2 * Asterisk -- An open source telephony toolkit.
4 * Copyright (C) 1999 - 2006, Digium, Inc.
6 * Mark Spencer <markster@digium.com>
8 * See http://www.asterisk.org for more information about
9 * the Asterisk project. Please do not directly contact
10 * any of the maintainers of this project for assistance;
11 * the project provides a web site, mailing lists and IRC
12 * channels for your use.
14 * This program is free software, distributed under the terms of
15 * the GNU General Public License Version 2. See the LICENSE file
16 * at the top of the source tree.
21 * \brief Implementation of Session Initiation Protocol
23 * \author Mark Spencer <markster@digium.com>
26 * \arg \ref AstCREDITS
28 * Implementation of RFC 3261 - without S/MIME, TCP and TLS support
29 * Configuration file \link Config_sip sip.conf \endlink
34 * \todo Better support of forking
35 * \todo VIA branch tag transaction checking
36 * \todo Transaction support
38 * \ingroup channel_drivers
40 * \par Overview of the handling of SIP sessions
41 * The SIP channel handles several types of SIP sessions, or dialogs,
42 * not all of them being "telephone calls".
43 * - Incoming calls that will be sent to the PBX core
44 * - Outgoing calls, generated by the PBX
45 * - SIP subscriptions and notifications of states and voicemail messages
46 * - SIP registrations, both inbound and outbound
47 * - SIP peer management (peerpoke, OPTIONS)
50 * In the SIP channel, there's a list of active SIP dialogs, which includes
51 * all of these when they are active. "sip show channels" in the CLI will
52 * show most of these, excluding subscriptions which are shown by
53 * "sip show subscriptions"
55 * \par incoming packets
56 * Incoming packets are received in the monitoring thread, then handled by
57 * sipsock_read(). This function parses the packet and matches an existing
58 * dialog or starts a new SIP dialog.
60 * sipsock_read sends the packet to handle_request(), that parses a bit more.
61 * if it's a response to an outbound request, it's sent to handle_response().
62 * If it is a request, handle_request sends it to one of a list of functions
63 * depending on the request type - INVITE, OPTIONS, REFER, BYE, CANCEL etc
64 * sipsock_read locks the ast_channel if it exists (an active call) and
65 * unlocks it after we have processed the SIP message.
67 * A new INVITE is sent to handle_request_invite(), that will end up
68 * starting a new channel in the PBX, the new channel after that executing
69 * in a separate channel thread. This is an incoming "call".
70 * When the call is answered, either by a bridged channel or the PBX itself
71 * the sip_answer() function is called.
73 * The actual media - Video or Audio - is mostly handled by the RTP subsystem
77 * Outbound calls are set up by the PBX through the sip_request_call()
78 * function. After that, they are activated by sip_call().
81 * The PBX issues a hangup on both incoming and outgoing calls through
82 * the sip_hangup() function
88 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
94 #include <sys/socket.h>
95 #include <sys/ioctl.h>
102 #include <sys/signal.h>
103 #include <netinet/in.h>
104 #include <netinet/in_systm.h>
105 #include <arpa/inet.h>
106 #include <netinet/ip.h>
109 #include "asterisk/lock.h"
110 #include "asterisk/channel.h"
111 #include "asterisk/config.h"
112 #include "asterisk/logger.h"
113 #include "asterisk/module.h"
114 #include "asterisk/pbx.h"
115 #include "asterisk/options.h"
116 #include "asterisk/lock.h"
117 #include "asterisk/sched.h"
118 #include "asterisk/io.h"
119 #include "asterisk/rtp.h"
120 #include "asterisk/udptl.h"
121 #include "asterisk/acl.h"
122 #include "asterisk/manager.h"
123 #include "asterisk/callerid.h"
124 #include "asterisk/cli.h"
125 #include "asterisk/app.h"
126 #include "asterisk/musiconhold.h"
127 #include "asterisk/dsp.h"
128 #include "asterisk/features.h"
129 #include "asterisk/acl.h"
130 #include "asterisk/srv.h"
131 #include "asterisk/astdb.h"
132 #include "asterisk/causes.h"
133 #include "asterisk/utils.h"
134 #include "asterisk/file.h"
135 #include "asterisk/astobj.h"
136 #include "asterisk/dnsmgr.h"
137 #include "asterisk/devicestate.h"
138 #include "asterisk/linkedlists.h"
139 #include "asterisk/stringfields.h"
140 #include "asterisk/monitor.h"
141 #include "asterisk/localtime.h"
142 #include "asterisk/abstract_jb.h"
143 #include "asterisk/compiler.h"
144 #include "asterisk/threadstorage.h"
145 #include "asterisk/translate.h"
155 #define VIDEO_CODEC_MASK 0x1fc0000 /*!< Video codecs from H.261 thru AST_FORMAT_MAX_VIDEO */
156 #ifndef IPTOS_MINCOST
157 #define IPTOS_MINCOST 0x02
160 /* #define VOCAL_DATA_HACK */
162 #define DEFAULT_DEFAULT_EXPIRY 120
163 #define DEFAULT_MIN_EXPIRY 60
164 #define DEFAULT_MAX_EXPIRY 3600
165 #define DEFAULT_REGISTRATION_TIMEOUT 20
166 #define DEFAULT_MAX_FORWARDS "70"
168 /* guard limit must be larger than guard secs */
169 /* guard min must be < 1000, and should be >= 250 */
170 #define EXPIRY_GUARD_SECS 15 /*!< How long before expiry do we reregister */
171 #define EXPIRY_GUARD_LIMIT 30 /*!< Below here, we use EXPIRY_GUARD_PCT instead of
173 #define EXPIRY_GUARD_MIN 500 /*!< This is the minimum guard time applied. If
174 GUARD_PCT turns out to be lower than this, it
175 will use this time instead.
176 This is in milliseconds. */
177 #define EXPIRY_GUARD_PCT 0.20 /*!< Percentage of expires timeout to use when
178 below EXPIRY_GUARD_LIMIT */
179 #define DEFAULT_EXPIRY 900 /*!< Expire slowly */
181 static int min_expiry = DEFAULT_MIN_EXPIRY; /*!< Minimum accepted registration time */
182 static int max_expiry = DEFAULT_MAX_EXPIRY; /*!< Maximum accepted registration time */
183 static int default_expiry = DEFAULT_DEFAULT_EXPIRY;
184 static int expiry = DEFAULT_EXPIRY;
187 #define MAX(a,b) ((a) > (b) ? (a) : (b))
190 #define CALLERID_UNKNOWN "Unknown"
192 #define DEFAULT_MAXMS 2000 /*!< Qualification: Must be faster than 2 seconds by default */
193 #define DEFAULT_FREQ_OK 60 * 1000 /*!< Qualification: How often to check for the host to be up */
194 #define DEFAULT_FREQ_NOTOK 10 * 1000 /*!< Qualification: How often to check, if the host is down... */
196 #define DEFAULT_RETRANS 1000 /*!< How frequently to retransmit Default: 2 * 500 ms in RFC 3261 */
197 #define MAX_RETRANS 6 /*!< Try only 6 times for retransmissions, a total of 7 transmissions */
198 #define SIP_TRANS_TIMEOUT 32000 /*!< SIP request timeout (rfc 3261) 64*T1
199 \todo Use known T1 for timeout (peerpoke)
201 #define DEFAULT_TRANS_TIMEOUT -1 /* Use default SIP transaction timeout */
202 #define MAX_AUTHTRIES 3 /*!< Try authentication three times, then fail */
204 #define SIP_MAX_HEADERS 64 /*!< Max amount of SIP headers to read */
205 #define SIP_MAX_LINES 64 /*!< Max amount of lines in SIP attachment (like SDP) */
206 #define SIP_MAX_PACKET 4096 /*!< Also from RFC 3261 (2543), should sub headers tho */
208 #define INITIAL_CSEQ 101 /*!< our initial sip sequence number */
210 /*! \brief Global jitterbuffer configuration - by default, jb is disabled */
211 static struct ast_jb_conf default_jbconf =
215 .resync_threshold = -1,
218 static struct ast_jb_conf global_jbconf;
220 static const char config[] = "sip.conf";
221 static const char notify_config[] = "sip_notify.conf";
222 static int usecnt = 0;
228 /*! \brief Authorization scheme for call transfers
229 \note Not a bitfield flag, since there are plans for other modes,
230 like "only allow transfers for authenticated devices" */
232 TRANSFER_OPENFORALL, /*!< Allow all SIP transfers */
233 TRANSFER_CLOSED, /*!< Allow no SIP transfers */
242 /* Do _NOT_ make any changes to this enum, or the array following it;
243 if you think you are doing the right thing, you are probably
244 not doing the right thing. If you think there are changes
245 needed, get someone else to review them first _before_
246 submitting a patch. If these two lists do not match properly
247 bad things will happen.
251 XMIT_CRITICAL = 2, /*!< Transmit critical SIP message reliably, with re-transmits.
252 If it fails, it's critical and will cause a teardown of the session */
253 XMIT_RELIABLE = 1, /*!< Transmit SIP message reliably, with re-transmits */
254 XMIT_UNRELIABLE = 0, /*!< Transmit SIP message without bothering with re-transmits */
257 enum parse_register_result {
258 PARSE_REGISTER_FAILED,
259 PARSE_REGISTER_UPDATE,
260 PARSE_REGISTER_QUERY,
263 enum subscriptiontype {
272 static const struct cfsubscription_types {
273 enum subscriptiontype type;
274 const char * const event;
275 const char * const mediatype;
276 const char * const text;
277 } subscription_types[] = {
278 { NONE, "-", "unknown", "unknown" },
279 /* RFC 4235: SIP Dialog event package */
280 { DIALOG_INFO_XML, "dialog", "application/dialog-info+xml", "dialog-info+xml" },
281 { CPIM_PIDF_XML, "presence", "application/cpim-pidf+xml", "cpim-pidf+xml" }, /* RFC 3863 */
282 { PIDF_XML, "presence", "application/pidf+xml", "pidf+xml" }, /* RFC 3863 */
283 { XPIDF_XML, "presence", "application/xpidf+xml", "xpidf+xml" }, /* Pre-RFC 3863 with MS additions */
284 { MWI_NOTIFICATION, "message-summary", "application/simple-message-summary", "mwi" } /* RFC 3842: Mailbox notification */
287 /*! \brief SIP Request methods known by Asterisk */
289 SIP_UNKNOWN, /* Unknown response */
290 SIP_RESPONSE, /* Not request, response to outbound request */
296 SIP_PRACK, /* Not supported at all */
301 SIP_UPDATE, /* We can send UPDATE; but not accept it */
304 SIP_PUBLISH, /* Not supported at all */
305 SIP_PING, /* Not supported at all, no standard but still implemented out there */
308 /*! \brief Authentication types - proxy or www authentication
309 \note Endpoints, like Asterisk, should always use WWW authentication to
310 allow multiple authentications in the same call - to the proxy and
318 /*! \brief Authentication result from check_auth* functions */
319 enum check_auth_result {
320 AUTH_DONT_KNOW = -100, /*!< no result, need to check further */
321 /* XXX maybe this is the same as AUTH_NOT_FOUND */
324 AUTH_CHALLENGE_SENT = 1,
325 AUTH_SECRET_FAILED = -1,
326 AUTH_USERNAME_MISMATCH = -2,
327 AUTH_NOT_FOUND = -3, /* returned by register_verify */
329 AUTH_UNKNOWN_DOMAIN = -5,
332 /*! \brief States for outbound registrations (with register= lines in sip.conf */
333 enum sipregistrystate {
334 REG_STATE_UNREGISTERED = 0, /*!< We are not registred */
335 REG_STATE_REGSENT, /*!< Registration request sent */
336 REG_STATE_AUTHSENT, /*!< We have tried to authenticate */
337 REG_STATE_REGISTERED, /*!< Registred and done */
338 REG_STATE_REJECTED, /*!< Registration rejected */
339 REG_STATE_TIMEOUT, /*!< Registration timed out */
340 REG_STATE_NOAUTH, /*!< We have no accepted credentials */
341 REG_STATE_FAILED, /*!< Registration failed after several tries */
344 enum can_create_dialog {
345 CAN_NOT_CREATE_DIALOG,
347 CAN_CREATE_DIALOG_UNSUPPORTED_METHOD,
350 /*! XXX Note that sip_methods[i].id == i must hold or the code breaks */
351 static const struct cfsip_methods {
353 int need_rtp; /*!< when this is the 'primary' use for a pvt structure, does it need RTP? */
355 enum can_create_dialog can_create;
357 { SIP_UNKNOWN, RTP, "-UNKNOWN-", CAN_CREATE_DIALOG },
358 { SIP_RESPONSE, NO_RTP, "SIP/2.0", CAN_NOT_CREATE_DIALOG },
359 { SIP_REGISTER, NO_RTP, "REGISTER", CAN_CREATE_DIALOG },
360 { SIP_OPTIONS, NO_RTP, "OPTIONS", CAN_CREATE_DIALOG },
361 { SIP_NOTIFY, NO_RTP, "NOTIFY", CAN_CREATE_DIALOG },
362 { SIP_INVITE, RTP, "INVITE", CAN_CREATE_DIALOG },
363 { SIP_ACK, NO_RTP, "ACK", CAN_NOT_CREATE_DIALOG },
364 { SIP_PRACK, NO_RTP, "PRACK", CAN_NOT_CREATE_DIALOG },
365 { SIP_BYE, NO_RTP, "BYE", CAN_NOT_CREATE_DIALOG },
366 { SIP_REFER, NO_RTP, "REFER", CAN_CREATE_DIALOG },
367 { SIP_SUBSCRIBE, NO_RTP, "SUBSCRIBE", CAN_CREATE_DIALOG },
368 { SIP_MESSAGE, NO_RTP, "MESSAGE", CAN_CREATE_DIALOG },
369 { SIP_UPDATE, NO_RTP, "UPDATE", CAN_NOT_CREATE_DIALOG },
370 { SIP_INFO, NO_RTP, "INFO", CAN_NOT_CREATE_DIALOG },
371 { SIP_CANCEL, NO_RTP, "CANCEL", CAN_NOT_CREATE_DIALOG },
372 { SIP_PUBLISH, NO_RTP, "PUBLISH", CAN_CREATE_DIALOG_UNSUPPORTED_METHOD },
373 { SIP_PING, NO_RTP, "PING", CAN_CREATE_DIALOG_UNSUPPORTED_METHOD }
376 /*! Define SIP option tags, used in Require: and Supported: headers
377 We need to be aware of these properties in the phones to use
378 the replace: header. We should not do that without knowing
379 that the other end supports it...
380 This is nothing we can configure, we learn by the dialog
381 Supported: header on the REGISTER (peer) or the INVITE
383 We are not using many of these today, but will in the future.
384 This is documented in RFC 3261
387 #define NOT_SUPPORTED 0
389 #define SIP_OPT_REPLACES (1 << 0)
390 #define SIP_OPT_100REL (1 << 1)
391 #define SIP_OPT_TIMER (1 << 2)
392 #define SIP_OPT_EARLY_SESSION (1 << 3)
393 #define SIP_OPT_JOIN (1 << 4)
394 #define SIP_OPT_PATH (1 << 5)
395 #define SIP_OPT_PREF (1 << 6)
396 #define SIP_OPT_PRECONDITION (1 << 7)
397 #define SIP_OPT_PRIVACY (1 << 8)
398 #define SIP_OPT_SDP_ANAT (1 << 9)
399 #define SIP_OPT_SEC_AGREE (1 << 10)
400 #define SIP_OPT_EVENTLIST (1 << 11)
401 #define SIP_OPT_GRUU (1 << 12)
402 #define SIP_OPT_TARGET_DIALOG (1 << 13)
403 #define SIP_OPT_NOREFERSUB (1 << 14)
404 #define SIP_OPT_HISTINFO (1 << 15)
405 #define SIP_OPT_RESPRIORITY (1 << 16)
407 /*! \brief List of well-known SIP options. If we get this in a require,
408 we should check the list and answer accordingly. */
409 static const struct cfsip_options {
410 int id; /*!< Bitmap ID */
411 int supported; /*!< Supported by Asterisk ? */
412 char * const text; /*!< Text id, as in standard */
413 } sip_options[] = { /* XXX used in 3 places */
414 /* RFC3891: Replaces: header for transfer */
415 { SIP_OPT_REPLACES, SUPPORTED, "replaces" },
416 /* One version of Polycom firmware has the wrong label */
417 { SIP_OPT_REPLACES, SUPPORTED, "replace" },
418 /* RFC3262: PRACK 100% reliability */
419 { SIP_OPT_100REL, NOT_SUPPORTED, "100rel" },
420 /* RFC4028: SIP Session Timers */
421 { SIP_OPT_TIMER, NOT_SUPPORTED, "timer" },
422 /* RFC3959: SIP Early session support */
423 { SIP_OPT_EARLY_SESSION, NOT_SUPPORTED, "early-session" },
424 /* RFC3911: SIP Join header support */
425 { SIP_OPT_JOIN, NOT_SUPPORTED, "join" },
426 /* RFC3327: Path support */
427 { SIP_OPT_PATH, NOT_SUPPORTED, "path" },
428 /* RFC3840: Callee preferences */
429 { SIP_OPT_PREF, NOT_SUPPORTED, "pref" },
430 /* RFC3312: Precondition support */
431 { SIP_OPT_PRECONDITION, NOT_SUPPORTED, "precondition" },
432 /* RFC3323: Privacy with proxies*/
433 { SIP_OPT_PRIVACY, NOT_SUPPORTED, "privacy" },
434 /* RFC4092: Usage of the SDP ANAT Semantics in the SIP */
435 { SIP_OPT_SDP_ANAT, NOT_SUPPORTED, "sdp-anat" },
436 /* RFC3329: Security agreement mechanism */
437 { SIP_OPT_SEC_AGREE, NOT_SUPPORTED, "sec_agree" },
438 /* SIMPLE events: draft-ietf-simple-event-list-07.txt */
439 { SIP_OPT_EVENTLIST, NOT_SUPPORTED, "eventlist" },
440 /* GRUU: Globally Routable User Agent URI's */
441 { SIP_OPT_GRUU, NOT_SUPPORTED, "gruu" },
442 /* Target-dialog: draft-ietf-sip-target-dialog-03.txt */
443 { SIP_OPT_TARGET_DIALOG,NOT_SUPPORTED, "tdialog" },
444 /* Disable the REFER subscription, RFC 4488 */
445 { SIP_OPT_NOREFERSUB, NOT_SUPPORTED, "norefersub" },
446 /* ietf-sip-history-info-06.txt */
447 { SIP_OPT_HISTINFO, NOT_SUPPORTED, "histinfo" },
448 /* ietf-sip-resource-priority-10.txt */
449 { SIP_OPT_RESPRIORITY, NOT_SUPPORTED, "resource-priority" },
453 /*! \brief SIP Methods we support */
454 #define ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY"
456 /*! \brief SIP Extensions we support */
457 #define SUPPORTED_EXTENSIONS "replaces"
459 /*! \brief Standard SIP port from RFC 3261. DO NOT CHANGE THIS */
460 #define STANDARD_SIP_PORT 5060
461 /* Note: in many SIP headers, absence of a port number implies port 5060,
462 * and this is why we cannot change the above constant.
