2 * Asterisk -- An open source telephony toolkit.
4 * Copyright (C) 1999 - 2012, 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, and experimental TCP and TLS support
29 * Configuration file \link Config_sip sip.conf \endlink
31 * ********** IMPORTANT *
32 * \note TCP/TLS support is EXPERIMENTAL and WILL CHANGE. This applies to configuration
33 * settings, dialplan commands and dialplans apps/functions
34 * See \ref sip_tcp_tls
37 * ******** General TODO:s
38 * \todo Better support of forking
39 * \todo VIA branch tag transaction checking
40 * \todo Transaction support
42 * ******** Wishlist: Improvements
43 * - Support of SIP domains for devices, so that we match on username\@domain in the From: header
44 * - Connect registrations with a specific device on the incoming call. It's not done
45 * automatically in Asterisk
47 * \ingroup channel_drivers
49 * \par Overview of the handling of SIP sessions
50 * The SIP channel handles several types of SIP sessions, or dialogs,
51 * not all of them being "telephone calls".
52 * - Incoming calls that will be sent to the PBX core
53 * - Outgoing calls, generated by the PBX
54 * - SIP subscriptions and notifications of states and voicemail messages
55 * - SIP registrations, both inbound and outbound
56 * - SIP peer management (peerpoke, OPTIONS)
59 * In the SIP channel, there's a list of active SIP dialogs, which includes
60 * all of these when they are active. "sip show channels" in the CLI will
61 * show most of these, excluding subscriptions which are shown by
62 * "sip show subscriptions"
64 * \par incoming packets
65 * Incoming packets are received in the monitoring thread, then handled by
66 * sipsock_read() for udp only. In tcp, packets are read by the tcp_helper thread.
67 * sipsock_read() function parses the packet and matches an existing
68 * dialog or starts a new SIP dialog.
70 * sipsock_read sends the packet to handle_incoming(), that parses a bit more.
71 * If it is a response to an outbound request, the packet is sent to handle_response().
72 * If it is a request, handle_incoming() sends it to one of a list of functions
73 * depending on the request type - INVITE, OPTIONS, REFER, BYE, CANCEL etc
74 * sipsock_read locks the ast_channel if it exists (an active call) and
75 * unlocks it after we have processed the SIP message.
77 * A new INVITE is sent to handle_request_invite(), that will end up
78 * starting a new channel in the PBX, the new channel after that executing
79 * in a separate channel thread. This is an incoming "call".
80 * When the call is answered, either by a bridged channel or the PBX itself
81 * the sip_answer() function is called.
83 * The actual media - Video or Audio - is mostly handled by the RTP subsystem
87 * Outbound calls are set up by the PBX through the sip_request_call()
88 * function. After that, they are activated by sip_call().
91 * The PBX issues a hangup on both incoming and outgoing calls through
92 * the sip_hangup() function
95 /*! \li \ref chan_sip.c uses configuration files \ref sip.conf and \ref sip_notify.conf
96 * \addtogroup configuration_file
99 /*! \page sip.conf sip.conf
100 * \verbinclude sip.conf.sample
103 /*! \page sip_notify.conf sip_notify.conf
104 * \verbinclude sip_notify.conf.sample
108 * \page sip_tcp_tls SIP TCP and TLS support
110 * \par tcpfixes TCP implementation changes needed
111 * \todo Fix TCP/TLS handling in dialplan, SRV records, transfers and much more
112 * \todo Save TCP/TLS sessions in registry
113 * If someone registers a SIPS uri, this forces us to set up a TLS connection back.
114 * \todo Add TCP/TLS information to function SIPPEER and SIPCHANINFO
115 * \todo If tcpenable=yes, we must open a TCP socket on the same address as the IP for UDP.
116 * The tcpbindaddr config option should only be used to open ADDITIONAL ports
117 * So we should propably go back to
118 * bindaddr= the default address to bind to. If tcpenable=yes, then bind this to both udp and TCP
119 * if tlsenable=yes, open TLS port (provided we also have cert)
120 * tcpbindaddr = extra address for additional TCP connections
121 * tlsbindaddr = extra address for additional TCP/TLS connections
122 * udpbindaddr = extra address for additional UDP connections
123 * These three options should take multiple IP/port pairs
124 * Note: Since opening additional listen sockets is a *new* feature we do not have today
125 * the XXXbindaddr options needs to be disabled until we have support for it
127 * \todo re-evaluate the transport= setting in sip.conf. This is right now not well
128 * thought of. If a device in sip.conf contacts us via TCP, we should not switch transport,
129 * even if udp is the configured first transport.
131 * \todo Be prepared for one outbound and another incoming socket per pvt. This applies
132 * specially to communication with other peers (proxies).
133 * \todo We need to test TCP sessions with SIP proxies and in regards
134 * to the SIP outbound specs.
135 * \todo ;transport=tls was deprecated in RFC3261 and should not be used at all. See section 26.2.2.
137 * \todo If the message is smaller than the given Content-length, the request should get a 400 Bad request
138 * message. If it's a response, it should be dropped. (RFC 3261, Section 18.3)
139 * \todo Since we have had multidomain support in Asterisk for quite a while, we need to support
140 * multiple domains in our TLS implementation, meaning one socket and one cert per domain
141 * \todo Selection of transport for a request needs to be done after we've parsed all route headers,
142 * also considering outbound proxy options.
143 * First request: Outboundproxy, routes, (reg contact or URI. If URI doesn't have port: DNS naptr, srv, AAA)
144 * Intermediate requests: Outboundproxy(only when forced), routes, contact/uri
145 * DNS naptr support is crucial. A SIP uri might lead to a TLS connection.
146 * Also note that due to outbound proxy settings, a SIPS uri might have to be sent on UDP (not to recommend though)
147 * \todo Default transports are set to UDP, which cause the wrong behaviour when contacting remote
148 * devices directly from the dialplan. UDP is only a fallback if no other method works,
149 * in order to be compatible with RFC2543 (SIP/1.0) devices. For transactions that exceed the
150 * MTU (like INIVTE with video, audio and RTT) TCP should be preferred.
152 * When dialling unconfigured peers (with no port number) or devices in external domains
153 * NAPTR records MUST be consulted to find configured transport. If they are not found,
154 * SRV records for both TCP and UDP should be checked. If there's a record for TCP, use that.
155 * If there's no record for TCP, then use UDP as a last resort. If there's no SRV records,
156 * \note this only applies if there's no outbound proxy configured for the session. If an outbound
157 * proxy is configured, these procedures might apply for locating the proxy and determining
158 * the transport to use for communication with the proxy.
159 * \par Other bugs to fix ----
160 * __set_address_from_contact(const char *fullcontact, struct sockaddr_in *sin, int tcp)
161 * - sets TLS port as default for all TCP connections, unless other port is given in contact.
162 * parse_register_contact(struct sip_pvt *pvt, struct sip_peer *peer, struct sip_request *req)
163 * - assumes that the contact the UA registers is using the same transport as the REGISTER request, which is
165 * - Does not save any information about TCP/TLS connected devices, which is a severe BUG, as discussed on the mailing list.
166 * get_destination(struct sip_pvt *p, struct sip_request *oreq)
167 * - Doesn't store the information that we got an incoming SIPS request in the channel, so that
168 * we can require a secure signalling path OUT of Asterisk (on SIP or IAX2). Possibly, the call should
169 * fail on in-secure signalling paths if there's no override in our configuration. At least, provide a
170 * channel variable in the dialplan.
171 * get_refer_info(struct sip_pvt *transferer, struct sip_request *outgoing_req)
172 * - As above, if we have a SIPS: uri in the refer-to header
173 * - Does not check transport in refer_to uri.
177 <use type="module">res_crypto</use>
178 <use type="module">res_http_websocket</use>
179 <support_level>core</support_level>
182 /*! \page sip_session_timers SIP Session Timers in Asterisk Chan_sip
184 The SIP Session-Timers is an extension of the SIP protocol that allows end-points and proxies to
185 refresh a session periodically. The sessions are kept alive by sending a RE-INVITE or UPDATE
186 request at a negotiated interval. If a session refresh fails then all the entities that support Session-
187 Timers clear their internal session state. In addition, UAs generate a BYE request in order to clear
188 the state in the proxies and the remote UA (this is done for the benefit of SIP entities in the path
189 that do not support Session-Timers).
191 The Session-Timers can be configured on a system-wide, per-user, or per-peer basis. The peruser/
192 per-peer settings override the global settings. The following new parameters have been
193 added to the sip.conf file.
194 session-timers=["accept", "originate", "refuse"]
195 session-expires=[integer]
196 session-minse=[integer]
197 session-refresher=["uas", "uac"]
199 The session-timers parameter in sip.conf defines the mode of operation of SIP session-timers feature in
200 Asterisk. The Asterisk can be configured in one of the following three modes:
202 1. Accept :: In the "accept" mode, the Asterisk server honors session-timers requests
203 made by remote end-points. A remote end-point can request Asterisk to engage
204 session-timers by either sending it an INVITE request with a "Supported: timer"
205 header in it or by responding to Asterisk's INVITE with a 200 OK that contains
206 Session-Expires: header in it. In this mode, the Asterisk server does not
207 request session-timers from remote end-points. This is the default mode.
208 2. Originate :: In the "originate" mode, the Asterisk server requests the remote
209 end-points to activate session-timers in addition to honoring such requests
210 made by the remote end-pints. In order to get as much protection as possible
211 against hanging SIP channels due to network or end-point failures, Asterisk
212 resends periodic re-INVITEs even if a remote end-point does not support
213 the session-timers feature.
214 3. Refuse :: In the "refuse" mode, Asterisk acts as if it does not support session-
215 timers for inbound or outbound requests. If a remote end-point requests
216 session-timers in a dialog, then Asterisk ignores that request unless it's
217 noted as a requirement (Require: header), in which case the INVITE is
218 rejected with a 420 Bad Extension response.
222 #include "asterisk.h"
224 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
227 #include <sys/signal.h>
229 #include <inttypes.h>
231 #include "asterisk/network.h"
232 #include "asterisk/paths.h" /* need ast_config_AST_SYSTEM_NAME */
234 Uncomment the define below, if you are having refcount related memory leaks.
235 With this uncommented, this module will generate a file, /tmp/refs, which contains
236 a history of the ao2_ref() calls. To be useful, all calls to ao2_* functions should
237 be modified to ao2_t_* calls, and include a tag describing what is happening with
238 enough detail, to make pairing up a reference count increment with its corresponding decrement.
239 The refcounter program in utils/ can be invaluable in highlighting objects that are not
240 balanced, along with the complete history for that object.
241 In normal operation, the macros defined will throw away the tags, so they do not
242 affect the speed of the program at all. They can be considered to be documentation.
244 Note: This must also be enabled in channels/sip/security_events.c
246 /* #define REF_DEBUG 1 */
248 #include "asterisk/lock.h"
249 #include "asterisk/config.h"
250 #include "asterisk/module.h"
251 #include "asterisk/pbx.h"
252 #include "asterisk/sched.h"
253 #include "asterisk/io.h"
254 #include "asterisk/rtp_engine.h"
255 #include "asterisk/udptl.h"
256 #include "asterisk/acl.h"
257 #include "asterisk/manager.h"
258 #include "asterisk/callerid.h"
259 #include "asterisk/cli.h"
260 #include "asterisk/musiconhold.h"
261 #include "asterisk/dsp.h"
262 #include "asterisk/pickup.h"
263 #include "asterisk/parking.h"
264 #include "asterisk/srv.h"
265 #include "asterisk/astdb.h"
266 #include "asterisk/causes.h"
267 #include "asterisk/utils.h"
268 #include "asterisk/file.h"
269 #include "asterisk/astobj2.h"
270 #include "asterisk/dnsmgr.h"
271 #include "asterisk/devicestate.h"
272 #include "asterisk/monitor.h"
273 #include "asterisk/netsock2.h"
274 #include "asterisk/localtime.h"
275 #include "asterisk/abstract_jb.h"
276 #include "asterisk/threadstorage.h"
277 #include "asterisk/translate.h"
278 #include "asterisk/ast_version.h"
279 #include "asterisk/data.h"
280 #include "asterisk/aoc.h"
281 #include "asterisk/message.h"
282 #include "sip/include/sip.h"
283 #include "sip/include/globals.h"
284 #include "sip/include/config_parser.h"
285 #include "sip/include/reqresp_parser.h"
286 #include "sip/include/sip_utils.h"
287 #include "asterisk/sdp_srtp.h"
288 #include "asterisk/ccss.h"
289 #include "asterisk/xml.h"
290 #include "sip/include/dialog.h"
291 #include "sip/include/dialplan_functions.h"
292 #include "sip/include/security_events.h"
293 #include "sip/include/route.h"
294 #include "asterisk/sip_api.h"
295 #include "asterisk/app.h"
296 #include "asterisk/bridge.h"
297 #include "asterisk/stasis.h"
298 #include "asterisk/stasis_endpoints.h"
299 #include "asterisk/stasis_system.h"
300 #include "asterisk/stasis_channels.h"
301 #include "asterisk/features_config.h"
302 #include "asterisk/http_websocket.h"
305 <application name="SIPDtmfMode" language="en_US">
307 Change the dtmfmode for a SIP call.
310 <parameter name="mode" required="true">
312 <enum name="inband" />
314 <enum name="rfc2833" />
319 <para>Changes the dtmfmode for a SIP call.</para>
322 <application name="SIPAddHeader" language="en_US">
324 Add a SIP header to the outbound call.
327 <parameter name="Header" required="true" />
328 <parameter name="Content" required="true" />
331 <para>Adds a header to a SIP call placed with DIAL.</para>
332 <para>Remember to use the X-header if you are adding non-standard SIP
333 headers, like <literal>X-Asterisk-Accountcode:</literal>. Use this with care.
