diff options
author | tilghman <tilghman@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-08-01 16:44:32 +0000 |
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committer | tilghman <tilghman@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-08-01 16:44:32 +0000 |
commit | 756596c58bb344c6d3dd12fc41b4b6a68bf5bae5 (patch) | |
tree | 495156e7ad307b4270b6d22cf5f9f79918e39e4d /channels/chan_sip.c | |
parent | 74561c1a36a0e600422462c36ef1afa8d1579c55 (diff) |
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
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@135127 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels/chan_sip.c')
-rw-r--r-- | channels/chan_sip.c | 156 |
1 files changed, 112 insertions, 44 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 0d1746f60..07f738a15 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -1821,6 +1821,37 @@ static int sip_addrcmp(char *name, struct sockaddr_in *sin); /* Support for peer 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 void copy_socket_data(struct sip_socket *to_sock, const struct sip_socket */ 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 @@ static int create_addr(struct sip_pvt *dialog, const char *opeer) 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 @@ static int create_addr(struct sip_pvt *dialog, const char *opeer) */ 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 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout) 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 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout) 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 @@ static enum check_auth_result register_verify(struct sip_pvt *p, struct sockaddr 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 struct ast_custom_function sipchaninfo_function = { 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 @@ static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req) 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 @@ static struct ast_channel *sip_request_call(const char *type, int format, void * 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. @@ -19540,23 +19589,42 @@ static struct ast_channel *sip_request_call(const char *type, int format, void * *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'; - } + } + 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; } - } else { + } + + 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) @@ -19574,7 +19642,7 @@ static struct ast_channel *sip_request_call(const char *type, int format, void * 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 |