463 * There is a limited number of places in asterisk where we could,
464 * in principle, use a different "default" port number, but
465 * we do not support this feature at the moment.
468 /* Default values, set and reset in reload_config before reading configuration */
469 /* These are default values in the source. There are other recommended values in the
470 sip.conf.sample for new installations. These may differ to keep backwards compatibility,
471 yet encouraging new behaviour on new installations
473 #define DEFAULT_CONTEXT "default"
474 #define DEFAULT_MOHINTERPRET "default"
475 #define DEFAULT_MOHSUGGEST ""
476 #define DEFAULT_VMEXTEN "asterisk"
477 #define DEFAULT_CALLERID "asterisk"
478 #define DEFAULT_NOTIFYMIME "application/simple-message-summary"
479 #define DEFAULT_MWITIME 10
480 #define DEFAULT_ALLOWGUEST TRUE
481 #define DEFAULT_SRVLOOKUP FALSE /*!< Recommended setting is ON */
482 #define DEFAULT_COMPACTHEADERS FALSE
483 #define DEFAULT_TOS_SIP 0 /*!< Call signalling packets should be marked as DSCP CS3, but the default is 0 to be compatible with previous versions. */
484 #define DEFAULT_TOS_AUDIO 0 /*!< Audio packets should be marked as DSCP EF (Expedited Forwarding), but the default is 0 to be compatible with previous versions. */
485 #define DEFAULT_TOS_VIDEO 0 /*!< Video packets should be marked as DSCP AF41, but the default is 0 to be compatible with previous versions. */
486 #define DEFAULT_ALLOW_EXT_DOM TRUE
487 #define DEFAULT_REALM "asterisk"
488 #define DEFAULT_NOTIFYRINGING TRUE
489 #define DEFAULT_PEDANTIC FALSE
490 #define DEFAULT_AUTOCREATEPEER FALSE
491 #define DEFAULT_QUALIFY FALSE
492 #define DEFAULT_T1MIN 100 /*!< 100 MS for minimal roundtrip time */
493 #define DEFAULT_MAX_CALL_BITRATE (384) /*!< Max bitrate for video */
494 #ifndef DEFAULT_USERAGENT
495 #define DEFAULT_USERAGENT "Asterisk PBX" /*!< Default Useragent: header unless re-defined in sip.conf */
499 /* Default setttings are used as a channel setting and as a default when
500 configuring devices */
501 static char default_context[AST_MAX_CONTEXT];
502 static char default_subscribecontext[AST_MAX_CONTEXT];
503 static char default_language[MAX_LANGUAGE];
504 static char default_callerid[AST_MAX_EXTENSION];
505 static char default_fromdomain[AST_MAX_EXTENSION];
506 static char default_notifymime[AST_MAX_EXTENSION];
507 static int default_qualify; /*!< Default Qualify= setting */
508 static char default_vmexten[AST_MAX_EXTENSION];
509 static char default_mohinterpret[MAX_MUSICCLASS]; /*!< Global setting for moh class to use when put on hold */
510 static char default_mohsuggest[MAX_MUSICCLASS]; /*!< Global setting for moh class to suggest when putting
511 * a bridged channel on hold */
512 static int default_maxcallbitrate; /*!< Maximum bitrate for call */
513 static struct ast_codec_pref default_prefs; /*!< Default codec prefs */
515 /* Global settings only apply to the channel */
516 static int global_rtautoclear;
517 static int global_notifyringing; /*!< Send notifications on ringing */
518 static int global_alwaysauthreject; /*!< Send 401 Unauthorized for all failing requests */
519 static int srvlookup; /*!< SRV Lookup on or off. Default is off, RFC behavior is on */
520 static int pedanticsipchecking; /*!< Extra checking ? Default off */
521 static int autocreatepeer; /*!< Auto creation of peers at registration? Default off. */
522 static int global_match_auth_username; /*!< Match auth username if available instead of From: Default off. */
523 static int global_relaxdtmf; /*!< Relax DTMF */
524 static int global_rtptimeout; /*!< Time out call if no RTP */
525 static int global_rtpholdtimeout;
526 static int global_rtpkeepalive; /*!< Send RTP keepalives */
527 static int global_reg_timeout;
528 static int global_regattempts_max; /*!< Registration attempts before giving up */
529 static int global_allowguest; /*!< allow unauthenticated users/peers to connect? */
530 static int global_allowsubscribe; /*!< Flag for disabling ALL subscriptions, this is FALSE only if all peers are FALSE
531 the global setting is in globals_flags[1] */
532 static int global_mwitime; /*!< Time between MWI checks for peers */
533 static unsigned int global_tos_sip; /*!< IP type of service for SIP packets */
534 static unsigned int global_tos_audio; /*!< IP type of service for audio RTP packets */
535 static unsigned int global_tos_video; /*!< IP type of service for video RTP packets */
536 static int compactheaders; /*!< send compact sip headers */
537 static int recordhistory; /*!< Record SIP history. Off by default */
538 static int dumphistory; /*!< Dump history to verbose before destroying SIP dialog */
539 static char global_realm[MAXHOSTNAMELEN]; /*!< Default realm */
540 static char global_regcontext[AST_MAX_CONTEXT]; /*!< Context for auto-extensions */
541 static char global_useragent[AST_MAX_EXTENSION]; /*!< Useragent for the SIP channel */
542 static int allow_external_domains; /*!< Accept calls to external SIP domains? */
543 static int global_callevents; /*!< Whether we send manager events or not */
544 static int global_t1min; /*!< T1 roundtrip time minimum */
545 static int global_autoframing; /*!< ?????????? */
546 static enum transfermodes global_allowtransfer; /*!< SIP Refer restriction scheme */
548 /*! \brief Codecs that we support by default: */
549 static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
550 static int noncodeccapability = AST_RTP_DTMF;
552 /* Object counters */
553 static int suserobjs = 0; /*!< Static users */
554 static int ruserobjs = 0; /*!< Realtime users */
555 static int speerobjs = 0; /*!< Statis peers */
556 static int rpeerobjs = 0; /*!< Realtime peers */
557 static int apeerobjs = 0; /*!< Autocreated peer objects */
558 static int regobjs = 0; /*!< Registry objects */
560 static struct ast_flags global_flags[2] = {{0}}; /*!< global SIP_ flags */
562 /*! \brief Protect the SIP dialog list (of sip_pvt's) */
563 AST_MUTEX_DEFINE_STATIC(dialoglock);
565 AST_MUTEX_DEFINE_STATIC(netlock);
567 /*! \brief Protect the monitoring thread, so only one process can kill or start it, and not
568 when it's doing something critical. */
570 AST_MUTEX_DEFINE_STATIC(monlock);
572 AST_MUTEX_DEFINE_STATIC(sip_reload_lock);
574 /*! \brief This is the thread for the monitor which checks for input on the channels
575 which are not currently in use. */
576 static pthread_t monitor_thread = AST_PTHREADT_NULL;
578 static int sip_reloading = FALSE; /*!< Flag for avoiding multiple reloads at the same time */
579 static enum channelreloadreason sip_reloadreason; /*!< Reason for last reload/load of configuration */
581 static struct sched_context *sched; /*!< The scheduling context */
582 static struct io_context *io; /*!< The IO context */
583 static int *sipsock_read_id; /*!< ID of IO entry for sipsock FD */
585 #define DEC_CALL_LIMIT 0
586 #define INC_CALL_LIMIT 1
587 #define DEC_CALL_RINGING 2
588 #define INC_CALL_RINGING 3
590 /*! \brief sip_request: The data grabbed from the UDP socket */
592 char *rlPart1; /*!< SIP Method Name or "SIP/2.0" protocol version */
593 char *rlPart2; /*!< The Request URI or Response Status */
594 int len; /*!< Length */
595 int headers; /*!< # of SIP Headers */
596 int method; /*!< Method of this request */
597 int lines; /*!< Body Content */
598 unsigned int flags; /*!< SIP_PKT Flags for this packet */
599 char *header[SIP_MAX_HEADERS];
600 char *line[SIP_MAX_LINES];
601 char data[SIP_MAX_PACKET];
602 unsigned int sdp_start; /*!< the line number where the SDP begins */
603 unsigned int sdp_end; /*!< the line number where the SDP ends */
607 * A sip packet is stored into the data[] buffer, with the header followed
608 * by an empty line and the body of the message.
609 * On outgoing packets, data is accumulated in data[] with len reflecting
610 * the next available byte, headers and lines count the number of lines
611 * in both parts. There are no '\0' in data[0..len-1].
613 * On received packet, the input read from the socket is copied into data[],
614 * len is set and the string is NUL-terminated. Then a parser fills up
615 * the other fields -header[] and line[] to point to the lines of the
616 * message, rlPart1 and rlPart2 parse the first lnie as below:
618 * Requests have in the first line METHOD URI SIP/2.0
619 * rlPart1 = method; rlPart2 = uri;
620 * Responses have in the first line SIP/2.0 code description
621 * rlPart1 = SIP/2.0; rlPart2 = code + description;
625 /*! \brief structure used in transfers */
627 struct ast_channel *chan1; /*!< First channel involved */
628 struct ast_channel *chan2; /*!< Second channel involved */
629 struct sip_request req; /*!< Request that caused the transfer (REFER) */
630 int seqno; /*!< Sequence number */
635 /*! \brief Parameters to the transmit_invite function */
636 struct sip_invite_param {
637 int addsipheaders; /*!< Add extra SIP headers */
638 const char *uri_options; /*!< URI options to add to the URI */
639 const char *vxml_url; /*!< VXML url for Cisco phones */
640 char *auth; /*!< Authentication */
641 char *authheader; /*!< Auth header */
642 enum sip_auth_type auth_type; /*!< Authentication type */
643 const char *replaces; /*!< Replaces header for call transfers */
644 int transfer; /*!< Flag - is this Invite part of a SIP transfer? (invite/replaces) */
647 /*! \brief Structure to save routing information for a SIP session */
649 struct sip_route *next;
653 /*! \brief Modes for SIP domain handling in the PBX */
655 SIP_DOMAIN_AUTO, /*!< This domain is auto-configured */
656 SIP_DOMAIN_CONFIG, /*!< This domain is from configuration */
659 /*! \brief Domain data structure.