334 Adding the wrong headers may jeopardize the SIP dialog.</para>
335 <para>Always returns <literal>0</literal>.</para>
338 <application name="SIPRemoveHeader" language="en_US">
340 Remove SIP headers previously added with SIPAddHeader
343 <parameter name="Header" required="false" />
346 <para>SIPRemoveHeader() allows you to remove headers which were previously
347 added with SIPAddHeader(). If no parameter is supplied, all previously added
348 headers will be removed. If a parameter is supplied, only the matching headers
349 will be removed.</para>
350 <para>For example you have added these 2 headers:</para>
351 <para>SIPAddHeader(P-Asserted-Identity: sip:foo@bar);</para>
352 <para>SIPAddHeader(P-Preferred-Identity: sip:bar@foo);</para>
354 <para>// remove all headers</para>
355 <para>SIPRemoveHeader();</para>
356 <para>// remove all P- headers</para>
357 <para>SIPRemoveHeader(P-);</para>
358 <para>// remove only the PAI header (note the : at the end)</para>
359 <para>SIPRemoveHeader(P-Asserted-Identity:);</para>
361 <para>Always returns <literal>0</literal>.</para>
364 <application name="SIPSendCustomINFO" language="en_US">
366 Send a custom INFO frame on specified channels.
369 <parameter name="Data" required="true" />
370 <parameter name="UserAgent" required="false" />
373 <para>SIPSendCustomINFO() allows you to send a custom INFO message on all
374 active SIP channels or on channels with the specified User Agent. This
375 application is only available if TEST_FRAMEWORK is defined.</para>
378 <function name="SIP_HEADER" language="en_US">
380 Gets the specified SIP header from an incoming INVITE message.
383 <parameter name="name" required="true" />
384 <parameter name="number">
385 <para>If not specified, defaults to <literal>1</literal>.</para>
389 <para>Since there are several headers (such as Via) which can occur multiple
390 times, SIP_HEADER takes an optional second argument to specify which header with
391 that name to retrieve. Headers start at offset <literal>1</literal>.</para>
392 <para>Please observe that contents of the SDP (an attachment to the
393 SIP request) can't be accessed with this function.</para>
396 <function name="SIPPEER" language="en_US">
398 Gets SIP peer information.
401 <parameter name="peername" required="true" />
402 <parameter name="item">
405 <para>(default) The IP address.</para>
408 <para>The port number.</para>
410 <enum name="mailbox">
411 <para>The configured mailbox.</para>
413 <enum name="context">
414 <para>The configured context.</para>
417 <para>The epoch time of the next expire.</para>
419 <enum name="dynamic">
420 <para>Is it dynamic? (yes/no).</para>
422 <enum name="callerid_name">
423 <para>The configured Caller ID name.</para>
425 <enum name="callerid_num">
426 <para>The configured Caller ID number.</para>
428 <enum name="callgroup">
429 <para>The configured Callgroup.</para>
431 <enum name="pickupgroup">
432 <para>The configured Pickupgroup.</para>
434 <enum name="namedcallgroup">
435 <para>The configured Named Callgroup.</para>
437 <enum name="namedpickupgroup">
438 <para>The configured Named Pickupgroup.</para>
441 <para>The configured codecs.</para>
444 <para>Status (if qualify=yes).</para>
446 <enum name="regexten">
447 <para>Extension activated at registration.</para>
450 <para>Call limit (call-limit).</para>
452 <enum name="busylevel">
453 <para>Configured call level for signalling busy.</para>
455 <enum name="curcalls">
456 <para>Current amount of calls. Only available if call-limit is set.</para>
458 <enum name="language">
459 <para>Default language for peer.</para>
461 <enum name="accountcode">
462 <para>Account code for this peer.</para>
464 <enum name="useragent">
465 <para>Current user agent header used by peer.</para>
467 <enum name="maxforwards">
468 <para>The value used for SIP loop prevention in outbound requests</para>
470 <enum name="chanvar[name]">
471 <para>A channel variable configured with setvar for this peer.</para>
473 <enum name="codec[x]">
474 <para>Preferred codec index number <replaceable>x</replaceable> (beginning with zero).</para>
479 <description></description>
481 <function name="SIPCHANINFO" language="en_US">
483 Gets the specified SIP parameter from the current channel.
486 <parameter name="item" required="true">
489 <para>The IP address of the peer.</para>
492 <para>The source IP address of the peer.</para>
495 <para>The SIP URI from the <literal>From:</literal> header.</para>
498 <para>The SIP URI from the <literal>Contact:</literal> header.</para>
500 <enum name="useragent">
501 <para>The Useragent header used by the peer.</para>
503 <enum name="peername">
504 <para>The name of the peer.</para>
506 <enum name="t38passthrough">
507 <para><literal>1</literal> if T38 is offered or enabled in this channel,
508 otherwise <literal>0</literal>.</para>
513 <description></description>
515 <function name="CHECKSIPDOMAIN" language="en_US">
517 Checks if domain is a local domain.
520 <parameter name="domain" required="true" />
523 <para>This function checks if the <replaceable>domain</replaceable> in the argument is configured
524 as a local SIP domain that this Asterisk server is configured to handle.
525 Returns the domain name if it is locally handled, otherwise an empty string.
526 Check the <literal>domain=</literal> configuration in <filename>sip.conf</filename>.</para>
529 <manager name="SIPpeers" language="en_US">
531 List SIP peers (text format).
534 <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
537 <para>Lists SIP peers in text format with details on current status.
538 <literal>Peerlist</literal> will follow as separate events, followed by a final event called
539 <literal>PeerlistComplete</literal>.</para>
542 <manager name="SIPshowpeer" language="en_US">
544 show SIP peer (text format).
547 <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
548 <parameter name="Peer" required="true">
549 <para>The peer name you want to check.</para>
553 <para>Show one SIP peer with details on current status.</para>
556 <manager name="SIPqualifypeer" language="en_US">
561 <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
562 <parameter name="Peer" required="true">
563 <para>The peer name you want to qualify.</para>
567 <para>Qualify a SIP peer.</para>
570 <ref type="managerEvent">SIPQualifyPeerDone</ref>
573 <manager name="SIPshowregistry" language="en_US">
575 Show SIP registrations (text format).
578 <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
581 <para>Lists all registration requests and status. Registrations will follow as separate
582 events followed by a final event called <literal>RegistrationsComplete</literal>.</para>
585 <manager name="SIPnotify" language="en_US">
590 <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
591 <parameter name="Channel" required="true">
592 <para>Peer to receive the notify.</para>
594 <parameter name="Variable" required="true">
595 <para>At least one variable pair must be specified.
596 <replaceable>name</replaceable>=<replaceable>value</replaceable></para>
600 <para>Sends a SIP Notify event.</para>
601 <para>All parameters for this event must be specified in the body of this request
602 via multiple <literal>Variable: name=value</literal> sequences.</para>
605 <manager name="SIPpeerstatus" language="en_US">
607 Show the status of one or all of the sip peers.
610 <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
611 <parameter name="Peer" required="false">
612 <para>The peer name you want to check.</para>
616 <para>Retrieves the status of one or all of the sip peers. If no peer name is specified, status
617 for all of the sip peers will be retrieved.</para>
620 <info name="SIPMessageFromInfo" language="en_US" tech="SIP">
621 <para>The <literal>from</literal> parameter can be a configured peer name
622 or in the form of "display-name" <URI>.</para>
624 <info name="SIPMessageToInfo" language="en_US" tech="SIP">
625 <para>Specifying a prefix of <literal>sip:</literal> will send the
626 message as a SIP MESSAGE request.</para>
628 <managerEvent language="en_US" name="SIPQualifyPeerDone">
629 <managerEventInstance class="EVENT_FLAG_CALL">
630 <synopsis>Raised when SIPQualifyPeer has finished qualifying the specified peer.</synopsis>
632 <parameter name="Peer">
633 <para>The name of the peer.</para>
635 <parameter name="ActionID">
636 <para>This is only included if an ActionID Header was sent with the action request, in which case it will be that ActionID.</para>
640 <ref type="manager">SIPqualifypeer</ref>
642 </managerEventInstance>
644 <managerEvent language="en_US" name="SessionTimeout">
645 <managerEventInstance class="EVENT_FLAG_CALL">
646 <synopsis>Raised when a SIP session times out.</synopsis>
649 <parameter name="Source">
650 <para>The source of the session timeout.</para>
652 <enum name="RTPTimeout" />
653 <enum name="SIPSessionTimer" />
657 </managerEventInstance>
661 static int min_expiry = DEFAULT_MIN_EXPIRY; /*!< Minimum accepted registration time */
662 static int max_expiry = DEFAULT_MAX_EXPIRY; /*!< Maximum accepted registration time */
663 static int default_expiry = DEFAULT_DEFAULT_EXPIRY;
664 static int min_subexpiry = DEFAULT_MIN_EXPIRY; /*!< Minimum accepted subscription time */
665 static int max_subexpiry = DEFAULT_MAX_EXPIRY; /*!< Maximum accepted subscription time */
666 static int mwi_expiry = DEFAULT_MWI_EXPIRY;
668 static int unauth_sessions = 0;
669 static int authlimit = DEFAULT_AUTHLIMIT;
670 static int authtimeout = DEFAULT_AUTHTIMEOUT;
672 /*! \brief Global jitterbuffer configuration - by default, jb is disabled
673 * \note Values shown here match the defaults shown in sip.conf.sample */
674 static struct ast_jb_conf default_jbconf =
678 .resync_threshold = 1000,
682 static struct ast_jb_conf global_jbconf; /*!< Global jitterbuffer configuration */
684 static const char config[] = "sip.conf"; /*!< Main configuration file */
685 static const char notify_config[] = "sip_notify.conf"; /*!< Configuration file for sending Notify with CLI commands to reconfigure or reboot phones */
687 /*! \brief Readable descriptions of device states.
688 * \note Should be aligned to above table as index */
689 static const struct invstate2stringtable {
690 const enum invitestates state;
692 } invitestate2string[] = {
694 {INV_CALLING, "Calling (Trying)"},
695 {INV_PROCEEDING, "Proceeding "},
696 {INV_EARLY_MEDIA, "Early media"},
697 {INV_COMPLETED, "Completed (done)"},
698 {INV_CONFIRMED, "Confirmed (up)"},
699 {INV_TERMINATED, "Done"},
700 {INV_CANCELLED, "Cancelled"}
703 /*! \brief Subscription types that we support. We support
704 * - dialoginfo updates (really device status, not dialog info as was the original intent of the standard)
705 * - SIMPLE presence used for device status
706 * - Voicemail notification subscriptions
708 static const struct cfsubscription_types {
709 enum subscriptiontype type;
710 const char * const event;
711 const char * const mediatype;
712 const char * const text;
713 } subscription_types[] = {
714 { NONE, "-", "unknown", "unknown" },
715 /* RFC 4235: SIP Dialog event package */
716 { DIALOG_INFO_XML, "dialog", "application/dialog-info+xml", "dialog-info+xml" },
717 { CPIM_PIDF_XML, "presence", "application/cpim-pidf+xml", "cpim-pidf+xml" }, /* RFC 3863 */
718 { PIDF_XML, "presence", "application/pidf+xml", "pidf+xml" }, /* RFC 3863 */
719 { XPIDF_XML, "presence", "application/xpidf+xml", "xpidf+xml" }, /* Pre-RFC 3863 with MS additions */
720 { MWI_NOTIFICATION, "message-summary", "application/simple-message-summary", "mwi" } /* RFC 3842: Mailbox notification */
723 /*! \brief The core structure to setup dialogs. We parse incoming messages by using
724 * structure and then route the messages according to the type.
726 * \note Note that sip_methods[i].id == i must hold or the code breaks
728 static const struct cfsip_methods {
730 int need_rtp; /*!< when this is the 'primary' use for a pvt structure, does it need RTP? */
732 enum can_create_dialog can_create;
734 { SIP_UNKNOWN, RTP, "-UNKNOWN-",CAN_CREATE_DIALOG },
735 { SIP_RESPONSE, NO_RTP, "SIP/2.0", CAN_NOT_CREATE_DIALOG },
736 { SIP_REGISTER, NO_RTP, "REGISTER", CAN_CREATE_DIALOG },
737 { SIP_OPTIONS, NO_RTP, "OPTIONS", CAN_CREATE_DIALOG },
738 { SIP_NOTIFY, NO_RTP, "NOTIFY", CAN_CREATE_DIALOG },
739 { SIP_INVITE, RTP, "INVITE", CAN_CREATE_DIALOG },
740 { SIP_ACK, NO_RTP, "ACK", CAN_NOT_CREATE_DIALOG },
741 { SIP_PRACK, NO_RTP, "PRACK", CAN_NOT_CREATE_DIALOG },
742 { SIP_BYE, NO_RTP, "BYE", CAN_NOT_CREATE_DIALOG },
743 { SIP_REFER, NO_RTP, "REFER", CAN_CREATE_DIALOG },
744 { SIP_SUBSCRIBE, NO_RTP, "SUBSCRIBE",CAN_CREATE_DIALOG },
745 { SIP_MESSAGE, NO_RTP, "MESSAGE", CAN_CREATE_DIALOG },
746 { SIP_UPDATE, NO_RTP, "UPDATE", CAN_NOT_CREATE_DIALOG },
747 { SIP_INFO, NO_RTP, "INFO", CAN_NOT_CREATE_DIALOG },
748 { SIP_CANCEL, NO_RTP, "CANCEL", CAN_NOT_CREATE_DIALOG },
749 { SIP_PUBLISH, NO_RTP, "PUBLISH", CAN_CREATE_DIALOG },
750 { SIP_PING, NO_RTP, "PING", CAN_CREATE_DIALOG_UNSUPPORTED_METHOD }
753 /*! \brief Diversion header reasons
755 * The core defines a bunch of constants used to define
756 * redirecting reasons. This provides a translation table
757 * between those and the strings which may be present in
758 * a SIP Diversion header
760 static const struct sip_reasons {
761 enum AST_REDIRECTING_REASON code;
763 } sip_reason_table[] = {
764 { AST_REDIRECTING_REASON_UNKNOWN, "unknown" },
765 { AST_REDIRECTING_REASON_USER_BUSY, "user-busy" },
766 { AST_REDIRECTING_REASON_NO_ANSWER, "no-answer" },
767 { AST_REDIRECTING_REASON_UNAVAILABLE, "unavailable" },
768 { AST_REDIRECTING_REASON_UNCONDITIONAL, "unconditional" },
769 { AST_REDIRECTING_REASON_TIME_OF_DAY, "time-of-day" },
770 { AST_REDIRECTING_REASON_DO_NOT_DISTURB, "do-not-disturb" },
771 { AST_REDIRECTING_REASON_DEFLECTION, "deflection" },
772 { AST_REDIRECTING_REASON_FOLLOW_ME, "follow-me" },
773 { AST_REDIRECTING_REASON_OUT_OF_ORDER, "out-of-service" },
774 { AST_REDIRECTING_REASON_AWAY, "away" },
775 { AST_REDIRECTING_REASON_CALL_FWD_DTE, "unknown"},
776 { AST_REDIRECTING_REASON_SEND_TO_VM, "send_to_vm"},
780 /*! \name DefaultSettings
781 Default setttings are used as a channel setting and as a default when
784 static char default_language[MAX_LANGUAGE]; /*!< Default language setting for new channels */
785 static char default_callerid[AST_MAX_EXTENSION]; /*!< Default caller ID for sip messages */
786 static char default_mwi_from[80]; /*!< Default caller ID for MWI updates */
787 static char default_fromdomain[AST_MAX_EXTENSION]; /*!< Default domain on outound messages */
788 static int default_fromdomainport; /*!< Default domain port on outbound messages */
789 static char default_notifymime[AST_MAX_EXTENSION]; /*!< Default MIME media type for MWI notify messages */
790 static char default_vmexten[AST_MAX_EXTENSION]; /*!< Default From Username on MWI updates */
791 static int default_qualify; /*!< Default Qualify= setting */
792 static int default_keepalive; /*!< Default keepalive= setting */
793 static char default_mohinterpret[MAX_MUSICCLASS]; /*!< Global setting for moh class to use when put on hold */
794 static char default_mohsuggest[MAX_MUSICCLASS]; /*!< Global setting for moh class to suggest when putting
795 * a bridged channel on hold */
796 static char default_parkinglot[AST_MAX_CONTEXT]; /*!< Parkinglot */
797 static char default_engine[256]; /*!< Default RTP engine */
798 static int default_maxcallbitrate; /*!< Maximum bitrate for call */
799 static struct ast_codec_pref default_prefs; /*!< Default codec prefs */
800 static char default_zone[MAX_TONEZONE_COUNTRY]; /*!< Default tone zone for channels created from the SIP driver */
801 static unsigned int default_transports; /*!< Default Transports (enum ast_transport) that are acceptable */
802 static unsigned int default_primary_transport; /*!< Default primary Transport (enum ast_transport) for outbound connections to devices */
804 static struct sip_settings sip_cfg; /*!< SIP configuration data.
