27d446296874bfff9568ba659a22974891886290
[asterisk/asterisk.git] / res / res_pjsip.c
1 /*
2  * Asterisk -- An open source telephony toolkit.
3  *
4  * Copyright (C) 2013, Digium, Inc.
5  *
6  * Mark Michelson <mmichelson@digium.com>
7  *
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.
13  *
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.
17  */
18
19 #include "asterisk.h"
20
21 #include <pjsip.h>
22 /* Needed for SUBSCRIBE, NOTIFY, and PUBLISH method definitions */
23 #include <pjsip_simple.h>
24 #include <pjlib.h>
25
26 #include "asterisk/res_pjsip.h"
27 #include "res_pjsip/include/res_pjsip_private.h"
28 #include "asterisk/linkedlists.h"
29 #include "asterisk/logger.h"
30 #include "asterisk/lock.h"
31 #include "asterisk/utils.h"
32 #include "asterisk/astobj2.h"
33 #include "asterisk/module.h"
34 #include "asterisk/threadpool.h"
35 #include "asterisk/taskprocessor.h"
36 #include "asterisk/uuid.h"
37 #include "asterisk/sorcery.h"
38
39 /*** MODULEINFO
40         <depend>pjproject</depend>
41         <depend>res_sorcery_config</depend>
42         <support_level>core</support_level>
43  ***/
44
45 /*** DOCUMENTATION
46         <configInfo name="res_pjsip" language="en_US">
47                 <synopsis>SIP Resource using PJProject</synopsis>
48                 <configFile name="pjsip.conf">
49                         <configObject name="endpoint">
50                                 <synopsis>Endpoint</synopsis>
51                                 <description><para>
52                                         The <emphasis>Endpoint</emphasis> is the primary configuration object.
53                                         It contains the core SIP related options only, endpoints are <emphasis>NOT</emphasis>
54                                         dialable entries of their own. Communication with another SIP device is
55                                         accomplished via Addresses of Record (AoRs) which have one or more
56                                         contacts assicated with them. Endpoints <emphasis>NOT</emphasis> configured to
57                                         use a <literal>transport</literal> will default to first transport found
58                                         in <filename>pjsip.conf</filename> that matches its type.
59                                         </para>
60                                         <para>Example: An Endpoint has been configured with no transport.
61                                         When it comes time to call an AoR, PJSIP will find the
62                                         first transport that matches the type. A SIP URI of <literal>sip:5000@[11::33]</literal>
63                                         will use the first IPv6 transport and try to send the request.
64                                         </para>
65                                         <para>If the anonymous endpoint identifier is in use an endpoint with the name
66                                         "anonymous@domain" will be searched for as a last resort. If this is not found
67                                         it will fall back to searching for "anonymous". If neither endpoints are found
68                                         the anonymous endpoint identifier will not return an endpoint and anonymous
69                                         calling will not be possible.
70                                         </para>
71                                 </description>
72                                 <configOption name="100rel" default="yes">
73                                         <synopsis>Allow support for RFC3262 provisional ACK tags</synopsis>
74                                         <description>
75                                                 <enumlist>
76                                                         <enum name="no" />
77                                                         <enum name="required" />
78                                                         <enum name="yes" />
79                                                 </enumlist>
80                                         </description>
81                                 </configOption>
82                                 <configOption name="aggregate_mwi" default="yes">
83                                         <synopsis></synopsis>
84                                         <description><para>When enabled, <replaceable>aggregate_mwi</replaceable> condenses message
85                                         waiting notifications from multiple mailboxes into a single NOTIFY. If it is disabled,
86                                         individual NOTIFYs are sent for each mailbox.</para></description>
87                                 </configOption>
88                                 <configOption name="allow">
89                                         <synopsis>Media Codec(s) to allow</synopsis>
90                                 </configOption>
91                                 <configOption name="aors">
92                                         <synopsis>AoR(s) to be used with the endpoint</synopsis>
93                                         <description><para>
94                                                 List of comma separated AoRs that the endpoint should be associated with.
95                                         </para></description>
96                                 </configOption>
97                                 <configOption name="auth">
98                                         <synopsis>Authentication Object(s) associated with the endpoint</synopsis>
99                                         <description><para>
100                                                 This is a comma-delimited list of <replaceable>auth</replaceable> sections defined
101                                                 in <filename>pjsip.conf</filename> to be used to verify inbound connection attempts.
102                                                 </para><para>
103                                                 Endpoints without an <literal>authentication</literal> object
104                                                 configured will allow connections without vertification.
105                                         </para></description>
106                                 </configOption>
107                                 <configOption name="callerid">
108                                         <synopsis>CallerID information for the endpoint</synopsis>
109                                         <description><para>
110                                                 Must be in the format <literal>Name &lt;Number&gt;</literal>,
111                                                 or only <literal>&lt;Number&gt;</literal>.
112                                         </para></description>
113                                 </configOption>
114                                 <configOption name="callerid_privacy">
115                                         <synopsis>Default privacy level</synopsis>
116                                         <description>
117                                                 <enumlist>
118                                                         <enum name="allowed_not_screened" />
119                                                         <enum name="allowed_passed_screened" />
120                                                         <enum name="allowed_failed_screened" />
121                                                         <enum name="allowed" />
122                                                         <enum name="prohib_not_screened" />
123                                                         <enum name="prohib_passed_screened" />
124                                                         <enum name="prohib_failed_screened" />
125                                                         <enum name="prohib" />
126                                                         <enum name="unavailable" />
127                                                 </enumlist>
128                                         </description>
129                                 </configOption>
130                                 <configOption name="callerid_tag">
131                                         <synopsis>Internal id_tag for the endpoint</synopsis>
132                                 </configOption>
133                                 <configOption name="context">
134                                         <synopsis>Dialplan context for inbound sessions</synopsis>
135                                 </configOption>
136                                 <configOption name="direct_media_glare_mitigation" default="none">
137                                         <synopsis>Mitigation of direct media (re)INVITE glare</synopsis>
138                                         <description>
139                                                 <para>
140                                                 This setting attempts to avoid creating INVITE glare scenarios
141                                                 by disabling direct media reINVITEs in one direction thereby allowing
142                                                 designated servers (according to this option) to initiate direct
143                                                 media reINVITEs without contention and significantly reducing call
144                                                 setup time.
145                                                 </para>
146                                                 <para>
147                                                 A more detailed description of how this option functions can be found on
148                                                 the Asterisk wiki https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance
149                                                 </para>
150                                                 <enumlist>
151                                                         <enum name="none" />
152                                                         <enum name="outgoing" />
153                                                         <enum name="incoming" />
154                                                 </enumlist>
155                                         </description>
156                                 </configOption>
157                                 <configOption name="direct_media_method" default="invite">
158                                         <synopsis>Direct Media method type</synopsis>
159                                         <description>
160                                                 <para>Method for setting up Direct Media between endpoints.</para>
161                                                 <enumlist>
162                                                         <enum name="invite" />
163                                                         <enum name="reinvite">
164                                                                 <para>Alias for the <literal>invite</literal> value.</para>
165                                                         </enum>
166                                                         <enum name="update" />
167                                                 </enumlist>
168                                         </description>
169                                 </configOption>
170                                 <configOption name="connected_line_method" default="invite">
171                                         <synopsis>Connected line method type</synopsis>
172                                         <description>
173                                                 <para>Method used when updating connected line information.</para>
174                                                 <enumlist>
175                                                         <enum name="invite" />
176                                                         <enum name="reinvite">
177                                                                 <para>Alias for the <literal>invite</literal> value.</para>
178                                                         </enum>
179                                                         <enum name="update" />
180                                                 </enumlist>
181                                         </description>
182                                 </configOption>
183                                 <configOption name="direct_media" default="yes">
184                                         <synopsis>Determines whether media may flow directly between endpoints.</synopsis>
185                                 </configOption>
186                                 <configOption name="disable_direct_media_on_nat" default="no">
187                                         <synopsis>Disable direct media session refreshes when NAT obstructs the media session</synopsis>
188                                 </configOption>
189                                 <configOption name="disallow">
190                                         <synopsis>Media Codec(s) to disallow</synopsis>
191                                 </configOption>
192                                 <configOption name="dtmfmode" default="rfc4733">
193                                         <synopsis>DTMF mode</synopsis>
194                                         <description>
195                                                 <para>This setting allows to choose the DTMF mode for endpoint communication.</para>
196                                                 <enumlist>
197                                                         <enum name="rfc4733">
198                                                                 <para>DTMF is sent out of band of the main audio stream.This
199                                                                 supercedes the older <emphasis>RFC-2833</emphasis> used within
200                                                                 the older <literal>chan_sip</literal>.</para>
201                                                         </enum>
202                                                         <enum name="inband">
203                                                                 <para>DTMF is sent as part of audio stream.</para>
204                                                         </enum>
205                                                         <enum name="info">
206                                                                 <para>DTMF is sent as SIP INFO packets.</para>
207                                                         </enum>
208                                                 </enumlist>
209                                         </description>
210                                 </configOption>
211                                 <configOption name="external_media_address">
212                                         <synopsis>IP used for External Media handling</synopsis>
213                                 </configOption>
214                                 <configOption name="force_rport" default="yes">
215                                         <synopsis>Force use of return port</synopsis>
216                                 </configOption>
217                                 <configOption name="ice_support" default="no">
218                                         <synopsis>Enable the ICE mechanism to help traverse NAT</synopsis>
219                                 </configOption>
220                                 <configOption name="identify_by" default="username,location">
221                                         <synopsis>Way(s) for Endpoint to be identified</synopsis>
222                                         <description><para>
223                                                 There are currently two methods to identify an endpoint. By default
224                                                 both are used to identify an endpoint.