660 \note In the future, we will connect this to a configuration tree specific
664 char domain[MAXHOSTNAMELEN]; /*!< SIP domain we are responsible for */
665 char context[AST_MAX_EXTENSION]; /*!< Incoming context for this domain */
666 enum domain_mode mode; /*!< How did we find this domain? */
667 AST_LIST_ENTRY(domain) list; /*!< List mechanics */
670 static AST_LIST_HEAD_STATIC(domain_list, domain); /*!< The SIP domain list */
673 /*! \brief sip_history: Structure for saving transactions within a SIP dialog */
675 AST_LIST_ENTRY(sip_history) list;
676 char event[0]; /* actually more, depending on needs */
679 AST_LIST_HEAD_NOLOCK(sip_history_head, sip_history); /*!< history list, entry in sip_pvt */
681 /*! \brief sip_auth: Credentials for authentication to other SIP services */
683 char realm[AST_MAX_EXTENSION]; /*!< Realm in which these credentials are valid */
684 char username[256]; /*!< Username */
685 char secret[256]; /*!< Secret */
686 char md5secret[256]; /*!< MD5Secret */
687 struct sip_auth *next; /*!< Next auth structure in list */
690 /*--- Various flags for the flags field in the pvt structure */
691 #define SIP_ALREADYGONE (1 << 0) /*!< Whether or not we've already been destroyed by our peer */
692 #define SIP_NEEDDESTROY (1 << 1) /*!< if we need to be destroyed by the monitor thread */
693 #define SIP_NOVIDEO (1 << 2) /*!< Didn't get video in invite, don't offer */
694 #define SIP_RINGING (1 << 3) /*!< Have sent 180 ringing */
695 #define SIP_PROGRESS_SENT (1 << 4) /*!< Have sent 183 message progress */
696 #define SIP_NEEDREINVITE (1 << 5) /*!< Do we need to send another reinvite? */
697 #define SIP_PENDINGBYE (1 << 6) /*!< Need to send bye after we ack? */
698 #define SIP_GOTREFER (1 << 7) /*!< Got a refer? */
699 #define SIP_PROMISCREDIR (1 << 8) /*!< Promiscuous redirection */
700 #define SIP_TRUSTRPID (1 << 9) /*!< Trust RPID headers? */
701 #define SIP_USEREQPHONE (1 << 10) /*!< Add user=phone to numeric URI. Default off */
702 #define SIP_REALTIME (1 << 11) /*!< Flag for realtime users */
703 #define SIP_USECLIENTCODE (1 << 12) /*!< Trust X-ClientCode info message */
704 #define SIP_OUTGOING (1 << 13) /*!< Direction of the last transaction in this dialog */
705 #define SIP_CAN_BYE (1 << 14) /*!< Can we send BYE on this dialog? */
706 #define SIP_DEFER_BYE_ON_TRANSFER (1 << 15) /*!< Do not hangup at first ast_hangup */
707 #define SIP_DTMF (3 << 16) /*!< DTMF Support: four settings, uses two bits */
708 #define SIP_DTMF_RFC2833 (0 << 16) /*!< DTMF Support: RTP DTMF - "rfc2833" */
709 #define SIP_DTMF_INBAND (1 << 16) /*!< DTMF Support: Inband audio, only for ULAW/ALAW - "inband" */
710 #define SIP_DTMF_INFO (2 << 16) /*!< DTMF Support: SIP Info messages - "info" */
711 #define SIP_DTMF_AUTO (3 << 16) /*!< DTMF Support: AUTO switch between rfc2833 and in-band DTMF */
713 #define SIP_NAT (3 << 18) /*!< four settings, uses two bits */
714 #define SIP_NAT_NEVER (0 << 18) /*!< No nat support */
715 #define SIP_NAT_RFC3581 (1 << 18) /*!< NAT RFC3581 */
716 #define SIP_NAT_ROUTE (2 << 18) /*!< NAT Only ROUTE */
717 #define SIP_NAT_ALWAYS (3 << 18) /*!< NAT Both ROUTE and RFC3581 */
718 /* re-INVITE related settings */
719 #define SIP_REINVITE (7 << 20) /*!< three bits used */
720 #define SIP_CAN_REINVITE (1 << 20) /*!< allow peers to be reinvited to send media directly p2p */
721 #define SIP_CAN_REINVITE_NAT (2 << 20) /*!< allow media reinvite when new peer is behind NAT */
722 #define SIP_REINVITE_UPDATE (4 << 20) /*!< use UPDATE (RFC3311) when reinviting this peer */
723 /* "insecure" settings */
724 #define SIP_INSECURE_PORT (1 << 23) /*!< don't require matching port for incoming requests */
725 #define SIP_INSECURE_INVITE (1 << 24) /*!< don't require authentication for incoming INVITEs */
726 /* Sending PROGRESS in-band settings */
727 #define SIP_PROG_INBAND (3 << 25) /*!< three settings, uses two bits */
728 #define SIP_PROG_INBAND_NEVER (0 << 25)
729 #define SIP_PROG_INBAND_NO (1 << 25)
730 #define SIP_PROG_INBAND_YES (2 << 25)
731 #define SIP_NO_HISTORY (1 << 27) /*!< Suppress recording request/response history */
732 #define SIP_CALL_LIMIT (1 << 28) /*!< Call limit enforced for this call */
733 #define SIP_SENDRPID (1 << 29) /*!< Remote Party-ID Support */
734 #define SIP_INC_COUNT (1 << 30) /*!< Did this connection increment the counter of in-use calls? */
735 #define SIP_G726_NONSTANDARD (1 << 31) /*!< Use non-standard packing for G726-32 data */
737 #define SIP_FLAGS_TO_COPY \
738 (SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_SENDRPID | SIP_DTMF | SIP_REINVITE | \
739 SIP_PROG_INBAND | SIP_USECLIENTCODE | SIP_NAT | SIP_G726_NONSTANDARD | \
740 SIP_USEREQPHONE | SIP_INSECURE_PORT | SIP_INSECURE_INVITE)
742 /*--- a new page of flags (for flags[1] */
744 #define SIP_PAGE2_RTCACHEFRIENDS (1 << 0)
745 #define SIP_PAGE2_RTUPDATE (1 << 1)
746 #define SIP_PAGE2_RTAUTOCLEAR (1 << 2)
747 #define SIP_PAGE2_RT_FROMCONTACT (1 << 4)
748 #define SIP_PAGE2_RTSAVE_SYSNAME (1 << 5)
749 /* Space for addition of other realtime flags in the future */
750 #define SIP_PAGE2_IGNOREREGEXPIRE (1 << 10)
751 #define SIP_PAGE2_DEBUG (3 << 11)
752 #define SIP_PAGE2_DEBUG_CONFIG (1 << 11)
753 #define SIP_PAGE2_DEBUG_CONSOLE (1 << 12)
754 #define SIP_PAGE2_DYNAMIC (1 << 13) /*!< Dynamic Peers register with Asterisk */
755 #define SIP_PAGE2_SELFDESTRUCT (1 << 14) /*!< Automatic peers need to destruct themselves */
756 #define SIP_PAGE2_VIDEOSUPPORT (1 << 15)
757 #define SIP_PAGE2_ALLOWSUBSCRIBE (1 << 16) /*!< Allow subscriptions from this peer? */
758 #define SIP_PAGE2_ALLOWOVERLAP (1 << 17) /*!< Allow overlap dialing ? */
759 #define SIP_PAGE2_SUBSCRIBEMWIONLY (1 << 18) /*!< Only issue MWI notification if subscribed to */
760 #define SIP_PAGE2_INC_RINGING (1 << 19) /*!< Did this connection increment the counter of in-use calls? */
761 #define SIP_PAGE2_T38SUPPORT (7 << 20) /*!< T38 Fax Passthrough Support */
762 #define SIP_PAGE2_T38SUPPORT_UDPTL (1 << 20) /*!< 20: T38 Fax Passthrough Support */
763 #define SIP_PAGE2_T38SUPPORT_RTP (2 << 20) /*!< 21: T38 Fax Passthrough Support */
764 #define SIP_PAGE2_T38SUPPORT_TCP (4 << 20) /*!< 22: T38 Fax Passthrough Support */
765 #define SIP_PAGE2_CALL_ONHOLD (3 << 23) /*!< Call states */
766 #define SIP_PAGE2_CALL_ONHOLD_ONEDIR (1 << 23) /*!< 23: One directional hold */
767 #define SIP_PAGE2_CALL_ONHOLD_INACTIVE (2 << 24) /*!< 24: Inactive */
768 #define SIP_PAGE2_RFC2833_COMPENSATE (1 << 26)
770 #define SIP_PAGE2_FLAGS_TO_COPY \
771 (SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT | SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE)
773 /* SIP packet flags */
774 #define SIP_PKT_DEBUG (1 << 0) /*!< Debug this packet */
775 #define SIP_PKT_WITH_TOTAG (1 << 1) /*!< This packet has a to-tag */
776 #define SIP_PKT_IGNORE (1 << 2) /*!< This is a re-transmit, ignore it */
777 #define SIP_PKT_IGNORE_RESP (1 << 3) /*!< Resp ignore - ??? */
778 #define SIP_PKT_IGNORE_REQ (1 << 4) /*!< Req ignore - ??? */
780 /* T.38 set of flags */
781 #define T38FAX_FILL_BIT_REMOVAL (1 << 0) /*!< Default: 0 (unset)*/
782 #define T38FAX_TRANSCODING_MMR (1 << 1) /*!< Default: 0 (unset)*/
783 #define T38FAX_TRANSCODING_JBIG (1 << 2) /*!< Default: 0 (unset)*/
784 /* Rate management */
785 #define T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF (0 << 3)
786 #define T38FAX_RATE_MANAGEMENT_LOCAL_TCF (1 << 3) /*!< Unset for transferredTCF (UDPTL), set for localTCF (TPKT) */
787 /* UDP Error correction */
788 #define T38FAX_UDP_EC_NONE (0 << 4) /*!< two bits, if unset NO t38UDPEC field in T38 SDP*/
789 #define T38FAX_UDP_EC_FEC (1 << 4) /*!< Set for t38UDPFEC */
790 #define T38FAX_UDP_EC_REDUNDANCY (2 << 4) /*!< Set for t38UDPRedundancy */
791 /* T38 Spec version */
792 #define T38FAX_VERSION (3 << 6) /*!< two bits, 2 values so far, up to 4 values max */
793 #define T38FAX_VERSION_0 (0 << 6) /*!< Version 0 */
794 #define T38FAX_VERSION_1 (1 << 6) /*!< Version 1 */
795 /* Maximum Fax Rate */
796 #define T38FAX_RATE_2400 (1 << 8) /*!< 2400 bps t38FaxRate */
797 #define T38FAX_RATE_4800 (1 << 9) /*!< 4800 bps t38FaxRate */
798 #define T38FAX_RATE_7200 (1 << 10) /*!< 7200 bps t38FaxRate */
799 #define T38FAX_RATE_9600 (1 << 11) /*!< 9600 bps t38FaxRate */
800 #define T38FAX_RATE_12000 (1 << 12) /*!< 12000 bps t38FaxRate */
801 #define T38FAX_RATE_14400 (1 << 13) /*!< 14400 bps t38FaxRate */
803 /*!< This is default: NO MMR and JBIG trancoding, NO fill bit removal, transferredTCF TCF, UDP FEC, Version 0 and 9600 max fax rate */
804 static int global_t38_capability = T38FAX_VERSION_0 | T38FAX_RATE_2400 | T38FAX_RATE_4800 | T38FAX_RATE_7200 | T38FAX_RATE_9600;
806 #define sipdebug ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG)
807 #define sipdebug_config ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONFIG)
808 #define sipdebug_console ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONSOLE)
810 /*! \brief T38 States for a call */
812 T38_DISABLED = 0, /*!< Not enabled */
813 T38_LOCAL_DIRECT, /*!< Offered from local */
814 T38_LOCAL_REINVITE, /*!< Offered from local - REINVITE */
815 T38_PEER_DIRECT, /*!< Offered from peer */
816 T38_PEER_REINVITE, /*!< Offered from peer - REINVITE */
817 T38_ENABLED /*!< Negotiated (enabled) */
820 /*! \brief T.38 channel settings (at some point we need to make this alloc'ed */
821 struct t38properties {
822 struct ast_flags t38support; /*!< Flag for udptl, rtp or tcp support for this session */
823 int capability; /*!< Our T38 capability */
824 int peercapability; /*!< Peers T38 capability */
825 int jointcapability; /*!< Supported T38 capability at both ends */
826 enum t38state state; /*!< T.38 state */
829 /*! \brief Parameters to know status of transfer */
831 REFER_IDLE, /*!< No REFER is in progress */
832 REFER_SENT, /*!< Sent REFER to transferee */
833 REFER_RECEIVED, /*!< Received REFER from transferer */
834 REFER_CONFIRMED, /*!< Refer confirmed with a 100 TRYING */
835 REFER_ACCEPTED, /*!< Accepted by transferee */
836 REFER_RINGING, /*!< Target Ringing */
837 REFER_200OK, /*!< Answered by transfer target */
838 REFER_FAILED, /*!< REFER declined - go on */
839 REFER_NOAUTH /*!< We had no auth for REFER */
842 static const struct c_referstatusstring {
843 enum referstatus status;
845 } referstatusstrings[] = {
846 { REFER_IDLE, "<none>" },
847 { REFER_SENT, "Request sent" },
848 { REFER_RECEIVED, "Request received" },
849 { REFER_ACCEPTED, "Accepted" },
850 { REFER_RINGING, "Target ringing" },
851 { REFER_200OK, "Done" },
852 { REFER_FAILED, "Failed" },
853 { REFER_NOAUTH, "Failed - auth failure" }
856 /*! \brief Structure to handle SIP transfers. Dynamically allocated when needed */
857 /* OEJ: Should be moved to string fields */
859 char refer_to[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO extension */
860 char refer_to_domain[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO domain */
861 char refer_to_urioption[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO uri options */
862 char refer_to_context[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO context */
863 char referred_by[AST_MAX_EXTENSION]; /*!< Place to store REFERRED-BY extension */
864 char referred_by_name[AST_MAX_EXTENSION]; /*!< Place to store REFERRED-BY extension */
865 char refer_contact[AST_MAX_EXTENSION]; /*!< Place to store Contact info from a REFER extension */
866 char replaces_callid[BUFSIZ]; /*!< Replace info: callid */
867 char replaces_callid_totag[BUFSIZ/2]; /*!< Replace info: to-tag */
868 char replaces_callid_fromtag[BUFSIZ/2]; /*!< Replace info: from-tag */
869 struct sip_pvt *refer_call; /*!< Call we are referring */
870 int attendedtransfer; /*!< Attended or blind transfer? */
871 int localtransfer; /*!< Transfer to local domain? */
872 enum referstatus status; /*!< REFER status */
875 /*! \brief sip_pvt: PVT structures are used for each SIP dialog, ie. a call, a registration, a subscribe */
877 ast_mutex_t pvt_lock; /*!< Dialog private lock */
878 int method; /*!< SIP method that opened this dialog */
879 AST_DECLARE_STRING_FIELDS(
880 AST_STRING_FIELD(callid); /*!< Global CallID */
881 AST_STRING_FIELD(randdata); /*!< Random data */
882 AST_STRING_FIELD(accountcode); /*!< Account code */
883 AST_STRING_FIELD(realm); /*!< Authorization realm */
884 AST_STRING_FIELD(nonce); /*!< Authorization nonce */
885 AST_STRING_FIELD(opaque); /*!< Opaque nonsense */
886 AST_STRING_FIELD(qop); /*!< Quality of Protection, since SIP wasn't complicated enough yet. */
887 AST_STRING_FIELD(domain); /*!< Authorization domain */
888 AST_STRING_FIELD(from); /*!< The From: header */
889 AST_STRING_FIELD(useragent); /*!< User agent in SIP request */
890 AST_STRING_FIELD(exten); /*!< Extension where to start */
891 AST_STRING_FIELD(context); /*!< Context for this call */
892 AST_STRING_FIELD(subscribecontext); /*!< Subscribecontext */
893 AST_STRING_FIELD(subscribeuri); /*!< Subscribecontext */
894 AST_STRING_FIELD(fromdomain); /*!< Domain to show in the from field */
895 AST_STRING_FIELD(fromuser); /*!< User to show in the user field */
896 AST_STRING_FIELD(fromname); /*!< Name to show in the user field */
897 AST_STRING_FIELD(tohost); /*!< Host we should put in the "to" field */
898 AST_STRING_FIELD(language); /*!< Default language for this call */
899 AST_STRING_FIELD(mohinterpret); /*!< MOH class to use when put on hold */
900 AST_STRING_FIELD(mohsuggest); /*!< MOH class to suggest when putting a peer on hold */
901 AST_STRING_FIELD(rdnis); /*!< Referring DNIS */
902 AST_STRING_FIELD(redircause); /*!< Referring cause */
903 AST_STRING_FIELD(theirtag); /*!< Their tag */
904 AST_STRING_FIELD(username); /*!< [user] name */
905 AST_STRING_FIELD(peername); /*!< [peer] name, not set if [user] */
906 AST_STRING_FIELD(authname); /*!< Who we use for authentication */
907 AST_STRING_FIELD(uri); /*!< Original requested URI */
908 AST_STRING_FIELD(okcontacturi); /*!< URI from the 200 OK on INVITE */
909 AST_STRING_FIELD(peersecret); /*!< Password */
910 AST_STRING_FIELD(peermd5secret);
911 AST_STRING_FIELD(cid_num); /*!< Caller*ID number */
912 AST_STRING_FIELD(cid_name); /*!< Caller*ID name */
913 AST_STRING_FIELD(via); /*!< Via: header */
914 AST_STRING_FIELD(fullcontact); /*!< The Contact: that the UA registers with us */
915 /* we only store the part in <brackets> in this field. */
916 AST_STRING_FIELD(our_contact); /*!< Our contact header */
917 AST_STRING_FIELD(rpid); /*!< Our RPID header */
918 AST_STRING_FIELD(rpid_from); /*!< Our RPID From header */
920 unsigned int ocseq; /*!< Current outgoing seqno */
921 unsigned int icseq; /*!< Current incoming seqno */
922 ast_group_t callgroup; /*!< Call group */
923 ast_group_t pickupgroup; /*!< Pickup group */
924 int lastinvite; /*!< Last Cseq of invite */
925 struct ast_flags flags[2]; /*!< SIP_ flags */
926 int timer_t1; /*!< SIP timer T1, ms rtt */
927 unsigned int sipoptions; /*!< Supported SIP options on the other end */
928 struct ast_codec_pref prefs; /*!< codec prefs */
929 int capability; /*!< Special capability (codec) */
930 int jointcapability; /*!< Supported capability at both ends (codecs) */
931 int peercapability; /*!< Supported peer capability */
932 int prefcodec; /*!< Preferred codec (outbound only) */
933 int noncodeccapability; /*!< DTMF RFC2833 telephony-event */
934 int redircodecs; /*!< Redirect codecs */
935 int maxcallbitrate; /*!< Maximum Call Bitrate for Video Calls */
936 struct t38properties t38; /*!< T38 settings */
937 struct sockaddr_in udptlredirip; /*!< Where our T.38 UDPTL should be going if not to us */
938 struct ast_udptl *udptl; /*!< T.38 UDPTL session */
939 int callingpres; /*!< Calling presentation */
940 int authtries; /*!< Times we've tried to authenticate */
941 int expiry; /*!< How long we take to expire */
942 long branch; /*!< The branch identifier of this session */
943 char tag[11]; /*!< Our tag for this session */
944 int sessionid; /*!< SDP Session ID */
945 int sessionversion; /*!< SDP Session Version */
946 struct sockaddr_in sa; /*!< Our peer */
947 struct sockaddr_in redirip; /*!< Where our RTP should be going if not to us */
948 struct sockaddr_in vredirip; /*!< Where our Video RTP should be going if not to us */
949 time_t lastrtprx; /*!< Last RTP received */
950 time_t lastrtptx; /*!< Last RTP sent */
951 int rtptimeout; /*!< RTP timeout time */
952 int rtpholdtimeout; /*!< RTP timeout when on hold */
953 int rtpkeepalive; /*!< Send RTP packets for keepalive */
954 struct sockaddr_in recv; /*!< Received as */
955 struct in_addr ourip; /*!< Our IP */
956 struct ast_channel *owner; /*!< Who owns us (if we have an owner) */
957 struct sip_route *route; /*!< Head of linked list of routing steps (fm Record-Route) */
958 int route_persistant; /*!< Is this the "real" route? */
959 struct sip_auth *peerauth; /*!< Realm authentication */
960 int noncecount; /*!< Nonce-count */
961 char lastmsg[256]; /*!< Last Message sent/received */
962 int amaflags; /*!< AMA Flags */
963 int pendinginvite; /*!< Any pending invite ? (seqno of this) */
964 struct sip_request initreq; /*!< Latest request that opened a new transaction
966 NOT the request that opened the dialog
969 int initid; /*!< Auto-congest ID if appropriate (scheduler) */
970 int autokillid; /*!< Auto-kill ID (scheduler) */
971 enum transfermodes allowtransfer; /*!< REFER: restriction scheme */
972 struct sip_refer *refer; /*!< REFER: SIP transfer data structure */
973 enum subscriptiontype subscribed; /*!< SUBSCRIBE: Is this dialog a subscription? */
974 int stateid; /*!< SUBSCRIBE: ID for devicestate subscriptions */
975 int laststate; /*!< SUBSCRIBE: Last known extension state */
976 int dialogver; /*!< SUBSCRIBE: Version for subscription dialog-info */
978 struct ast_dsp *vad; /*!< Inband DTMF Detection dsp */
980 struct sip_peer *relatedpeer; /*!< If this dialog is related to a peer, which one
981 Used in peerpoke, mwi subscriptions */
982 struct sip_registry *registry; /*!< If this is a REGISTER dialog, to which registry */