805 \note in the future we could have multiple of these (per domain, per device group etc) */
807 /*!< use this macro when ast_uri_decode is dependent on pedantic checking to be on. */
808 #define SIP_PEDANTIC_DECODE(str) \
809 if (sip_cfg.pedanticsipchecking && !ast_strlen_zero(str)) { \
810 ast_uri_decode(str, ast_uri_sip_user); \
813 static unsigned int chan_idx; /*!< used in naming sip channel */
814 static int global_match_auth_username; /*!< Match auth username if available instead of From: Default off. */
816 static int global_relaxdtmf; /*!< Relax DTMF */
817 static int global_prematuremediafilter; /*!< Enable/disable premature frames in a call (causing 183 early media) */
818 static int global_rtptimeout; /*!< Time out call if no RTP */
819 static int global_rtpholdtimeout; /*!< Time out call if no RTP during hold */
820 static int global_rtpkeepalive; /*!< Send RTP keepalives */
821 static int global_reg_timeout; /*!< Global time between attempts for outbound registrations */
822 static int global_regattempts_max; /*!< Registration attempts before giving up */
823 static int global_reg_retry_403; /*!< Treat 403 responses to registrations as 401 responses */
824 static int global_shrinkcallerid; /*!< enable or disable shrinking of caller id */
825 static int global_callcounter; /*!< Enable call counters for all devices. This is currently enabled by setting the peer
826 * call-limit to INT_MAX. When we remove the call-limit from the code, we can make it
827 * with just a boolean flag in the device structure */
828 static unsigned int global_tos_sip; /*!< IP type of service for SIP packets */
829 static unsigned int global_tos_audio; /*!< IP type of service for audio RTP packets */
830 static unsigned int global_tos_video; /*!< IP type of service for video RTP packets */
831 static unsigned int global_tos_text; /*!< IP type of service for text RTP packets */
832 static unsigned int global_cos_sip; /*!< 802.1p class of service for SIP packets */
833 static unsigned int global_cos_audio; /*!< 802.1p class of service for audio RTP packets */
834 static unsigned int global_cos_video; /*!< 802.1p class of service for video RTP packets */
835 static unsigned int global_cos_text; /*!< 802.1p class of service for text RTP packets */
836 static unsigned int recordhistory; /*!< Record SIP history. Off by default */
837 static unsigned int dumphistory; /*!< Dump history to verbose before destroying SIP dialog */
838 static char global_useragent[AST_MAX_EXTENSION]; /*!< Useragent for the SIP channel */
839 static char global_sdpsession[AST_MAX_EXTENSION]; /*!< SDP session name for the SIP channel */
840 static char global_sdpowner[AST_MAX_EXTENSION]; /*!< SDP owner name for the SIP channel */
841 static int global_authfailureevents; /*!< Whether we send authentication failure manager events or not. Default no. */
842 static int global_t1; /*!< T1 time */
843 static int global_t1min; /*!< T1 roundtrip time minimum */
844 static int global_timer_b; /*!< Timer B - RFC 3261 Section 17.1.1.2 */
845 static unsigned int global_autoframing; /*!< Turn autoframing on or off. */
846 static int global_qualifyfreq; /*!< Qualify frequency */
847 static int global_qualify_gap; /*!< Time between our group of peer pokes */
848 static int global_qualify_peers; /*!< Number of peers to poke at a given time */
850 static enum st_mode global_st_mode; /*!< Mode of operation for Session-Timers */
851 static enum st_refresher_param global_st_refresher; /*!< Session-Timer refresher */
852 static int global_min_se; /*!< Lowest threshold for session refresh interval */
853 static int global_max_se; /*!< Highest threshold for session refresh interval */
855 static int global_store_sip_cause; /*!< Whether the MASTER_CHANNEL(HASH(SIP_CAUSE,[chan_name])) var should be set */
857 static int global_dynamic_exclude_static = 0; /*!< Exclude static peers from contact registrations */
858 static unsigned char global_refer_addheaders; /*!< Add extra headers to outgoing REFER */
862 * We use libxml2 in order to parse XML that may appear in the body of a SIP message. Currently,
863 * the only usage is for parsing PIDF bodies of incoming PUBLISH requests in the call-completion
864 * event package. This variable is set at module load time and may be checked at runtime to determine
865 * if XML parsing support was found.
867 static int can_parse_xml;
869 /*! \name Object counters @{
871 * \bug These counters are not handled in a thread-safe way ast_atomic_fetchadd_int()
872 * should be used to modify these values.
874 static int speerobjs = 0; /*!< Static peers */
875 static int rpeerobjs = 0; /*!< Realtime peers */
876 static int apeerobjs = 0; /*!< Autocreated peer objects */
877 static int regobjs = 0; /*!< Registry objects */
880 static struct ast_flags global_flags[3] = {{0}}; /*!< global SIP_ flags */
881 static int global_t38_maxdatagram; /*!< global T.38 FaxMaxDatagram override */
883 static struct stasis_subscription *network_change_sub; /*!< subscription id for network change events */
884 static struct stasis_subscription *acl_change_sub; /*!< subscription id for named ACL system change events */
885 static int network_change_sched_id = -1;
887 static char used_context[AST_MAX_CONTEXT]; /*!< name of automatically created context for unloading */
889 AST_MUTEX_DEFINE_STATIC(netlock);
891 /*! \brief Protect the monitoring thread, so only one process can kill or start it, and not
892 when it's doing something critical. */
893 AST_MUTEX_DEFINE_STATIC(monlock);
895 AST_MUTEX_DEFINE_STATIC(sip_reload_lock);
897 /*! \brief This is the thread for the monitor which checks for input on the channels
898 which are not currently in use. */
899 static pthread_t monitor_thread = AST_PTHREADT_NULL;
901 static int sip_reloading = FALSE; /*!< Flag for avoiding multiple reloads at the same time */
902 static enum channelreloadreason sip_reloadreason; /*!< Reason for last reload/load of configuration */
904 struct ast_sched_context *sched; /*!< The scheduling context */
905 static struct io_context *io; /*!< The IO context */
906 static int *sipsock_read_id; /*!< ID of IO entry for sipsock FD */
908 static AST_LIST_HEAD_STATIC(domain_list, domain); /*!< The SIP domain list */
910 AST_LIST_HEAD_NOLOCK(sip_history_head, sip_history); /*!< history list, entry in sip_pvt */
912 static enum sip_debug_e sipdebug;
914 /*! \brief extra debugging for 'text' related events.
915 * At the moment this is set together with sip_debug_console.
916 * \note It should either go away or be implemented properly.
918 static int sipdebug_text;
920 static const struct _map_x_s referstatusstrings[] = {
921 { REFER_IDLE, "<none>" },
922 { REFER_SENT, "Request sent" },
923 { REFER_RECEIVED, "Request received" },
924 { REFER_CONFIRMED, "Confirmed" },
925 { REFER_ACCEPTED, "Accepted" },
926 { REFER_RINGING, "Target ringing" },
927 { REFER_200OK, "Done" },
928 { REFER_FAILED, "Failed" },
929 { REFER_NOAUTH, "Failed - auth failure" },
930 { -1, NULL} /* terminator */
933 /* --- Hash tables of various objects --------*/
935 static const int HASH_PEER_SIZE = 17;
936 static const int HASH_DIALOG_SIZE = 17;
938 static const int HASH_PEER_SIZE = 563; /*!< Size of peer hash table, prime number preferred! */
939 static const int HASH_DIALOG_SIZE = 563;
942 static const struct {
943 enum ast_cc_service_type service;
944 const char *service_string;
945 } sip_cc_service_map [] = {
946 [AST_CC_NONE] = { AST_CC_NONE, "" },
947 [AST_CC_CCBS] = { AST_CC_CCBS, "BS" },
948 [AST_CC_CCNR] = { AST_CC_CCNR, "NR" },
949 [AST_CC_CCNL] = { AST_CC_CCNL, "NL" },
952 static const struct {
953 enum sip_cc_notify_state state;
954 const char *state_string;
955 } sip_cc_notify_state_map [] = {
956 [CC_QUEUED] = {CC_QUEUED, "cc-state: queued"},
957 [CC_READY] = {CC_READY, "cc-state: ready"},
960 AST_LIST_HEAD_STATIC(epa_static_data_list, epa_backend);
964 * Used to create new entity IDs by ESCs.
966 static int esc_etag_counter;
967 static const int DEFAULT_PUBLISH_EXPIRES = 3600;
970 static int cc_esc_publish_handler(struct sip_pvt *pvt, struct sip_request *req, struct event_state_compositor *esc, struct sip_esc_entry *esc_entry);
972 static const struct sip_esc_publish_callbacks cc_esc_publish_callbacks = {
973 .initial_handler = cc_esc_publish_handler,
974 .modify_handler = cc_esc_publish_handler,
979 * \brief The Event State Compositors
981 * An Event State Compositor is an entity which
982 * accepts PUBLISH requests and acts appropriately
983 * based on these requests.
985 * The actual event_state_compositor structure is simply
986 * an ao2_container of sip_esc_entrys. When an incoming
987 * PUBLISH is received, we can match the appropriate sip_esc_entry
988 * using the entity ID of the incoming PUBLISH.
990 static struct event_state_compositor {
991 enum subscriptiontype event;
993 const struct sip_esc_publish_callbacks *callbacks;
994 struct ao2_container *compositor;
995 } event_state_compositors [] = {
997 {CALL_COMPLETION, "call-completion", &cc_esc_publish_callbacks},
1001 struct state_notify_data {
1003 struct ao2_container *device_state_info;
1005 const char *presence_subtype;
1006 const char *presence_message;
1010 static const int ESC_MAX_BUCKETS = 37;
1014 * Here we implement the container for dialogs which are in the
1015 * dialog_needdestroy state to iterate only through the dialogs
1016 * unlink them instead of iterate through all dialogs
1018 struct ao2_container *dialogs_needdestroy;
1022 * Here we implement the container for dialogs which have rtp
1023 * traffic and rtptimeout, rtpholdtimeout or rtpkeepalive
1024 * set. We use this container instead the whole dialog list.
1026 struct ao2_container *dialogs_rtpcheck;
1030 * Here we implement the container for dialogs (sip_pvt), defining
1031 * generic wrapper functions to ease the transition from the current
1032 * implementation (a single linked list) to a different container.
1033 * In addition to a reference to the container, we need functions to lock/unlock
1034 * the container and individual items, and functions to add/remove
1035 * references to the individual items.
1037 static struct ao2_container *dialogs;
1038 #define sip_pvt_lock(x) ao2_lock(x)
1039 #define sip_pvt_trylock(x) ao2_trylock(x)
1040 #define sip_pvt_unlock(x) ao2_unlock(x)
1042 /*! \brief The table of TCP threads */
1043 static struct ao2_container *threadt;
1045 /*! \brief The peer list: Users, Peers and Friends */
1046 static struct ao2_container *peers;
1047 static struct ao2_container *peers_by_ip;
1049 /*! \brief A bogus peer, to be used when authentication should fail */
1050 static struct sip_peer *bogus_peer;
1051 /*! \brief We can recognise the bogus peer by this invalid MD5 hash */
1052 #define BOGUS_PEER_MD5SECRET "intentionally_invalid_md5_string"
1054 /*! \brief The register list: Other SIP proxies we register with and receive calls from */
1055 static struct ast_register_list {
1056 ASTOBJ_CONTAINER_COMPONENTS(struct sip_registry);
1060 /*! \brief The MWI subscription list */
1061 static struct ast_subscription_mwi_list {
1062 ASTOBJ_CONTAINER_COMPONENTS(struct sip_subscription_mwi);
1064 static int temp_pvt_init(void *);
1065 static void temp_pvt_cleanup(void *);
1067 /*! \brief A per-thread temporary pvt structure */
1068 AST_THREADSTORAGE_CUSTOM(ts_temp_pvt, temp_pvt_init, temp_pvt_cleanup);
1070 /*! \brief A per-thread buffer for transport to string conversion */
1071 AST_THREADSTORAGE(sip_transport_str_buf);
1073 /*! \brief Size of the SIP transport buffer */
1074 #define SIP_TRANSPORT_STR_BUFSIZE 128
1076 /*! \brief Authentication container for realm authentication */
1077 static struct sip_auth_container *authl = NULL;
1078 /*! \brief Global authentication container protection while adjusting the references. */
1079 AST_MUTEX_DEFINE_STATIC(authl_lock);
1081 static struct ast_manager_event_blob *session_timeout_to_ami(struct stasis_message *msg);
1082 STASIS_MESSAGE_TYPE_DEFN_LOCAL(session_timeout_type,
1083 .to_ami = session_timeout_to_ami,
1086 /* --- Sockets and networking --------------*/
1088 /*! \brief Main socket for UDP SIP communication.
1090 * sipsock is shared between the SIP manager thread (which handles reload
1091 * requests), the udp io handler (sipsock_read()) and the user routines that
1092 * issue udp writes (using __sip_xmit()).
1093 * The socket is -1 only when opening fails (this is a permanent condition),
1094 * or when we are handling a reload() that changes its address (this is
1095 * a transient situation during which we might have a harmless race, see
1096 * below). Because the conditions for the race to be possible are extremely
1097 * rare, we don't want to pay the cost of locking on every I/O.
1098 * Rather, we remember that when the race may occur, communication is
1099 * bound to fail anyways, so we just live with this event and let
1100 * the protocol handle this above us.
1102 static int sipsock = -1;
1104 struct ast_sockaddr bindaddr; /*!< UDP: The address we bind to */
1106 /*! \brief our (internal) default address/port to put in SIP/SDP messages
1107 * internip is initialized picking a suitable address from one of the
1108 * interfaces, and the same port number we bind to. It is used as the
1109 * default address/port in SIP messages, and as the default address
1110 * (but not port) in SDP messages.
1112 static struct ast_sockaddr internip;
1114 /*! \brief our external IP address/port for SIP sessions.
1115 * externaddr.sin_addr is only set when we know we might be behind
1116 * a NAT, and this is done using a variety of (mutually exclusive)
1117 * ways from the config file:
1119 * + with "externaddr = host[:port]" we specify the address/port explicitly.
1120 * The address is looked up only once when (re)loading the config file;
1122 * + with "externhost = host[:port]" we do a similar thing, but the
1123 * hostname is stored in externhost, and the hostname->IP mapping
1124 * is refreshed every 'externrefresh' seconds;
1126 * Other variables (externhost, externexpire, externrefresh) are used
1127 * to support the above functions.
1129 static struct ast_sockaddr externaddr; /*!< External IP address if we are behind NAT */
1130 static struct ast_sockaddr media_address; /*!< External RTP IP address if we are behind NAT */
1132 static char externhost[MAXHOSTNAMELEN]; /*!< External host name */
1133 static time_t externexpire; /*!< Expiration counter for re-resolving external host name in dynamic DNS */
1134 static int externrefresh = 10; /*!< Refresh timer for DNS-based external address (dyndns) */
1135 static uint16_t externtcpport; /*!< external tcp port */
1136 static uint16_t externtlsport; /*!< external tls port */
1138 /*! \brief List of local networks
1139 * We store "localnet" addresses from the config file into an access list,
1140 * marked as 'DENY', so the call to ast_apply_ha() will return
1141 * AST_SENSE_DENY for 'local' addresses, and AST_SENSE_ALLOW for 'non local'
1142 * (i.e. presumably public) addresses.