225                                                 </para>
226                                                 <enumlist>
227                                                         <enum name="username" />
228                                                         <enum name="location" />
229                                                         <enum name="username,location" />
230                                                 </enumlist>
231                                         </description>
232                                 </configOption>
233                                 <configOption name="mailboxes">
234                                         <synopsis>Mailbox(es) to be associated with</synopsis>
235                                 </configOption>
236                                 <configOption name="mohsuggest" default="default">
237                                         <synopsis>Default Music On Hold class</synopsis>
238                                 </configOption>
239                                 <configOption name="outbound_auth">
240                                         <synopsis>Authentication object used for outbound requests</synopsis>
241                                 </configOption>
242                                 <configOption name="outbound_proxy">
243                                         <synopsis>Proxy through which to send requests</synopsis>
244                                 </configOption>
245                                 <configOption name="rewrite_contact">
246                                         <synopsis>Allow Contact header to be rewritten with the source IP address-port</synopsis>
247                                 </configOption>
248                                 <configOption name="rtp_ipv6" default="no">
249                                         <synopsis>Allow use of IPv6 for RTP traffic</synopsis>
250                                 </configOption>
251                                 <configOption name="rtp_symmetric" default="no">
252                                         <synopsis>Enforce that RTP must be symmetric</synopsis>
253                                 </configOption>
254                                 <configOption name="send_pai" default="no">
255                                         <synopsis>Send the P-Asserted-Identity header</synopsis>
256                                 </configOption>
257                                 <configOption name="send_rpid" default="no">
258                                         <synopsis>Send the Remote-Party-ID header</synopsis>
259                                 </configOption>
260                                 <configOption name="timers_min_se" default="90">
261                                         <synopsis>Minimum session timers expiration period</synopsis>
262                                         <description><para>
263                                                 Minimium session timer expiration period. Time in seconds.
264                                         </para></description>
265                                 </configOption>
266                                 <configOption name="timers" default="yes">
267                                         <synopsis>Session timers for SIP packets</synopsis>
268                                         <description>
269                                                 <enumlist>
270                                                         <enum name="forced" />
271                                                         <enum name="no" />
272                                                         <enum name="required" />
273                                                         <enum name="yes" />
274                                                 </enumlist>
275                                         </description>
276                                 </configOption>
277                                 <configOption name="timers_sess_expires" default="1800">
278                                         <synopsis>Maximum session timer expiration period</synopsis>
279                                         <description><para>
280                                                 Maximium session timer expiration period. Time in seconds.
281                                         </para></description>
282                                 </configOption>
283                                 <configOption name="transport">
284                                         <synopsis>Desired transport configuration</synopsis>
285                                         <description><para>
286                                                 This will set the desired transport configuration to send SIP data through.
287                                                 </para>
288                                                 <warning><para>Not specifying a transport will <emphasis>DEFAULT</emphasis>
289                                                 to the first configured transport in <filename>pjsip.conf</filename> which is
290                                                 valid for the URI we are trying to contact.
291                                                 </para></warning>
292                                         </description>
293                                 </configOption>
294                                 <configOption name="trust_id_inbound" default="no">
295                                         <synopsis>Accept identification information received from this endpoint</synopsis>
296                                         <description><para>This option determines whether Asterisk will accept
297                                         identification from the endpoint from headers such as P-Asserted-Identity
298                                         or Remote-Party-ID header. This option applies both to calls originating from the
299                                         endpoint and calls originating from Asterisk. If <literal>no</literal>, the
300                                         configured Caller-ID from pjsip.conf will always be used as the identity for
301                                         the endpoint.</para></description>
302                                 </configOption>
303                                 <configOption name="trust_id_outbound" default="no">
304                                         <synopsis>Send private identification details to the endpoint.</synopsis>
305                                         <description><para>This option determines whether res_pjsip will send private
306                                         identification information to the endpoint. If <literal>no</literal>,
307                                         private Caller-ID information will not be forwarded to the endpoint.
308                                         "Private" in this case refers to any method of restricting identification.
309                                         Example: setting <replaceable>callerid_privacy</replaceable> to any
310                                         <literal>prohib</literal> variation.
311                                         Example: If <replaceable>trust_id_inbound</replaceable> is set to
312                                         <literal>yes</literal>, the presence of a <literal>Privacy: id</literal>
313                                         header in a SIP request or response would indicate the identification
314                                         provided in the request is private.</para></description>
315                                 </configOption>
316                                 <configOption name="type">
317                                         <synopsis>Must be of type 'endpoint'.</synopsis>
318                                 </configOption>
319                                 <configOption name="use_ptime" default="no">
320                                         <synopsis>Use Endpoint's requested packetisation interval</synopsis>
321                                 </configOption>
322                                 <configOption name="use_avpf" default="no">
323                                         <synopsis>Determines whether res_pjsip will use and enforce usage of AVPF for this
324                                         endpoint.</synopsis>
325                                         <description><para>
326                                                 If set to <literal>yes</literal>, res_pjsip will use use the AVPF or SAVPF RTP
327                                                 profile for all media offers on outbound calls and media updates and will
328                                                 decline media offers not using the AVPF or SAVPF profile.
329                                         </para><para>
330                                                 If set to <literal>no</literal>, res_pjsip will use use the AVP or SAVP RTP
331                                                 profile for all media offers on outbound calls and media updates and will
332                                                 decline media offers not using the AVP or SAVP profile.
333                                         </para></description>
334                                 </configOption>
335                                 <configOption name="media_encryption" default="no">
336                                         <synopsis>Determines whether res_pjsip will use and enforce usage of media encryption
337                                         for this endpoint.</synopsis>
338                                         <description>
339                                                 <enumlist>
340                                                         <enum name="no"><para>
341                                                                 res_pjsip will offer no encryption and allow no encryption to be setup.
342                                                         </para></enum>
343                                                         <enum name="sdes"><para>
344                                                                 res_pjsip will offer standard SRTP setup via in-SDP keys. Encrypted SIP
345                                                                 transport should be used in conjunction with this option to prevent
346                                                                 exposure of media encryption keys.
347                                                         </para></enum>
348                                                         <enum name="dtls"><para>
349                                                                 res_pjsip will offer DTLS-SRTP setup.
350                                                         </para></enum>
351                                                 </enumlist>
352                                         </description>
353                                 </configOption>
354                                 <configOption name="inband_progress" default="no">
355                                         <synopsis>Determines whether chan_pjsip will indicate ringing using inband
356                                             progress.</synopsis>
357                                         <description><para>
358                                                 If set to <literal>yes</literal>, chan_pjsip will send a 183 Session Progress
359                                                 when told to indicate ringing and will immediately start sending ringing
360                                                 as audio.
361                                         </para><para>
362                                                 If set to <literal>no</literal>, chan_pjsip will send a 180 Ringing when told
363                                                 to indicate ringing and will NOT send it as audio.
364                                         </para></description>
365                                 </configOption>
366                                 <configOption name="callgroup">
367                                         <synopsis>The numeric pickup groups for a channel.</synopsis>
368                                         <description><para>
369                                                 Can be set to a comma separated list of numbers or ranges between the values
370                                                 of 0-63 (maximum of 64 groups).
371                                         </para></description>
372                                 </configOption>
373                                 <configOption name="pickupgroup">
374                                         <synopsis>The numeric pickup groups that a channel can pickup.</synopsis>
375                                         <description><para>
376                                                 Can be set to a comma separated list of numbers or ranges between the values
377                                                 of 0-63 (maximum of 64 groups).
378                                         </para></description>
379                                 </configOption>
380                                 <configOption name="namedcallgroup">
381                                         <synopsis>The named pickup groups for a channel.</synopsis>
382                                         <description><para>
383                                                 Can be set to a comma separated list of case sensitive strings limited by
384                                                 supported line length.
385                                         </para></description>
386                                 </configOption>
387                                 <configOption name="namedpickupgroup">
388                                         <synopsis>The named pickup groups that a channel can pickup.</synopsis>
389                                         <description><para>
390                                                 Can be set to a comma separated list of case sensitive strings limited by
391                                                 supported line length.
392                                         </para></description>
393                                 </configOption>
394                                 <configOption name="devicestate_busy_at" default="0">
395                                         <synopsis>The number of in-use channels which will cause busy to be returned as device state</synopsis>
396                                         <description><para>
397                                                 When the number of in-use channels for the endpoint matches the devicestate_busy_at setting the
398                                                 PJSIP channel driver will return busy as the device state instead of in use.
399                                         </para></description>
400                                 </configOption>
401                                 <configOption name="t38udptl" default="no">
402                                         <synopsis>Whether T.38 UDPTL support is enabled or not</synopsis>
403                                         <description><para>
404                                                 If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted
405                                                 and relayed.
406                                         </para></description>
407                                 </configOption>
408                                 <configOption name="t38udptl_ec" default="none">
409                                         <synopsis>T.38 UDPTL error correction method</synopsis>
410                                         <description>
411                                                 <enumlist>
412                                                         <enum name="none"><para>
413                                                                 No error correction should be used.
414                                                         </para></enum>
415                                                         <enum name="fec"><para>
416                                                                 Forward error correction should be used.
417                                                         </para></enum>
418                                                         <enum name="redundancy"><para>
419                                                                 Redundacy error correction should be used.
420                                                         </para></enum>
421                                                 </enumlist>
422                                         </description>
423                                 </configOption>
424                                 <configOption name="t38udptl_maxdatagram" default="0">
425                                         <synopsis>T.38 UDPTL maximum datagram size</synopsis>
426                                         <description><para>
427                                                 This option can be set to override the maximum datagram of a remote endpoint for broken
428                                                 endpoints.
429                                         </para></description>
430                                 </configOption>
431                                 <configOption name="faxdetect" default="no">
432                                         <synopsis>Whether CNG tone detection is enabled</synopsis>
433                                         <description><para>
434                                                 This option can be set to send the session to the fax extension when a CNG tone is
435                                                 detected.
