,11 @@
} else if (!strcasecmp(curi, "*") || !expiry) { /* Unregister this peer */
/* This means remove all registrations and return OK */
memset(&peer->addr, 0, sizeof(peer->addr));
+ set_peer_transport(peer, peer->default_outbound_transport);
AST_SCHED_DEL(sched, peer->expire);
destroy_association(peer);
-
+
register_peer_exten(peer, FALSE); /* Remove extension from regexten= setting in sip.conf */
peer->fullcontact[0] = '\0';
peer->useragent[0] = '\0';
@@ -10172,15 +10219,34 @@
/* Make sure it's a SIP URL */
if (pvt->socket.type == SIP_TRANSPORT_TLS) {
- if (parse_uri(curi, "sips:", &curi, NULL, &host, &pt, NULL)) {
- if (parse_uri(curi2, "sip:", &curi, NULL, &host, &pt, NULL))
+ if (parse_uri(curi, "sips:", &curi, NULL, &host, &pt, NULL, &transport)) {
+ if (parse_uri(curi2, "sip:", &curi, NULL, &host, &pt, NULL, &transport))
ast_log(LOG_NOTICE, "Not a valid SIP contact (missing sip:) trying to use anyway\n");
}
port = !ast_strlen_zero(pt) ? atoi(pt) : STANDARD_TLS_PORT;
} else {
- if (parse_uri(curi, "sip:", &curi, NULL, &host, &pt, NULL))
+ if (parse_uri(curi, "sip:", &curi, NULL, &host, &pt, NULL, &transport))
ast_log(LOG_NOTICE, "Not a valid SIP contact (missing sip:) trying to use anyway\n");
port = !ast_strlen_zero(pt) ? atoi(pt) : STANDARD_SIP_PORT;
+ }
+
+ /* handle the transport type specified in Contact header. */
+ if ((transport_type = get_transport_str2enum(transport))) {
+ /* if the port is not specified but the transport is, make sure to set the
+ * default port to match the specified transport. This may or may not be the
+ * same transport used by the pvt struct for the Register dialog. */
+ if (ast_strlen_zero(pt)) {
+ port = (transport_type == SIP_TRANSPORT_TLS) ? STANDARD_TLS_PORT : STANDARD_SIP_PORT;
+ }
+ } else {
+ transport_type = pvt->socket.type;
+ }
+
+ /* if the peer's socket type is different than the Registration
+ * transport type, change it. If it got this far, it is a
+ * supported type, but check just in case */
+ if ((peer->socket.type != transport_type) && (peer->transports & transport_type)) {
+ set_peer_transport(peer, transport_type);
}
oldsin = peer->addr;
@@ -10211,6 +10277,16 @@
/* Don't trust the contact field. Just use what they came to us
with */
peer->addr = pvt->recv;
+ }
+
+ /* if the Contact header information copied into peer->addr matches the
+ * received address, and the transport types are the same, then copy socket
+ * data into the peer struct */
+ if ((peer->socket.type == pvt->socket.type) &&
+ (peer->addr.sin_addr.s_addr == pvt->recv.sin_addr.s_addr) &&
+ (peer->addr.sin_port == pvt->recv.sin_port)){
+
+ copy_socket_data(&peer->socket, &pvt->socket);
}
/* Save SIP options profile */
@@ -10232,7 +10308,7 @@
XXX WHY???? XXX
\todo check this
*/
- if (!peer->rt_fromcontact && (peer->socket.type & SIP_TRANSPORT_UDP))
+ if (!peer->rt_fromcontact && (peer->socket.type & SIP_TRANSPORT_UDP))
ast_db_put("SIP/Registry", peer->name, data);
manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Registered\r\n", peer->name);
@@ -11987,12 +12063,12 @@
/* ignore all fields but name */
if (p->socket.type == SIP_TRANSPORT_TLS) {
- if (parse_uri(of, "sips:", &of, &dummy, &domain, &dummy, &dummy)) {
- if (parse_uri(of2, "sip:", &of, &dummy, &domain, &dummy, &dummy))
+ if (parse_uri(of, "sips:", &of, &dummy, &domain, &dummy, &dummy, NULL)) {
+ if (parse_uri(of2, "sip:", &of, &dummy, &domain, &dummy, &dummy, NULL))
ast_log(LOG_NOTICE, "From address missing 'sip:', using it anyway\n");
}
} else {
- if (parse_uri(of, "sip:", &of, &dummy, &domain, &dummy, &dummy))
+ if (parse_uri(of, "sip:", &of, &dummy, &domain, &dummy, &dummy, NULL))