1144 static struct ast_ha *localaddr; /*!< List of local networks, on the same side of NAT as this Asterisk */
1146 static int ourport_tcp; /*!< The port used for TCP connections */
1147 static int ourport_tls; /*!< The port used for TCP/TLS connections */
1148 static struct ast_sockaddr debugaddr;
1150 static struct ast_config *notify_types = NULL; /*!< The list of manual NOTIFY types we know how to send */
1152 /*! some list management macros. */
1154 #define UNLINK(element, head, prev) do { \
1156 (prev)->next = (element)->next; \
1158 (head) = (element)->next; \
1161 struct ao2_container *sip_monitor_instances;
1163 struct show_peers_context;
1165 /*---------------------------- Forward declarations of functions in chan_sip.c */
1166 /* Note: This is added to help splitting up chan_sip.c into several files
1167 in coming releases. */
1169 /*--- PBX interface functions */
1170 static struct ast_channel *sip_request_call(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, const char *dest, int *cause);
1171 static int sip_devicestate(const char *data);
1172 static int sip_sendtext(struct ast_channel *ast, const char *text);
1173 static int sip_call(struct ast_channel *ast, const char *dest, int timeout);
1174 static int sip_sendhtml(struct ast_channel *chan, int subclass, const char *data, int datalen);
1175 static int sip_hangup(struct ast_channel *ast);
1176 static int sip_answer(struct ast_channel *ast);
1177 static struct ast_frame *sip_read(struct ast_channel *ast);
1178 static int sip_write(struct ast_channel *ast, struct ast_frame *frame);
1179 static int sip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen);
1180 static int sip_transfer(struct ast_channel *ast, const char *dest);
1181 static int sip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
1182 static int sip_senddigit_begin(struct ast_channel *ast, char digit);
1183 static int sip_senddigit_end(struct ast_channel *ast, char digit, unsigned int duration);
1184 static int sip_setoption(struct ast_channel *chan, int option, void *data, int datalen);
1185 static int sip_queryoption(struct ast_channel *chan, int option, void *data, int *datalen);
1186 static const char *sip_get_callid(struct ast_channel *chan);
1188 static int handle_request_do(struct sip_request *req, struct ast_sockaddr *addr);
1189 static int sip_standard_port(enum ast_transport type, int port);
1190 static int sip_prepare_socket(struct sip_pvt *p);
1191 static int get_address_family_filter(unsigned int transport);
1193 /*--- Transmitting responses and requests */
1194 static int sipsock_read(int *id, int fd, short events, void *ignore);
1195 static int __sip_xmit(struct sip_pvt *p, struct ast_str *data);
1196 static int __sip_reliable_xmit(struct sip_pvt *p, uint32_t seqno, int resp, struct ast_str *data, int fatal, int sipmethod);
1197 static void add_cc_call_info_to_response(struct sip_pvt *p, struct sip_request *resp);
1198 static int __transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
1199 static int retrans_pkt(const void *data);
1200 static int transmit_response_using_temp(ast_string_field callid, struct ast_sockaddr *addr, int useglobal_nat, const int intended_method, const struct sip_request *req, const char *msg);
1201 static int transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req);
1202 static int transmit_response_reliable(struct sip_pvt *p, const char *msg, const struct sip_request *req);
1203 static int transmit_response_with_date(struct sip_pvt *p, const char *msg, const struct sip_request *req);
1204 static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable, int oldsdp, int rpid);
1205 static int transmit_response_with_unsupported(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *unsupported);
1206 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);
1207 static int transmit_provisional_response(struct sip_pvt *p, const char *msg, const struct sip_request *req, int with_sdp);
1208 static int transmit_response_with_allow(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
1209 static void transmit_fake_auth_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable);
1210 static int transmit_request(struct sip_pvt *p, int sipmethod, uint32_t seqno, enum xmittype reliable, int newbranch);
1211 static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, uint32_t seqno, enum xmittype reliable, int newbranch);
1212 static int transmit_publish(struct sip_epa_entry *epa_entry, enum sip_publish_type publish_type, const char * const explicit_uri);
1213 static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init, const char * const explicit_uri);
1214 static int transmit_reinvite_with_sdp(struct sip_pvt *p, int t38version, int oldsdp);
1215 static int transmit_info_with_aoc(struct sip_pvt *p, struct ast_aoc_decoded *decoded);
1216 static int transmit_info_with_digit(struct sip_pvt *p, const char digit, unsigned int duration);
1217 static int transmit_info_with_vidupdate(struct sip_pvt *p);
1218 static int transmit_message(struct sip_pvt *p, int init, int auth);
1219 static int transmit_refer(struct sip_pvt *p, const char *dest);
1220 static int transmit_notify_with_mwi(struct sip_pvt *p, int newmsgs, int oldmsgs, const char *vmexten);
1221 static int transmit_notify_with_sipfrag(struct sip_pvt *p, int cseq, char *message, int terminate);
1222 static int transmit_cc_notify(struct ast_cc_agent *agent, struct sip_pvt *subscription, enum sip_cc_notify_state state);
1223 static int transmit_register(struct sip_registry *r, int sipmethod, const char *auth, const char *authheader);
1224 static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, uint32_t seqno);
1225 static int send_request(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, uint32_t seqno);
1226 static void copy_request(struct sip_request *dst, const struct sip_request *src);
1227 static void receive_message(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const char *e);
1228 static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req, char **name, char **number, int set_call_forward);
1229 static int sip_send_mwi_to_peer(struct sip_peer *peer, int cache_only);
1231 /* Misc dialog routines */
1232 static int __sip_autodestruct(const void *data);
1233 static void *registry_unref(struct sip_registry *reg, char *tag);
1234 static int update_call_counter(struct sip_pvt *fup, int event);
1235 static int auto_congest(const void *arg);
1236 static struct sip_pvt *find_call(struct sip_request *req, struct ast_sockaddr *addr, const int intended_method);
1237 static void build_route(struct sip_pvt *p, struct sip_request *req, int backwards, int resp);
1238 static int build_path(struct sip_pvt *p, struct sip_peer *peer, struct sip_request *req, const char *pathbuf);
1239 static enum check_auth_result register_verify(struct sip_pvt *p, struct ast_sockaddr *addr,
1240 struct sip_request *req, const char *uri);
1241 static int get_sip_pvt_from_replaces(const char *callid, const char *totag, const char *fromtag,
1242 struct sip_pvt **out_pvt, struct ast_channel **out_chan);
1243 static void check_pendings(struct sip_pvt *p);
1244 static void sip_set_owner(struct sip_pvt *p, struct ast_channel *chan);
1246 static void *sip_pickup_thread(void *stuff);
1247 static int sip_pickup(struct ast_channel *chan);
1249 static int sip_sipredirect(struct sip_pvt *p, const char *dest);
1250 static int is_method_allowed(unsigned int *allowed_methods, enum sipmethod method);
1252 /*--- Codec handling / SDP */
1253 static void try_suggested_sip_codec(struct sip_pvt *p);
1254 static const char *get_sdp_iterate(int* start, struct sip_request *req, const char *name);
1255 static char get_sdp_line(int *start, int stop, struct sip_request *req, const char **value);
1256 static int find_sdp(struct sip_request *req);
1257 static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action);
1258 static int process_sdp_o(const char *o, struct sip_pvt *p);
1259 static int process_sdp_c(const char *c, struct ast_sockaddr *addr);
1260 static int process_sdp_a_sendonly(const char *a, int *sendonly);
1261 static int process_sdp_a_ice(const char *a, struct sip_pvt *p, struct ast_rtp_instance *instance);
1262 static int process_sdp_a_dtls(const char *a, struct sip_pvt *p, struct ast_rtp_instance *instance);
1263 static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newaudiortp, int *last_rtpmap_codec);
1264 static int process_sdp_a_video(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newvideortp, int *last_rtpmap_codec);
1265 static int process_sdp_a_text(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newtextrtp, char *red_fmtp, int *red_num_gen, int *red_data_pt, int *last_rtpmap_codec);
1266 static int process_sdp_a_image(const char *a, struct sip_pvt *p);
1267 static void add_ice_to_sdp(struct ast_rtp_instance *instance, struct ast_str **a_buf);
1268 static void add_dtls_to_sdp(struct ast_rtp_instance *instance, struct ast_str **a_buf);
1269 static void start_ice(struct ast_rtp_instance *instance);
1270 static void add_codec_to_sdp(const struct sip_pvt *p, struct ast_format *codec,
1271 struct ast_str **m_buf, struct ast_str **a_buf,
1272 int debug, int *min_packet_size, int *max_packet_size);
1273 static void add_noncodec_to_sdp(const struct sip_pvt *p, int format,
1274 struct ast_str **m_buf, struct ast_str **a_buf,
1276 static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int oldsdp, int add_audio, int add_t38);
1277 static void do_setnat(struct sip_pvt *p);
1278 static void stop_media_flows(struct sip_pvt *p);
1280 /*--- Authentication stuff */
1281 static int reply_digest(struct sip_pvt *p, struct sip_request *req, char *header, int sipmethod, char *digest, int digest_len);
1282 static int build_reply_digest(struct sip_pvt *p, int method, char *digest, int digest_len);
1283 static enum check_auth_result check_auth(struct sip_pvt *p, struct sip_request *req, const char *username,
1284 const char *secret, const char *md5secret, int sipmethod,
1285 const char *uri, enum xmittype reliable);
1286 static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_request *req,
1287 int sipmethod, const char *uri, enum xmittype reliable,
1288 struct ast_sockaddr *addr, struct sip_peer **authpeer);
1289 static int check_user(struct sip_pvt *p, struct sip_request *req, int sipmethod, const char *uri, enum xmittype reliable, struct ast_sockaddr *addr);
1291 /*--- Domain handling */
1292 static int check_sip_domain(const char *domain, char *context, size_t len); /* Check if domain is one of our local domains */
1293 static int add_sip_domain(const char *domain, const enum domain_mode mode, const char *context);
1294 static void clear_sip_domains(void);
1296 /*--- SIP realm authentication */
1297 static void add_realm_authentication(struct sip_auth_container **credentials, const char *configuration, int lineno);
1298 static struct sip_auth *find_realm_authentication(struct sip_auth_container *credentials, const char *realm);
1300 /*--- Misc functions */
1301 static int check_rtp_timeout(struct sip_pvt *dialog, time_t t);
1302 static int reload_config(enum channelreloadreason reason);
1303 static void add_diversion(struct sip_request *req, struct sip_pvt *pvt);
1304 static int expire_register(const void *data);
1305 static void *do_monitor(void *data);
1306 static int restart_monitor(void);
1307 static void peer_mailboxes_to_str(struct ast_str **mailbox_str, struct sip_peer *peer);
1308 static struct ast_variable *copy_vars(struct ast_variable *src);
1309 static int dialog_find_multiple(void *obj, void *arg, int flags);
1310 static struct ast_channel *sip_pvt_lock_full(struct sip_pvt *pvt);
1311 /* static int sip_addrcmp(char *name, struct sockaddr_in *sin); Support for peer matching */
1312 static int sip_refer_alloc(struct sip_pvt *p);
1313 static int sip_notify_alloc(struct sip_pvt *p);
1314 static int do_magic_pickup(struct ast_channel *channel, const char *extension, const char *context);
1315 static void set_peer_nat(const struct sip_pvt *p, struct sip_peer *peer);
1316 static void check_for_nat(const struct ast_sockaddr *them, struct sip_pvt *p);
1318 /*--- Device monitoring and Device/extension state/event handling */
1319 static int extensionstate_update(const char *context, const char *exten, struct state_notify_data *data, struct sip_pvt *p, int force);
1320 static int cb_extensionstate(char *context, char *exten, struct ast_state_cb_info *info, void *data);
1321 static int sip_poke_noanswer(const void *data);
1322 static int sip_poke_peer(struct sip_peer *peer, int force);
1323 static void sip_poke_all_peers(void);
1324 static void sip_peer_hold(struct sip_pvt *p, int hold);
1325 static void mwi_event_cb(void *, struct stasis_subscription *, struct stasis_message *);
1326 static void network_change_stasis_cb(void *data, struct stasis_subscription *sub, struct stasis_message *message);
1327 static void acl_change_stasis_cb(void *data, struct stasis_subscription *sub, struct stasis_message *message);
1328 static void sip_keepalive_all_peers(void);
1330 /*--- Applications, functions, CLI and manager command helpers */
1331 static const char *sip_nat_mode(const struct sip_pvt *p);
1332 static char *sip_show_inuse(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
1333 static char *transfermode2str(enum transfermodes mode) attribute_const;
1334 static int peer_status(struct sip_peer *peer, char *status, int statuslen);
1335 static char *sip_show_sched(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
1336 static char * _sip_show_peers(int fd, int *total, struct mansession *s, const struct message *m, int argc, const char *argv[]);
1337 static struct sip_peer *_sip_show_peers_one(int fd, struct mansession *s, struct show_peers_context *cont, struct sip_peer *peer);
1338 static char *sip_show_peers(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
1339 static char *sip_show_objects(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
1340 static void print_group(int fd, ast_group_t group, int crlf);
1341 static void print_named_groups(int fd, struct ast_namedgroups *groups, int crlf);
1342 static const char *dtmfmode2str(int mode) attribute_const;
1343 static int str2dtmfmode(const char *str) attribute_unused;
1344 static const char *insecure2str(int mode) attribute_const;
1345 static const char *allowoverlap2str(int mode) attribute_const;
1346 static void cleanup_stale_contexts(char *new, char *old);
1347 static void print_codec_to_cli(int fd, struct ast_codec_pref *pref);
1348 static const char *domain_mode_to_text(const enum domain_mode mode);
1349 static char *sip_show_domains(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
1350 static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[]);
1351 static char *sip_show_peer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
1352 static char *_sip_qualify_peer(int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[]);
1353 static char *sip_qualify_peer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
1354 static char *sip_show_registry(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
1355 static char *sip_unregister(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
1356 static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
1357 static char *sip_show_mwi(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
1358 static const char *subscription_type2str(enum subscriptiontype subtype) attribute_pure;
1359 static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype);
1360 static char *complete_sip_peer(const char *word, int state, int flags2);
1361 static char *complete_sip_registered_peer(const char *word, int state, int flags2);
1362 static char *complete_sip_show_history(const char *line, const char *word, int pos, int state);