436                                         </para></description>
437                                 </configOption>
438                                 <configOption name="t38udptl_nat" default="no">
439                                         <synopsis>Whether NAT support is enabled on UDPTL sessions</synopsis>
440                                         <description><para>
441                                                 When enabled the UDPTL stack will send UDPTL packets to the source address of
442                                                 received packets.
443                                         </para></description>
444                                 </configOption>
445                                 <configOption name="t38udptl_ipv6" default="no">
446                                         <synopsis>Whether IPv6 is used for UDPTL Sessions</synopsis>
447                                         <description><para>
448                                                 When enabled the UDPTL stack will use IPv6.
449                                         </para></description>
450                                 </configOption>
451                                 <configOption name="tonezone">
452                                         <synopsis>Set which country's indications to use for channels created for this endpoint.</synopsis>
453                                 </configOption>
454                                 <configOption name="language">
455                                         <synopsis>Set the default language to use for channels created for this endpoint.</synopsis>
456                                 </configOption>
457                                 <configOption name="one_touch_recording" default="no">
458                                         <synopsis>Determines whether one-touch recording is allowed for this endpoint.</synopsis>
459                                         <see-also>
460                                                 <ref type="configOption">recordonfeature</ref>
461                                                 <ref type="configOption">recordofffeature</ref>
462                                         </see-also>
463                                 </configOption>
464                                 <configOption name="recordonfeature" default="automixmon">
465                                         <synopsis>The feature to enact when one-touch recording is turned on.</synopsis>
466                                         <description>
467                                                 <para>When an INFO request for one-touch recording arrives with a Record header set to "on", this
468                                                 feature will be enabled for the channel. The feature designated here can be any built-in
469                                                 or dynamic feature defined in features.conf.</para>
470                                                 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
471                                         </description>
472                                         <see-also>
473                                                 <ref type="configOption">one_touch_recording</ref>
474                                                 <ref type="configOption">recordofffeature</ref>
475                                         </see-also>
476                                 </configOption>
477                                 <configOption name="recordofffeature" default="automixmon">
478                                         <synopsis>The feature to enact when one-touch recording is turned off.</synopsis>
479                                         <description>
480                                                 <para>When an INFO request for one-touch recording arrives with a Record header set to "off", this
481                                                 feature will be enabled for the channel. The feature designated here can be any built-in
482                                                 or dynamic feature defined in features.conf.</para>
483                                                 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
484                                         </description>
485                                         <see-also>
486                                                 <ref type="configOption">one_touch_recording</ref>
487                                                 <ref type="configOption">recordonfeature</ref>
488                                         </see-also>
489                                 </configOption>
490                                 <configOption name="rtpengine" default="asterisk">
491                                         <synopsis>Name of the RTP engine to use for channels created for this endpoint</synopsis>
492                                 </configOption>
493                                 <configOption name="allowtransfer" default="yes">
494                                         <synopsis>Determines whether SIP REFER transfers are allowed for this endpoint</synopsis>
495                                 </configOption>
496                                 <configOption name="sdpowner" default="-">
497                                         <synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis>
498                                 </configOption>
499                                 <configOption name="sdpsession" default="Asterisk">
500                                         <synopsis>String used for the SDP session (s=) line.</synopsis>
501                                 </configOption>
502                                 <configOption name="tos_audio">
503                                         <synopsis>DSCP TOS bits for audio streams</synopsis>
504                                         <description><para>
505                                                 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
506                                         </para></description>
507                                 </configOption>
508                                 <configOption name="tos_video">
509                                         <synopsis>DSCP TOS bits for video streams</synopsis>
510                                         <description><para>
511                                                 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
512                                         </para></description>
513                                 </configOption>
514                                 <configOption name="cos_audio">
515                                         <synopsis>Priority for audio streams</synopsis>
516                                         <description><para>
517                                                 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
518                                         </para></description>
519                                 </configOption>
520                                 <configOption name="cos_video">
521                                         <synopsis>Priority for video streams</synopsis>
522                                         <description><para>
523                                                 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
524                                         </para></description>
525                                 </configOption>
526                                 <configOption name="allowsubscribe" default="yes">
527                                         <synopsis>Determines if endpoint is allowed to initiate subscriptions with Asterisk.</synopsis>
528                                 </configOption>
529                                 <configOption name="subminexpiry" default="60">
530                                         <synopsis>The minimum allowed expiry time for subscriptions initiated by the endpoint.</synopsis>
531                                 </configOption>
532                                 <configOption name="fromuser">
533                                         <synopsis>Username to use in From header for requests to this endpoint.</synopsis>
534                                 </configOption>
535                                 <configOption name="mwifromuser">
536                                         <synopsis>Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.</synopsis>
537                                 </configOption>
538                                 <configOption name="fromdomain">
539                                         <synopsis>Domain to user in From header for requests to this endpoint.</synopsis>
540                                 </configOption>
541                                 <configOption name="dtlsverify">
542                                         <synopsis>Verify that the provided peer certificate is valid</synopsis>
543                                         <description><para>
544                                                 This option only applies if <replaceable>media_encryption</replaceable> is
545                                                 set to <literal>dtls</literal>.
546                                         </para></description>
547                                 </configOption>
548                                 <configOption name="dtlsrekey">
549                                         <synopsis>Interval at which to renegotiate the TLS session and rekey the SRTP session</synopsis>
550                                         <description><para>
551                                                 This option only applies if <replaceable>media_encryption</replaceable> is
552                                                 set to <literal>dtls</literal>.
553                                         </para><para>
554                                                 If this is not set or the value provided is 0 rekeying will be disabled.
555                                         </para></description>
556                                 </configOption>
557                                 <configOption name="dtlscertfile">
558                                         <synopsis>Path to certificate file to present to peer</synopsis>
559                                         <description><para>
560                                                 This option only applies if <replaceable>media_encryption</replaceable> is
561                                                 set to <literal>dtls</literal>.
562                                         </para></description>
563                                 </configOption>
564                                 <configOption name="dtlsprivatekey">
565                                         <synopsis>Path to private key for certificate file</synopsis>
566                                         <description><para>
567                                                 This option only applies if <replaceable>media_encryption</replaceable> is
568                                                 set to <literal>dtls</literal>.
569                                         </para></description>
570                                 </configOption>
571                                 <configOption name="dtlscipher">
572                                         <synopsis>Cipher to use for DTLS negotiation</synopsis>
573                                         <description><para>
574                                                 This option only applies if <replaceable>media_encryption</replaceable> is
575                                                 set to <literal>dtls</literal>.
576                                         </para><para>
577                                                 Many options for acceptable ciphers. See link for more:
578                                                 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
579                                         </para></description>
580                                 </configOption>
581                                 <configOption name="dtlscafile">
582                                         <synopsis>Path to certificate authority certificate</synopsis>
583                                         <description><para>
584                                                 This option only applies if <replaceable>media_encryption</replaceable> is
585                                                 set to <literal>dtls</literal>.
586                                         </para></description>
587                                 </configOption>
588                                 <configOption name="dtlscapath">
589                                         <synopsis>Path to a directory containing certificate authority certificates</synopsis>
590                                         <description><para>
591                                                 This option only applies if <replaceable>media_encryption</replaceable> is
592                                                 set to <literal>dtls</literal>.
593                                         </para></description>
594                                 </configOption>
595                                 <configOption name="dtlssetup">
596                                         <synopsis>Whether we are willing to accept connections, connect to the other party, or both.</synopsis>
597                                         <description>
598                                                 <para>
599                                                         This option only applies if <replaceable>media_encryption</replaceable> is
600                                                         set to <literal>dtls</literal>.
601                                                 </para>
602                                                 <enumlist>
603                                                         <enum name="active"><para>
604                                                                 res_pjsip will make a connection to the peer.
605                                                         </para></enum>
606                                                         <enum name="passive"><para>
607                                                                 res_pjsip will accept connections from the peer.
608                                                         </para></enum>
609                                                         <enum name="actpass"><para>
610                                                                 res_pjsip will offer and accept connections from the peer.
611                                                         </para></enum>
612                                                 </enumlist>
613                                         </description>
614                                 </configOption>
615                                 <configOption name="srtp_tag_32">
616                                         <synopsis>Determines whether 32 byte tags should be used instead of 80 byte tags.</synopsis>
617                                         <description><para>
618                                                 This option only applies if <replaceable>media_encryption</replaceable> is
619                                                 set to <literal>sdes</literal> or <literal>dtls</literal>.
620                                         </para></description>
621                                 </configOption>
622                         </configObject>
623                         <configObject name="auth">
624                                 <synopsis>Authentication type</synopsis>
625                                 <description><para>
626                                         Authentication objects hold the authenitcation information for use
627                                         by <literal>endpoints</literal>. This also allows for multiple <literal>
628                                         endpoints</literal> to use the same information. Choice of MD5/plaintext
629                                         and setting of username.
630                                 </para></description>
631                                 <configOption name="auth_type" default="userpass">
632                                         <synopsis>Authentication type</synopsis>
633                                         <description><para>
634                                                 This option specifies which of the password style config options should be read,
635                                                 either 'password' or 'md5_cred' when trying to authenticate an endpoint inbound request.