983 struct ast_rtp *rtp; /*!< RTP Session */
984 struct ast_rtp *vrtp; /*!< Video RTP session */
985 struct sip_pkt *packets; /*!< Packets scheduled for re-transmission */
986 struct sip_history_head *history; /*!< History of this SIP dialog */
987 struct ast_variable *chanvars; /*!< Channel variables to set for inbound call */
988 struct sip_pvt *next; /*!< Next dialog in chain */
989 struct sip_invite_param *options; /*!< Options for INVITE */
990 int autoframing; /*!< The number of Asters we group in a Pyroflax
991 before strolling to the Grokyzpå
992 (A bit unsure of this, please correct if
996 static struct sip_pvt *dialoglist = NULL;
998 #define FLAG_RESPONSE (1 << 0)
999 #define FLAG_FATAL (1 << 1)
1001 /*! \brief sip packet - raw format for outbound packets that are sent or scheduled for transmission */
1003 struct sip_pkt *next; /*!< Next packet in linked list */
1004 int retrans; /*!< Retransmission number */
1005 int method; /*!< SIP method for this packet */
1006 int seqno; /*!< Sequence number */
1007 unsigned int flags; /*!< non-zero if this is a response packet (e.g. 200 OK) */
1008 struct sip_pvt *owner; /*!< Owner AST call */
1009 int retransid; /*!< Retransmission ID */
1010 int timer_a; /*!< SIP timer A, retransmission timer */
1011 int timer_t1; /*!< SIP Timer T1, estimated RTT or 500 ms */
1012 int packetlen; /*!< Length of packet */
1016 /*! \brief Structure for SIP user data. User's place calls to us */
1018 /* Users who can access various contexts */
1019 ASTOBJ_COMPONENTS(struct sip_user);
1020 char secret[80]; /*!< Password */
1021 char md5secret[80]; /*!< Password in md5 */
1022 char context[AST_MAX_CONTEXT]; /*!< Default context for incoming calls */
1023 char subscribecontext[AST_MAX_CONTEXT]; /* Default context for subscriptions */
1024 char cid_num[80]; /*!< Caller ID num */
1025 char cid_name[80]; /*!< Caller ID name */
1026 char accountcode[AST_MAX_ACCOUNT_CODE]; /* Account code */
1027 char language[MAX_LANGUAGE]; /*!< Default language for this user */
1028 char mohinterpret[MAX_MUSICCLASS];/*!< Music on Hold class */
1029 char mohsuggest[MAX_MUSICCLASS];/*!< Music on Hold class */
1030 char useragent[256]; /*!< User agent in SIP request */
1031 struct ast_codec_pref prefs; /*!< codec prefs */
1032 ast_group_t callgroup; /*!< Call group */
1033 ast_group_t pickupgroup; /*!< Pickup Group */
1034 unsigned int sipoptions; /*!< Supported SIP options */
1035 struct ast_flags flags[2]; /*!< SIP_ flags */
1036 int amaflags; /*!< AMA flags for billing */
1037 int callingpres; /*!< Calling id presentation */
1038 int capability; /*!< Codec capability */
1039 int inUse; /*!< Number of calls in use */
1040 int call_limit; /*!< Limit of concurrent calls */
1041 enum transfermodes allowtransfer; /*! SIP Refer restriction scheme */
1042 struct ast_ha *ha; /*!< ACL setting */
1043 struct ast_variable *chanvars; /*!< Variables to set for channel created by user */
1044 int maxcallbitrate; /*!< Maximum Bitrate for a video call */
1048 /*! \brief Structure for SIP peer data, we place calls to peers if registered or fixed IP address (host) */
1049 /* XXX field 'name' must be first otherwise sip_addrcmp() will fail */
1051 ASTOBJ_COMPONENTS(struct sip_peer); /*!< name, refcount, objflags, object pointers */
1052 /*!< peer->name is the unique name of this object */
1053 char secret[80]; /*!< Password */
1054 char md5secret[80]; /*!< Password in MD5 */
1055 struct sip_auth *auth; /*!< Realm authentication list */
1056 char context[AST_MAX_CONTEXT]; /*!< Default context for incoming calls */
1057 char subscribecontext[AST_MAX_CONTEXT]; /*!< Default context for subscriptions */
1058 char username[80]; /*!< Temporary username until registration */
1059 char accountcode[AST_MAX_ACCOUNT_CODE]; /*!< Account code */
1060 int amaflags; /*!< AMA Flags (for billing) */
1061 char tohost[MAXHOSTNAMELEN]; /*!< If not dynamic, IP address */
1062 char regexten[AST_MAX_EXTENSION]; /*!< Extension to register (if regcontext is used) */
1063 char fromuser[80]; /*!< From: user when calling this peer */
1064 char fromdomain[MAXHOSTNAMELEN]; /*!< From: domain when calling this peer */
1065 char fullcontact[256]; /*!< Contact registered with us (not in sip.conf) */
1066 char cid_num[80]; /*!< Caller ID num */
1067 char cid_name[80]; /*!< Caller ID name */
1068 int callingpres; /*!< Calling id presentation */
1069 int inUse; /*!< Number of calls in use */
1070 int inRinging; /*!< Number of calls ringing */
1071 int onHold; /*!< Peer has someone on hold */
1072 int call_limit; /*!< Limit of concurrent calls */
1073 enum transfermodes allowtransfer; /*! SIP Refer restriction scheme */
1074 char vmexten[AST_MAX_EXTENSION]; /*!< Dialplan extension for MWI notify message*/
1075 char mailbox[AST_MAX_EXTENSION]; /*!< Mailbox setting for MWI checks */
1076 char language[MAX_LANGUAGE]; /*!< Default language for prompts */
1077 char mohinterpret[MAX_MUSICCLASS];/*!< Music on Hold class */
1078 char mohsuggest[MAX_MUSICCLASS];/*!< Music on Hold class */
1079 char useragent[256]; /*!< User agent in SIP request (saved from registration) */
1080 struct ast_codec_pref prefs; /*!< codec prefs */
1082 time_t lastmsgcheck; /*!< Last time we checked for MWI */
1083 unsigned int sipoptions; /*!< Supported SIP options */
1084 struct ast_flags flags[2]; /*!< SIP_ flags */
1085 int expire; /*!< When to expire this peer registration */
1086 int capability; /*!< Codec capability */
1087 int rtptimeout; /*!< RTP timeout */
1088 int rtpholdtimeout; /*!< RTP Hold Timeout */
1089 int rtpkeepalive; /*!< Send RTP packets for keepalive */
1090 ast_group_t callgroup; /*!< Call group */
1091 ast_group_t pickupgroup; /*!< Pickup group */
1092 struct ast_dnsmgr_entry *dnsmgr;/*!< DNS refresh manager for peer */
1093 struct sockaddr_in addr; /*!< IP address of peer */
1094 int maxcallbitrate; /*!< Maximum Bitrate for a video call */
1097 struct sip_pvt *call; /*!< Call pointer */
1098 int pokeexpire; /*!< When to expire poke (qualify= checking) */
1099 int lastms; /*!< How long last response took (in ms), or -1 for no response */
1100 int maxms; /*!< Max ms we will accept for the host to be up, 0 to not monitor */
1101 struct timeval ps; /*!< Ping send time */
1103 struct sockaddr_in defaddr; /*!< Default IP address, used until registration */
1104 struct ast_ha *ha; /*!< Access control list */
1105 struct ast_variable *chanvars; /*!< Variables to set for channel created by user */
1106 struct sip_pvt *mwipvt; /*!< Subscription for MWI */
1113 /*! \brief Registrations with other SIP proxies */
1114 struct sip_registry {
1115 ASTOBJ_COMPONENTS_FULL(struct sip_registry,1,1);
1116 AST_DECLARE_STRING_FIELDS(
1117 AST_STRING_FIELD(callid); /*!< Global Call-ID */
1118 AST_STRING_FIELD(realm); /*!< Authorization realm */
1119 AST_STRING_FIELD(nonce); /*!< Authorization nonce */
1120 AST_STRING_FIELD(opaque); /*!< Opaque nonsense */
1121 AST_STRING_FIELD(qop); /*!< Quality of Protection, since SIP wasn't complicated enough yet. */
1122 AST_STRING_FIELD(domain); /*!< Authorization domain */
1123 AST_STRING_FIELD(username); /*!< Who we are registering as */
1124 AST_STRING_FIELD(authuser); /*!< Who we *authenticate* as */
1125 AST_STRING_FIELD(hostname); /*!< Domain or host we register to */
1126 AST_STRING_FIELD(secret); /*!< Password in clear text */
1127 AST_STRING_FIELD(md5secret); /*!< Password in md5 */
1128 AST_STRING_FIELD(callback); /*!< Contact extension */
1129 AST_STRING_FIELD(random);
1131 int portno; /*!< Optional port override */
1132 int expire; /*!< Sched ID of expiration */
1133 int expiry; /*!< Value to use for the Expires header */
1134 int regattempts; /*!< Number of attempts (since the last success) */
1135 int timeout; /*!< sched id of sip_reg_timeout */
1136 int refresh; /*!< How often to refresh */
1137 struct sip_pvt *call; /*!< create a sip_pvt structure for each outbound "registration dialog" in progress */
1138 enum sipregistrystate regstate; /*!< Registration state (see above) */
1139 time_t regtime; /*!< Last succesful registration time */
1140 int callid_valid; /*!< 0 means we haven't chosen callid for this registry yet. */
1141 unsigned int ocseq; /*!< Sequence number we got to for REGISTERs for this registry */
1142 struct sockaddr_in us; /*!< Who the server thinks we are */
1143 int noncecount; /*!< Nonce-count */
1144 char lastmsg[256]; /*!< Last Message sent/received */
1147 /* --- Linked lists of various objects --------*/
1149 /*! \brief The user list: Users and friends */
1150 static struct ast_user_list {
1151 ASTOBJ_CONTAINER_COMPONENTS(struct sip_user);
1154 /*! \brief The peer list: Peers and Friends */
1155 static struct ast_peer_list {
1156 ASTOBJ_CONTAINER_COMPONENTS(struct sip_peer);
1159 /*! \brief The register list: Other SIP proxys we register with and place calls to */
1160 static struct ast_register_list {
1161 ASTOBJ_CONTAINER_COMPONENTS(struct sip_registry);
1165 static int temp_pvt_init(void *);
1166 static void temp_pvt_cleanup(void *);
1168 /*! \brief A per-thread temporary pvt structure */
1169 AST_THREADSTORAGE_CUSTOM(ts_temp_pvt, temp_pvt_init, temp_pvt_cleanup);
1171 /*! \todo Move the sip_auth list to AST_LIST */
1172 static struct sip_auth *authl = NULL; /*!< Authentication list for realm authentication */
1175 /* --- Sockets and networking --------------*/
1176 static int sipsock = -1; /*!< Main socket for SIP network communication */
1177 static struct sockaddr_in bindaddr = { 0, }; /*!< The address we bind to */
1178 static struct sockaddr_in externip; /*!< External IP address if we are behind NAT */
1179 static char externhost[MAXHOSTNAMELEN]; /*!< External host name (possibly with dynamic DNS and DHCP */
1180 static time_t externexpire = 0; /*!< Expiration counter for re-resolving external host name in dynamic DNS */
1181 static int externrefresh = 10;
1182 static struct ast_ha *localaddr; /*!< List of local networks, on the same side of NAT as this Asterisk */
1183 static struct in_addr __ourip;
1184 static struct sockaddr_in outboundproxyip;
1186 static struct sockaddr_in debugaddr;
1188 static struct ast_config *notify_types; /*!< The list of manual NOTIFY types we know how to send */
1190 /*---------------------------- Forward declarations of functions in chan_sip.c */
1191 /*! \note This is added to help splitting up chan_sip.c into several files
1192 in coming releases */
1194 /*--- PBX interface functions */
1195 static struct ast_channel *sip_request_call(const char *type, int format, void *data, int *cause);
1196 static int sip_devicestate(void *data);
1197 static int sip_sendtext(struct ast_channel *ast, const char *text);
1198 static int sip_call(struct ast_channel *ast, char *dest, int timeout);
1199 static int sip_hangup(struct ast_channel *ast);
1200 static int sip_answer(struct ast_channel *ast);
1201 static struct ast_frame *sip_read(struct ast_channel *ast);
1202 static int sip_write(struct ast_channel *ast, struct ast_frame *frame);
1203 static int sip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen);
1204 static int sip_transfer(struct ast_channel *ast, const char *dest);
1205 static int sip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
1206 static int sip_senddigit_begin(struct ast_channel *ast, char digit);
1207 static int sip_senddigit_end(struct ast_channel *ast, char digit);
1209 /*--- Transmitting responses and requests */
1210 static int sipsock_read(int *id, int fd, short events, void *ignore);
1211 static int __sip_xmit(struct sip_pvt *p, char *data, int len);
1212 static int __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, char *data, int len, int fatal, int sipmethod);
1213 static int __transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
1214 static int retrans_pkt(void *data);
1215 static int transmit_sip_request(struct sip_pvt *p, struct sip_request *req);
1216 static int transmit_response_using_temp(ast_string_field callid, struct sockaddr_in *sin, int useglobal_nat, const int intended_method, const struct sip_request *req, const char *msg);
1217 static int transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req);
1218 static int transmit_response_reliable(struct sip_pvt *p, const char *msg, const struct sip_request *req);
1219 static int transmit_response_with_date(struct sip_pvt *p, const char *msg, const struct sip_request *req);
1220 static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
1221 static int transmit_response_with_unsupported(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *unsupported);
1222 static int transmit_response_with_auth(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *rand, enum xmittype reliable, const char *header, int stale);
1223 static int transmit_response_with_allow(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
1224 static void transmit_fake_auth_response(struct sip_pvt *p, struct sip_request *req, int reliable);
1225 static int transmit_request(struct sip_pvt *p, int sipmethod, int inc, enum xmittype reliable, int newbranch);
1226 static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch);
1227 static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init);
1228 static int transmit_reinvite_with_sdp(struct sip_pvt *p, int t38version);
1229 static int transmit_info_with_digit(struct sip_pvt *p, const char digit);
1230 static int transmit_info_with_vidupdate(struct sip_pvt *p);
1231 static int transmit_message_with_text(struct sip_pvt *p, const char *text);
1232 static int transmit_refer(struct sip_pvt *p, const char *dest);
1233 static int transmit_notify_with_mwi(struct sip_pvt *p, int newmsgs, int oldmsgs, char *vmexten);
1234 static int transmit_notify_with_sipfrag(struct sip_pvt *p, int cseq, char *message, int terminate);
1235 static int transmit_register(struct sip_registry *r, int sipmethod, const char *auth, const char *authheader);
1236 static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno);
1237 static int send_request(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno);
1238 static void copy_request(struct sip_request *dst, const struct sip_request *src);
1239 static void receive_message(struct sip_pvt *p, struct sip_request *req);
1240 static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req);
1241 static int sip_send_mwi_to_peer(struct sip_peer *peer);
1242 static int does_peer_need_mwi(struct sip_peer *peer);
1244 /*--- Dialog management */
1245 static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *sin,
1246 int useglobal_nat, const int intended_method);
1247 static int __sip_autodestruct(void *data);
1248 static void sip_scheddestroy(struct sip_pvt *p, int ms);
1249 static void sip_cancel_destroy(struct sip_pvt *p);
1250 static void sip_destroy(struct sip_pvt *p);
1251 static void __sip_destroy(struct sip_pvt *p, int lockowner);
1252 static void __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod, int reset);
1253 static void __sip_pretend_ack(struct sip_pvt *p);
1254 static int __sip_semi_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod);
1255 static int auto_congest(void *nothing);
1256 static int update_call_counter(struct sip_pvt *fup, int event);
1257 static int hangup_sip2cause(int cause);
1258 static const char *hangup_cause2sip(int cause);
1259 static struct sip_pvt *find_call(struct sip_request *req, struct sockaddr_in *sin, const int intended_method);
1260 static void free_old_route(struct sip_route *route);
1261 static void list_route(struct sip_route *route);
1262 static void build_route(struct sip_pvt *p, struct sip_request *req, int backwards);
1263 static enum check_auth_result register_verify(struct sip_pvt *p, struct sockaddr_in *sin,
1264 struct sip_request *req, char *uri);
1265 static struct sip_pvt *get_sip_pvt_byid_locked(const char *callid, const char *totag, const char *fromtag);
1266 static void check_pendings(struct sip_pvt *p);
1267 static void *sip_park_thread(void *stuff);
1268 static int sip_park(struct ast_channel *chan1, struct ast_channel *chan2, struct sip_request *req, int seqno);
1269 static int sip_sipredirect(struct sip_pvt *p, const char *dest);
1271 /*--- Codec handling / SDP */
1272 static void try_suggested_sip_codec(struct sip_pvt *p);
1273 static const char* get_sdp_iterate(int* start, struct sip_request *req, const char *name);
1274 static const char *get_sdp(struct sip_request *req, const char *name);
1275 static int find_sdp(struct sip_request *req);
1276 static int process_sdp(struct sip_pvt *p, struct sip_request *req);
1277 static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate,
1278 char **m_buf, size_t *m_size, char **a_buf, size_t *a_size,
1279 int debug, int *min_packet_size);
1280 static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_rate,
1281 char **m_buf, size_t *m_size, char **a_buf, size_t *a_size,
1283 static int add_sdp(struct sip_request *resp, struct sip_pvt *p);
1284 static void do_setnat(struct sip_pvt *p, int natflags);
1286 /*--- Authentication stuff */
1287 static int reply_digest(struct sip_pvt *p, struct sip_request *req, char *header, int sipmethod, char *digest, int digest_len);
1288 static int build_reply_digest(struct sip_pvt *p, int method, char *digest, int digest_len);
1289 static enum check_auth_result check_auth(struct sip_pvt *p, struct sip_request *req, const char *username,
1290 const char *secret, const char *md5secret, int sipmethod,
1291 char *uri, enum xmittype reliable, int ignore);
1292 static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_request *req,
1293 int sipmethod, char *uri, enum xmittype reliable,
1294 struct sockaddr_in *sin, struct sip_peer **authpeer);
1295 static int check_user(struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, enum xmittype reliable, struct sockaddr_in *sin);
1297 /*--- Domain handling */
1298 static int check_sip_domain(const char *domain, char *context, size_t len); /* Check if domain is one of our local domains */