1363 static char *complete_sip_show_peer(const char *line, const char *word, int pos, int state);
1364 static char *complete_sip_unregister(const char *line, const char *word, int pos, int state);
1365 static char *complete_sip_notify(const char *line, const char *word, int pos, int state);
1366 static char *sip_show_channel(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
1367 static char *sip_show_channelstats(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
1368 static char *sip_show_history(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
1369 static char *sip_do_debug_ip(int fd, const char *arg);
1370 static char *sip_do_debug_peer(int fd, const char *arg);
1371 static char *sip_do_debug(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
1372 static char *sip_cli_notify(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
1373 static char *sip_set_history(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
1374 static int sip_dtmfmode(struct ast_channel *chan, const char *data);
1375 static int sip_addheader(struct ast_channel *chan, const char *data);
1376 static int sip_do_reload(enum channelreloadreason reason);
1377 static char *sip_reload(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
1378 static int ast_sockaddr_resolve_first_af(struct ast_sockaddr *addr,
1379 const char *name, int flag, int family);
1380 static int ast_sockaddr_resolve_first(struct ast_sockaddr *addr,
1381 const char *name, int flag);
1382 static int ast_sockaddr_resolve_first_transport(struct ast_sockaddr *addr,
1383 const char *name, int flag, unsigned int transport);
1386 Functions for enabling debug per IP or fully, or enabling history logging for
1389 static void sip_dump_history(struct sip_pvt *dialog); /* Dump history to debuglog at end of dialog, before destroying data */
1390 static inline int sip_debug_test_addr(const struct ast_sockaddr *addr);
1391 static inline int sip_debug_test_pvt(struct sip_pvt *p);
1392 static void append_history_full(struct sip_pvt *p, const char *fmt, ...);
1393 static void sip_dump_history(struct sip_pvt *dialog);
1395 /*--- Device object handling */
1396 static struct sip_peer *build_peer(const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime, int devstate_only);
1397 static int update_call_counter(struct sip_pvt *fup, int event);
1398 static void sip_destroy_peer(struct sip_peer *peer);
1399 static void sip_destroy_peer_fn(void *peer);
1400 static void set_peer_defaults(struct sip_peer *peer);
1401 static struct sip_peer *temp_peer(const char *name);
1402 static void register_peer_exten(struct sip_peer *peer, int onoff);
1403 static int sip_poke_peer_s(const void *data);
1404 static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, struct sip_peer *p, struct sip_request *req);
1405 static void reg_source_db(struct sip_peer *peer);
1406 static void destroy_association(struct sip_peer *peer);
1407 static void set_insecure_flags(struct ast_flags *flags, const char *value, int lineno);
1408 static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask, struct ast_variable *v);
1409 static void set_socket_transport(struct sip_socket *socket, int transport);
1410 static int peer_ipcmp_cb_full(void *obj, void *arg, void *data, int flags);
1412 /* Realtime device support */
1413 static void realtime_update_peer(const char *peername, struct ast_sockaddr *addr, const char *username, const char *fullcontact, const char *useragent, int expirey, unsigned short deprecated_username, int lastms, const char *path);
1414 static void update_peer(struct sip_peer *p, int expire);
1415 static struct ast_variable *get_insecure_variable_from_config(struct ast_config *config);
1416 static const char *get_name_from_variable(const struct ast_variable *var);
1417 static struct sip_peer *realtime_peer(const char *peername, struct ast_sockaddr *sin, char *callbackexten, int devstate_only, int which_objects);
1418 static char *sip_prune_realtime(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
1420 /*--- Internal UA client handling (outbound registrations) */
1421 static void ast_sip_ouraddrfor(const struct ast_sockaddr *them, struct ast_sockaddr *us, struct sip_pvt *p);
1422 static void sip_registry_destroy(struct sip_registry *reg);
1423 static int sip_register(const char *value, int lineno);
1424 static const char *regstate2str(enum sipregistrystate regstate) attribute_const;
1425 static int sip_reregister(const void *data);
1426 static int __sip_do_register(struct sip_registry *r);
1427 static int sip_reg_timeout(const void *data);
1428 static void sip_send_all_registers(void);
1429 static int sip_reinvite_retry(const void *data);
1431 /*--- Parsing SIP requests and responses */
1432 static int determine_firstline_parts(struct sip_request *req);
1433 static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype);
1434 static const char *gettag(const struct sip_request *req, const char *header, char *tagbuf, int tagbufsize);
1435 static int find_sip_method(const char *msg);
1436 static unsigned int parse_allowed_methods(struct sip_request *req);
1437 static unsigned int set_pvt_allowed_methods(struct sip_pvt *pvt, struct sip_request *req);
1438 static int parse_request(struct sip_request *req);
1439 static const char *referstatus2str(enum referstatus rstatus) attribute_pure;
1440 static int method_match(enum sipmethod id, const char *name);
1441 static void parse_copy(struct sip_request *dst, const struct sip_request *src);
1442 static void parse_oli(struct sip_request *req, struct ast_channel *chan);
1443 static const char *find_alias(const char *name, const char *_default);
1444 static const char *__get_header(const struct sip_request *req, const char *name, int *start);
1445 static void lws2sws(struct ast_str *msgbuf);
1446 static void extract_uri(struct sip_pvt *p, struct sip_request *req);
1447 static char *remove_uri_parameters(char *uri);
1448 static int get_refer_info(struct sip_pvt *transferer, struct sip_request *outgoing_req);
1449 static int get_also_info(struct sip_pvt *p, struct sip_request *oreq);
1450 static int parse_ok_contact(struct sip_pvt *pvt, struct sip_request *req);
1451 static int set_address_from_contact(struct sip_pvt *pvt);
1452 static void check_via(struct sip_pvt *p, const struct sip_request *req);
1453 static int get_rpid(struct sip_pvt *p, struct sip_request *oreq);
1454 static int get_rdnis(struct sip_pvt *p, struct sip_request *oreq, char **name, char **number, int *reason, char **reason_str);
1455 static enum sip_get_dest_result get_destination(struct sip_pvt *p, struct sip_request *oreq, int *cc_recall_core_id);
1456 static int transmit_state_notify(struct sip_pvt *p, struct state_notify_data *data, int full, int timeout);
1457 static void update_connectedline(struct sip_pvt *p, const void *data, size_t datalen);
1458 static void update_redirecting(struct sip_pvt *p, const void *data, size_t datalen);
1459 static int get_domain(const char *str, char *domain, int len);
1460 static void get_realm(struct sip_pvt *p, const struct sip_request *req);
1461 static char *get_content(struct sip_request *req);
1463 /*-- TCP connection handling ---*/
1464 static void *_sip_tcp_helper_thread(struct ast_tcptls_session_instance *tcptls_session);
1465 static void *sip_tcp_worker_fn(void *);
1467 /*--- Constructing requests and responses */
1468 static void initialize_initreq(struct sip_pvt *p, struct sip_request *req);
1469 static int init_req(struct sip_request *req, int sipmethod, const char *recip);
1470 static void deinit_req(struct sip_request *req);
1471 static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, uint32_t seqno, int newbranch);
1472 static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, const char * const explicit_uri);
1473 static int init_resp(struct sip_request *resp, const char *msg);
1474 static inline int resp_needs_contact(const char *msg, enum sipmethod method);
1475 static int respprep(struct sip_request *resp, struct sip_pvt *p, const char *msg, const struct sip_request *req);
1476 static const struct ast_sockaddr *sip_real_dst(const struct sip_pvt *p);
1477 static void build_via(struct sip_pvt *p);
1478 static int create_addr_from_peer(struct sip_pvt *r, struct sip_peer *peer);
1479 static int create_addr(struct sip_pvt *dialog, const char *opeer, struct ast_sockaddr *addr, int newdialog);
1480 static char *generate_random_string(char *buf, size_t size);
1481 static void build_callid_pvt(struct sip_pvt *pvt);
1482 static void change_callid_pvt(struct sip_pvt *pvt, const char *callid);
1483 static void build_callid_registry(struct sip_registry *reg, const struct ast_sockaddr *ourip, const char *fromdomain);
1484 static void build_localtag_registry(struct sip_registry *reg);
1485 static void make_our_tag(struct sip_pvt *pvt);
1486 static int add_header(struct sip_request *req, const char *var, const char *value);
1487 static int add_max_forwards(struct sip_pvt *dialog, struct sip_request *req);
1488 static int add_content(struct sip_request *req, const char *line);
1489 static int finalize_content(struct sip_request *req);
1490 static void destroy_msg_headers(struct sip_pvt *pvt);
1491 static int add_text(struct sip_request *req, struct sip_pvt *p);
1492 static int add_digit(struct sip_request *req, char digit, unsigned int duration, int mode);
1493 static int add_rpid(struct sip_request *req, struct sip_pvt *p);
1494 static int add_vidupdate(struct sip_request *req);
1495 static void add_route(struct sip_request *req, struct sip_route *route, int skip);
1496 static int copy_header(struct sip_request *req, const struct sip_request *orig, const char *field);
1497 static int copy_all_header(struct sip_request *req, const struct sip_request *orig, const char *field);
1498 static int copy_via_headers(struct sip_pvt *p, struct sip_request *req, const struct sip_request *orig, const char *field);
1499 static void set_destination(struct sip_pvt *p, const char *uri);
1500 static void add_date(struct sip_request *req);
1501 static void add_expires(struct sip_request *req, int expires);
1502 static void build_contact(struct sip_pvt *p);
1504 /*------Request handling functions */
1505 static int handle_incoming(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, int *recount, int *nounlock);
1506 static int handle_request_update(struct sip_pvt *p, struct sip_request *req);
1507 static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, uint32_t seqno, int *recount, const char *e, int *nounlock);
1508 static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, uint32_t seqno, int *nounlock);
1509 static int handle_request_bye(struct sip_pvt *p, struct sip_request *req);
1510 static int handle_request_register(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *sin, const char *e);
1511 static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req);
1512 static int handle_request_message(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const char *e);
1513 static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, uint32_t seqno, const char *e);
1514 static void handle_request_info(struct sip_pvt *p, struct sip_request *req);
1515 static int handle_request_options(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const char *e);
1516 static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req,
1517 int *nounlock, struct sip_pvt *replaces_pvt, struct ast_channel *replaces_chan);
1518 static int handle_request_notify(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, uint32_t seqno, const char *e);
1519 static int local_attended_transfer(struct sip_pvt *transferer, struct ast_channel *transferer_chan, uint32_t seqno, int *nounlock);
1521 /*------Response handling functions */
1522 static void handle_response_publish(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno);
1523 static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno);
1524 static void handle_response_notify(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno);
1525 static void handle_response_refer(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno);
1526 static void handle_response_subscribe(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno);
1527 static int handle_response_register(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno);
1528 static void handle_response(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno);
1530 /*------ SRTP Support -------- */
1531 static int process_crypto(struct sip_pvt *p, struct ast_rtp_instance *rtp, struct ast_sdp_srtp **srtp, const char *a);
1533 /*------ T38 Support --------- */
1534 static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans);
1535 static void change_t38_state(struct sip_pvt *p, int state);
1537 /*------ Session-Timers functions --------- */
1538 static void proc_422_rsp(struct sip_pvt *p, struct sip_request *rsp);
1539 static int proc_session_timer(const void *vp);
1540 static void stop_session_timer(struct sip_pvt *p);
1541 static void start_session_timer(struct sip_pvt *p);
1542 static void restart_session_timer(struct sip_pvt *p);
1543 static const char *strefresherparam2str(enum st_refresher r);
1544 static int parse_session_expires(const char *p_hdrval, int *const p_interval, enum st_refresher_param *const p_ref);
1545 static int parse_minse(const char *p_hdrval, int *const p_interval);
1546 static int st_get_se(struct sip_pvt *, int max);
1547 static enum st_refresher st_get_refresher(struct sip_pvt *);
1548 static enum st_mode st_get_mode(struct sip_pvt *, int no_cached);
1549 static struct sip_st_dlg* sip_st_alloc(struct sip_pvt *const p);
1551 /*------- RTP Glue functions -------- */
1552 static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_rtp_instance *vinstance, struct ast_rtp_instance *tinstance, const struct ast_format_cap *cap, int nat_active);
1554 /*!--- SIP MWI Subscription support */
1555 static int sip_subscribe_mwi(const char *value, int lineno);
1556 static void sip_subscribe_mwi_destroy(struct sip_subscription_mwi *mwi);
1557 static void sip_send_all_mwi_subscriptions(void);
1558 static int sip_subscribe_mwi_do(const void *data);
1559 static int __sip_subscribe_mwi_do(struct sip_subscription_mwi *mwi);
1561 /*! \brief Definition of this channel for PBX channel registration */
1562 struct ast_channel_tech sip_tech = {
1564 .description = "Session Initiation Protocol (SIP)",
1565 .properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,
1566 .requester = sip_request_call, /* called with chan unlocked */
1567 .devicestate = sip_devicestate, /* called with chan unlocked (not chan-specific) */
1568 .call = sip_call, /* called with chan locked */
1569 .send_html = sip_sendhtml,
1570 .hangup = sip_hangup, /* called with chan locked */
1571 .answer = sip_answer, /* called with chan locked */
1572 .read = sip_read, /* called with chan locked */
1573 .write = sip_write, /* called with chan locked */
1574 .write_video = sip_write, /* called with chan locked */
1575 .write_text = sip_write,
1576 .indicate = sip_indicate, /* called with chan locked */
1577 .transfer = sip_transfer, /* called with chan locked */
1578 .fixup = sip_fixup, /* called with chan locked */
1579 .send_digit_begin = sip_senddigit_begin, /* called with chan unlocked */
1580 .send_digit_end = sip_senddigit_end,
1581 .early_bridge = ast_rtp_instance_early_bridge,
1582 .send_text = sip_sendtext, /* called with chan locked */
1583 .func_channel_read = sip_acf_channel_read,
1584 .setoption = sip_setoption,
1585 .queryoption = sip_queryoption,
1586 .get_pvt_uniqueid = sip_get_callid,
1589 /*! \brief This version of the sip channel tech has no send_digit_begin
1590 * callback so that the core knows that the channel does not want
1591 * DTMF BEGIN frames.
1592 * The struct is initialized just before registering the channel driver,
1593 * and is for use with channels using SIP INFO DTMF.