636                                                 </para>
637                                                 <enumlist>
638                                                         <enum name="md5"/>
639                                                         <enum name="userpass"/>
640                                                 </enumlist>
641                                         </description>
642                                 </configOption>
643                                 <configOption name="nonce_lifetime" default="32">
644                                         <synopsis>Lifetime of a nonce associated with this authentication config.</synopsis>
645                                 </configOption>
646                                 <configOption name="md5_cred">
647                                         <synopsis>MD5 Hash used for authentication.</synopsis>
648                                         <description><para>Only used when auth_type is <literal>md5</literal>.</para></description>
649                                 </configOption>
650                                 <configOption name="password">
651                                         <synopsis>PlainText password used for authentication.</synopsis>
652                                         <description><para>Only used when auth_type is <literal>userpass</literal>.</para></description>
653                                 </configOption>
654                                 <configOption name="realm" default="asterisk">
655                                         <synopsis>SIP realm for endpoint</synopsis>
656                                 </configOption>
657                                 <configOption name="type">
658                                         <synopsis>Must be 'auth'</synopsis>
659                                 </configOption>
660                                 <configOption name="username">
661                                         <synopsis>Username to use for account</synopsis>
662                                 </configOption>
663                         </configObject>
664                         <configObject name="nat_hook">
665                                 <synopsis>XXX This exists only to prevent XML documentation errors.</synopsis>
666                                 <configOption name="external_media_address">
667                                         <synopsis>I should be undocumented or hidden</synopsis>
668                                 </configOption>
669                                 <configOption name="method">
670                                         <synopsis>I should be undocumented or hidden</synopsis>
671                                 </configOption>
672                         </configObject>
673                         <configObject name="domain_alias">
674                                 <synopsis>Domain Alias</synopsis>
675                                 <description><para>
676                                         Signifies that a domain is an alias. Used for checking the domain of
677                                         the AoR to which the endpoint is binding.
678                                 </para></description>
679                                 <configOption name="type">
680                                         <synopsis>Must be of type 'domain_alias'.</synopsis>
681                                 </configOption>
682                                 <configOption name="domain">
683                                         <synopsis>Domain to be aliased</synopsis>
684                                 </configOption>
685                         </configObject>
686                         <configObject name="transport">
687                                 <synopsis>SIP Transport</synopsis>
688                                 <description><para>
689                                         <emphasis>Transports</emphasis>
690                                         </para>
691                                         <para>There are different transports and protocol derivatives
692                                                 supported by <literal>res_pjsip</literal>. They are in order of
693                                                 preference: UDP, TCP, and WebSocket (WS).</para>
694                                         <note><para>Changes to transport configuration in pjsip.conf will only be
695                                                 effected on a complete restart of Asterisk. A module reload
696                                                 will not suffice.</para></note>
697                                 </description>
698                                 <configOption name="async_operations" default="1">
699                                         <synopsis>Number of simultaneous Asynchronous Operations</synopsis>
700                                 </configOption>
701                                 <configOption name="bind">
702                                         <synopsis>IP Address and optional port to bind to for this transport</synopsis>
703                                 </configOption>
704                                 <configOption name="ca_list_file">
705                                         <synopsis>File containing a list of certificates to read (TLS ONLY)</synopsis>
706                                 </configOption>
707                                 <configOption name="cert_file">
708                                         <synopsis>Certificate file for endpoint (TLS ONLY)</synopsis>
709                                 </configOption>
710                                 <configOption name="cipher">
711                                         <synopsis>Preferred Cryptography Cipher (TLS ONLY)</synopsis>
712                                         <description><para>
713                                                 Many options for acceptable ciphers see link for more:
714                                                 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
715                                         </para></description>
716                                 </configOption>
717                                 <configOption name="domain">
718                                         <synopsis>Domain the transport comes from</synopsis>
719                                 </configOption>
720                                 <configOption name="external_media_address">
721                                         <synopsis>External Address to use in RTP handling</synopsis>
722                                 </configOption>
723                                 <configOption name="external_signaling_address">
724                                         <synopsis>External address for SIP signalling</synopsis>
725                                 </configOption>
726                                 <configOption name="external_signaling_port" default="0">
727                                         <synopsis>External port for SIP signalling</synopsis>
728                                 </configOption>
729                                 <configOption name="method">
730                                         <synopsis>Method of SSL transport (TLS ONLY)</synopsis>
731                                         <description>
732                                                 <enumlist>
733                                                         <enum name="default" />
734                                                         <enum name="unspecified" />
735                                                         <enum name="tlsv1" />
736                                                         <enum name="sslv2" />
737                                                         <enum name="sslv3" />
738                                                         <enum name="sslv23" />
739                                                 </enumlist>
740                                         </description>
741                                 </configOption>
742                                 <configOption name="localnet">
743                                         <synopsis>Network to consider local (used for NAT purposes).</synopsis>
744                                         <description><para>This must be in CIDR or dotted decimal format with the IP
745                                         and mask separated with a slash ('/').</para></description>
746                                 </configOption>
747                                 <configOption name="password">
748                                         <synopsis>Password required for transport</synopsis>
749                                 </configOption>
750                                 <configOption name="privkey_file">
751                                         <synopsis>Private key file (TLS ONLY)</synopsis>
752                                 </configOption>
753                                 <configOption name="protocol" default="udp">
754                                         <synopsis>Protocol to use for SIP traffic</synopsis>
755                                         <description>
756                                                 <enumlist>
757                                                         <enum name="udp" />
758                                                         <enum name="tcp" />
759                                                         <enum name="tls" />
760                                                 </enumlist>
761                                         </description>
762                                 </configOption>
763                                 <configOption name="require_client_cert" default="false">
764                                         <synopsis>Require client certificate (TLS ONLY)</synopsis>
765                                 </configOption>
766                                 <configOption name="type">
767                                         <synopsis>Must be of type 'transport'.</synopsis>
768                                 </configOption>
769                                 <configOption name="verify_client" default="false">
770                                         <synopsis>Require verification of client certificate (TLS ONLY)</synopsis>
771                                 </configOption>
772                                 <configOption name="verify_server" default="false">
773                                         <synopsis>Require verification of server certificate (TLS ONLY)</synopsis>
774                                 </configOption>
775                         </configObject>
776                         <configObject name="contact">
777                                 <synopsis>A way of creating an aliased name to a SIP URI</synopsis>
778                                 <description><para>
779                                         Contacts are a way to hide SIP URIs from the dialplan directly.
780                                         They are also used to make a group of contactable parties when
781                                         in use with <literal>AoR</literal> lists.
782                                 </para></description>
783                                 <configOption name="type">
784                                         <synopsis>Must be of type 'contact'.</synopsis>
785                                 </configOption>
786                                 <configOption name="uri">
787                                         <synopsis>SIP URI to contact peer</synopsis>
788                                 </configOption>
789                                 <configOption name="expiration_time">
790                                         <synopsis>Time to keep alive a contact</synopsis>
791                                         <description><para>
792                                                 Time to keep alive a contact. String style specification.
793                                         </para></description>
794                                 </configOption>
795                                 <configOption name="qualify_frequency" default="0">
796                                         <synopsis>Interval at which to qualify a contact</synopsis>
797                                         <description><para>
798                                                 Interval between attempts to qualify the contact for reachability.
799                                                 If <literal>0</literal> never qualify. Time in seconds.
800                                         </para></description>
801                                 </configOption>
802                         </configObject>
803                         <configObject name="contact_status">
804                                 <synopsis>Status for a contact</synopsis>
805                                 <description><para>
806                                         The contact status keeps track of whether or not a contact is reachable
807                                         and how long it took to qualify the contact (round trip time).
808                                 </para></description>
809                                 <configOption name="status">
810                                         <synopsis>A contact's status</synopsis>
811                                         <description>
812                                                 <enumlist>
813                                                         <enum name="AVAILABLE" />
814                                                         <enum name="UNAVAILABLE" />
815                                                 </enumlist>
816                                         </description>
817                                 </configOption>
818                                 <configOption name="rtt">
819                                         <synopsis>Round trip time</synopsis>
820                                         <description><para>
821                                                 The time, in microseconds, it took to qualify the contact.
822                                         </para></description>
823                                 </configOption>
824                         </configObject>
825                         <configObject name="aor">
826                                 <synopsis>The configuration for a location of an endpoint</synopsis>
827                                 <description><para>
828                                         An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
829                                         AoRs are specified, an endpoint will not be reachable by Asterisk.
830                                         Beyond that, an AoR has other uses within Asterisk.
831                                         </para><para>
832                                         An <literal>AoR</literal> is a way to allow dialing a group
833                                         of <literal>Contacts</literal> that all use the same
834                                         <literal>endpoint</literal> for calls.
835                                         </para><para>
836                                         This can be used as another way of grouping a list of contacts to dial
837                                         rather than specifing them each directly when dialing via the dialplan.
838                                         This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
839                                 </para></description>
840                                 <configOption name="contact">
841                                         <synopsis>Permanent contacts assigned to AoR</synopsis>
842                                         <description><para>
843                                                 Contacts included in this list will be called whenever referenced
844                                                 by <literal>chan_pjsip</literal>.
845                                         </para></description>
846                                 </configOption>
847                                 <configOption name="default_expiration" default="3600">
848                                         <synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis>
849                                 </configOption>
850                                 <configOption name="mailboxes">
851                                         <synopsis>Mailbox(es) to be associated with</synopsis>
852                                         <description><para>This option applies when an external entity subscribes to an AoR
853                                         for message waiting indications. The mailboxes specified here will be
854                                         subscribed to.</para></description>
855                                 </configOption>
856                                 <configOption name="maximum_expiration" default="7200">
857                                         <synopsis>Maximum time to keep an AoR</synopsis>
858                                         <description><para>
859                                                 Maximium time to keep a peer with explicit expiration. Time in seconds.
860                                         </para></description>
861                                 </configOption>
862                                 <configOption name="max_contacts" default="0">
863                                         <synopsis>Maximum number of contacts that can bind to an AoR</synopsis>
864                                         <description><para>
865                                                 Maximum number of contacts that can associate with this AoR.
866                                                 </para>
867                                                 <note><para>This should be set to <literal>1</literal> and
868                                                 <replaceable>remove_existing</replaceable> set to <literal>yes</literal> if you
869                                                 wish to stick with the older <literal>chan_sip</literal> behaviour.
870                                                 </para></note>
871                                         </description>
872                                 </configOption>
873                                 <configOption name="minimum_expiration" default="60">
874                                         <synopsis>Minimum keep alive time for an AoR</synopsis>
875                                         <description><para>
876                                                 Minimum time to keep a peer with an explict expiration. Time in seconds.