1299 static int add_sip_domain(const char *domain, const enum domain_mode mode, const char *context);
1300 static void clear_sip_domains(void);
1302 /*--- SIP realm authentication */
1303 static struct sip_auth *add_realm_authentication(struct sip_auth *authlist, char *configuration, int lineno);
1304 static int clear_realm_authentication(struct sip_auth *authlist); /* Clear realm authentication list (at reload) */
1305 static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, const char *realm);
1307 /*--- Misc functions */
1308 static int sip_do_reload(enum channelreloadreason reason);
1309 static int reload_config(enum channelreloadreason reason);
1310 static int expire_register(void *data);
1311 static void *do_monitor(void *data);
1312 static int restart_monitor(void);
1313 static int sip_send_mwi_to_peer(struct sip_peer *peer);
1314 static void sip_destroy(struct sip_pvt *p);
1315 static int sip_addrcmp(char *name, struct sockaddr_in *sin); /* Support for peer matching */
1316 static int sip_refer_allocate(struct sip_pvt *p);
1317 static void ast_quiet_chan(struct ast_channel *chan);
1318 static int attempt_transfer(struct sip_dual *transferer, struct sip_dual *target);
1320 /*--- Device monitoring and Device/extension state handling */
1321 static int cb_extensionstate(char *context, char* exten, int state, void *data);
1322 static int sip_devicestate(void *data);
1323 static int sip_poke_noanswer(void *data);
1324 static int sip_poke_peer(struct sip_peer *peer);
1325 static void sip_poke_all_peers(void);
1326 static void sip_peer_hold(struct sip_pvt *p, int hold);
1328 /*--- Applications, functions, CLI and manager command helpers */
1329 static const char *sip_nat_mode(const struct sip_pvt *p);
1330 static int sip_show_inuse(int fd, int argc, char *argv[]);
1331 static char *transfermode2str(enum transfermodes mode) attribute_const;
1332 static char *nat2str(int nat) attribute_const;
1333 static int peer_status(struct sip_peer *peer, char *status, int statuslen);
1334 static int sip_show_users(int fd, int argc, char *argv[]);
1335 static int _sip_show_peers(int fd, int *total, struct mansession *s, struct message *m, int argc, char *argv[]);
1336 static int manager_sip_show_peers( struct mansession *s, struct message *m );
1337 static int sip_show_peers(int fd, int argc, char *argv[]);
1338 static int sip_show_objects(int fd, int argc, char *argv[]);
1339 static void print_group(int fd, ast_group_t group, int crlf);
1340 static const char *dtmfmode2str(int mode) attribute_const;
1341 static const char *insecure2str(int port, int invite) attribute_const;
1342 static void cleanup_stale_contexts(char *new, char *old);
1343 static void print_codec_to_cli(int fd, struct ast_codec_pref *pref);
1344 static const char *domain_mode_to_text(const enum domain_mode mode);
1345 static int sip_show_domains(int fd, int argc, char *argv[]);
1346 static int _sip_show_peer(int type, int fd, struct mansession *s, struct message *m, int argc, char *argv[]);
1347 static int manager_sip_show_peer( struct mansession *s, struct message *m);
1348 static int sip_show_peer(int fd, int argc, char *argv[]);
1349 static int _sip_show_peer(int type, int fd, struct mansession *s, struct message *m, int argc, char *argv[]);
1350 static int sip_show_user(int fd, int argc, char *argv[]);
1351 static int sip_show_registry(int fd, int argc, char *argv[]);
1352 static int sip_show_settings(int fd, int argc, char *argv[]);
1353 static const char *subscription_type2str(enum subscriptiontype subtype) attribute_pure;
1354 static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype);
1355 static int __sip_show_channels(int fd, int argc, char *argv[], int subscriptions);
1356 static int sip_show_channels(int fd, int argc, char *argv[]);
1357 static int sip_show_subscriptions(int fd, int argc, char *argv[]);
1358 static int __sip_show_channels(int fd, int argc, char *argv[], int subscriptions);
1359 static char *complete_sipch(const char *line, const char *word, int pos, int state);
1360 static char *complete_sip_peer(const char *word, int state, int flags2);
1361 static char *complete_sip_show_peer(const char *line, const char *word, int pos, int state);
1362 static char *complete_sip_debug_peer(const char *line, const char *word, int pos, int state);
1363 static char *complete_sip_user(const char *word, int state, int flags2);
1364 static char *complete_sip_show_user(const char *line, const char *word, int pos, int state);
1365 static char *complete_sipnotify(const char *line, const char *word, int pos, int state);
1366 static char *complete_sip_prune_realtime_peer(const char *line, const char *word, int pos, int state);
1367 static char *complete_sip_prune_realtime_user(const char *line, const char *word, int pos, int state);
1368 static int sip_show_channel(int fd, int argc, char *argv[]);
1369 static int sip_show_history(int fd, int argc, char *argv[]);
1370 static int sip_do_debug_ip(int fd, int argc, char *argv[]);
1371 static int sip_do_debug_peer(int fd, int argc, char *argv[]);
1372 static int sip_do_debug(int fd, int argc, char *argv[]);
1373 static int sip_no_debug(int fd, int argc, char *argv[]);
1374 static int sip_notify(int fd, int argc, char *argv[]);
1375 static int sip_do_history(int fd, int argc, char *argv[]);
1376 static int sip_no_history(int fd, int argc, char *argv[]);
1377 static int func_header_read(struct ast_channel *chan, char *function, char *data, char *buf, size_t len);
1378 static int func_check_sipdomain(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len);
1379 static int function_sippeer(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len);
1380 static int function_sipchaninfo_read(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len);
1381 static int sip_dtmfmode(struct ast_channel *chan, void *data);
1382 static int sip_addheader(struct ast_channel *chan, void *data);
1383 static int sip_do_reload(enum channelreloadreason reason);
1384 static int sip_reload(int fd, int argc, char *argv[]);
1387 Functions for enabling debug per IP or fully, or enabling history logging for
1390 static void sip_dump_history(struct sip_pvt *dialog); /* Dump history to LOG_DEBUG at end of dialog, before destroying data */
1391 static inline int sip_debug_test_addr(const struct sockaddr_in *addr);
1392 static inline int sip_debug_test_pvt(struct sip_pvt *p);
1393 static void append_history_full(struct sip_pvt *p, const char *fmt, ...);
1394 static void sip_dump_history(struct sip_pvt *dialog);
1396 /*--- Device object handling */
1397 static struct sip_peer *temp_peer(const char *name);
1398 static struct sip_peer *build_peer(const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime);
1399 static struct sip_user *build_user(const char *name, struct ast_variable *v, int realtime);
1400 static int update_call_counter(struct sip_pvt *fup, int event);
1401 static void sip_destroy_peer(struct sip_peer *peer);
1402 static void sip_destroy_user(struct sip_user *user);
1403 static int sip_poke_peer(struct sip_peer *peer);
1404 static void set_peer_defaults(struct sip_peer *peer);
1405 static struct sip_peer *temp_peer(const char *name);
1406 static void register_peer_exten(struct sip_peer *peer, int onoff);
1407 static void sip_destroy_peer(struct sip_peer *peer);
1408 static void sip_destroy_user(struct sip_user *user);
1409 static struct sip_peer *find_peer(const char *peer, struct sockaddr_in *sin, int realtime);
1410 static struct sip_user *find_user(const char *name, int realtime);
1411 static int sip_poke_peer_s(void *data);
1412 static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, struct sip_peer *p, struct sip_request *req);
1413 static void reg_source_db(struct sip_peer *peer);
1414 static void destroy_association(struct sip_peer *peer);
1415 static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask, struct ast_variable *v);
1417 /* Realtime device support */
1418 static void realtime_update_peer(const char *peername, struct sockaddr_in *sin, const char *username, const char *fullcontact, int expirey);
1419 static struct sip_user *realtime_user(const char *username);
1420 static void update_peer(struct sip_peer *p, int expiry);
1421 static struct sip_peer *realtime_peer(const char *peername, struct sockaddr_in *sin);
1422 static int sip_prune_realtime(int fd, int argc, char *argv[]);
1424 /*--- Internal UA client handling (outbound registrations) */
1425 static int ast_sip_ouraddrfor(struct in_addr *them, struct in_addr *us);
1426 static void sip_registry_destroy(struct sip_registry *reg);
1427 static int sip_register(char *value, int lineno);
1428 static char *regstate2str(enum sipregistrystate regstate) attribute_const;
1429 static int sip_reregister(void *data);
1430 static int __sip_do_register(struct sip_registry *r);
1431 static int sip_reg_timeout(void *data);
1432 static void sip_send_all_registers(void);
1434 /*--- Parsing SIP requests and responses */
1435 static void append_date(struct sip_request *req); /* Append date to SIP packet */
1436 static int determine_firstline_parts(struct sip_request *req);
1437 static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype);
1438 static const char *gettag(const struct sip_request *req, const char *header, char *tagbuf, int tagbufsize);
1439 static int find_sip_method(const char *msg);
1440 static unsigned int parse_sip_options(struct sip_pvt *pvt, const char *supported);
1441 static void parse_request(struct sip_request *req);
1442 static const char *get_header(const struct sip_request *req, const char *name);
1443 static const char *referstatus2str(enum referstatus rstatus) attribute_pure;
1444 static int method_match(enum sipmethod id, const char *name);
1445 static void parse_copy(struct sip_request *dst, const struct sip_request *src);
1446 static char *get_in_brackets(char *tmp);
1447 static const char *find_alias(const char *name, const char *_default);
1448 static const char *__get_header(const struct sip_request *req, const char *name, int *start);
1449 static int lws2sws(char *msgbuf, int len);
1450 static void extract_uri(struct sip_pvt *p, struct sip_request *req);
1451 static int get_refer_info(struct sip_pvt *transferer, struct sip_request *outgoing_req);
1452 static int get_also_info(struct sip_pvt *p, struct sip_request *oreq);
1453 static int parse_ok_contact(struct sip_pvt *pvt, struct sip_request *req);
1454 static int set_address_from_contact(struct sip_pvt *pvt);
1455 static void check_via(struct sip_pvt *p, struct sip_request *req);
1456 static char *get_calleridname(const char *input, char *output, size_t outputsize);
1457 static int get_rpid_num(const char *input, char *output, int maxlen);
1458 static int get_rdnis(struct sip_pvt *p, struct sip_request *oreq);
1459 static int get_destination(struct sip_pvt *p, struct sip_request *oreq);
1460 static int get_msg_text(char *buf, int len, struct sip_request *req);
1461 static void free_old_route(struct sip_route *route);
1462 static int transmit_state_notify(struct sip_pvt *p, int state, int full, int timeout);
1464 /*--- Constructing requests and responses */
1465 static void initialize_initreq(struct sip_pvt *p, struct sip_request *req);
1466 static int init_req(struct sip_request *req, int sipmethod, const char *recip);
1467 static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, int seqno, int newbranch);
1468 static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod);
1469 static int init_resp(struct sip_request *resp, const char *msg);
1470 static int respprep(struct sip_request *resp, struct sip_pvt *p, const char *msg, const struct sip_request *req);
1471 static const struct sockaddr_in *sip_real_dst(const struct sip_pvt *p);
1472 static void build_via(struct sip_pvt *p);
1473 static int create_addr_from_peer(struct sip_pvt *r, struct sip_peer *peer);
1474 static int create_addr(struct sip_pvt *dialog, const char *opeer);
1475 static char *generate_random_string(char *buf, size_t size);
1476 static void build_callid_pvt(struct sip_pvt *pvt);
1477 static void build_callid_registry(struct sip_registry *reg, struct in_addr ourip, const char *fromdomain);
1478 static void make_our_tag(char *tagbuf, size_t len);
1479 static int add_header(struct sip_request *req, const char *var, const char *value);
1480 static int add_header_contentLength(struct sip_request *req, int len);
1481 static int add_line(struct sip_request *req, const char *line);
1482 static int add_text(struct sip_request *req, const char *text);
1483 static int add_digit(struct sip_request *req, char digit);
1484 static int add_vidupdate(struct sip_request *req);
1485 static void add_route(struct sip_request *req, struct sip_route *route);
1486 static int copy_header(struct sip_request *req, const struct sip_request *orig, const char *field);
1487 static int copy_all_header(struct sip_request *req, const struct sip_request *orig, const char *field);
1488 static int copy_via_headers(struct sip_pvt *p, struct sip_request *req, const struct sip_request *orig, const char *field);
1489 static void set_destination(struct sip_pvt *p, char *uri);
1490 static void append_date(struct sip_request *req);
1491 static void build_contact(struct sip_pvt *p);
1492 static void build_rpid(struct sip_pvt *p);
1494 /*------Request handling functions */
1495 static int handle_request(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int *recount, int *nounlock);
1496 static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int debug, int seqno, struct sockaddr_in *sin, int *recount, char *e);
1497 static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int debug, int seqno, int *nounlock);
1498 static int handle_request_bye(struct sip_pvt *p, struct sip_request *req);
1499 static int handle_request_register(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, char *e);
1500 static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req);
1501 static int handle_request_message(struct sip_pvt *p, struct sip_request *req);
1502 static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int seqno, char *e);
1503 static void handle_request_info(struct sip_pvt *p, struct sip_request *req);
1504 static int handle_request_options(struct sip_pvt *p, struct sip_request *req);
1505 static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, int debug, int ignore, int seqno, struct sockaddr_in *sin);
1506 static int handle_request_notify(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int seqno, char *e);
1507 static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, int debug, int ignore, int seqno, struct sockaddr_in *sin);
1508 static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual *current, struct sip_request *req, int seqno);
1510 /*------Response handling functions */
1511 static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
1512 static void handle_response_refer(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
1513 static int handle_response_register(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
1514 static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
1516 /*----- RTP interface functions */
1517 static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs, int nat_active);
1518 static enum ast_rtp_get_result sip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
1519 static enum ast_rtp_get_result sip_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
1520 static int sip_get_codec(struct ast_channel *chan);
1521 static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p, int *faxdetect);
1523 /*------ T38 Support --------- */
1524 static int sip_handle_t38_reinvite(struct ast_channel *chan, struct sip_pvt *pvt, int reinvite); /*!< T38 negotiation helper function */
1525 static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans);
1526 static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan);
1527 static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl);
1529 /*! \brief Definition of this channel for PBX channel registration */
1530 static const struct ast_channel_tech sip_tech = {
1532 .description = "Session Initiation Protocol (SIP)",
1533 .capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
1534 .properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,
1535 .requester = sip_request_call,
1536 .devicestate = sip_devicestate,
1538 .hangup = sip_hangup,
1539 .answer = sip_answer,
1542 .write_video = sip_write,
1543 .indicate = sip_indicate,
1544 .transfer = sip_transfer,
1546 .send_digit_begin = sip_senddigit_begin,
1547 .send_digit_end = sip_senddigit_end,
1548 .bridge = ast_rtp_bridge,
1549 .early_bridge = ast_rtp_early_bridge,
1550 .send_text = sip_sendtext,
1553 /**--- some list management macros. **/
1555 #define UNLINK(element, head, prev) do { \
1557 (prev)->next = (element)->next; \
1559 (head) = (element)->next; \
1562 /*! \brief Interface structure with callbacks used to connect to RTP module */
1563 static struct ast_rtp_protocol sip_rtp = {
1565 get_rtp_info: sip_get_rtp_peer,
1566 get_vrtp_info: sip_get_vrtp_peer,
1567 set_rtp_peer: sip_set_rtp_peer,
1568 get_codec: sip_get_codec,
1572 * Helper functions to lock/unlock pvt, hiding the
1573 * underlying locking mechanism.
1575 static void sip_pvt_lock(struct sip_pvt *pvt)
1577 ast_mutex_lock(&pvt->pvt_lock);
1580 static void sip_pvt_unlock(struct sip_pvt *pvt)
1582 ast_mutex_unlock(&pvt->pvt_lock);
1585 /*! \brief Interface structure with callbacks used to connect to UDPTL module*/
1586 static struct ast_udptl_protocol sip_udptl = {
1588 get_udptl_info: sip_get_udptl_peer,
1589 set_udptl_peer: sip_set_udptl_peer,
1592 /*! \brief Convert transfer status to string */
1593 static const char *referstatus2str(enum referstatus rstatus)
1595 int i = (sizeof(referstatusstrings) / sizeof(referstatusstrings[0]));
1598 for (x = 0; x < i; x++) {
1599 if (referstatusstrings[x].status == rstatus)
1600 return referstatusstrings[x].text;
1605 /*! \brief Initialize the initital request packet in the pvt structure.
