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Change the way that the transport option works for sip users. transport will now take multiple arguments, the first one listed will be the one used

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for new dialogs, and the rest listed will be acceptable ways for that peer to contact us. This fixes a minor bug where, because SIP TCP/UDP run on 
the same port, could cause a TCP peer to be saved in the ast_db. There will also be warnings when a transport is changed for an unexpected reason.

(issue #12799)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125891 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.6.1
Brett Bryant 17 years ago
parent 67e2b82951
commit 12d5cebea2
2 changed files with 72 additions and 5 deletions
Show Stats Download Patch File Download Diff File

70
channels/chan_sip.c
Unescape View File

@ -378,6 +378,7 @@ enum check_auth_result {
AUTH_UNKNOWN_DOMAIN = -5,
AUTH_PEER_NOT_DYNAMIC = -6,
AUTH_ACL_FAILED = -7,
AUTH_BAD_TRANSPORT = -8,
};
/*! \brief States for outbound registrations (with register= lines in sip.conf */
@ -1461,6 +1462,7 @@ struct sip_mailbox {
struct sip_peer {
char name[80]; /*!< peer->name is the unique name of this object */
struct sip_socket socket; /*!< Socket used for this peer */
unsigned int transports:3; /*!< Transports (enum sip_transport) that are acceptable for this peer */
char secret[80]; /*!< Password */
char md5secret[80]; /*!< Password in MD5 */
struct sip_auth *auth; /*!< Realm authentication list */
@ -2714,6 +2716,27 @@ static inline int sip_debug_test_pvt(struct sip_pvt *p)
return sip_debug_test_addr(sip_real_dst(p));
}
static inline const char *get_transport_list(struct sip_peer *peer) {
switch (peer->transports) {
case SIP_TRANSPORT_UDP:
return "UDP";
case SIP_TRANSPORT_TCP:
return "TCP";
case SIP_TRANSPORT_TLS:
return "TLS";
}
if (peer->transports & (SIP_TRANSPORT_TLS | SIP_TRANSPORT_TCP))
return "TLS,TCP";
if (peer->transports & (SIP_TRANSPORT_TLS | SIP_TRANSPORT_UDP))
return "TLS,UDP";
if (peer->transports & (SIP_TRANSPORT_UDP | SIP_TRANSPORT_TCP))
return "TCP,UDP";
return peer->transports ?
"TLS,TCP,UDP" : "UNKNOWN";
}
static inline const char *get_transport(enum sip_transport t)
{
switch (t) {
@ -10321,8 +10344,9 @@ static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, st
}
}
copy_socket_data(&peer->socket, &req->socket);
copy_socket_data(&pvt->socket, &peer->socket);
if (peer->socket.type == req->socket.type)
copy_socket_data(&peer->socket, &req->socket);
copy_socket_data(&pvt->socket, &req->socket);
/* Look for brackets */
curi = contact;
@ -10941,6 +10965,29 @@ static enum check_auth_result register_verify(struct sip_pvt *p, struct sockaddr
res = 0;
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)
);
}
}
}
}
}
@ -11019,6 +11066,7 @@ static enum check_auth_result register_verify(struct sip_pvt *p, struct sockaddr
name, ast_inet_ntoa(sin->sin_addr), ntohs(sin->sin_port));
}
break;
case AUTH_BAD_TRANSPORT:
default:
break;
}
@ -19165,6 +19213,9 @@ static int handle_request_register(struct sip_pvt *p, struct sip_request *req, s
case AUTH_ACL_FAILED:
reason = "Device does not match ACL";
break;
case AUTH_BAD_TRANSPORT:
reason = "Device not configured to use this transport type";
break;
default:
reason = "Unknown failure";
break;
@ -21198,17 +21249,21 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, str
/* If we have realm authentication information, remove them (reload) */
clear_realm_authentication(peer->auth);
peer->auth = NULL;
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))
continue;
if (!strcasecmp(v->name, "transport")) {
if (!strcasecmp(v->value, "udp"))
peer->socket.type = SIP_TRANSPORT_UDP;
peer->transports &= SIP_TRANSPORT_UDP;
else if (!strcasecmp(v->value, "tcp"))
peer->socket.type = SIP_TRANSPORT_TCP;
peer->transports &= SIP_TRANSPORT_TCP;
else if (!strcasecmp(v->value, "tls"))
peer->socket.type = SIP_TRANSPORT_TLS;
peer->transports &= SIP_TRANSPORT_TLS;
if (!peer->socket.type) /*!< The first transport listed should be used for outgoing */
peer->socket.type = peer->transports;
} else if (realtime && !strcasecmp(v->name, "regseconds")) {
ast_get_time_t(v->value, &regseconds, 0, NULL);
} else if (realtime && !strcasecmp(v->name, "ipaddr") && !ast_strlen_zero(v->value) ) {
@ -21450,6 +21505,11 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, str
}
}
if (!peer->socket.type) {
peer->transports = SIP_TRANSPORT_UDP;
peer->socket.type = SIP_TRANSPORT_UDP;
}
if (fullcontact->used > 0) {
ast_copy_string(peer->fullcontact, fullcontact->str, sizeof(peer->fullcontact));
peer->rt_fromcontact = TRUE;

7
configs/sip.conf.sample
Unescape View File

@ -685,6 +685,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; deny deny
; secret secret
; md5secret md5secret
; transport transport
; dtmfmode dtmfmode
; canreinvite canreinvite
; nat nat
@ -753,6 +754,9 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;fromuser=yourusername ; Many SIP providers require this!
;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.
;usereqphone=yes ; This provider requires ";user=phone" on URI
;callcounter=yes ; Enable call counter
;busylevel=2 ; Signal busy at 2 or more calls
@ -767,6 +771,9 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;fromuser=4015552299 ; how your provider knows you
;secret=youwillneverguessit
;callbackextension=123 ; Register with this server and require calls coming back to this extension
;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.
;------------------------------------------------------------------------------
; Definitions of locally connected SIP devices

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