877                                         </para></description>
878                                 </configOption>
879                                 <configOption name="remove_existing" default="no">
880                                         <synopsis>Determines whether new contacts replace existing ones.</synopsis>
881                                         <description><para>
882                                                 On receiving a new registration to the AoR should it remove
883                                                 the existing contact that was registered against it?
884                                                 </para>
885                                                 <note><para>This should be set to <literal>yes</literal> and
886                                                 <replaceable>max_contacts</replaceable> set to <literal>1</literal> if you
887                                                 wish to stick with the older <literal>chan_sip</literal> behaviour.
888                                                 </para></note>
889                                         </description>
890                                 </configOption>
891                                 <configOption name="type">
892                                         <synopsis>Must be of type 'aor'.</synopsis>
893                                 </configOption>
894                                 <configOption name="qualify_frequency" default="0">
895                                         <synopsis>Interval at which to qualify an AoR</synopsis>
896                                         <description><para>
897                                                 Interval between attempts to qualify the AoR for reachability.
898                                                 If <literal>0</literal> never qualify. Time in seconds.
899                                         </para></description>
900                                 </configOption>
901                                 <configOption name="authenticate_qualify" default="no">
902                                         <synopsis>Authenticates a qualify request if needed</synopsis>
903                                         <description><para>
904                                                 If true and a qualify request receives a challenge or authenticate response
905                                                 authentication is attempted before declaring the contact available.
906                                         </para></description>
907                                 </configOption>
908                         </configObject>
909                         <configObject name="system">
910                                 <synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis>
911                                 <description><para>
912                                         The settings in this section are global. In addition to being global, the values will
913                                         not be re-evaluated when a reload is performed. This is because the values must be set
914                                         before the SIP stack is initialized. The only way to reset these values is to either 
915                                         restart Asterisk, or unload res_pjsip.so and then load it again.
916                                 </para></description>
917                                 <configOption name="timert1" default="500">
918                                         <synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
919                                         <description><para>
920                                                 Timer T1 is the base for determining how long to wait before retransmitting
921                                                 requests that receive no response when using an unreliable transport (e.g. UDP).
922                                                 For more information on this timer, see RFC 3261, Section 17.1.1.1.
923                                         </para></description>
924                                 </configOption>
925                                 <configOption name="timerb" default="32000">
926                                         <synopsis>Set transaction timer B value (milliseconds).</synopsis>
927                                         <description><para>
928                                                 Timer B determines the maximum amount of time to wait after sending an INVITE
929                                                 request before terminating the transaction. It is recommended that this be set
930                                                 to 64 * Timer T1, but it may be set higher if desired. For more information on
931                                                 this timer, see RFC 3261, Section 17.1.1.1.
932                                         </para></description>
933                                 </configOption>
934                                 <configOption name="compactheaders" default="no">
935                                         <synopsis>Use the short forms of common SIP header names.</synopsis>
936                                 </configOption>
937                                 <configOption name="threadpool_initial_size" default="0">
938                                         <synopsis>Initial number of threads in the res_pjsip threadpool.</synopsis>
939                                 </configOption>
940                                 <configOption name="threadpool_auto_increment" default="5">
941                                         <synopsis>The amount by which the number of threads is incremented when necessary.</synopsis>
942                                 </configOption>
943                                 <configOption name="threadpool_idle_timeout" default="60">
944                                         <synopsis>Number of seconds before an idle thread should be disposed of.</synopsis>
945                                 </configOption>
946                                 <configOption name="threadpool_max_size" default="0">
947                                         <synopsis>Maximum number of threads in the res_pjsip threadpool.
948                                         A value of 0 indicates no maximum.</synopsis>
949                                 </configOption>
950                         </configObject>
951                         <configObject name="global">
952                                 <synopsis>Options that apply globally to all SIP communications</synopsis>
953                                 <description><para>
954                                         The settings in this section are global. Unlike options in the <literal>system</literal>
955                                         section, these options can be refreshed by performing a reload.
956                                 </para></description>
957                                 <configOption name="maxforwards" default="70">
958                                         <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
959                                 </configOption>
960                                 <configOption name="useragent" default="Asterisk &lt;Asterisk Version&gt;">
961                                         <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
962                                 </configOption>
963                         </configObject>
964                 </configFile>
965         </configInfo>
966         <manager name="PJSIPQualify" language="en_US">
967                 <synopsis>
968                         Qualify a chan_pjsip endpoint.
969                 </synopsis>
970                 <syntax>
971                         <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
972                         <parameter name="Endpoint" required="true">
973                                 <para>The endpoint you want to qualify.</para>
974                         </parameter>
975                 </syntax>
976                 <description>
977                         <para>Qualify a chan_pjsip endpoint.</para>
978                 </description>
979         </manager>
980  ***/
981
982
983 static pjsip_endpoint *ast_pjsip_endpoint;
984
985 static struct ast_threadpool *sip_threadpool;
986
987 static int register_service(void *data)
988 {
989         pjsip_module **module = data;
990         if (!ast_pjsip_endpoint) {
991                 ast_log(LOG_ERROR, "There is no PJSIP endpoint. Unable to register services\n");
992                 return -1;
993         }
994         if (pjsip_endpt_register_module(ast_pjsip_endpoint, *module) != PJ_SUCCESS) {
995                 ast_log(LOG_ERROR, "Unable to register module %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
996                 return -1;
997         }
998         ast_debug(1, "Registered SIP service %.*s (%p)\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name), *module);
999         ast_module_ref(ast_module_info->self);
1000         return 0;
1001 }
1002
1003 int ast_sip_register_service(pjsip_module *module)
1004 {
1005         return ast_sip_push_task_synchronous(NULL, register_service, &module);
1006 }
1007
1008 static int unregister_service(void *data)
1009 {
1010         pjsip_module **module = data;
1011         ast_module_unref(ast_module_info->self);
1012         if (!ast_pjsip_endpoint) {
1013                 return -1;
1014         }
1015         pjsip_endpt_unregister_module(ast_pjsip_endpoint, *module);
1016         ast_debug(1, "Unregistered SIP service %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1017         return 0;
1018 }
1019
1020 void ast_sip_unregister_service(pjsip_module *module)
1021 {
1022         ast_sip_push_task_synchronous(NULL, unregister_service, &module);
1023 }
1024
1025 static struct ast_sip_authenticator *registered_authenticator;
1026
1027 int ast_sip_register_authenticator(struct ast_sip_authenticator *auth)
1028 {
1029         if (registered_authenticator) {
1030                 ast_log(LOG_WARNING, "Authenticator %p is already registered. Cannot register a new one\n", registered_authenticator);
1031                 return -1;
1032         }
1033         registered_authenticator = auth;
1034         ast_debug(1, "Registered SIP authenticator module %p\n", auth);
1035         ast_module_ref(ast_module_info->self);
1036         return 0;
1037 }
1038
1039 void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth)
1040 {
1041         if (registered_authenticator != auth) {
1042                 ast_log(LOG_WARNING, "Trying to unregister authenticator %p but authenticator %p registered\n",
1043                                 auth, registered_authenticator);
1044                 return;
1045         }
1046         registered_authenticator = NULL;
1047         ast_debug(1, "Unregistered SIP authenticator %p\n", auth);
1048         ast_module_unref(ast_module_info->self);
1049 }
1050
1051 int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1052 {
1053         if (!registered_authenticator) {
1054                 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is not required\n");
1055                 return 0;
1056         }
1057
1058         return registered_authenticator->requires_authentication(endpoint, rdata);
1059 }
1060
1061 enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint,