1606 This packet is used for creating replies and future requests in
1608 static void initialize_initreq(struct sip_pvt *p, struct sip_request *req)
1610 if (p->initreq.headers && option_debug) {
1611 ast_log(LOG_DEBUG, "Initializing already initialized SIP dialog %s (presumably reinvite)\n", p->callid);
1613 /* Use this as the basis */
1614 copy_request(&p->initreq, req);
1615 parse_request(&p->initreq);
1616 if (ast_test_flag(req, SIP_PKT_DEBUG))
1617 ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines);
1621 /*! \brief returns true if 'name' (with optional trailing whitespace)
1622 * matches the sip method 'id'.
1623 * Strictly speaking, SIP methods are case SENSITIVE, but we do
1624 * a case-insensitive comparison to be more tolerant.
1625 * following Jon Postel's rule: Be gentle in what you accept, strict with what you send
1627 static int method_match(enum sipmethod id, const char *name)
1629 int len = strlen(sip_methods[id].text);
1630 int l_name = name ? strlen(name) : 0;
1631 /* true if the string is long enough, and ends with whitespace, and matches */
1632 return (l_name >= len && name[len] < 33 &&
1633 !strncasecmp(sip_methods[id].text, name, len));
1636 /*! \brief find_sip_method: Find SIP method from header */
1637 static int find_sip_method(const char *msg)
1641 if (ast_strlen_zero(msg))
1643 for (i = 1; i < (sizeof(sip_methods) / sizeof(sip_methods[0])) && !res; i++) {
1644 if (method_match(i, msg))
1645 res = sip_methods[i].id;
1650 /*! \brief Parse supported header in incoming packet */
1651 static unsigned int parse_sip_options(struct sip_pvt *pvt, const char *supported)
1655 unsigned int profile = 0;
1658 if (ast_strlen_zero(supported) )
1660 temp = ast_strdupa(supported);
1662 if (option_debug > 2 && sipdebug)
1663 ast_log(LOG_DEBUG, "Begin: parsing SIP \"Supported: %s\"\n", supported);
1665 for (next = temp; next; next = sep) {
1667 if ( (sep = strchr(next, ',')) != NULL)
1669 next = ast_skip_blanks(next);
1670 if (option_debug > 2 && sipdebug)
1671 ast_log(LOG_DEBUG, "Found SIP option: -%s-\n", next);
1672 for (i=0; i < (sizeof(sip_options) / sizeof(sip_options[0])); i++) {
1673 if (!strcasecmp(next, sip_options[i].text)) {
1674 profile |= sip_options[i].id;
1676 if (option_debug > 2 && sipdebug)
1677 ast_log(LOG_DEBUG, "Matched SIP option: %s\n", next);
1681 if (!found && option_debug > 2 && sipdebug) {
1682 if (!strncasecmp(next, "x-", 2))
1683 ast_log(LOG_DEBUG, "Found private SIP option, not supported: %s\n", next);
1685 ast_log(LOG_DEBUG, "Found no match for SIP option: %s (Please file bug report!)\n", next);
1690 pvt->sipoptions = profile;
1694 /*! \brief See if we pass debug IP filter */
1695 static inline int sip_debug_test_addr(const struct sockaddr_in *addr)
1699 if (debugaddr.sin_addr.s_addr) {
1700 if (((ntohs(debugaddr.sin_port) != 0)
1701 && (debugaddr.sin_port != addr->sin_port))
1702 || (debugaddr.sin_addr.s_addr != addr->sin_addr.s_addr))
1708 /*! \brief The real destination address for a write */
1709 static const struct sockaddr_in *sip_real_dst(const struct sip_pvt *p)
1711 return ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_ROUTE ? &p->recv : &p->sa;
1714 /*! \brief Display SIP nat mode */
1715 static const char *sip_nat_mode(const struct sip_pvt *p)
1717 return ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_ROUTE ? "NAT" : "no NAT";
1720 /*! \brief Test PVT for debugging output */
1721 static inline int sip_debug_test_pvt(struct sip_pvt *p)
1725 return sip_debug_test_addr(sip_real_dst(p));
1728 /*! \brief Transmit SIP message */
1729 static int __sip_xmit(struct sip_pvt *p, char *data, int len)
1732 const struct sockaddr_in *dst = sip_real_dst(p);
1733 res = sendto(sipsock, data, len, 0, (const struct sockaddr *)dst, sizeof(struct sockaddr_in));
1736 ast_log(LOG_WARNING, "sip_xmit of %p (len %d) to %s:%d returned %d: %s\n", data, len, ast_inet_ntoa(dst->sin_addr), ntohs(dst->sin_port), res, strerror(errno));
1741 /*! \brief Build a Via header for a request */
1742 static void build_via(struct sip_pvt *p)
1744 /* Work around buggy UNIDEN UIP200 firmware */
1745 const char *rport = ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_RFC3581 ? ";rport" : "";
1747 /* z9hG4bK is a magic cookie. See RFC 3261 section 8.1.1.7 */
1748 ast_string_field_build(p, via, "SIP/2.0/UDP %s:%d;branch=z9hG4bK%08x%s",
1749 ast_inet_ntoa(p->ourip), ourport, p->branch, rport);
1752 /*! \brief NAT fix - decide which IP address to use for ASterisk server?
1754 * Using the localaddr structure built up with localnet statements in sip.conf
1755 * apply it to their address to see if we need to substitute our
1756 * externip or can get away with our internal bindaddr
1758 static enum sip_result ast_sip_ouraddrfor(struct in_addr *them, struct in_addr *us)
1760 struct sockaddr_in theirs, ours;
1762 /* Get our local information */
1763 ast_ouraddrfor(them, us);
1764 theirs.sin_addr = *them;
1765 ours.sin_addr = *us;
1767 if (localaddr && externip.sin_addr.s_addr &&
1768 ast_apply_ha(localaddr, &theirs) &&
1769 !ast_apply_ha(localaddr, &ours)) {
1770 if (externexpire && time(NULL) >= externexpire) {
1771 struct ast_hostent ahp;
1774 externexpire = time(NULL) + externrefresh;
1775 if ((hp = ast_gethostbyname(externhost, &ahp))) {
1776 memcpy(&externip.sin_addr, hp->h_addr, sizeof(externip.sin_addr));
1778 ast_log(LOG_NOTICE, "Warning: Re-lookup of '%s' failed!\n", externhost);
1780 *us = externip.sin_addr;
1782 ast_log(LOG_DEBUG, "Target address %s is not local, substituting externip\n",
1783 ast_inet_ntoa(*(struct in_addr *)&them->s_addr));
1785 } else if (bindaddr.sin_addr.s_addr)
1786 *us = bindaddr.sin_addr;
1790 /*! \brief Append to SIP dialog history
1791 \return Always returns 0 */
1792 #define append_history(p, event, fmt , args... ) append_history_full(p, "%-15s " fmt, event, ## args)
1794 static void append_history_full(struct sip_pvt *p, const char *fmt, ...)
1795 __attribute__ ((format (printf, 2, 3)));
1797 /*! \brief Append to SIP dialog history with arg list */
1798 static void append_history_va(struct sip_pvt *p, const char *fmt, va_list ap)
1800 char buf[80], *c = buf; /* max history length */
1801 struct sip_history *hist;
1804 vsnprintf(buf, sizeof(buf), fmt, ap);
1805 strsep(&c, "\r\n"); /* Trim up everything after \r or \n */
1806 l = strlen(buf) + 1;
1807 if (!(hist = ast_calloc(1, sizeof(*hist) + l)))
1809 if (!p->history && !(p->history = ast_calloc(1, sizeof(*p->history)))) {
1813 memcpy(hist->event, buf, l);
1814 AST_LIST_INSERT_TAIL(p->history, hist, list);
1817 /*! \brief Append to SIP dialog history with arg list */
1818 static void append_history_full(struct sip_pvt *p, const char *fmt, ...)
1825 append_history_va(p, fmt, ap);
1831 /*! \brief Retransmit SIP message if no answer (Called from scheduler) */
1832 static int retrans_pkt(void *data)
1834 struct sip_pkt *pkt = data, *prev, *cur = NULL;
1835 int reschedule = DEFAULT_RETRANS;
1837 /* Lock channel PVT */
1838 sip_pvt_lock(pkt->owner);
1840 if (pkt->retrans < MAX_RETRANS) {
1842 if (!pkt->timer_t1) { /* Re-schedule using timer_a and timer_t1 */
1843 if (sipdebug && option_debug > 3)
1844 ast_log(LOG_DEBUG, "SIP TIMER: Not rescheduling id #%d:%s (Method %d) (No timer T1)\n", pkt->retransid, sip_methods[pkt->method].text, pkt->method);
1848 if (sipdebug && option_debug > 3)
1849 ast_log(LOG_DEBUG, "SIP TIMER: Rescheduling retransmission #%d (%d) %s - %d\n", pkt->retransid, pkt->retrans, sip_methods[pkt->method].text, pkt->method);
1853 pkt->timer_a = 2 * pkt->timer_a;
1855 /* For non-invites, a maximum of 4 secs */
1856 siptimer_a = pkt->timer_t1 * pkt->timer_a; /* Double each time */
1857 if (pkt->method != SIP_INVITE && siptimer_a > 4000)
1860 /* Reschedule re-transmit */
1861 reschedule = siptimer_a;
1862 if (option_debug > 3)
1863 ast_log(LOG_DEBUG, "** SIP timers: Rescheduling retransmission %d to %d ms (t1 %d ms (Retrans id #%d)) \n", pkt->retrans +1, siptimer_a, pkt->timer_t1, pkt->retransid);
1866 if (sip_debug_test_pvt(pkt->owner)) {
1867 const struct sockaddr_in *dst = sip_real_dst(pkt->owner);
1868 ast_verbose("Retransmitting #%d (%s) to %s:%d:\n%s\n---\n",
1869 pkt->retrans, sip_nat_mode(pkt->owner),
1870 ast_inet_ntoa(dst->sin_addr),
1871 ntohs(dst->sin_port), pkt->data);
1874 append_history(pkt->owner, "ReTx", "%d %s", reschedule, pkt->data);
1875 __sip_xmit(pkt->owner, pkt->data, pkt->packetlen);
1876 sip_pvt_unlock(pkt->owner);
1879 /* Too many retries */
1880 if (pkt->owner && pkt->method != SIP_OPTIONS) {
1881 if (ast_test_flag(pkt, FLAG_FATAL) || sipdebug) /* Tell us if it's critical or if we're debugging */
1882 ast_log(LOG_WARNING, "Maximum retries exceeded on transmission %s for seqno %d (%s %s)\n", pkt->owner->callid, pkt->seqno, (ast_test_flag(pkt, FLAG_FATAL)) ? "Critical" : "Non-critical", (ast_test_flag(pkt, FLAG_RESPONSE)) ? "Response" : "Request");
1884 if ((pkt->method == SIP_OPTIONS) && sipdebug)
1885 ast_log(LOG_WARNING, "Cancelling retransmit of OPTIONs (call id %s) \n", pkt->owner->callid);
1887 append_history(pkt->owner, "MaxRetries", "%s", (ast_test_flag(pkt, FLAG_FATAL)) ? "(Critical)" : "(Non-critical)");
1889 pkt->retransid = -1;
1891 if (ast_test_flag(pkt, FLAG_FATAL)) {
1892 while(pkt->owner->owner && ast_channel_trylock(pkt->owner->owner)) {
1893 sip_pvt_unlock(pkt->owner); /* SIP_PVT, not channel */
1895 sip_pvt_lock(pkt->owner);
1897 if (pkt->owner->owner) {
1898 ast_set_flag(&pkt->owner->flags[0], SIP_ALREADYGONE);
1899 ast_log(LOG_WARNING, "Hanging up call %s - no reply to our critical packet.\n", pkt->owner->callid);
1900 ast_queue_hangup(pkt->owner->owner);
1901 ast_channel_unlock(pkt->owner->owner);
1903 /* If no channel owner, destroy now */
1904 ast_set_flag(&pkt->owner->flags[0], SIP_NEEDDESTROY);
1907 /* In any case, go ahead and remove the packet */
1908 for (prev = NULL, cur = pkt->owner->packets; cur; prev = cur, cur = cur->next) {
1914 prev->next = cur->next;
1916 pkt->owner->packets = cur->next;
1917 sip_pvt_unlock(pkt->owner);
1921 ast_log(LOG_WARNING, "Weird, couldn't find packet owner!\n");
1923 sip_pvt_unlock(pkt->owner);
1927 /*! \brief Transmit packet with retransmits
1928 \return 0 on success, -1 on failure to allocate packet
1930 static enum sip_result __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, char *data, int len, int fatal, int sipmethod)
1932 struct sip_pkt *pkt;
1933 int siptimer_a = DEFAULT_RETRANS;
1935 if (!(pkt = ast_calloc(1, sizeof(*pkt) + len + 1)))
1937 memcpy(pkt->data, data, len);
1938 pkt->method = sipmethod;
1939 pkt->packetlen = len;
1940 pkt->next = p->packets;
1944 pkt->data[len] = '\0';
1945 pkt->timer_t1 = p->timer_t1; /* Set SIP timer T1 */
1947 ast_set_flag(pkt, FLAG_FATAL);
1949 siptimer_a = pkt->timer_t1 * 2;
1951 /* Schedule retransmission */
1952 pkt->retransid = ast_sched_add_variable(sched, siptimer_a, retrans_pkt, pkt, 1);
1953 if (option_debug > 3 && sipdebug)
1954 ast_log(LOG_DEBUG, "*** SIP TIMER: Initalizing retransmit timer on packet: Id #%d\n", pkt->retransid);
1955 pkt->next = p->packets;
1958 __sip_xmit(pkt->owner, pkt->data, pkt->packetlen); /* Send packet */
1959 if (sipmethod == SIP_INVITE) {
1960 /* Note this is a pending invite */
1961 p->pendinginvite = seqno;
1966 /*! \brief Kill a SIP dialog (called by scheduler) */
1967 static int __sip_autodestruct(void *data)
1969 struct sip_pvt *p = data;
1971 /* If this is a subscription, tell the phone that we got a timeout */
1972 if (p->subscribed) {
1973 transmit_state_notify(p, AST_EXTENSION_DEACTIVATED, 1, TRUE); /* Send last notification */
1974 p->subscribed = NONE;
1975 append_history(p, "Subscribestatus", "timeout");
1976 if (option_debug > 2)
1977 ast_log(LOG_DEBUG, "Re-scheduled destruction of SIP subsription %s\n", p->callid ? p->callid : "<unknown>");
1978 return 10000; /* Reschedule this destruction so that we know that it's gone */
1981 if (p->subscribed == MWI_NOTIFICATION)
1983 ASTOBJ_UNREF(p->relatedpeer,sip_destroy_peer); /* Remove link to peer. If it's realtime, make sure it's gone from memory) */
1985 /* Reset schedule ID */
1989 ast_log(LOG_DEBUG, "Auto destroying SIP dialog '%s'\n", p->callid);
1990 append_history(p, "AutoDestroy", "%s", p->callid);
1992 ast_log(LOG_WARNING, "Autodestruct on dialog '%s' with owner in place (Method: %s)\n", p->callid, sip_methods[p->method].text);
1993 ast_queue_hangup(p->owner);
1994 } else if (p->refer)
1995 transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
2001 /*! \brief Schedule destruction of SIP dialog */
2002 static void sip_scheddestroy(struct sip_pvt *p, int ms)
2005 if (p->timer_t1 == 0)
2006 p->timer_t1 = 500; /* Set timer T1 if not set (RFC 3261) */
2007 ms = p->timer_t1 * 64;
2009 if (sip_debug_test_pvt(p))
2010 ast_verbose("Scheduling destruction of SIP dialog '%s' in %d ms (Method: %s)\n", p->callid, ms, sip_methods[p->method].text);
2011 if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
2012 append_history(p, "SchedDestroy", "%d ms", ms);
2014 if (p->autokillid > -1)
2015 ast_sched_del(sched, p->autokillid);
2016 p->autokillid = ast_sched_add(sched, ms, __sip_autodestruct, p);
2019 /*! \brief Cancel destruction of SIP dialog */
2020 static void sip_cancel_destroy(struct sip_pvt *p)
2022 if (p->autokillid > -1) {
2023 ast_sched_del(sched, p->autokillid);
2024 append_history(p, "CancelDestroy", "");
2029 /*! \brief Acknowledges receipt of a packet and stops retransmission */
2030 static void __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod, int reset)
2032 struct sip_pkt *cur, *prev = NULL;
2034 /* Just in case... */
2038 msg = sip_methods[sipmethod].text;
2041 for (cur = p->packets; cur; prev = cur, cur = cur->next) {
2042 if ((cur->seqno == seqno) && ((ast_test_flag(cur, FLAG_RESPONSE)) == resp) &&
2043 ((ast_test_flag(cur, FLAG_RESPONSE)) ||
2044 (!strncasecmp(msg, cur->data, strlen(msg)) && (cur->data[strlen(msg)] < 33)))) {
2045 if (!resp && (seqno == p->pendinginvite)) {
2047 ast_log(LOG_DEBUG, "Acked pending invite %d\n", p->pendinginvite);
2048 p->pendinginvite = 0;
2050 /* this is our baby */
2052 UNLINK(cur, p->packets, prev);
2053 if (cur->retransid > -1) {
2054 if (sipdebug && option_debug > 3)
2055 ast_log(LOG_DEBUG, "** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #%d\n", cur->retransid);
2056 ast_sched_del(sched, cur->retransid);
2065 ast_log(LOG_DEBUG, "Stopping retransmission on '%s' of %s %d: Match %s\n", p->callid, resp ? "Response" : "Request", seqno, res ? "Not Found" : "Found");
2068 /*! \brief Pretend to ack all packets
2069 * maybe the lock on p is not strictly necessary but there might be a race */
2070 static void __sip_pretend_ack(struct sip_pvt *p)
2072 struct sip_pkt *cur = NULL;
2074 while (p->packets) {
2076 if (cur == p->packets) {
2077 ast_log(LOG_WARNING, "Have a packet that doesn't want to give up! %s\n", sip_methods[cur->method].text);
2081 method = (cur->method) ? cur->method : find_sip_method(cur->data);
2082 __sip_ack(p, cur->seqno, ast_test_flag(cur, FLAG_RESPONSE), method, FALSE);
2086 /*! \brief Acks receipt of packet, keep it around (used for provisional responses) */
2087 static int __sip_semi_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod)
2089 struct sip_pkt *cur;
2092 for (cur = p->packets; cur; cur = cur->next) {
2093 if (cur->seqno == seqno && ast_test_flag(cur, FLAG_RESPONSE) == resp &&
2094 (ast_test_flag(cur, FLAG_RESPONSE) || method_match(sipmethod, cur->data))) {
2095 /* this is our baby */
2096 if (cur->retransid > -1) {
2097 if (option_debug > 3 && sipdebug)
2098 ast_log(LOG_DEBUG, "*** SIP TIMER: Cancelling retransmission #%d - %s (got response)\n", cur->retransid, sip_methods[sipmethod].text);
2099 ast_sched_del(sched, cur->retransid);
2101 cur->retransid = -1;
2107 ast_log(LOG_DEBUG, "(Provisional) Stopping retransmission (but retaining packet) on '%s' %s %d: %s\n", p->callid, resp ? "Response" : "Request", seqno, res ? "Not Found" : "Found");
2112 /*! \brief Copy SIP request, parse it */
2113 static void parse_copy(struct sip_request *dst, const struct sip_request *src)
2115 memset(dst, 0, sizeof(*dst));
2116 memcpy(dst->data, src->data, sizeof(dst->data));
2117 dst->len = src->len;
2121 /*! \brief add a blank line if no body */
2122 static void add_blank(struct sip_request *req)
2125 /* Add extra empty return. add_header() reserves 4 bytes so cannot be truncated */
2126 snprintf(req->data + req->len, sizeof(req->data) - req->len, "\r\n");
2127 req->len += strlen(req->data + req->len);
2131 /*! \brief Transmit response on SIP request*/
2132 static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno)
2137 if (sip_debug_test_pvt(p)) {
2138 const struct sockaddr_in *dst = sip_real_dst(p);
2140 ast_verbose("\n<--- %sTransmitting (%s) to %s:%d --->\n%s\n<------------>\n",
2141 reliable ? "Reliably " : "", sip_nat_mode(p),
2142 ast_inet_ntoa(dst->sin_addr),
2143 ntohs(dst->sin_port), req->data);
2145 if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) {
2146 struct sip_request tmp;
2147 parse_copy(&tmp, req);
2148 append_history(p, reliable ? "TxRespRel" : "TxResp", "%s / %s - %s", tmp.data, get_header(&tmp, "CSeq"),
2149 (tmp.method == SIP_RESPONSE || tmp.method == SIP_UNKNOWN) ? tmp.rlPart2 : sip_methods[tmp.method].text);
2152 __sip_reliable_xmit(p, seqno, 1, req->data, req->len, (reliable == XMIT_CRITICAL), req->method) :
2153 __sip_xmit(p, req->data, req->len);
2159 /*! \brief Send SIP Request to the other part of the dialogue */
2160 static int send_request(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno)
2165 if (sip_debug_test_pvt(p)) {
2166 if (ast_test_flag(&p->flags[0], SIP_NAT_ROUTE))
2167 ast_verbose("%sTransmitting (NAT) to %s:%d:\n%s\n---\n", reliable ? "Reliably " : "", ast_inet_ntoa(p->recv.sin_addr), ntohs(p->recv.sin_port), req->data);
2169 ast_verbose("%sTransmitting (no NAT) to %s:%d:\n%s\n---\n", reliable ? "Reliably " : "", ast_inet_ntoa(p->sa.sin_addr), ntohs(p->sa.sin_port), req->data);
2171 if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) {
2172 struct sip_request tmp;
2173 parse_copy(&tmp, req);
2174 append_history(p, reliable ? "TxReqRel" : "TxReq", "%s / %s - %s", tmp.data, get_header(&tmp, "CSeq"), sip_methods[tmp.method].text);
2177 __sip_reliable_xmit(p, seqno, 0, req->data, req->len, (reliable > 1), req->method) :
2178 __sip_xmit(p, req->data, req->len);
2182 /*! \brief Locate closing quote in a string, skipping escaped quotes.