1595 struct ast_channel_tech sip_tech_info;
1597 /*------- CC Support -------- */
1598 static int sip_cc_agent_init(struct ast_cc_agent *agent, struct ast_channel *chan);
1599 static int sip_cc_agent_start_offer_timer(struct ast_cc_agent *agent);
1600 static int sip_cc_agent_stop_offer_timer(struct ast_cc_agent *agent);
1601 static void sip_cc_agent_respond(struct ast_cc_agent *agent, enum ast_cc_agent_response_reason reason);
1602 static int sip_cc_agent_status_request(struct ast_cc_agent *agent);
1603 static int sip_cc_agent_start_monitoring(struct ast_cc_agent *agent);
1604 static int sip_cc_agent_recall(struct ast_cc_agent *agent);
1605 static void sip_cc_agent_destructor(struct ast_cc_agent *agent);
1607 static struct ast_cc_agent_callbacks sip_cc_agent_callbacks = {
1609 .init = sip_cc_agent_init,
1610 .start_offer_timer = sip_cc_agent_start_offer_timer,
1611 .stop_offer_timer = sip_cc_agent_stop_offer_timer,
1612 .respond = sip_cc_agent_respond,
1613 .status_request = sip_cc_agent_status_request,
1614 .start_monitoring = sip_cc_agent_start_monitoring,
1615 .callee_available = sip_cc_agent_recall,
1616 .destructor = sip_cc_agent_destructor,
1619 /* -------- End of declarations of structures, constants and forward declarations of functions
1620 Below starts actual code
1621 ------------------------
1624 static int sip_epa_register(const struct epa_static_data *static_data)
1626 struct epa_backend *backend = ast_calloc(1, sizeof(*backend));
1632 backend->static_data = static_data;
1634 AST_LIST_LOCK(&epa_static_data_list);
1635 AST_LIST_INSERT_TAIL(&epa_static_data_list, backend, next);
1636 AST_LIST_UNLOCK(&epa_static_data_list);
1640 static void sip_epa_unregister_all(void)
1642 struct epa_backend *backend;
1644 AST_LIST_LOCK(&epa_static_data_list);
1645 while ((backend = AST_LIST_REMOVE_HEAD(&epa_static_data_list, next))) {
1648 AST_LIST_UNLOCK(&epa_static_data_list);
1651 static void cc_handle_publish_error(struct sip_pvt *pvt, const int resp, struct sip_request *req, struct sip_epa_entry *epa_entry);
1653 static void cc_epa_destructor(void *data)
1655 struct sip_epa_entry *epa_entry = data;
1656 struct cc_epa_entry *cc_entry = epa_entry->instance_data;
1660 static const struct epa_static_data cc_epa_static_data = {
1661 .event = CALL_COMPLETION,
1662 .name = "call-completion",
1663 .handle_error = cc_handle_publish_error,
1664 .destructor = cc_epa_destructor,
1667 static const struct epa_static_data *find_static_data(const char * const event_package)
1669 const struct epa_backend *backend = NULL;
1671 AST_LIST_LOCK(&epa_static_data_list);
1672 AST_LIST_TRAVERSE(&epa_static_data_list, backend, next) {
1673 if (!strcmp(backend->static_data->name, event_package)) {
1677 AST_LIST_UNLOCK(&epa_static_data_list);
1678 return backend ? backend->static_data : NULL;
1681 static struct sip_epa_entry *create_epa_entry (const char * const event_package, const char * const destination)
1683 struct sip_epa_entry *epa_entry;
1684 const struct epa_static_data *static_data;
1686 if (!(static_data = find_static_data(event_package))) {
1690 if (!(epa_entry = ao2_t_alloc(sizeof(*epa_entry), static_data->destructor, "Allocate new EPA entry"))) {
1694 epa_entry->static_data = static_data;
1695 ast_copy_string(epa_entry->destination, destination, sizeof(epa_entry->destination));
1698 static enum ast_cc_service_type service_string_to_service_type(const char * const service_string)
1700 enum ast_cc_service_type service;
1701 for (service = AST_CC_CCBS; service <= AST_CC_CCNL; ++service) {
1702 if (!strcasecmp(service_string, sip_cc_service_map[service].service_string)) {
1709 /* Even state compositors code */
1710 static void esc_entry_destructor(void *obj)
1712 struct sip_esc_entry *esc_entry = obj;
1713 if (esc_entry->sched_id > -1) {
1714 AST_SCHED_DEL(sched, esc_entry->sched_id);
1718 static int esc_hash_fn(const void *obj, const int flags)
1720 const struct sip_esc_entry *entry = obj;
1721 return ast_str_hash(entry->entity_tag);
1724 static int esc_cmp_fn(void *obj, void *arg, int flags)
1726 struct sip_esc_entry *entry1 = obj;
1727 struct sip_esc_entry *entry2 = arg;
1729 return (!strcmp(entry1->entity_tag, entry2->entity_tag)) ? (CMP_MATCH | CMP_STOP) : 0;
1732 static struct event_state_compositor *get_esc(const char * const event_package) {
1734 for (i = 0; i < ARRAY_LEN(event_state_compositors); i++) {
1735 if (!strcasecmp(event_package, event_state_compositors[i].name)) {
1736 return &event_state_compositors[i];
1742 static struct sip_esc_entry *get_esc_entry(const char * entity_tag, struct event_state_compositor *esc) {
1743 struct sip_esc_entry *entry;
1744 struct sip_esc_entry finder;
1746 ast_copy_string(finder.entity_tag, entity_tag, sizeof(finder.entity_tag));
1748 entry = ao2_find(esc->compositor, &finder, OBJ_POINTER);
1753 static int publish_expire(const void *data)
1755 struct sip_esc_entry *esc_entry = (struct sip_esc_entry *) data;
1756 struct event_state_compositor *esc = get_esc(esc_entry->event);
1758 ast_assert(esc != NULL);
1760 ao2_unlink(esc->compositor, esc_entry);
1761 ao2_ref(esc_entry, -1);
1765 static void create_new_sip_etag(struct sip_esc_entry *esc_entry, int is_linked)
1767 int new_etag = ast_atomic_fetchadd_int(&esc_etag_counter, +1);
1768 struct event_state_compositor *esc = get_esc(esc_entry->event);
1770 ast_assert(esc != NULL);
1772 ao2_unlink(esc->compositor, esc_entry);
1774 snprintf(esc_entry->entity_tag, sizeof(esc_entry->entity_tag), "%d", new_etag);
1775 ao2_link(esc->compositor, esc_entry);
1778 static struct sip_esc_entry *create_esc_entry(struct event_state_compositor *esc, struct sip_request *req, const int expires)
1780 struct sip_esc_entry *esc_entry;
1783 if (!(esc_entry = ao2_alloc(sizeof(*esc_entry), esc_entry_destructor))) {
1787 esc_entry->event = esc->name;
1789 expires_ms = expires * 1000;
1790 /* Bump refcount for scheduler */
1791 ao2_ref(esc_entry, +1);
1792 esc_entry->sched_id = ast_sched_add(sched, expires_ms, publish_expire, esc_entry);
1794 /* Note: This links the esc_entry into the ESC properly */
1795 create_new_sip_etag(esc_entry, 0);
1800 static int initialize_escs(void)
1803 for (i = 0; i < ARRAY_LEN(event_state_compositors); i++) {
1804 if (!((event_state_compositors[i].compositor) =
1805 ao2_container_alloc(ESC_MAX_BUCKETS, esc_hash_fn, esc_cmp_fn))) {
1812 static void destroy_escs(void)
1815 for (i = 0; i < ARRAY_LEN(event_state_compositors); i++) {
1816 ao2_ref(event_state_compositors[i].compositor, -1);
1821 static int find_by_notify_uri_helper(void *obj, void *arg, int flags)
1823 struct ast_cc_agent *agent = obj;
1824 struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
1825 const char *uri = arg;
1827 return !sip_uri_cmp(agent_pvt->notify_uri, uri) ? CMP_MATCH | CMP_STOP : 0;
1830 static struct ast_cc_agent *find_sip_cc_agent_by_notify_uri(const char * const uri)
1832 struct ast_cc_agent *agent = ast_cc_agent_callback(0, find_by_notify_uri_helper, (char *)uri, "SIP");
1836 static int find_by_subscribe_uri_helper(void *obj, void *arg, int flags)
1838 struct ast_cc_agent *agent = obj;
1839 struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
1840 const char *uri = arg;
1842 return !sip_uri_cmp(agent_pvt->subscribe_uri, uri) ? CMP_MATCH | CMP_STOP : 0;
1845 static struct ast_cc_agent *find_sip_cc_agent_by_subscribe_uri(const char * const uri)
1847 struct ast_cc_agent *agent = ast_cc_agent_callback(0, find_by_subscribe_uri_helper, (char *)uri, "SIP");
1851 static int find_by_callid_helper(void *obj, void *arg, int flags)
1853 struct ast_cc_agent *agent = obj;
1854 struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
1855 struct sip_pvt *call_pvt = arg;
1857 return !strcmp(agent_pvt->original_callid, call_pvt->callid) ? CMP_MATCH | CMP_STOP : 0;
1860 static struct ast_cc_agent *find_sip_cc_agent_by_original_callid(struct sip_pvt *pvt)
1862 struct ast_cc_agent *agent = ast_cc_agent_callback(0, find_by_callid_helper, pvt, "SIP");
1866 static int sip_cc_agent_init(struct ast_cc_agent *agent, struct ast_channel *chan)
1868 struct sip_cc_agent_pvt *agent_pvt = ast_calloc(1, sizeof(*agent_pvt));
1869 struct sip_pvt *call_pvt = ast_channel_tech_pvt(chan);
1875 ast_assert(!strcmp(ast_channel_tech(chan)->type, "SIP"));
1877 ast_copy_string(agent_pvt->original_callid, call_pvt->callid, sizeof(agent_pvt->original_callid));
1878 ast_copy_string(agent_pvt->original_exten, call_pvt->exten, sizeof(agent_pvt->original_exten));
1879 agent_pvt->offer_timer_id = -1;
1880 agent->private_data = agent_pvt;
1881 sip_pvt_lock(call_pvt);
1882 ast_set_flag(&call_pvt->flags[0], SIP_OFFER_CC);
1883 sip_pvt_unlock(call_pvt);
1887 static int sip_offer_timer_expire(const void *data)
1889 struct ast_cc_agent *agent = (struct ast_cc_agent *) data;
1890 struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
1892 agent_pvt->offer_timer_id = -1;
1894 return ast_cc_failed(agent->core_id, "SIP agent %s's offer timer expired", agent->device_name);
1897 static int sip_cc_agent_start_offer_timer(struct ast_cc_agent *agent)
1899 struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
1902 when = ast_get_cc_offer_timer(agent->cc_params) * 1000;
1903 agent_pvt->offer_timer_id = ast_sched_add(sched, when, sip_offer_timer_expire, agent);
1907 static int sip_cc_agent_stop_offer_timer(struct ast_cc_agent *agent)
1909 struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
1911 AST_SCHED_DEL(sched, agent_pvt->offer_timer_id);
1915 static void sip_cc_agent_respond(struct ast_cc_agent *agent, enum ast_cc_agent_response_reason reason)
1917 struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
1919 sip_pvt_lock(agent_pvt->subscribe_pvt);
1920 ast_set_flag(&agent_pvt->subscribe_pvt->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
1921 if (reason == AST_CC_AGENT_RESPONSE_SUCCESS || !ast_strlen_zero(agent_pvt->notify_uri)) {
1922 /* The second half of this if statement may be a bit hard to grasp,
1923 * so here's an explanation. When a subscription comes into
1924 * chan_sip, as long as it is not malformed, it will be passed
1925 * to the CC core. If the core senses an out-of-order state transition,
1926 * then the core will call this callback with the "reason" set to a
1927 * failure condition.
1928 * However, an out-of-order state transition will occur during a resubscription
1929 * for CC. In such a case, we can see that we have already generated a notify_uri
1930 * and so we can detect that this isn't a *real* failure. Rather, it is just
1931 * something the core doesn't recognize as a legitimate SIP state transition.
1932 * Thus we respond with happiness and flowers.
1934 transmit_response(agent_pvt->subscribe_pvt, "200 OK", &agent_pvt->subscribe_pvt->initreq);
1935 transmit_cc_notify(agent, agent_pvt->subscribe_pvt, CC_QUEUED);
1937 transmit_response(agent_pvt->subscribe_pvt, "500 Internal Error", &agent_pvt->subscribe_pvt->initreq);
1939 sip_pvt_unlock(agent_pvt->subscribe_pvt);
1940 agent_pvt->is_available = TRUE;
1943 static int sip_cc_agent_status_request(struct ast_cc_agent *agent)
1945 struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
1946 enum ast_device_state state = agent_pvt->is_available ? AST_DEVICE_NOT_INUSE : AST_DEVICE_INUSE;
1947 return ast_cc_agent_status_response(agent->core_id, state);
1950 static int sip_cc_agent_start_monitoring(struct ast_cc_agent *agent)
1952 /* To start monitoring just means to wait for an incoming PUBLISH
1953 * to tell us that the caller has become available again. No special
1959 static int sip_cc_agent_recall(struct ast_cc_agent *agent)
1961 struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
1962 /* If we have received a PUBLISH beforehand stating that the caller in question
1963 * is not available, we can save ourself a bit of effort here and just report
1964 * the caller as busy
1966 if (!agent_pvt->is_available) {
1967 return ast_cc_agent_caller_busy(agent->core_id, "Caller %s is busy, reporting to the core",
1968 agent->device_name);
1970 /* Otherwise, we transmit a NOTIFY to the caller and await either
1971 * a PUBLISH or an INVITE
1973 sip_pvt_lock(agent_pvt->subscribe_pvt);
1974 transmit_cc_notify(agent, agent_pvt->subscribe_pvt, CC_READY);
1975 sip_pvt_unlock(agent_pvt->subscribe_pvt);
1979 static void sip_cc_agent_destructor(struct ast_cc_agent *agent)
1981 struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
1984 /* The agent constructor probably failed. */
1988 sip_cc_agent_stop_offer_timer(agent);
1989 if (agent_pvt->subscribe_pvt) {
1990 sip_pvt_lock(agent_pvt->subscribe_pvt);
1991 if (!ast_test_flag(&agent_pvt->subscribe_pvt->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED)) {
1992 /* If we haven't sent a 200 OK for the SUBSCRIBE dialog yet, then we need to send a response letting
1993 * the subscriber know something went wrong
1995 transmit_response(agent_pvt->subscribe_pvt, "500 Internal Server Error", &agent_pvt->subscribe_pvt->initreq);
1997 sip_pvt_unlock(agent_pvt->subscribe_pvt);
1998 agent_pvt->subscribe_pvt = dialog_unref(agent_pvt->subscribe_pvt, "SIP CC agent destructor: Remove ref to subscription");
2000 ast_free(agent_pvt);
2004 static int sip_monitor_instance_hash_fn(const void *obj, const int flags)
2006 const struct sip_monitor_instance *monitor_instance = obj;
2007 return monitor_instance->core_id;
2010 static int sip_monitor_instance_cmp_fn(void *obj, void *arg, int flags)
2012 struct sip_monitor_instance *monitor_instance1 = obj;
2013 struct sip_monitor_instance *monitor_instance2 = arg;
2015 return monitor_instance1->core_id == monitor_instance2->core_id ? CMP_MATCH | CMP_STOP : 0;
2018 static void sip_monitor_instance_destructor(void *data)
2020 struct sip_monitor_instance *monitor_instance = data;
2021 if (monitor_instance->subscription_pvt) {
2022 sip_pvt_lock(monitor_instance->subscription_pvt);
2023 monitor_instance->subscription_pvt->expiry = 0;
2024 transmit_invite(monitor_instance->subscription_pvt, SIP_SUBSCRIBE, FALSE, 0, monitor_instance->subscribe_uri);
2025 sip_pvt_unlock(monitor_instance->subscription_pvt);
2026 dialog_unref(monitor_instance->subscription_pvt, "Unref monitor instance ref of subscription pvt");
2028 if (monitor_instance->suspension_entry) {
2029 monitor_instance->suspension_entry->body[0] = '\0';
2030 transmit_publish(monitor_instance->suspension_entry, SIP_PUBLISH_REMOVE ,monitor_instance->notify_uri);
2031 ao2_t_ref(monitor_instance->suspension_entry, -1, "Decrementing suspension entry refcount in sip_monitor_instance_destructor");
2033 ast_string_field_free_memory(monitor_instance);
2036 static struct sip_monitor_instance *sip_monitor_instance_init(int core_id, const char * const subscribe_uri, const char * const peername, const char * const device_name)
2038 struct sip_monitor_instance *monitor_instance = ao2_alloc(sizeof(*monitor_instance), sip_monitor_instance_destructor);
2040 if (!monitor_instance) {
2044 if (ast_string_field_init(monitor_instance, 256)) {
2045 ao2_ref(monitor_instance, -1);
2049 ast_string_field_set(monitor_instance, subscribe_uri, subscribe_uri);
2050 ast_string_field_set(monitor_instance, peername, peername);
2051 ast_string_field_set(monitor_instance, device_name, device_name);
2052 monitor_instance->core_id = core_id;
2053 ao2_link(sip_monitor_instances, monitor_instance);
2054 return monitor_instance;
2057 static int find_sip_monitor_instance_by_subscription_pvt(void *obj, void *arg, int flags)
2059 struct sip_monitor_instance *monitor_instance = obj;
2060 return monitor_instance->subscription_pvt == arg ? CMP_MATCH | CMP_STOP : 0;
2063 static int find_sip_monitor_instance_by_suspension_entry(void *obj, void *arg, int flags)
2065 struct sip_monitor_instance *monitor_instance = obj;
2066 return monitor_instance->suspension_entry == arg ? CMP_MATCH | CMP_STOP : 0;
2069 static int sip_cc_monitor_request_cc(struct ast_cc_monitor *monitor, int *available_timer_id);
2070 static int sip_cc_monitor_suspend(struct ast_cc_monitor *monitor);
2071 static int sip_cc_monitor_unsuspend(struct ast_cc_monitor *monitor);
2072 static int sip_cc_monitor_cancel_available_timer(struct ast_cc_monitor *monitor, int *sched_id);
2073 static void sip_cc_monitor_destructor(void *private_data);
2075 static struct ast_cc_monitor_callbacks sip_cc_monitor_callbacks = {
2077 .request_cc = sip_cc_monitor_request_cc,
2078 .suspend = sip_cc_monitor_suspend,
2079 .unsuspend = sip_cc_monitor_unsuspend,
2080 .cancel_available_timer = sip_cc_monitor_cancel_available_timer,
2081 .destructor = sip_cc_monitor_destructor,
2084 static int sip_cc_monitor_request_cc(struct ast_cc_monitor *monitor, int *available_timer_id)
2086 struct sip_monitor_instance *monitor_instance = monitor->private_data;
2087 enum ast_cc_service_type service = monitor->service_offered;
2090 if (!monitor_instance) {
2094 if (!(monitor_instance->subscription_pvt = sip_alloc(NULL, NULL, 0, SIP_SUBSCRIBE, NULL, NULL))) {
2098 when = service == AST_CC_CCBS ? ast_get_ccbs_available_timer(monitor->interface->config_params) :
2099 ast_get_ccnr_available_timer(monitor->interface->config_params);
2101 sip_pvt_lock(monitor_instance->subscription_pvt);
2102 ast_set_flag(&monitor_instance->subscription_pvt->flags[0], SIP_OUTGOING);
2103 create_addr(monitor_instance->subscription_pvt, monitor_instance->peername, 0, 1);
2104 ast_sip_ouraddrfor(&monitor_instance->subscription_pvt->sa, &monitor_instance->subscription_pvt->ourip, monitor_instance->subscription_pvt);
2105 monitor_instance->subscription_pvt->subscribed = CALL_COMPLETION;
2106 monitor_instance->subscription_pvt->expiry = when;
2108 transmit_invite(monitor_instance->subscription_pvt, SIP_SUBSCRIBE, FALSE, 2, monitor_instance->subscribe_uri);
2109 sip_pvt_unlock(monitor_instance->subscription_pvt);
2111 ao2_t_ref(monitor, +1, "Adding a ref to the monitor for the scheduler");
2112 *available_timer_id = ast_sched_add(sched, when * 1000, ast_cc_available_timer_expire, monitor);
2116 static int construct_pidf_body(enum sip_cc_publish_state state, char *pidf_body, size_t size, const char *presentity)
2118 struct ast_str *body = ast_str_alloca(size);
2121 generate_random_string(tuple_id, sizeof(tuple_id));
2123 /* We'll make this a bare-bones pidf body. In state_notify_build_xml, the PIDF
2124 * body gets a lot more extra junk that isn't necessary, so we'll leave it out here.