1062                 pjsip_rx_data *rdata, pjsip_tx_data *tdata)
1063 {
1064         if (!registered_authenticator) {
1065                 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is successful\n");
1066                 return 0;
1067         }
1068         return registered_authenticator->check_authentication(endpoint, rdata, tdata);
1069 }
1070
1071 static struct ast_sip_outbound_authenticator *registered_outbound_authenticator;
1072
1073 int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1074 {
1075         if (registered_outbound_authenticator) {
1076                 ast_log(LOG_WARNING, "Outbound authenticator %p is already registered. Cannot register a new one\n", registered_outbound_authenticator);
1077                 return -1;
1078         }
1079         registered_outbound_authenticator = auth;
1080         ast_debug(1, "Registered SIP outbound authenticator module %p\n", auth);
1081         ast_module_ref(ast_module_info->self);
1082         return 0;
1083 }
1084
1085 void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1086 {
1087         if (registered_outbound_authenticator != auth) {
1088                 ast_log(LOG_WARNING, "Trying to unregister outbound authenticator %p but outbound authenticator %p registered\n",
1089                                 auth, registered_outbound_authenticator);
1090                 return;
1091         }
1092         registered_outbound_authenticator = NULL;
1093         ast_debug(1, "Unregistered SIP outbound authenticator %p\n", auth);
1094         ast_module_unref(ast_module_info->self);
1095 }
1096
1097 int ast_sip_create_request_with_auth(const struct ast_sip_auth_array *auths, pjsip_rx_data *challenge,
1098                 pjsip_transaction *tsx, pjsip_tx_data **new_request)
1099 {
1100         if (!registered_outbound_authenticator) {
1101                 ast_log(LOG_WARNING, "No SIP outbound authenticator registered. Cannot respond to authentication challenge\n");
1102                 return -1;
1103         }
1104         return registered_outbound_authenticator->create_request_with_auth(auths, challenge, tsx, new_request);
1105 }
1106
1107 struct endpoint_identifier_list {
1108         struct ast_sip_endpoint_identifier *identifier;
1109         AST_RWLIST_ENTRY(endpoint_identifier_list) list;
1110 };
1111
1112 static AST_RWLIST_HEAD_STATIC(endpoint_identifiers, endpoint_identifier_list);
1113
1114 int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1115 {
1116         struct endpoint_identifier_list *id_list_item;
1117         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1118
1119         id_list_item = ast_calloc(1, sizeof(*id_list_item));
1120         if (!id_list_item) {
1121                 ast_log(LOG_ERROR, "Unabled to add endpoint identifier. Out of memory.\n");
1122                 return -1;
1123         }
1124         id_list_item->identifier = identifier;
1125
1126         AST_RWLIST_INSERT_TAIL(&endpoint_identifiers, id_list_item, list);
1127         ast_debug(1, "Registered endpoint identifier %p\n", identifier);
1128
1129         ast_module_ref(ast_module_info->self);
1130         return 0;
1131 }
1132
1133 void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1134 {
1135         struct endpoint_identifier_list *iter;
1136         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1137         AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_identifiers, iter, list) {
1138                 if (iter->identifier == identifier) {
1139                         AST_RWLIST_REMOVE_CURRENT(list);
1140                         ast_free(iter);
1141                         ast_debug(1, "Unregistered endpoint identifier %p\n", identifier);
1142                         ast_module_unref(ast_module_info->self);
1143                         break;
1144                 }
1145         }
1146         AST_RWLIST_TRAVERSE_SAFE_END;
1147 }
1148
1149 struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata)
1150 {
1151         struct endpoint_identifier_list *iter;
1152         struct ast_sip_endpoint *endpoint = NULL;
1153         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1154         AST_RWLIST_TRAVERSE(&endpoint_identifiers, iter, list) {
1155                 ast_assert(iter->identifier->identify_endpoint != NULL);
1156                 endpoint = iter->identifier->identify_endpoint(rdata);
1157                 if (endpoint) {
1158                         break;
1159                 }
1160         }
1161         return endpoint;
1162 }
1163
1164 pjsip_endpoint *ast_sip_get_pjsip_endpoint(void)
1165 {
1166         return ast_pjsip_endpoint;
1167 }
1168
1169 static int sip_dialog_create_from(pj_pool_t *pool, pj_str_t *from, const char *user, const char *domain, const pj_str_t *target, pjsip_tpselector *selector)
1170 {
1171         pj_str_t tmp, local_addr;
1172         pjsip_uri *uri;
1173         pjsip_sip_uri *sip_uri;
1174         pjsip_transport_type_e type = PJSIP_TRANSPORT_UNSPECIFIED;
1175         int local_port;
1176         char uuid_str[AST_UUID_STR_LEN];
1177
1178         if (ast_strlen_zero(user)) {
1179                 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1180                 if (!uuid) {
1181                         return -1;
1182                 }
1183                 user = ast_uuid_to_str(uuid, uuid_str, sizeof(uuid_str));
1184         }
1185
1186         /* Parse the provided target URI so we can determine what transport it will end up using */
1187         pj_strdup_with_null(pool, &tmp, target);
1188
1189         if (!(uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0)) ||
1190             (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
1191                 return -1;
1192         }
1193
1194         sip_uri = pjsip_uri_get_uri(uri);
1195
1196         /* Determine the transport type to use */
1197         if (PJSIP_URI_SCHEME_IS_SIPS(sip_uri)) {
1198                 type = PJSIP_TRANSPORT_TLS;
1199         } else if (!sip_uri->transport_param.slen) {
1200                 type = PJSIP_TRANSPORT_UDP;
1201         } else {
1202                 type = pjsip_transport_get_type_from_name(&sip_uri->transport_param);
1203         }
1204
1205         if (type == PJSIP_TRANSPORT_UNSPECIFIED) {
1206                 return -1;
1207         }
1208
1209         /* If the host is IPv6 turn the transport into an IPv6 version */
1210         if (pj_strchr(&sip_uri->host, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1211                 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1212         }
1213
1214         if (!ast_strlen_zero(domain)) {
1215                 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1216                 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1217                                 "<%s:%s@%s%s%s>",
1218                                 (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1219                                 user,
1220                                 domain,
1221                                 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1222                                 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1223                 return 0;
1224         }
1225
1226         /* Get the local bound address for the transport that will be used when communicating with the provided URI */
1227         if (pjsip_tpmgr_find_local_addr(pjsip_endpt_get_tpmgr(ast_sip_get_pjsip_endpoint()), pool, type, selector,
1228                                                               &local_addr, &local_port) != PJ_SUCCESS) {
1229                 return -1;
1230         }
1231
1232         /* If IPv6 was specified in the transport, set the proper type */
1233         if (pj_strchr(&local_addr, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1234                 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1235         }
1236
1237         from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1238         from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1239                                       "<%s:%s@%s%.*s%s:%d%s%s>",
1240                                       (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1241                                       user,
1242                                       (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1243                                       (int)local_addr.slen,
1244                                       local_addr.ptr,
1245                                       (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1246                                       local_port,
1247                                       (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1248                                       (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1249
1250         return 0;
1251 }
1252
1253 static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpoint, pjsip_tpselector *selector)
1254 {
1255         RAII_VAR(struct ast_sip_transport *, transport, NULL, ao2_cleanup);
1256         const char *transport_name = endpoint->transport;
1257
1258         if (ast_strlen_zero(transport_name)) {
1259                 return 0;
1260         }
1261
1262         transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_name);
1263
1264         if (!transport || !transport->state) {
1265                 return -1;
1266         }
1267
1268         if (transport->state->transport) {
1269                 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1270                 selector->u.transport = transport->state->transport;
1271         } else if (transport->state->factory) {
1272                 selector->type = PJSIP_TPSELECTOR_LISTENER;
1273                 selector->u.listener = transport->state->factory;
1274         } else {
1275                 return -1;
1276         }
1277
1278         return 0;
1279 }
1280
1281 static int sip_get_tpselector_from_uri(const char *uri, pjsip_tpselector *selector)
1282 {
1283         RAII_VAR(struct ast_sip_contact_transport *, contact_transport, NULL, ao2_cleanup);
1284
1285         contact_transport = ast_sip_location_retrieve_contact_transport_by_uri(uri);
1286
1287         if (!contact_transport) {
1288                 return -1;
1289         }
1290
1291         selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1292         selector->u.transport = contact_transport->transport;
1293
1294         return 0;
1295 }
1296
1297 pjsip_dialog *ast_sip_create_dialog(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user)
1298 {
1299         pj_str_t local_uri = { "sip:temp@temp", 13 }, remote_uri;
1300         pjsip_dialog *dlg = NULL;
1301         const char *outbound_proxy = endpoint->outbound_proxy;
1302         pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1303         static const pj_str_t HCONTACT = { "Contact", 7 };
1304
1305         pj_cstr(&remote_uri, uri);
1306
1307         if (pjsip_dlg_create_uac(pjsip_ua_instance(), &local_uri, NULL, &remote_uri, NULL, &dlg) != PJ_SUCCESS) {
1308                 return NULL;
1309         }
1310
1311         if (sip_get_tpselector_from_uri(uri, &selector) && sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1312                 pjsip_dlg_terminate(dlg);
1313                 return NULL;
1314         }
1315
1316         if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
1317                 pjsip_dlg_terminate(dlg);
1318                 return NULL;
1319         }
1320