2183 * optionally with a limit on the search.
2184 * start must be past the first quote.
2186 static const char *find_closing_quote(const char *start, const char *lim)
2188 char last_char = '\0';
2190 for (s = start; *s && s != lim; last_char = *s++) {
2191 if (*s == '"' && last_char != '\\')
2197 /*! \brief Pick out text in brackets from character string
2198 \return pointer to terminated stripped string
2199 \param tmp input string that will be modified
2202 "foo" <bar> valid input, returns bar
2203 foo returns the whole string
2204 < "foo ... > returns the string between brackets
2205 < "foo... bogus (missing closing bracket), returns the whole string
2206 XXX maybe should still skip the opening bracket
2208 static char *get_in_brackets(char *tmp)
2210 const char *parse = tmp;
2211 char *first_bracket;
2214 * Skip any quoted text until we find the part in brackets.
2215 * On any error give up and return the full string.
2217 while ( (first_bracket = strchr(parse, '<')) ) {
2218 char *first_quote = strchr(parse, '"');
2220 if (!first_quote || first_quote > first_bracket)
2221 break; /* no need to look at quoted part */
2222 /* the bracket is within quotes, so ignore it */
2223 parse = find_closing_quote(first_quote + 1, NULL);
2224 if (!*parse) { /* not found, return full string ? */
2225 /* XXX or be robust and return in-bracket part ? */
2226 ast_log(LOG_WARNING, "No closing quote found in '%s'\n", tmp);
2231 if (first_bracket) {
2232 char *second_bracket = strchr(first_bracket + 1, '>');
2233 if (second_bracket) {
2234 *second_bracket = '\0';
2235 tmp = first_bracket + 1;
2237 ast_log(LOG_WARNING, "No closing bracket found in '%s'\n", tmp);
2244 * parses a URI in its components.
2245 * If scheme is specified, drop it from the top.
2246 * If a component is not requested, do not split around it.
2247 * This means that if we don't have domain, we cannot split
2248 * name:pass and domain:port.
2249 * It is safe to call with ret_name, pass, domain, port
2250 * pointing all to the same place.
2251 * Init pointers to empty string so we never get NULL dereferencing.
2252 * Overwrites the string.
2253 * return 0 on success, other values on error.
2255 static int parse_uri(char *uri, char *scheme,
2256 char **ret_name, char **pass, char **domain, char **port, char **options)
2261 /* init field as required */
2266 name = strsep(&uri, ";"); /* remove options */
2268 int l = strlen(scheme);
2269 if (!strncmp(name, scheme, l))
2272 ast_log(LOG_NOTICE, "Missing scheme '%s' in '%s'\n", scheme, name);
2277 /* if we don't want to split around domain, keep everything as a name,
2278 * so we need to do nothing here, except remember why.
2281 /* store the result in a temp. variable to avoid it being
2282 * overwritten if arguments point to the same place.
2286 if ((c = strchr(name, '@')) == NULL) {
2287 /* domain-only URI, according to the SIP RFC. */
2294 if (port && (c = strchr(dom, ':'))) { /* Remove :port */
2298 if (pass && (c = strchr(name, ':'))) { /* user:password */
2304 if (ret_name) /* same as for domain, store the result only at the end */
2307 *options = uri ? uri : "";
2312 /*! \brief Send SIP MESSAGE text within a call
2313 Called from PBX core sendtext() application */
2314 static int sip_sendtext(struct ast_channel *ast, const char *text)
2316 struct sip_pvt *p = ast->tech_pvt;
2317 int debug = sip_debug_test_pvt(p);
2320 ast_verbose("Sending text %s on %s\n", text, ast->name);
2323 if (ast_strlen_zero(text))
2326 ast_verbose("Really sending text %s on %s\n", text, ast->name);
2327 transmit_message_with_text(p, text);
2331 /*! \brief Update peer object in realtime storage
2332 If the Asterisk system name is set in asterisk.conf, we will use
2333 that name and store that in the "regserver" field in the sippeers
2334 table to facilitate multi-server setups.
2336 static void realtime_update_peer(const char *peername, struct sockaddr_in *sin, const char *username, const char *fullcontact, int expirey)
2339 char ipaddr[INET_ADDRSTRLEN];
2340 char regseconds[20];
2342 char *sysname = ast_config_AST_SYSTEM_NAME;
2343 char *syslabel = NULL;
2345 time_t nowtime = time(NULL) + expirey;
2346 const char *fc = fullcontact ? "fullcontact" : NULL;
2348 snprintf(regseconds, sizeof(regseconds), "%d", (int)nowtime); /* Expiration time */
2349 ast_copy_string(ipaddr, ast_inet_ntoa(sin->sin_addr), sizeof(ipaddr));
2350 snprintf(port, sizeof(port), "%d", ntohs(sin->sin_port));
2352 if (ast_strlen_zero(sysname)) /* No system name, disable this */
2354 else if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTSAVE_SYSNAME))
2355 syslabel = "regserver";
2358 ast_update_realtime("sippeers", "name", peername, "ipaddr", ipaddr,
2359 "port", port, "regseconds", regseconds,
2360 "username", username, fc, fullcontact, syslabel, sysname, NULL); /* note fc and syslabel _can_ be NULL */
2362 ast_update_realtime("sippeers", "name", peername, "ipaddr", ipaddr,
2363 "port", port, "regseconds", regseconds,
2364 "username", username, syslabel, sysname, NULL); /* note syslabel _can_ be NULL */
2367 /*! \brief Automatically add peer extension to dial plan */
2368 static void register_peer_exten(struct sip_peer *peer, int onoff)
2371 char *stringp, *ext, *context;
2373 /* XXX note that global_regcontext is both a global 'enable' flag and
2374 * the name of the global regexten context, if not specified
2377 if (ast_strlen_zero(global_regcontext))
2380 ast_copy_string(multi, S_OR(peer->regexten, peer->name), sizeof(multi));
2382 while ((ext = strsep(&stringp, "&"))) {
2383 if ((context = strchr(ext, '@'))) {
2384 *context++ = '\0'; /* split ext@context */
2385 if (!ast_context_find(context)) {
2386 ast_log(LOG_WARNING, "Context %s must exist in regcontext= in sip.conf!\n", context);
2390 context = global_regcontext;
2393 ast_add_extension(context, 1, ext, 1, NULL, NULL, "Noop",
2394 ast_strdup(peer->name), ast_free, "SIP");
2396 ast_context_remove_extension(context, ext, 1, NULL);
2400 /*! \brief Destroy peer object from memory */
2401 static void sip_destroy_peer(struct sip_peer *peer)
2403 if (option_debug > 2)
2404 ast_log(LOG_DEBUG, "Destroying SIP peer %s\n", peer->name);
2406 /* Delete it, it needs to disappear */
2408 sip_destroy(peer->call);
2410 if (peer->mwipvt) /* We have an active subscription, delete it */
2411 sip_destroy(peer->mwipvt);
2413 if (peer->chanvars) {
2414 ast_variables_destroy(peer->chanvars);
2415 peer->chanvars = NULL;
2417 if (peer->expire > -1)
2418 ast_sched_del(sched, peer->expire);
2419 if (peer->pokeexpire > -1)
2420 ast_sched_del(sched, peer->pokeexpire);
2421 register_peer_exten(peer, FALSE);
2422 ast_free_ha(peer->ha);
2423 if (ast_test_flag(&peer->flags[1], SIP_PAGE2_SELFDESTRUCT))
2425 else if (ast_test_flag(&peer->flags[0], SIP_REALTIME))
2429 clear_realm_authentication(peer->auth);
2432 ast_dnsmgr_release(peer->dnsmgr);
2436 /*! \brief Update peer data in database (if used) */
2437 static void update_peer(struct sip_peer *p, int expiry)
2439 int rtcachefriends = ast_test_flag(&p->flags[1], SIP_PAGE2_RTCACHEFRIENDS);
2440 if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTUPDATE) &&
2441 (ast_test_flag(&p->flags[0], SIP_REALTIME) || rtcachefriends)) {
2442 realtime_update_peer(p->name, &p->addr, p->username, rtcachefriends ? p->fullcontact : NULL, expiry);
2447 /*! \brief realtime_peer: Get peer from realtime storage
2448 * Checks the "sippeers" realtime family from extconfig.conf
2449 * \todo Consider adding check of port address when matching here to follow the same
2450 * algorithm as for static peers. Will we break anything by adding that?
2452 static struct sip_peer *realtime_peer(const char *newpeername, struct sockaddr_in *sin)
2454 struct sip_peer *peer;
2455 struct ast_variable *var = NULL;
2456 struct ast_variable *tmp;
2457 char ipaddr[INET_ADDRSTRLEN];
2459 /* First check on peer name */
2461 var = ast_load_realtime("sippeers", "name", newpeername, NULL);
2462 else if (sin) { /* Then check on IP address for dynamic peers */
2463 ast_copy_string(ipaddr, ast_inet_ntoa(sin->sin_addr), sizeof(ipaddr));
2464 var = ast_load_realtime("sippeers", "host", ipaddr, NULL); /* First check for fixed IP hosts */
2466 var = ast_load_realtime("sippeers", "ipaddr", ipaddr, NULL); /* Then check for registred hosts */
2472 for (tmp = var; tmp; tmp = tmp->next) {
2473 /* If this is type=user, then skip this object. */
2474 if (!strcasecmp(tmp->name, "type") &&
2475 !strcasecmp(tmp->value, "user")) {
2476 ast_variables_destroy(var);
2478 } else if (!newpeername && !strcasecmp(tmp->name, "name")) {
2479 newpeername = tmp->value;
2483 if (!newpeername) { /* Did not find peer in realtime */
2484 ast_log(LOG_WARNING, "Cannot Determine peer name ip=%s\n", ipaddr);
2485 ast_variables_destroy(var);
2489 /* Peer found in realtime, now build it in memory */
2490 peer = build_peer(newpeername, var, NULL, !ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS));
2492 ast_variables_destroy(var);
2496 if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
2498 ast_copy_flags(&peer->flags[1],&global_flags[1], SIP_PAGE2_RTAUTOCLEAR|SIP_PAGE2_RTCACHEFRIENDS);
2499 if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTAUTOCLEAR)) {
2500 if (peer->expire > -1) {
2501 ast_sched_del(sched, peer->expire);
2503 peer->expire = ast_sched_add(sched, (global_rtautoclear) * 1000, expire_register, (void *)peer);
2505 ASTOBJ_CONTAINER_LINK(&peerl,peer);
2507 ast_set_flag(&peer->flags[0], SIP_REALTIME);
2509 ast_variables_destroy(var);
2514 /*! \brief Support routine for find_peer */
2515 static int sip_addrcmp(char *name, struct sockaddr_in *sin)
2517 /* We know name is the first field, so we can cast */
2518 struct sip_peer *p = (struct sip_peer *) name;
2519 return !(!inaddrcmp(&p->addr, sin) ||
2520 (ast_test_flag(&p->flags[0], SIP_INSECURE_PORT) &&
2521 (p->addr.sin_addr.s_addr == sin->sin_addr.s_addr)));
2524 /*! \brief Locate peer by name or ip address
2525 * This is used on incoming SIP message to find matching peer on ip
2526 or outgoing message to find matching peer on name */
2527 static struct sip_peer *find_peer(const char *peer, struct sockaddr_in *sin, int realtime)
2529 struct sip_peer *p = NULL;
2532 p = ASTOBJ_CONTAINER_FIND(&peerl, peer);
2534 p = ASTOBJ_CONTAINER_FIND_FULL(&peerl, sin, name, sip_addr_hashfunc, 1, sip_addrcmp);
2537 p = realtime_peer(peer, sin);
2542 /*! \brief Remove user object from in-memory storage */
2543 static void sip_destroy_user(struct sip_user *user)
2545 if (option_debug > 2)
2546 ast_log(LOG_DEBUG, "Destroying user object from memory: %s\n", user->name);
2547 ast_free_ha(user->ha);
2548 if (user->chanvars) {
2549 ast_variables_destroy(user->chanvars);
2550 user->chanvars = NULL;
2552 if (ast_test_flag(&user->flags[0], SIP_REALTIME))
2559 /*! \brief Load user from realtime storage
2560 * Loads user from "sipusers" category in realtime (extconfig.conf)
2561 * Users are matched on From: user name (the domain in skipped) */
2562 static struct sip_user *realtime_user(const char *username)
2564 struct ast_variable *var;
2565 struct ast_variable *tmp;
2566 struct sip_user *user = NULL;
2568 var = ast_load_realtime("sipusers", "name", username, NULL);
2573 for (tmp = var; tmp; tmp = tmp->next) {
2574 if (!strcasecmp(tmp->name, "type") &&
2575 !strcasecmp(tmp->value, "peer")) {
2576 ast_variables_destroy(var);
2581 user = build_user(username, var, !ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS));
2583 if (!user) { /* No user found */
2584 ast_variables_destroy(var);
2588 if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
2589 ast_set_flag(&user->flags[1], SIP_PAGE2_RTCACHEFRIENDS);
2591 ASTOBJ_CONTAINER_LINK(&userl,user);
2593 /* Move counter from s to r... */
2596 ast_set_flag(&user->flags[0], SIP_REALTIME);
2598 ast_variables_destroy(var);
2602 /*! \brief Locate user by name
2603 * Locates user by name (From: sip uri user name part) first
2604 * from in-memory list (static configuration) then from
2605 * realtime storage (defined in extconfig.conf) */
2606 static struct sip_user *find_user(const char *name, int realtime)
2608 struct sip_user *u = ASTOBJ_CONTAINER_FIND(&userl, name);
2610 u = realtime_user(name);
2614 /*! \brief Set nat mode on the various data sockets */
2615 static void do_setnat(struct sip_pvt *p, int natflags)
2617 const char *mode = natflags ? "On" : "Off";
2621 ast_log(LOG_DEBUG, "Setting NAT on RTP to %s\n", mode);
2622 ast_rtp_setnat(p->rtp, natflags);
2626 ast_log(LOG_DEBUG, "Setting NAT on VRTP to %s\n", mode);
2627 ast_rtp_setnat(p->vrtp, natflags);
2631 ast_log(LOG_DEBUG, "Setting NAT on UDPTL to %s\n", mode);
2632 ast_udptl_setnat(p->udptl, natflags);
2636 /*! \brief Create address structure from peer reference.