2126 ast_str_append(&body, 0, "<?xml version=\"1.0\" encoding=\"UTF-8\"?>\n");
2127 /* XXX The entity attribute is currently set to the peer name associated with the
2128 * dialog. This is because we currently only call this function for call-completion
2129 * PUBLISH bodies. In such cases, the entity is completely disregarded. For other
2130 * event packages, it may be crucial to have a proper URI as the presentity so this
2131 * should be revisited as support is expanded.
2133 ast_str_append(&body, 0, "<presence xmlns=\"urn:ietf:params:xml:ns:pidf\" entity=\"%s\">\n", presentity);
2134 ast_str_append(&body, 0, "<tuple id=\"%s\">\n", tuple_id);
2135 ast_str_append(&body, 0, "<status><basic>%s</basic></status>\n", state == CC_OPEN ? "open" : "closed");
2136 ast_str_append(&body, 0, "</tuple>\n");
2137 ast_str_append(&body, 0, "</presence>\n");
2138 ast_copy_string(pidf_body, ast_str_buffer(body), size);
2142 static int sip_cc_monitor_suspend(struct ast_cc_monitor *monitor)
2144 struct sip_monitor_instance *monitor_instance = monitor->private_data;
2145 enum sip_publish_type publish_type;
2146 struct cc_epa_entry *cc_entry;
2148 if (!monitor_instance) {
2152 if (!monitor_instance->suspension_entry) {
2153 /* We haven't yet allocated the suspension entry, so let's give it a shot */
2154 if (!(monitor_instance->suspension_entry = create_epa_entry("call-completion", monitor_instance->peername))) {
2155 ast_log(LOG_WARNING, "Unable to allocate sip EPA entry for call-completion\n");
2156 ao2_ref(monitor_instance, -1);
2159 if (!(cc_entry = ast_calloc(1, sizeof(*cc_entry)))) {
2160 ast_log(LOG_WARNING, "Unable to allocate space for instance data of EPA entry for call-completion\n");
2161 ao2_ref(monitor_instance, -1);
2164 cc_entry->core_id = monitor->core_id;
2165 monitor_instance->suspension_entry->instance_data = cc_entry;
2166 publish_type = SIP_PUBLISH_INITIAL;
2168 publish_type = SIP_PUBLISH_MODIFY;
2169 cc_entry = monitor_instance->suspension_entry->instance_data;
2172 cc_entry->current_state = CC_CLOSED;
2174 if (ast_strlen_zero(monitor_instance->notify_uri)) {
2175 /* If we have no set notify_uri, then what this means is that we have
2176 * not received a NOTIFY from this destination stating that he is
2177 * currently available.
2179 * This situation can arise when the core calls the suspend callbacks
2180 * of multiple destinations. If one of the other destinations aside
2181 * from this one notified Asterisk that he is available, then there
2182 * is no reason to take any suspension action on this device. Rather,
2183 * we should return now and if we receive a NOTIFY while monitoring
2184 * is still "suspended" then we can immediately respond with the
2185 * proper PUBLISH to let this endpoint know what is going on.
2189 construct_pidf_body(CC_CLOSED, monitor_instance->suspension_entry->body, sizeof(monitor_instance->suspension_entry->body), monitor_instance->peername);
2190 return transmit_publish(monitor_instance->suspension_entry, publish_type, monitor_instance->notify_uri);
2193 static int sip_cc_monitor_unsuspend(struct ast_cc_monitor *monitor)
2195 struct sip_monitor_instance *monitor_instance = monitor->private_data;
2196 struct cc_epa_entry *cc_entry;
2198 if (!monitor_instance) {
2202 ast_assert(monitor_instance->suspension_entry != NULL);
2204 cc_entry = monitor_instance->suspension_entry->instance_data;
2205 cc_entry->current_state = CC_OPEN;
2206 if (ast_strlen_zero(monitor_instance->notify_uri)) {
2207 /* This means we are being asked to unsuspend a call leg we never
2208 * sent a PUBLISH on. As such, there is no reason to send another
2209 * PUBLISH at this point either. We can just return instead.
2213 construct_pidf_body(CC_OPEN, monitor_instance->suspension_entry->body, sizeof(monitor_instance->suspension_entry->body), monitor_instance->peername);
2214 return transmit_publish(monitor_instance->suspension_entry, SIP_PUBLISH_MODIFY, monitor_instance->notify_uri);
2217 static int sip_cc_monitor_cancel_available_timer(struct ast_cc_monitor *monitor, int *sched_id)
2219 if (*sched_id != -1) {
2220 AST_SCHED_DEL(sched, *sched_id);
2221 ao2_t_ref(monitor, -1, "Removing scheduler's reference to the monitor");
2226 static void sip_cc_monitor_destructor(void *private_data)
2228 struct sip_monitor_instance *monitor_instance = private_data;
2229 ao2_unlink(sip_monitor_instances, monitor_instance);
2230 ast_module_unref(ast_module_info->self);
2233 static int sip_get_cc_information(struct sip_request *req, char *subscribe_uri, size_t size, enum ast_cc_service_type *service)
2235 char *call_info = ast_strdupa(sip_get_header(req, "Call-Info"));
2239 static const char cc_purpose[] = "purpose=call-completion";
2240 static const int cc_purpose_len = sizeof(cc_purpose) - 1;
2242 if (ast_strlen_zero(call_info)) {
2243 /* No Call-Info present. Definitely no CC offer */
2247 uri = strsep(&call_info, ";");
2249 while ((purpose = strsep(&call_info, ";"))) {
2250 if (!strncmp(purpose, cc_purpose, cc_purpose_len)) {
2255 /* We didn't find the appropriate purpose= parameter. Oh well */
2259 /* Okay, call-completion has been offered. Let's figure out what type of service this is */
2260 while ((service_str = strsep(&call_info, ";"))) {
2261 if (!strncmp(service_str, "m=", 2)) {
2266 /* So they didn't offer a particular service, We'll just go with CCBS since it really
2267 * doesn't matter anyway
2271 /* We already determined that there is an "m=" so no need to check
2272 * the result of this strsep
2274 strsep(&service_str, "=");
2277 if ((*service = service_string_to_service_type(service_str)) == AST_CC_NONE) {
2278 /* Invalid service offered */
2282 ast_copy_string(subscribe_uri, get_in_brackets(uri), size);
2288 * \brief Determine what, if any, CC has been offered and queue a CC frame if possible
2290 * After taking care of some formalities to be sure that this call is eligible for CC,
2291 * we first try to see if we can make use of native CC. We grab the information from
2292 * the passed-in sip_request (which is always a response to an INVITE). If we can
2293 * use native CC monitoring for the call, then so be it.
2295 * If native cc monitoring is not possible or not supported, then we will instead attempt
2296 * to use generic monitoring. Falling back to generic from a failed attempt at using native
2297 * monitoring will only work if the monitor policy of the endpoint is "always"
2299 * \param pvt The current dialog. Contains CC parameters for the endpoint
2300 * \param req The response to the INVITE we want to inspect
2301 * \param service The service to use if generic monitoring is to be used. For native
2302 * monitoring, we get the service from the SIP response itself
2304 static void sip_handle_cc(struct sip_pvt *pvt, struct sip_request *req, enum ast_cc_service_type service)
2306 enum ast_cc_monitor_policies monitor_policy = ast_get_cc_monitor_policy(pvt->cc_params);
2308 char interface_name[AST_CHANNEL_NAME];
2310 if (monitor_policy == AST_CC_MONITOR_NEVER) {
2311 /* Don't bother, just return */
2315 if ((core_id = ast_cc_get_current_core_id(pvt->owner)) == -1) {
2316 /* For some reason, CC is invalid, so don't try it! */
2320 ast_channel_get_device_name(pvt->owner, interface_name, sizeof(interface_name));
2322 if (monitor_policy == AST_CC_MONITOR_ALWAYS || monitor_policy == AST_CC_MONITOR_NATIVE) {
2323 char subscribe_uri[SIPBUFSIZE];
2324 char device_name[AST_CHANNEL_NAME];
2325 enum ast_cc_service_type offered_service;
2326 struct sip_monitor_instance *monitor_instance;
2327 if (sip_get_cc_information(req, subscribe_uri, sizeof(subscribe_uri), &offered_service)) {
2328 /* If CC isn't being offered to us, or for some reason the CC offer is
2329 * not formatted correctly, then it may still be possible to use generic
2330 * call completion since the monitor policy may be "always"
2334 ast_channel_get_device_name(pvt->owner, device_name, sizeof(device_name));
2335 if (!(monitor_instance = sip_monitor_instance_init(core_id, subscribe_uri, pvt->peername, device_name))) {
2336 /* Same deal. We can try using generic still */
2339 /* We bump the refcount of chan_sip because once we queue this frame, the CC core
2340 * will have a reference to callbacks in this module. We decrement the module
2341 * refcount once the monitor destructor is called
2343 ast_module_ref(ast_module_info->self);
2344 ast_queue_cc_frame(pvt->owner, "SIP", pvt->dialstring, offered_service, monitor_instance);
2345 ao2_ref(monitor_instance, -1);
2350 if (monitor_policy == AST_CC_MONITOR_GENERIC || monitor_policy == AST_CC_MONITOR_ALWAYS) {
2351 ast_queue_cc_frame(pvt->owner, AST_CC_GENERIC_MONITOR_TYPE, interface_name, service, NULL);
2355 /*! \brief Working TLS connection configuration */
2356 static struct ast_tls_config sip_tls_cfg;
2358 /*! \brief Default TLS connection configuration */
2359 static struct ast_tls_config default_tls_cfg;
2361 /*! \brief The TCP server definition */
2362 static struct ast_tcptls_session_args sip_tcp_desc = {
2364 .master = AST_PTHREADT_NULL,
2367 .name = "SIP TCP server",
2368 .accept_fn = ast_tcptls_server_root,
2369 .worker_fn = sip_tcp_worker_fn,
2372 /*! \brief The TCP/TLS server definition */
2373 static struct ast_tcptls_session_args sip_tls_desc = {
2375 .master = AST_PTHREADT_NULL,
2376 .tls_cfg = &sip_tls_cfg,
2378 .name = "SIP TLS server",
2379 .accept_fn = ast_tcptls_server_root,
2380 .worker_fn = sip_tcp_worker_fn,
2383 /*! \brief Append to SIP dialog history
2384 \return Always returns 0 */
2385 #define append_history(p, event, fmt , args... ) append_history_full(p, "%-15s " fmt, event, ## args)
2387 struct sip_pvt *dialog_ref_debug(struct sip_pvt *p, const char *tag, char *file, int line, const char *func)
2391 __ao2_ref_debug(p, 1, tag, file, line, func);
2396 ast_log(LOG_ERROR, "Attempt to Ref a null pointer\n");
2400 struct sip_pvt *dialog_unref_debug(struct sip_pvt *p, const char *tag, char *file, int line, const char *func)
2404 __ao2_ref_debug(p, -1, tag, file, line, func);
2411 /*! \brief map from an integer value to a string.
2412 * If no match is found, return errorstring
2414 static const char *map_x_s(const struct _map_x_s *table, int x, const char *errorstring)
2416 const struct _map_x_s *cur;
2418 for (cur = table; cur->s; cur++) {
2426 /*! \brief map from a string to an integer value, case insensitive.
2427 * If no match is found, return errorvalue.
2429 static int map_s_x(const struct _map_x_s *table, const char *s, int errorvalue)
2431 const struct _map_x_s *cur;
2433 for (cur = table; cur->s; cur++) {
2434 if (!strcasecmp(cur->s, s)) {
2441 static enum AST_REDIRECTING_REASON sip_reason_str_to_code(const char *text)
2443 enum AST_REDIRECTING_REASON ast = AST_REDIRECTING_REASON_UNKNOWN;
2446 for (i = 0; i < ARRAY_LEN(sip_reason_table); ++i) {
2447 if (!strcasecmp(text, sip_reason_table[i].text)) {
2448 ast = sip_reason_table[i].code;
2456 static const char *sip_reason_code_to_str(struct ast_party_redirecting_reason *reason, int *table_lookup)
2458 int code = reason->code;
2460 /* If there's a specific string set, then we just
2463 if (!ast_strlen_zero(reason->str)) {
2464 /* If we care about whether this can be found in
2465 * the table, then we need to check about that.
2468 /* If the string is literally "unknown" then don't bother with the lookup
2469 * because it can lead to a false negative.
2471 if (!strcasecmp(reason->str, "unknown") ||
2472 sip_reason_str_to_code(reason->str) != AST_REDIRECTING_REASON_UNKNOWN) {
2473 *table_lookup = TRUE;
2475 *table_lookup = FALSE;
2482 *table_lookup = TRUE;
2485 if (code >= 0 && code < ARRAY_LEN(sip_reason_table)) {
2486 return sip_reason_table[code].text;
2493 * \brief generic function for determining if a correct transport is being
2494 * used to contact a peer
2496 * this is done as a macro so that the "tmpl" var can be passed either a
2497 * sip_request or a sip_peer
2499 #define check_request_transport(peer, tmpl) ({ \
2501 if (peer->socket.type == tmpl->socket.type) \
2503 else if (!(peer->transports & tmpl->socket.type)) {\
2504 ast_log(LOG_ERROR, \
2505 "'%s' is not a valid transport for '%s'. we only use '%s'! ending call.\n", \
2506 sip_get_transport(tmpl->socket.type), peer->name, get_transport_list(peer->transports) \
2509 } else if (peer->socket.type & AST_TRANSPORT_TLS) { \
2510 ast_log(LOG_WARNING, \
2511 "peer '%s' HAS NOT USED (OR SWITCHED TO) TLS in favor of '%s' (but this was allowed in sip.conf)!\n", \
2512 peer->name, sip_get_transport(tmpl->socket.type) \
2516 "peer '%s' has contacted us over %s even though we prefer %s.\n", \
2517 peer->name, sip_get_transport(tmpl->socket.type), sip_get_transport(peer->socket.type) \
2524 * duplicate a list of channel variables, \return the copy.