1321         /* Update the dialog with the new local URI, we do it afterwards so we can use the dialog pool for construction */
1322         pj_strdup_with_null(dlg->pool, &dlg->local.info_str, &local_uri);
1323         dlg->local.info->uri = pjsip_parse_uri(dlg->pool, dlg->local.info_str.ptr, dlg->local.info_str.slen, 0);
1324         dlg->local.contact = pjsip_parse_hdr(dlg->pool, &HCONTACT, local_uri.ptr, local_uri.slen, NULL);
1325
1326         /* If a request user has been specified and we are permitted to change it, do so */
1327         if (!ast_strlen_zero(request_user) && (PJSIP_URI_SCHEME_IS_SIP(dlg->target) || PJSIP_URI_SCHEME_IS_SIPS(dlg->target))) {
1328                 pjsip_sip_uri *target = pjsip_uri_get_uri(dlg->target);
1329                 pj_strdup2(dlg->pool, &target->user, request_user);
1330         }
1331
1332         /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
1333         dlg->sess_count++;
1334
1335         pjsip_dlg_set_transport(dlg, &selector);
1336
1337         if (!ast_strlen_zero(outbound_proxy)) {
1338                 pjsip_route_hdr route_set, *route;
1339                 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1340                 pj_str_t tmp;
1341
1342                 pj_list_init(&route_set);
1343
1344                 pj_strdup2_with_null(dlg->pool, &tmp, outbound_proxy);
1345                 if (!(route = pjsip_parse_hdr(dlg->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1346                         pjsip_dlg_terminate(dlg);
1347                         return NULL;
1348                 }
1349                 pj_list_push_back(&route_set, route);
1350
1351                 pjsip_dlg_set_route_set(dlg, &route_set);
1352         }
1353
1354         dlg->sess_count--;
1355
1356         return dlg;
1357 }
1358
1359 /* PJSIP doesn't know about the INFO method, so we have to define it ourselves */
1360 const pjsip_method pjsip_info_method = {PJSIP_OTHER_METHOD, {"INFO", 4} };
1361 const pjsip_method pjsip_message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
1362
1363 static struct {
1364         const char *method;
1365         const pjsip_method *pmethod;
1366 } methods [] = {
1367         { "INVITE", &pjsip_invite_method },
1368         { "CANCEL", &pjsip_cancel_method },
1369         { "ACK", &pjsip_ack_method },
1370         { "BYE", &pjsip_bye_method },
1371         { "REGISTER", &pjsip_register_method },
1372         { "OPTIONS", &pjsip_options_method },
1373         { "SUBSCRIBE", &pjsip_subscribe_method },
1374         { "NOTIFY", &pjsip_notify_method },
1375         { "PUBLISH", &pjsip_publish_method },
1376         { "INFO", &pjsip_info_method },
1377         { "MESSAGE", &pjsip_message_method },
1378 };
1379
1380 static const pjsip_method *get_pjsip_method(const char *method)
1381 {
1382         int i;
1383         for (i = 0; i < ARRAY_LEN(methods); ++i) {
1384                 if (!strcmp(method, methods[i].method)) {
1385                         return methods[i].pmethod;
1386                 }
1387         }
1388         return NULL;
1389 }
1390
1391 static int create_in_dialog_request(const pjsip_method *method, struct pjsip_dialog *dlg, pjsip_tx_data **tdata)
1392 {
1393         if (pjsip_dlg_create_request(dlg, method, -1, tdata) != PJ_SUCCESS) {
1394                 ast_log(LOG_WARNING, "Unable to create in-dialog request.\n");
1395                 return -1;
1396         }
1397
1398         return 0;
1399 }
1400
1401 static int create_out_of_dialog_request(const pjsip_method *method, struct ast_sip_endpoint *endpoint,
1402                 const char *uri, pjsip_tx_data **tdata)
1403 {
1404         RAII_VAR(struct ast_sip_contact *, contact, NULL, ao2_cleanup);
1405         pj_str_t remote_uri;
1406         pj_str_t from;
1407         pj_pool_t *pool;
1408         pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1409
1410         if (ast_strlen_zero(uri)) {
1411                 if (!endpoint) {
1412                         ast_log(LOG_ERROR, "An endpoint and/or uri must be specified\n");
1413                         return -1;
1414                 }
1415
1416                 contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
1417                 if (!contact || ast_strlen_zero(contact->uri)) {
1418                         ast_log(LOG_ERROR, "Unable to retrieve contact for endpoint %s\n",
1419                                         ast_sorcery_object_get_id(endpoint));
1420                         return -1;
1421                 }
1422
1423                 pj_cstr(&remote_uri, contact->uri);
1424         } else {
1425                 pj_cstr(&remote_uri, uri);
1426         }
1427
1428         if (endpoint) {
1429                 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1430                         ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport selector for endpoint %s\n",
1431                                 ast_sorcery_object_get_id(endpoint));
1432                         return -1;
1433                 }
1434         }
1435
1436         pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "Outbound request", 256, 256);
1437
1438         if (!pool) {
1439                 ast_log(LOG_ERROR, "Unable to create PJLIB memory pool\n");
1440                 return -1;
1441         }
1442
1443         if (sip_dialog_create_from(pool, &from, endpoint ? endpoint->fromuser : NULL,
1444                                 endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) {
1445                 ast_log(LOG_ERROR, "Unable to create From header for %.*s request to endpoint %s\n",
1446                                 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1447                 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1448                 return -1;
1449         }
1450
1451         if (pjsip_endpt_create_request(ast_sip_get_pjsip_endpoint(), method, &remote_uri,
1452                         &from, &remote_uri, &from, NULL, -1, NULL, tdata) != PJ_SUCCESS) {
1453                 ast_log(LOG_ERROR, "Unable to create outbound %.*s request to endpoint %s\n",
1454                                 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1455                 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1456                 return -1;
1457         }
1458
1459         /* We can release this pool since request creation copied all the necessary
1460          * data into the outbound request's pool
1461          */
1462         pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1463         return 0;
1464 }
1465
1466 int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
1467                 struct ast_sip_endpoint *endpoint, const char *uri,
1468                 pjsip_tx_data **tdata)
1469 {
1470         const pjsip_method *pmethod = get_pjsip_method(method);
1471
1472         if (!pmethod) {
1473                 ast_log(LOG_WARNING, "Unknown method '%s'. Cannot send request\n", method);
1474                 return -1;
1475         }
1476
1477         if (dlg) {
1478                 return create_in_dialog_request(pmethod, dlg, tdata);
1479         } else {
1480                 return create_out_of_dialog_request(pmethod, endpoint, uri, tdata);
1481         }
1482 }
1483
1484 static int send_in_dialog_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg)
1485 {
1486         if (pjsip_dlg_send_request(dlg, tdata, -1, NULL) != PJ_SUCCESS) {
1487                 ast_log(LOG_WARNING, "Unable to send in-dialog request.\n");
1488                 return -1;
1489         }
1490         return 0;
1491 }
1492
1493 static void send_request_cb(void *token, pjsip_event *e)
1494 {
1495         RAII_VAR(struct ast_sip_endpoint *, endpoint, token, ao2_cleanup);
1496         pjsip_transaction *tsx = e->body.tsx_state.tsx;
1497         pjsip_rx_data *challenge = e->body.tsx_state.src.rdata;
1498         pjsip_tx_data *tdata;
1499
1500         if (tsx->status_code != 401 && tsx->status_code != 407) {
1501                 return;
1502         }
1503
1504         if (!ast_sip_create_request_with_auth(&endpoint->outbound_auths, challenge, tsx, &tdata)) {
1505                 pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, NULL, NULL);
1506         }
1507 }
1508
1509 static int send_out_of_dialog_request(pjsip_tx_data *tdata, struct ast_sip_endpoint *endpoint)
1510 {
1511         ao2_ref(endpoint, +1);
1512         if (pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, endpoint, send_request_cb) != PJ_SUCCESS) {
1513                 ast_log(LOG_ERROR, "Error attempting to send outbound %.*s request to endpoint %s\n",
1514                                 (int) pj_strlen(&tdata->msg->line.req.method.name),
1515                                 pj_strbuf(&tdata->msg->line.req.method.name),
1516                                 ast_sorcery_object_get_id(endpoint));
1517                 ao2_ref(endpoint, -1);
1518                 return -1;
1519         }
1520
1521         return 0;
1522 }
1523
1524 int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg, struct ast_sip_endpoint *endpoint)
1525 {
1526         ast_assert(tdata->msg->type == PJSIP_REQUEST_MSG);
1527
1528         if (dlg) {
1529                 return send_in_dialog_request(tdata, dlg);
1530         } else {
1531                 return send_out_of_dialog_request(tdata, endpoint);
1532         }
1533 }
1534
1535 int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value)
1536 {
1537         pj_str_t hdr_name;
1538         pj_str_t hdr_value;
1539         pjsip_generic_string_hdr *hdr;
1540
1541         pj_cstr(&hdr_name, name);
1542         pj_cstr(&hdr_value, value);
1543
1544         hdr = pjsip_generic_string_hdr_create(tdata->pool, &hdr_name, &hdr_value);
1545
1546         pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *) hdr);
1547         return 0;
1548 }
1549
1550 static pjsip_msg_body *ast_body_to_pjsip_body(pj_pool_t *pool, const struct ast_sip_body *body)
1551 {
1552         pj_str_t type;
1553         pj_str_t subtype;
1554         pj_str_t body_text;
1555
1556         pj_cstr(&type, body->type);
1557         pj_cstr(&subtype, body->subtype);
1558         pj_cstr(&body_text, body->body_text);
1559
1560         return pjsip_msg_body_create(pool, &type, &subtype, &body_text);
1561 }
1562
1563 int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body)
1564 {
1565         pjsip_msg_body *pjsip_body = ast_body_to_pjsip_body(tdata->pool, body);
1566         tdata->msg->body = pjsip_body;
1567         return 0;
1568 }
1569
1570 int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies)
1571 {
1572         int i;
1573         /* NULL for type and subtype automatically creates "multipart/mixed" */
1574         pjsip_msg_body *body = pjsip_multipart_create(tdata->pool, NULL, NULL);
1575
1576         for (i = 0; i < num_bodies; ++i) {
1577                 pjsip_multipart_part *part = pjsip_multipart_create_part(tdata->pool);
1578                 part->body = ast_body_to_pjsip_body(tdata->pool, bodies[i]);
1579                 pjsip_multipart_add_part(tdata->pool, body, part);
1580         }
1581
1582         tdata->msg->body = body;
1583         return 0;
1584 }
1585
1586 int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text)
1587 {
1588         size_t combined_size = strlen(body_text) + tdata->msg->body->len;
1589         struct ast_str *body_buffer = ast_str_alloca(combined_size);
1590
1591         ast_str_set(&body_buffer, 0, "%.*s%s", (int) tdata->msg->body->len, (char *) tdata->msg->body->data, body_text);