ast_log(LOG_NOTICE, "From address missing 'sip:', using it anyway\n");
}
@@ -20783,9 +20859,9 @@
{
struct sip_peer *peer = NULL;
struct ast_ha *oldha = NULL;
- int found=0;
- int firstpass=1;
- int format=0; /* Ama flags */
+ int found = 0;
+ int firstpass = 1;
+ int format = 0; /* Ama flags */
time_t regseconds = 0;
struct ast_flags peerflags[2] = {{(0)}};
struct ast_flags mask[2] = {{(0)}};
@@ -20840,8 +20916,8 @@
/* If we have realm authentication information, remove them (reload) */
clear_realm_authentication(peer->auth);
peer->auth = NULL;
+ peer->default_outbound_transport = 0;
peer->transports = 0;
- peer->socket.type = 0;
for (; v || ((v = alt) && !(alt=NULL)); v = v->next) {
if (handle_common_options(&peerflags[0], &mask[0], v))
@@ -20853,7 +20929,7 @@
while ((trans = strsep(&val, ","))) {
trans = ast_skip_blanks(trans);
- if (!strncasecmp(trans, "udp", 3))
+ if (!strncasecmp(trans, "udp", 3))
peer->transports |= SIP_TRANSPORT_UDP;
else if (!strncasecmp(trans, "tcp", 3))
peer->transports |= SIP_TRANSPORT_TCP;
@@ -20862,9 +20938,8 @@
else
ast_log(LOG_NOTICE, "'%s' is not a valid transport type. if no other is specified, udp will be used.\n", trans);
- if (!peer->socket.type) { /*!< The first transport listed should be used for outgoing */
- peer->socket.type = peer->transports;
- peer->socket.fd = -1;
+ if (!peer->default_outbound_transport) { /*!< The first transport listed should be default outbound */
+ peer->default_outbound_transport = peer->transports;
}
}
} else if (realtime && !strcasecmp(v->name, "regseconds")) {
@@ -21154,6 +21229,16 @@
peer->socket.type = SIP_TRANSPORT_UDP;
}
+ /* The default transport type set during build_peer should only replace the socket.type when...
+ * 1. Registration is not present and the socket.type and default transport types are different.
+ * 2. The socket.type is not an acceptable transport type after rebuilding peer.
+ * 3. The socket.type is not set yet. */
+ if (((peer->socket.type != peer->default_outbound_transport) && (peer->expire == -1)) ||
+ !(peer->socket.type & peer->transports) || !(peer->socket.type)) {
+
+ set_peer_transport(peer, peer->default_outbound_transport);
+ }
+
if (fullcontact->used > 0) {
ast_copy_string(peer->fullcontact, fullcontact->str, sizeof(peer->fullcontact));
peer->rt_fromcontact = TRUE;
Modified: branches/1.6.0/configs/sip.conf.sample
URL: http://svn.asterisk.org/svn-view/asterisk/branches/1.6.0/configs/sip.conf.sample?view=diff&rev=196454&r1=196453&r2=196454
==============================================================================
--- branches/1.6.0/configs/sip.conf.sample (original)
+++ branches/1.6.0/configs/sip.conf.sample Fri May 22 17:51:09 2009
@@ -774,11 +774,14 @@
;secret=guessit
;defaultuser=yourusername ; Authentication user for outbound proxies
;fromuser=yourusername ; Many SIP providers require this!
-;fromdomain=provider.sip.domain
+;fromdomain=provider.sip.domain
;host=box.provider.com
-;transport=udp,tcp ; This sets the transport type to udp for outgoing, and will
-; ; accept both tcp and udp. Default is udp. The first transport
-; ; listed will always be used for outgoing connections.
+;transport=udp,tcp ; This sets the default transport type to udp for outgoing, and will
+; ; accept both tcp and udp. The default transport type is only used for
+; ; outbound messages until a Registration takes place. During the
+; ; peer Registration the transport type may change to another supported
+; ; type if the peer requests so.
+
;usereqphone=yes ; This provider requires ";user=phone" on URI
;callcounter=yes ; Enable call counter
;busylevel=2 ; Signal busy at 2 or more calls
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