2526 static struct ast_variable *copy_vars(struct ast_variable *src)
2528 struct ast_variable *res = NULL, *tmp, *v = NULL;
2530 for (v = src ; v ; v = v->next) {
2531 if ((tmp = ast_variable_new(v->name, v->value, v->file))) {
2539 static void tcptls_packet_destructor(void *obj)
2541 struct tcptls_packet *packet = obj;
2543 ast_free(packet->data);
2546 static void sip_tcptls_client_args_destructor(void *obj)
2548 struct ast_tcptls_session_args *args = obj;
2549 if (args->tls_cfg) {
2550 ast_free(args->tls_cfg->certfile);
2551 ast_free(args->tls_cfg->pvtfile);
2552 ast_free(args->tls_cfg->cipher);
2553 ast_free(args->tls_cfg->cafile);
2554 ast_free(args->tls_cfg->capath);
2556 ast_ssl_teardown(args->tls_cfg);
2558 ast_free(args->tls_cfg);
2559 ast_free((char *) args->name);
2562 static void sip_threadinfo_destructor(void *obj)
2564 struct sip_threadinfo *th = obj;
2565 struct tcptls_packet *packet;
2567 if (th->alert_pipe[1] > -1) {
2568 close(th->alert_pipe[0]);
2570 if (th->alert_pipe[1] > -1) {
2571 close(th->alert_pipe[1]);
2573 th->alert_pipe[0] = th->alert_pipe[1] = -1;
2575 while ((packet = AST_LIST_REMOVE_HEAD(&th->packet_q, entry))) {
2576 ao2_t_ref(packet, -1, "thread destruction, removing packet from frame queue");
2579 if (th->tcptls_session) {
2580 ao2_t_ref(th->tcptls_session, -1, "remove tcptls_session for sip_threadinfo object");
2584 /*! \brief creates a sip_threadinfo object and links it into the threadt table. */
2585 static struct sip_threadinfo *sip_threadinfo_create(struct ast_tcptls_session_instance *tcptls_session, int transport)
2587 struct sip_threadinfo *th;
2589 if (!tcptls_session || !(th = ao2_alloc(sizeof(*th), sip_threadinfo_destructor))) {
2593 th->alert_pipe[0] = th->alert_pipe[1] = -1;
2595 if (pipe(th->alert_pipe) == -1) {
2596 ao2_t_ref(th, -1, "Failed to open alert pipe on sip_threadinfo");
2597 ast_log(LOG_ERROR, "Could not create sip alert pipe in tcptls thread, error %s\n", strerror(errno));
2600 ao2_t_ref(tcptls_session, +1, "tcptls_session ref for sip_threadinfo object");
2601 th->tcptls_session = tcptls_session;
2602 th->type = transport ? transport : (tcptls_session->ssl ? AST_TRANSPORT_TLS: AST_TRANSPORT_TCP);
2603 ao2_t_link(threadt, th, "Adding new tcptls helper thread");
2604 ao2_t_ref(th, -1, "Decrementing threadinfo ref from alloc, only table ref remains");
2608 /*! \brief used to indicate to a tcptls thread that data is ready to be written */
2609 static int sip_tcptls_write(struct ast_tcptls_session_instance *tcptls_session, const void *buf, size_t len)
2612 struct sip_threadinfo *th = NULL;
2613 struct tcptls_packet *packet = NULL;
2614 struct sip_threadinfo tmp = {
2615 .tcptls_session = tcptls_session,
2617 enum sip_tcptls_alert alert = TCPTLS_ALERT_DATA;
2619 if (!tcptls_session) {
2623 ao2_lock(tcptls_session);
2625 if ((tcptls_session->fd == -1) ||
2626 !(th = ao2_t_find(threadt, &tmp, OBJ_POINTER, "ao2_find, getting sip_threadinfo in tcp helper thread")) ||
2627 !(packet = ao2_alloc(sizeof(*packet), tcptls_packet_destructor)) ||
2628 !(packet->data = ast_str_create(len))) {
2629 goto tcptls_write_setup_error;
2632 /* goto tcptls_write_error should _NOT_ be used beyond this point */
2633 ast_str_set(&packet->data, 0, "%s", (char *) buf);
2636 /* alert tcptls thread handler that there is a packet to be sent.
2637 * must lock the thread info object to guarantee control of the
2640 if (write(th->alert_pipe[1], &alert, sizeof(alert)) == -1) {
2641 ast_log(LOG_ERROR, "write() to alert pipe failed: %s\n", strerror(errno));
2642 ao2_t_ref(packet, -1, "could not write to alert pipe, remove packet");
2645 } else { /* it is safe to queue the frame after issuing the alert when we hold the threadinfo lock */
2646 AST_LIST_INSERT_TAIL(&th->packet_q, packet, entry);
2650 ao2_unlock(tcptls_session);
2651 ao2_t_ref(th, -1, "In sip_tcptls_write, unref threadinfo object after finding it");
2654 tcptls_write_setup_error:
2656 ao2_t_ref(th, -1, "In sip_tcptls_write, unref threadinfo obj, could not create packet");
2659 ao2_t_ref(packet, -1, "could not allocate packet's data");
2661 ao2_unlock(tcptls_session);
2666 /*! \brief SIP TCP connection handler */
2667 static void *sip_tcp_worker_fn(void *data)
2669 struct ast_tcptls_session_instance *tcptls_session = data;
2671 return _sip_tcp_helper_thread(tcptls_session);
2674 /*! \brief SIP WebSocket connection handler */
2675 static void sip_websocket_callback(struct ast_websocket *session, struct ast_variable *parameters, struct ast_variable *headers)
2679 if (ast_websocket_set_nonblock(session)) {
2683 while ((res = ast_wait_for_input(ast_websocket_fd(session), -1)) > 0) {
2685 uint64_t payload_len;
2686 enum ast_websocket_opcode opcode;
2689 if (ast_websocket_read(session, &payload, &payload_len, &opcode, &fragmented)) {
2690 /* We err on the side of caution and terminate the session if any error occurs */
2694 if (opcode == AST_WEBSOCKET_OPCODE_TEXT || opcode == AST_WEBSOCKET_OPCODE_BINARY) {
2695 struct sip_request req = { 0, };
2697 if (!(req.data = ast_str_create(payload_len + 1))) {
2701 if (ast_str_set(&req.data, -1, "%s", payload) == AST_DYNSTR_BUILD_FAILED) {
2706 req.socket.fd = ast_websocket_fd(session);
2707 set_socket_transport(&req.socket, ast_websocket_is_secure(session) ? AST_TRANSPORT_WSS : AST_TRANSPORT_WS);
2708 req.socket.ws_session = session;
2710 handle_request_do(&req, ast_websocket_remote_address(session));
2713 } else if (opcode == AST_WEBSOCKET_OPCODE_CLOSE) {
2719 ast_websocket_unref(session);
2722 /*! \brief Check if the authtimeout has expired.
2723 * \param start the time when the session started
2725 * \retval 0 the timeout has expired
2727 * \return the number of milliseconds until the timeout will expire
2729 static int sip_check_authtimeout(time_t start)
2733 if(time(&now) == -1) {
2734 ast_log(LOG_ERROR, "error executing time(): %s\n", strerror(errno));
2738 timeout = (authtimeout - (now - start)) * 1000;
2740 /* we have timed out */
2748 * \brief Read a SIP request or response from a TLS connection
2750 * Because TLS operations are hidden from view via a FILE handle, the
2751 * logic for reading data is a bit complex, and we have to make periodic
2752 * checks to be sure we aren't taking too long to perform the necessary
2755 * \todo XXX This should be altered in the future not to use a FILE pointer
2757 * \param req The request structure to fill in
2758 * \param tcptls_session The TLS connection on which the data is being received
2759 * \param authenticated A flag indicating whether authentication has occurred yet.
2760 * This is only relevant in a server role.
2761 * \param start The time at which we started attempting to read data. Used in
2762 * determining if there has been a timeout.
2763 * \param me Thread info. Used as a means of determining if the session needs to be stoppped.
2764 * \retval -1 Failed to read data
2765 * \retval 0 Succeeded in reading data
2767 static int sip_tls_read(struct sip_request *req, struct sip_request *reqcpy, struct ast_tcptls_session_instance *tcptls_session,
2768 int authenticated, time_t start, struct sip_threadinfo *me)
2770 int res, content_length, after_poll = 1, need_poll = 1;
2771 size_t datalen = ast_str_strlen(req->data);
2772 char buf[1024] = "";
2775 /* Read in headers one line at a time */
2776 while (datalen < 4 || strncmp(REQ_OFFSET_TO_STR(req, data->used - 4), "\r\n\r\n", 4)) {
2777 if (!tcptls_session->client && !authenticated) {
2778 if ((timeout = sip_check_authtimeout(start)) < 0) {
2779 ast_debug(2, "SIP TLS server failed to determine authentication timeout\n");
2784 ast_debug(2, "SIP TLS server timed out\n");
2791 /* special polling behavior is required for TLS
2792 * sockets because of the buffering done in the
2797 res = ast_wait_for_input(tcptls_session->fd, timeout);
2799 ast_debug(2, "SIP TLS server :: ast_wait_for_input returned %d\n", res);
2801 } else if (res == 0) {
2803 ast_debug(2, "SIP TLS server timed out\n");
2808 ao2_lock(tcptls_session);
2809 if (!fgets(buf, sizeof(buf), tcptls_session->f)) {
2810 ao2_unlock(tcptls_session);
2818 ao2_unlock(tcptls_session);
2823 ast_str_append(&req->data, 0, "%s", buf);
2825 datalen = ast_str_strlen(req->data);
2826 if (datalen > SIP_MAX_PACKET_SIZE) {
2827 ast_log(LOG_WARNING, "Rejecting TLS packet from '%s' because way too large: %zu\n",
2828 ast_sockaddr_stringify(&tcptls_session->remote_address), datalen);
2832 copy_request(reqcpy, req);
2833 parse_request(reqcpy);
2834 /* In order to know how much to read, we need the content-length header */
2835 if (sscanf(sip_get_header(reqcpy, "Content-Length"), "%30d", &content_length)) {
2836 while (content_length > 0) {
2838 if (!tcptls_session->client && !authenticated) {
2839 if ((timeout = sip_check_authtimeout(start)) < 0) {
2844 ast_debug(2, "SIP TLS server timed out\n");
2854 res = ast_wait_for_input(tcptls_session->fd, timeout);
2856 ast_debug(2, "SIP TLS server :: ast_wait_for_input returned %d\n", res);
2858 } else if (res == 0) {
2860 ast_debug(2, "SIP TLS server timed out\n");
2865 ao2_lock(tcptls_session);
2866 if (!(bytes_read = fread(buf, 1, MIN(sizeof(buf) - 1, content_length), tcptls_session->f))) {
2867 ao2_unlock(tcptls_session);
2875 buf[bytes_read] = '\0';
2876 ao2_unlock(tcptls_session);
2881 content_length -= strlen(buf);
2882 ast_str_append(&req->data, 0, "%s", buf);
2884 datalen = ast_str_strlen(req->data);
2885 if (datalen > SIP_MAX_PACKET_SIZE) {
2886 ast_log(LOG_WARNING, "Rejecting TLS packet from '%s' because way too large: %zu\n",
2887 ast_sockaddr_stringify(&tcptls_session->remote_address), datalen);
2892 /*! \todo XXX If there's no Content-Length or if the content-length and what
2893 we receive is not the same - we should generate an error */
2898 * \brief Indication of a TCP message's integrity
2900 enum message_integrity {
2902 * The message has an error in it with
2903 * regards to its Content-Length header
2907 * The message is incomplete
2911 * The data contains a complete message
2912 * plus a fragment of another.
2914 MESSAGE_FRAGMENT_COMPLETE,
2916 * The message is complete
2923 * Get the content length from an unparsed SIP message
2925 * \param message The unparsed SIP message headers
2926 * \return The value of the Content-Length header or -1 if message is invalid
2928 static int read_raw_content_length(const char *message)
2930 char *content_length_str;
2931 int content_length = -1;
2933 struct ast_str *msg_copy;
2936 /* Using a ast_str because lws2sws takes one of those */
2937 if (!(msg_copy = ast_str_create(strlen(message) + 1))) {
2940 ast_str_set(&msg_copy, 0, "%s", message);
2942 if (sip_cfg.pedanticsipchecking) {
2946 msg = ast_str_buffer(msg_copy);
2948 /* Let's find a Content-Length header */
2949 if ((content_length_str = strcasestr(msg, "\nContent-Length:"))) {
2950 content_length_str += sizeof("\nContent-Length:") - 1;
2951 } else if ((content_length_str = strcasestr(msg, "\nl:"))) {
2952 content_length_str += sizeof("\nl:") - 1;
2955 * "In the case of stream-oriented transports such as TCP, the Content-
2956 * Length header field indicates the size of the body. The Content-
2957 * Length header field MUST be used with stream oriented transports."
2962 /* Double-check that this is a complete header */
2963 if (!strchr(content_length_str, '\n')) {
2967 if (sscanf(content_length_str, "%30d", &content_length) != 1) {
2968 content_length = -1;
2973 return content_length;
2977 * \brief Check that a message received over TCP is a full message
2979 * This will take the information read in and then determine if
2980 * 1) The message is a full SIP request
2981 * 2) The message is a partial SIP request
2982 * 3) The message contains a full SIP request along with another partial request
2983 * \param data The unparsed incoming SIP message.
2984 * \param request The resulting request with extra fragments removed.
2985 * \param overflow If the message contains more than a full request, this is the remainder of the message
2986 * \return The resulting integrity of the message
2988 static enum message_integrity check_message_integrity(struct ast_str **request, struct ast_str **overflow)
2990 char *message = ast_str_buffer(*request);
2993 int message_len = ast_str_strlen(*request);
2996 /* Important pieces to search for in a SIP request are \r\n\r\n. This
2998 * 1) The division between the headers and body
2999 * 2) The end of the SIP request
3001 body = strstr(message, "\r\n\r\n");
3003 /* This is clearly a partial message since we haven't reached an end
3006 return MESSAGE_FRAGMENT;
3008 body += sizeof("\r\n\r\n") - 1;
3009 body_len = message_len - (body - message);
3012 content_length = read_raw_content_length(message);
3015 if (content_length < 0) {
3016 return MESSAGE_INVALID;
3017 } else if (content_length == 0) {
3018 /* We've definitely received an entire message. We need
3019 * to check if there's also a fragment of another message
3022 if (body_len == 0) {
3023 return MESSAGE_COMPLETE;
3025 ast_str_append(overflow, 0, "%s", body);
3026 ast_str_truncate(*request, message_len - body_len);
3027 return MESSAGE_FRAGMENT_COMPLETE;
3030 /* Positive content length. Let's see what sort of
3031 * message body we're dealing with.
3033 if (body_len < content_length) {
3034 /* We don't have the full message body yet */
3035 return MESSAGE_FRAGMENT;
3036 } else if (body_len > content_length) {
3037 /* We have the full message plus a fragment of a further
3040 ast_str_append(overflow, 0, "%s", body + content_length);
3041 ast_str_truncate(*request, message_len - (body_len - content_length));
3042 return MESSAGE_FRAGMENT_COMPLETE;
3044 /* Yay! Full message with no extra content */
3045 return MESSAGE_COMPLETE;
3050 * \brief Read SIP request or response from a TCP connection
3052 * \param req The request structure to be filled in
3053 * \param tcptls_session The TCP connection from which to read
3054 * \retval -1 Failed to read data
3055 * \retval 0 Successfully read data
3057 static int sip_tcp_read(struct sip_request *req, struct ast_tcptls_session_instance *tcptls_session,
3058 int authenticated, time_t start)
3060 enum message_integrity message_integrity = MESSAGE_FRAGMENT;
3062 while (message_integrity == MESSAGE_FRAGMENT) {
3065 if (ast_str_strlen(tcptls_session->overflow_buf) == 0) {
3069 if (!tcptls_session->client && !authenticated) {
3070 if ((timeout = sip_check_authtimeout(start)) < 0) {
3075 ast_debug(2, "SIP TCP server timed out\n");
3081 res = ast_wait_for_input(tcptls_session->fd, timeout);
3083 ast_debug(2, "SIP TCP server :: ast_wait_for_input returned %d\n", res);
3085 } else if (res == 0) {
3086 ast_debug(2, "SIP TCP server timed out\n");
3090 res = recv(tcptls_session->fd, readbuf, sizeof(readbuf) - 1, 0);
3092 ast_debug(2, "SIP TCP server error when receiving data\n");
3094 } else if (res == 0) {
3095 ast_debug(2, "SIP TCP server has shut down\n");