1592
1593         tdata->msg->body->data = pj_pool_alloc(tdata->pool, combined_size);
1594         pj_memcpy(tdata->msg->body->data, ast_str_buffer(body_buffer), combined_size);
1595         tdata->msg->body->len = combined_size;
1596
1597         return 0;
1598 }
1599
1600 struct ast_taskprocessor *ast_sip_create_serializer(void)
1601 {
1602         struct ast_taskprocessor *serializer;
1603         RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1604         char name[AST_UUID_STR_LEN];
1605
1606         if (!uuid) {
1607                 return NULL;
1608         }
1609
1610         ast_uuid_to_str(uuid, name, sizeof(name));
1611
1612         serializer = ast_threadpool_serializer(name, sip_threadpool);
1613         if (!serializer) {
1614                 return NULL;
1615         }
1616         return serializer;
1617 }
1618
1619 int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1620 {
1621         if (serializer) {
1622                 return ast_taskprocessor_push(serializer, sip_task, task_data);
1623         } else {
1624                 return ast_threadpool_push(sip_threadpool, sip_task, task_data);
1625         }
1626 }
1627
1628 struct sync_task_data {
1629         ast_mutex_t lock;
1630         ast_cond_t cond;
1631         int complete;
1632         int fail;
1633         int (*task)(void *);
1634         void *task_data;
1635 };
1636
1637 static int sync_task(void *data)
1638 {
1639         struct sync_task_data *std = data;
1640         std->fail = std->task(std->task_data);
1641
1642         ast_mutex_lock(&std->lock);
1643         std->complete = 1;
1644         ast_cond_signal(&std->cond);
1645         ast_mutex_unlock(&std->lock);
1646         return std->fail;
1647 }
1648
1649 int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1650 {
1651         /* This method is an onion */
1652         struct sync_task_data std;
1653         ast_mutex_init(&std.lock);
1654         ast_cond_init(&std.cond, NULL);
1655         std.fail = std.complete = 0;
1656         std.task = sip_task;
1657         std.task_data = task_data;
1658
1659         if (serializer) {
1660                 if (ast_taskprocessor_push(serializer, sync_task, &std)) {
1661                         return -1;
1662                 }
1663         } else {
1664                 if (ast_threadpool_push(sip_threadpool, sync_task, &std)) {
1665                         return -1;
1666                 }
1667         }
1668
1669         ast_mutex_lock(&std.lock);
1670         while (!std.complete) {
1671                 ast_cond_wait(&std.cond, &std.lock);
1672         }
1673         ast_mutex_unlock(&std.lock);
1674
1675         ast_mutex_destroy(&std.lock);
1676         ast_cond_destroy(&std.cond);
1677         return std.fail;
1678 }
1679
1680 void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size)
1681 {
1682         size_t chars_to_copy = MIN(size - 1, pj_strlen(src));
1683         memcpy(dest, pj_strbuf(src), chars_to_copy);
1684         dest[chars_to_copy] = '\0';
1685 }
1686
1687 int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype)
1688 {
1689         pjsip_media_type compare;
1690
1691         if (!content_type) {
1692                 return 0;
1693         }
1694
1695         pjsip_media_type_init2(&compare, type, subtype);
1696
1697         return pjsip_media_type_cmp(content_type, &compare, 0) ? -1 : 0;
1698 }
1699
1700 pj_caching_pool caching_pool;
1701 pj_pool_t *memory_pool;
1702 pj_thread_t *monitor_thread;
1703 static int monitor_continue;
1704
1705 static void *monitor_thread_exec(void *endpt)
1706 {
1707         while (monitor_continue) {
1708                 const pj_time_val delay = {0, 10};
1709                 pjsip_endpt_handle_events(ast_pjsip_endpoint, &delay);
1710         }
1711         return NULL;
1712 }
1713
1714 static void stop_monitor_thread(void)
1715 {
1716         monitor_continue = 0;
1717         pj_thread_join(monitor_thread);
1718 }
1719
1720 AST_THREADSTORAGE(pj_thread_storage);
1721 AST_THREADSTORAGE(servant_id_storage);
1722 #define SIP_SERVANT_ID 0x5E2F1D
1723
1724 static void sip_thread_start(void)
1725 {
1726         pj_thread_desc *desc;
1727         pj_thread_t *thread;
1728         uint32_t *servant_id;
1729
1730         servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1731         if (!servant_id) {
1732                 ast_log(LOG_ERROR, "Could not set SIP servant ID in thread-local storage.\n");
1733                 return;
1734         }
1735         *servant_id = SIP_SERVANT_ID;
1736
1737         desc = ast_threadstorage_get(&pj_thread_storage, sizeof(pj_thread_desc));
1738         if (!desc) {
1739                 ast_log(LOG_ERROR, "Could not get thread desc from thread-local storage. Expect awful things to occur\n");
1740                 return;
1741         }
1742         pj_bzero(*desc, sizeof(*desc));
1743
1744         if (pj_thread_register("Asterisk Thread", *desc, &thread) != PJ_SUCCESS) {
1745                 ast_log(LOG_ERROR, "Couldn't register thread with PJLIB.\n");
1746         }
1747 }
1748
1749 int ast_sip_thread_is_servant(void)
1750 {
1751         uint32_t *servant_id;
1752
1753         servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1754         if (!servant_id) {
1755                 return 0;
1756         }
1757
1758         return *servant_id == SIP_SERVANT_ID;
1759 }
1760
1761 static void remove_request_headers(pjsip_endpoint *endpt)
1762 {
1763         const pjsip_hdr *request_headers = pjsip_endpt_get_request_headers(endpt);
1764         pjsip_hdr *iter = request_headers->next;
1765
1766         while (iter != request_headers) {
1767                 pjsip_hdr *to_erase = iter;
1768                 iter = iter->next;
1769                 pj_list_erase(to_erase);
1770         }
1771 }
1772
1773 static int load_module(void)
1774 {
1775         /* The third parameter is just copied from
1776          * example code from PJLIB. This can be adjusted
1777          * if necessary.
1778          */
1779         pj_status_t status;
1780         struct ast_threadpool_options options;
1781
1782         if (pj_init() != PJ_SUCCESS) {
1783                 return AST_MODULE_LOAD_DECLINE;
1784         }
1785
1786         if (pjlib_util_init() != PJ_SUCCESS) {
1787                 pj_shutdown();
1788                 return AST_MODULE_LOAD_DECLINE;
1789         }
1790
1791         pj_caching_pool_init(&caching_pool, NULL, 1024 * 1024);
1792         if (pjsip_endpt_create(&caching_pool.factory, "SIP", &ast_pjsip_endpoint) != PJ_SUCCESS) {
1793                 ast_log(LOG_ERROR, "Failed to create PJSIP endpoint structure. Aborting load\n");
1794                 goto error;
1795         }
1796
1797         /* PJSIP will automatically try to add a Max-Forwards header. Since we want to control that,
1798          * we need to stop PJSIP from doing it automatically
1799          */
1800         remove_request_headers(ast_pjsip_endpoint);
1801
1802         memory_pool = pj_pool_create(&caching_pool.factory, "SIP", 1024, 1024, NULL);
1803         if (!memory_pool) {
1804                 ast_log(LOG_ERROR, "Failed to create memory pool for SIP. Aborting load\n");
1805                 goto error;
1806         }
1807
1808         if (ast_sip_initialize_system()) {
1809                 ast_log(LOG_ERROR, "Failed to initialize SIP system configuration. Aborting load\n");
1810                 goto error;
1811         }
1812
1813         sip_get_threadpool_options(&options);
1814         options.thread_start = sip_thread_start;
1815         sip_threadpool = ast_threadpool_create("SIP", NULL, &options);
1816         if (!sip_threadpool) {
1817                 ast_log(LOG_ERROR, "Failed to create SIP threadpool. Aborting load\n");
1818                 goto error;
1819         }
1820
1821         pjsip_tsx_layer_init_module(ast_pjsip_endpoint);
1822         pjsip_ua_init_module(ast_pjsip_endpoint, NULL);
1823
1824         monitor_continue = 1;
1825         status = pj_thread_create(memory_pool, "SIP", (pj_thread_proc *) &monitor_thread_exec,
1826                         NULL, PJ_THREAD_DEFAULT_STACK_SIZE * 2, 0, &monitor_thread);
1827         if (status != PJ_SUCCESS) {
1828                 ast_log(LOG_ERROR, "Failed to start SIP monitor thread. Aborting load\n");
1829                 goto error;
1830         }
1831
1832         ast_sip_initialize_global_headers();
1833
1834         if (ast_res_pjsip_initialize_configuration()) {
1835                 ast_log(LOG_ERROR, "Failed to initialize SIP configuration. Aborting load\n");
1836                 goto error;
1837         }
1838
1839         if (ast_sip_initialize_distributor()) {
1840                 ast_log(LOG_ERROR, "Failed to register distributor module. Aborting load\n");
1841                 goto error;
1842         }
1843
1844         if (ast_sip_initialize_outbound_authentication()) {
1845                 ast_log(LOG_ERROR, "Failed to initialize outbound authentication. Aborting load\n");
1846                 goto error;
1847         }
1848
1849         ast_res_pjsip_init_options_handling(0);
1850
1851         ast_res_pjsip_init_contact_transports();
1852
1853 return AST_MODULE_LOAD_SUCCESS;
1854
1855 error:
1856         ast_sip_destroy_distributor();
1857         ast_res_pjsip_destroy_configuration();
1858         ast_sip_destroy_global_headers();
1859         if (monitor_thread) {
1860                 stop_monitor_thread();
1861         }
1862         if (memory_pool) {
1863                 pj_pool_release(memory_pool);
1864                 memory_pool = NULL;
1865         }
1866         if (ast_pjsip_endpoint) {
1867                 pjsip_endpt_destroy(ast_pjsip_endpoint);
1868                 ast_pjsip_endpoint = NULL;
1869         }
1870         pj_caching_pool_destroy(&caching_pool);
1871         return AST_MODULE_LOAD_DECLINE;
1872 }
1873
1874 static int reload_module(void)
1875 {
1876         if (ast_res_pjsip_reload_configuration()) {
1877                 return AST_MODULE_LOAD_DECLINE;
1878         }
1879         ast_res_pjsip_init_options_handling(1);
1880         return 0;
1881 }
1882
1883 static int unload_pjsip(void *data)
1884 {
1885         if (memory_pool) {
1886                 pj_pool_release(memory_pool);
1887                 memory_pool = NULL;
1888         }
1889         if (ast_pjsip_endpoint) {
1890                 pjsip_endpt_destroy(ast_pjsip_endpoint);
1891                 ast_pjsip_endpoint = NULL;
1892         }
1893         pj_caching_pool_destroy(&caching_pool);
1894         return 0;
1895 }
1896
1897 static int unload_module(void)
1898 {
1899         ast_res_pjsip_cleanup_options_handling();
1900         ast_sip_destroy_distributor();
1901         ast_res_pjsip_destroy_configuration();
1902         ast_sip_destroy_global_headers();
1903         if (monitor_thread) {
1904                 stop_monitor_thread();
1905         }
1906         /* The thread this is called from cannot call PJSIP/PJLIB functions,
1907          * so we have to push the work to the threadpool to handle
1908          */
1909         ast_sip_push_task_synchronous(NULL, unload_pjsip, NULL);
1910
1911         ast_threadpool_shutdown(sip_threadpool);
1912
1913         return 0;
1914 }
1915
1916 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Basic SIP resource",
1917                 .load = load_module,
1918                 .unload = unload_module,
1919                 .reload = reload_module,
1920                 .load_pri = AST_MODPRI_CHANNEL_DEPEND - 5,
1921 );