2637 * return -1 on error, 0 on success.
2639 static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
2641 if ((peer->addr.sin_addr.s_addr || peer->defaddr.sin_addr.s_addr) &&
2642 (!peer->maxms || ((peer->lastms >= 0) && (peer->lastms <= peer->maxms)))) {
2643 dialog->sa = (peer->addr.sin_addr.s_addr) ? peer->addr : peer->defaddr;
2644 dialog->recv = dialog->sa;
2648 ast_copy_flags(&dialog->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY);
2649 ast_copy_flags(&dialog->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
2650 dialog->capability = peer->capability;
2651 if (!ast_test_flag(&dialog->flags[1], SIP_PAGE2_VIDEOSUPPORT) && dialog->vrtp) {
2652 ast_rtp_destroy(dialog->vrtp);
2653 dialog->vrtp = NULL;
2655 dialog->prefs = peer->prefs;
2656 if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_T38SUPPORT)) {
2657 dialog->t38.capability = global_t38_capability;
2658 if (dialog->udptl) {
2659 if (ast_udptl_get_error_correction_scheme(dialog->udptl) == UDPTL_ERROR_CORRECTION_FEC )
2660 dialog->t38.capability |= T38FAX_UDP_EC_FEC;
2661 else if (ast_udptl_get_error_correction_scheme(dialog->udptl) == UDPTL_ERROR_CORRECTION_REDUNDANCY )
2662 dialog->t38.capability |= T38FAX_UDP_EC_REDUNDANCY;
2663 else if (ast_udptl_get_error_correction_scheme(dialog->udptl) == UDPTL_ERROR_CORRECTION_NONE )
2664 dialog->t38.capability |= T38FAX_UDP_EC_NONE;
2665 dialog->t38.capability |= T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF;
2666 if (option_debug > 1)
2667 ast_log(LOG_DEBUG,"Our T38 capability (%d)\n", dialog->t38.capability);
2669 dialog->t38.jointcapability = dialog->t38.capability;
2670 } else if (dialog->udptl) {
2671 ast_udptl_destroy(dialog->udptl);
2672 dialog->udptl = NULL;
2674 do_setnat(dialog, ast_test_flag(&dialog->flags[0], SIP_NAT) & SIP_NAT_ROUTE );
2677 ast_rtp_setdtmf(dialog->rtp, ast_test_flag(&dialog->flags[0], SIP_DTMF) != SIP_DTMF_INFO);
2678 ast_rtp_setdtmfcompensate(dialog->rtp, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
2681 ast_rtp_setdtmf(dialog->vrtp, 0);
2682 ast_rtp_setdtmfcompensate(dialog->vrtp, 0);
2685 /* Set Frame packetization */
2687 ast_rtp_codec_setpref(dialog->rtp, &dialog->prefs);
2688 dialog->autoframing = peer->autoframing;
2690 ast_string_field_set(dialog, peername, peer->username);
2691 ast_string_field_set(dialog, authname, peer->username);
2692 ast_string_field_set(dialog, username, peer->username);
2693 ast_string_field_set(dialog, peersecret, peer->secret);
2694 ast_string_field_set(dialog, peermd5secret, peer->md5secret);
2695 ast_string_field_set(dialog, tohost, peer->tohost);
2696 ast_string_field_set(dialog, fullcontact, peer->fullcontact);
2697 if (!dialog->initreq.headers && !ast_strlen_zero(peer->fromdomain)) {
2700 tmpcall = ast_strdupa(dialog->callid);
2701 c = strchr(tmpcall, '@');
2704 ast_string_field_build(dialog, callid, "%s@%s", tmpcall, peer->fromdomain);
2707 if (ast_strlen_zero(dialog->tohost))
2708 ast_string_field_set(dialog, tohost, ast_inet_ntoa(dialog->sa.sin_addr));
2709 if (!ast_strlen_zero(peer->fromdomain))
2710 ast_string_field_set(dialog, fromdomain, peer->fromdomain);
2711 if (!ast_strlen_zero(peer->fromuser))
2712 ast_string_field_set(dialog, fromuser, peer->fromuser);
2713 dialog->callgroup = peer->callgroup;
2714 dialog->pickupgroup = peer->pickupgroup;
2715 dialog->allowtransfer = peer->allowtransfer;
2716 /* Set timer T1 to RTT for this peer (if known by qualify=) */
2717 /* Minimum is settable or default to 100 ms */
2718 if (peer->maxms && peer->lastms)
2719 dialog->timer_t1 = peer->lastms < global_t1min ? global_t1min : peer->lastms;
2720 if ((ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
2721 (ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
2722 dialog->noncodeccapability |= AST_RTP_DTMF;
2724 dialog->noncodeccapability &= ~AST_RTP_DTMF;
2725 ast_string_field_set(dialog, context, peer->context);
2726 dialog->rtptimeout = peer->rtptimeout;
2727 dialog->rtpholdtimeout = peer->rtpholdtimeout;
2728 dialog->rtpkeepalive = peer->rtpkeepalive;
2729 if (peer->call_limit)
2730 ast_set_flag(&dialog->flags[0], SIP_CALL_LIMIT);
2731 dialog->maxcallbitrate = peer->maxcallbitrate;
2736 /*! \brief create address structure from peer name
2737 * Or, if peer not found, find it in the global DNS
2738 * returns TRUE (-1) on failure, FALSE on success */
2739 static int create_addr(struct sip_pvt *dialog, const char *opeer)
2742 struct ast_hostent ahp;
2746 char host[MAXHOSTNAMELEN], *hostn;
2749 ast_copy_string(peer, opeer, sizeof(peer));
2750 port = strchr(peer, ':');
2753 dialog->sa.sin_family = AF_INET;
2754 dialog->timer_t1 = 500; /* Default SIP retransmission timer T1 (RFC 3261) */
2755 p = find_peer(peer, NULL, 1);
2758 int res = create_addr_from_peer(dialog, p);
2759 ASTOBJ_UNREF(p, sip_destroy_peer);
2763 portno = port ? atoi(port) : STANDARD_SIP_PORT;
2765 char service[MAXHOSTNAMELEN];
2769 snprintf(service, sizeof(service), "_sip._udp.%s", peer);
2770 ret = ast_get_srv(NULL, host, sizeof(host), &tportno, service);
2776 hp = ast_gethostbyname(hostn, &ahp);
2778 ast_log(LOG_WARNING, "No such host: %s\n", peer);
2781 ast_string_field_set(dialog, tohost, peer);
2782 memcpy(&dialog->sa.sin_addr, hp->h_addr, sizeof(dialog->sa.sin_addr));
2783 dialog->sa.sin_port = htons(portno);
2784 dialog->recv = dialog->sa;
2788 /*! \brief Scheduled congestion on a call */
2789 static int auto_congest(void *nothing)
2791 struct sip_pvt *p = nothing;
2796 /* XXX fails on possible deadlock */
2797 if (!ast_channel_trylock(p->owner)) {
2798 ast_log(LOG_NOTICE, "Auto-congesting %s\n", p->owner->name);
2799 append_history(p, "Cong", "Auto-congesting (timer)");
2800 ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
2801 ast_channel_unlock(p->owner);
2809 /*! \brief Initiate SIP call from PBX
2810 * used from the dial() application */
2811 static int sip_call(struct ast_channel *ast, char *dest, int timeout)
2815 struct varshead *headp;
2816 struct ast_var_t *current;
2817 const char *referer = NULL; /* SIP refererer */
2820 if ((ast->_state != AST_STATE_DOWN) && (ast->_state != AST_STATE_RESERVED)) {
2821 ast_log(LOG_WARNING, "sip_call called on %s, neither down nor reserved\n", ast->name);
2825 /* Check whether there is vxml_url, distinctive ring variables */
2826 headp=&ast->varshead;
2827 AST_LIST_TRAVERSE(headp,current,entries) {
2828 /* Check whether there is a VXML_URL variable */
2829 if (!p->options->vxml_url && !strcasecmp(ast_var_name(current), "VXML_URL")) {
2830 p->options->vxml_url = ast_var_value(current);
2831 } else if (!p->options->uri_options && !strcasecmp(ast_var_name(current), "SIP_URI_OPTIONS")) {
2832 p->options->uri_options = ast_var_value(current);
2833 } else if (!p->options->addsipheaders && !strncasecmp(ast_var_name(current), "SIPADDHEADER", strlen("SIPADDHEADER"))) {
2834 /* Check whether there is a variable with a name starting with SIPADDHEADER */
2835 p->options->addsipheaders = 1;
2836 } else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER")) {
2837 /* This is a transfered call */
2838 p->options->transfer = 1;
2839 } else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER_REFERER")) {
2840 /* This is the referer */
2841 referer = ast_var_value(current);
2842 } else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER_REPLACES")) {
2843 /* We're replacing a call. */
2844 p->options->replaces = ast_var_value(current);
2845 } else if (!strcasecmp(ast_var_name(current), "T38CALL")) {
2846 p->t38.state = T38_LOCAL_DIRECT;
2848 ast_log(LOG_DEBUG,"T38State change to %d on channel %s\n", p->t38.state, ast->name);
2854 ast_set_flag(&p->flags[0], SIP_OUTGOING);
2856 if (p->options->transfer) {
2860 if (sipdebug && option_debug > 2)
2861 ast_log(LOG_DEBUG, "Call for %s transfered by %s\n", p->username, referer);
2862 snprintf(buf, sizeof(buf)-1, "-> %s (via %s)", p->cid_name, referer);
2864 snprintf(buf, sizeof(buf)-1, "-> %s", p->cid_name);
2865 ast_string_field_set(p, cid_name, buf);
2868 ast_log(LOG_DEBUG, "Outgoing Call for %s\n", p->username);
2870 res = update_call_counter(p, INC_CALL_RINGING);
2872 p->callingpres = ast->cid.cid_pres;
2873 p->jointcapability = p->capability;
2874 p->t38.jointcapability = p->t38.capability;
2876 ast_log(LOG_DEBUG,"Our T38 capability (%d), joint T38 capability (%d)\n", p->t38.capability, p->t38.jointcapability);
2877 transmit_invite(p, SIP_INVITE, 1, 2);
2878 p->initid = ast_sched_add(sched, SIP_TRANS_TIMEOUT, auto_congest, p);
2883 /*! \brief Destroy registry object
2884 Objects created with the register= statement in static configuration */
2885 static void sip_registry_destroy(struct sip_registry *reg)
2888 if (option_debug > 2)
2889 ast_log(LOG_DEBUG, "Destroying registry entry for %s@%s\n", reg->username, reg->hostname);
2892 /* Clear registry before destroying to ensure
2893 we don't get reentered trying to grab the registry lock */
2894 reg->call->registry = NULL;
2895 if (option_debug > 2)
2896 ast_log(LOG_DEBUG, "Destroying active SIP dialog for registry %s@%s\n", reg->username, reg->hostname);
2897 sip_destroy(reg->call);
2899 if (reg->expire > -1)
2900 ast_sched_del(sched, reg->expire);
2901 if (reg->timeout > -1)
2902 ast_sched_del(sched, reg->timeout);
2903 ast_string_field_free_pools(reg);
2909 /*! \brief Execute destruction of SIP dialog structure, release memory */
2910 static void __sip_destroy(struct sip_pvt *p, int lockowner)
2912 struct sip_pvt *cur, *prev = NULL;
2915 if (sip_debug_test_pvt(p) || option_debug > 2)
2916 ast_verbose("Really destroying SIP dialog '%s' Method: %s\n", p->callid, sip_methods[p->method].text);
2918 /* Remove link from peer to subscription of MWI */
2919 if (p->relatedpeer && p->relatedpeer->mwipvt)
2920 p->relatedpeer->mwipvt = NULL;
2923 sip_dump_history(p);
2928 if (p->stateid > -1)
2929 ast_extension_state_del(p->stateid, NULL);
2931 ast_sched_del(sched, p->initid);
2932 if (p->autokillid > -1)
2933 ast_sched_del(sched, p->autokillid);
2936 ast_rtp_destroy(p->rtp);
2938 ast_rtp_destroy(p->vrtp);
2940 ast_udptl_destroy(p->udptl);
2944 free_old_route(p->route);
2948 if (p->registry->call == p)
2949 p->registry->call = NULL;
2950 ASTOBJ_UNREF(p->registry, sip_registry_destroy);
2953 /* Unlink us from the owner if we have one */
2956 ast_channel_lock(p->owner);
2958 ast_log(LOG_DEBUG, "Detaching from %s\n", p->owner->name);
2959 p->owner->tech_pvt = NULL;
2961 ast_channel_unlock(p->owner);
2965 struct sip_history *hist;
2966 while( (hist = AST_LIST_REMOVE_HEAD(p->history, list)) )
2972 /* Lock dialog list before removing ourselves from the list */
2973 ast_mutex_lock(&dialoglock);
2974 for (prev = NULL, cur = dialoglist; cur; prev = cur, cur = cur->next) {
2976 UNLINK(cur, dialoglist, prev);
2980 ast_mutex_unlock(&dialoglock);
2982 ast_log(LOG_WARNING, "Trying to destroy \"%s\", not found in dialog list?!?! \n", p->callid);
2986 /* remove all current packets in this dialog */
2987 while((cp = p->packets)) {
2988 p->packets = p->packets->next;
2989 if (cp->retransid > -1)
2990 ast_sched_del(sched, cp->retransid);
2994 ast_variables_destroy(p->chanvars);
2997 ast_mutex_destroy(&p->pvt_lock);
2999 ast_string_field_free_pools(p);
3004 /*! \brief update_call_counter: Handle call_limit for SIP users
3005 * Setting a call-limit will cause calls above the limit not to be accepted.
3007 * Remember that for a type=friend, there's one limit for the user and
3008 * another for the peer, not a combined call limit.
3009 * This will cause unexpected behaviour in subscriptions, since a "friend"
3010 * is *two* devices in Asterisk, not one.
3012 * Thought: For realtime, we should propably update storage with inuse counter...
3014 * \return 0 if call is ok (no call limit, below treshold)
3015 * -1 on rejection of call
3018 static int update_call_counter(struct sip_pvt *fup, int event)
3021 int *inuse, *call_limit, *inringing;
3022 int outgoing = ast_test_flag(&fup->flags[0], SIP_OUTGOING);
3023 struct sip_user *u = NULL;
3024 struct sip_peer *p = NULL;
3026 if (option_debug > 2)
3027 ast_log(LOG_DEBUG, "Updating call counter for %s call\n", outgoing ? "outgoing" : "incoming");
3028 /* Test if we need to check call limits, in order to avoid
3029 realtime lookups if we do not need it */
3030 if (!ast_test_flag(&fup->flags[0], SIP_CALL_LIMIT))
3033 ast_copy_string(name, fup->username, sizeof(name));
3035 /* Check the list of users only for incoming calls */
3036 if (!outgoing && (u = find_user(name, 1)) ) {
3038 call_limit = &u->call_limit;
3040 } else if ( (p = find_peer(fup->peername, NULL, 1) ) ) { /* Try to find peer */
3042 call_limit = &p->call_limit;
3043 inringing = &p->inRinging;
3044 ast_copy_string(name, fup->peername, sizeof(name));
3046 if (option_debug > 1)
3047 ast_log(LOG_DEBUG, "%s is not a local device, no call limit\n", name);
3052 /* incoming and outgoing affects the inUse counter */
3053 case DEC_CALL_LIMIT: