Expose res_pjsip threadpool options
[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                                         <warning><para>
695                                                 Multiple endpoints using the same connection is <emphasis>NOT</emphasis>
696                                                 supported. Doing so may result in broken calls.
697                                         </para></warning>
698                                 </description>
699                                 <configOption name="async_operations" default="1">
700                                         <synopsis>Number of simultaneous Asynchronous Operations</synopsis>
701                                 </configOption>
702                                 <configOption name="bind">
703                                         <synopsis>IP Address and optional port to bind to for this transport</synopsis>
704                                 </configOption>
705                                 <configOption name="ca_list_file">
706                                         <synopsis>File containing a list of certificates to read (TLS ONLY)</synopsis>
707                                 </configOption>
708                                 <configOption name="cert_file">
709                                         <synopsis>Certificate file for endpoint (TLS ONLY)</synopsis>
710                                 </configOption>
711                                 <configOption name="cipher">
712                                         <synopsis>Preferred Cryptography Cipher (TLS ONLY)</synopsis>
713                                         <description><para>
714                                                 Many options for acceptable ciphers see link for more:
715                                                 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
716                                         </para></description>
717                                 </configOption>
718                                 <configOption name="domain">
719                                         <synopsis>Domain the transport comes from</synopsis>
720                                 </configOption>
721                                 <configOption name="external_media_address">
722                                         <synopsis>External Address to use in RTP handling</synopsis>
723                                 </configOption>
724                                 <configOption name="external_signaling_address">
725                                         <synopsis>External address for SIP signalling</synopsis>
726                                 </configOption>
727                                 <configOption name="external_signaling_port" default="0">
728                                         <synopsis>External port for SIP signalling</synopsis>
729                                 </configOption>
730                                 <configOption name="method">
731                                         <synopsis>Method of SSL transport (TLS ONLY)</synopsis>
732                                         <description>
733                                                 <enumlist>
734                                                         <enum name="default" />
735                                                         <enum name="unspecified" />
736                                                         <enum name="tlsv1" />
737                                                         <enum name="sslv2" />
738                                                         <enum name="sslv3" />
739                                                         <enum name="sslv23" />
740                                                 </enumlist>
741                                         </description>
742                                 </configOption>
743                                 <configOption name="localnet">
744                                         <synopsis>Network to consider local (used for NAT purposes).</synopsis>
745                                         <description><para>This must be in CIDR or dotted decimal format with the IP
746                                         and mask separated with a slash ('/').</para></description>
747                                 </configOption>
748                                 <configOption name="password">
749                                         <synopsis>Password required for transport</synopsis>
750                                 </configOption>
751                                 <configOption name="privkey_file">
752                                         <synopsis>Private key file (TLS ONLY)</synopsis>
753                                 </configOption>
754                                 <configOption name="protocol" default="udp">
755                                         <synopsis>Protocol to use for SIP traffic</synopsis>
756                                         <description>
757                                                 <enumlist>
758                                                         <enum name="udp" />
759                                                         <enum name="tcp" />
760                                                         <enum name="tls" />
761                                                 </enumlist>
762                                         </description>
763                                 </configOption>
764                                 <configOption name="require_client_cert" default="false">
765                                         <synopsis>Require client certificate (TLS ONLY)</synopsis>
766                                 </configOption>
767                                 <configOption name="type">
768                                         <synopsis>Must be of type 'transport'.</synopsis>
769                                 </configOption>
770                                 <configOption name="verify_client" default="false">
771                                         <synopsis>Require verification of client certificate (TLS ONLY)</synopsis>
772                                 </configOption>
773                                 <configOption name="verify_server" default="false">
774                                         <synopsis>Require verification of server certificate (TLS ONLY)</synopsis>
775                                 </configOption>
776                         </configObject>
777                         <configObject name="contact">
778                                 <synopsis>A way of creating an aliased name to a SIP URI</synopsis>
779                                 <description><para>
780                                         Contacts are a way to hide SIP URIs from the dialplan directly.
781                                         They are also used to make a group of contactable parties when
782                                         in use with <literal>AoR</literal> lists.
783                                 </para></description>
784                                 <configOption name="type">
785                                         <synopsis>Must be of type 'contact'.</synopsis>
786                                 </configOption>
787                                 <configOption name="uri">
788                                         <synopsis>SIP URI to contact peer</synopsis>
789                                 </configOption>
790                                 <configOption name="expiration_time">
791                                         <synopsis>Time to keep alive a contact</synopsis>
792                                         <description><para>
793                                                 Time to keep alive a contact. String style specification.
794                                         </para></description>
795                                 </configOption>
796                                 <configOption name="qualify_frequency" default="0">
797                                         <synopsis>Interval at which to qualify a contact</synopsis>
798                                         <description><para>
799                                                 Interval between attempts to qualify the contact for reachability.
800                                                 If <literal>0</literal> never qualify. Time in seconds.
801                                         </para></description>
802                                 </configOption>
803                         </configObject>
804                         <configObject name="contact_status">
805                                 <synopsis>Status for a contact</synopsis>
806                                 <description><para>
807                                         The contact status keeps track of whether or not a contact is reachable
808                                         and how long it took to qualify the contact (round trip time).
809                                 </para></description>
810                                 <configOption name="status">
811                                         <synopsis>A contact's status</synopsis>
812                                         <description>
813                                                 <enumlist>
814                                                         <enum name="AVAILABLE" />
815                                                         <enum name="UNAVAILABLE" />
816                                                 </enumlist>
817                                         </description>
818                                 </configOption>
819                                 <configOption name="rtt">
820                                         <synopsis>Round trip time</synopsis>
821                                         <description><para>
822                                                 The time, in microseconds, it took to qualify the contact.
823                                         </para></description>
824                                 </configOption>
825                         </configObject>
826                         <configObject name="aor">
827                                 <synopsis>The configuration for a location of an endpoint</synopsis>
828                                 <description><para>
829                                         An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
830                                         AoRs are specified, an endpoint will not be reachable by Asterisk.
831                                         Beyond that, an AoR has other uses within Asterisk.
832                                         </para><para>
833                                         An <literal>AoR</literal> is a way to allow dialing a group
834                                         of <literal>Contacts</literal> that all use the same
835                                         <literal>endpoint</literal> for calls.
836                                         </para><para>
837                                         This can be used as another way of grouping a list of contacts to dial
838                                         rather than specifing them each directly when dialing via the dialplan.
839                                         This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
840                                 </para></description>
841                                 <configOption name="contact">
842                                         <synopsis>Permanent contacts assigned to AoR</synopsis>
843                                         <description><para>
844                                                 Contacts included in this list will be called whenever referenced
845                                                 by <literal>chan_pjsip</literal>.
846                                         </para></description>
847                                 </configOption>
848                                 <configOption name="default_expiration" default="3600">
849                                         <synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis>
850                                 </configOption>
851                                 <configOption name="mailboxes">
852                                         <synopsis>Mailbox(es) to be associated with</synopsis>
853                                         <description><para>This option applies when an external entity subscribes to an AoR
854                                         for message waiting indications. The mailboxes specified here will be
855                                         subscribed to.</para></description>
856                                 </configOption>
857                                 <configOption name="maximum_expiration" default="7200">
858                                         <synopsis>Maximum time to keep an AoR</synopsis>
859                                         <description><para>
860                                                 Maximium time to keep a peer with explicit expiration. Time in seconds.
861                                         </para></description>
862                                 </configOption>
863                                 <configOption name="max_contacts" default="0">
864                                         <synopsis>Maximum number of contacts that can bind to an AoR</synopsis>
865                                         <description><para>
866                                                 Maximum number of contacts that can associate with this AoR.
867                                                 </para>
868                                                 <note><para>This should be set to <literal>1</literal> and
869                                                 <replaceable>remove_existing</replaceable> set to <literal>yes</literal> if you
870                                                 wish to stick with the older <literal>chan_sip</literal> behaviour.
871                                                 </para></note>
872                                         </description>
873                                 </configOption>
874                                 <configOption name="minimum_expiration" default="60">
875                                         <synopsis>Minimum keep alive time for an AoR</synopsis>
876                                         <description><para>
877                                                 Minimum time to keep a peer with an explict expiration. Time in seconds.
878                                         </para></description>
879                                 </configOption>
880                                 <configOption name="remove_existing" default="no">
881                                         <synopsis>Determines whether new contacts replace existing ones.</synopsis>
882                                         <description><para>
883                                                 On receiving a new registration to the AoR should it remove
884                                                 the existing contact that was registered against it?
885                                                 </para>
886                                                 <note><para>This should be set to <literal>yes</literal> and
887                                                 <replaceable>max_contacts</replaceable> set to <literal>1</literal> if you
888                                                 wish to stick with the older <literal>chan_sip</literal> behaviour.
889                                                 </para></note>
890                                         </description>
891                                 </configOption>
892                                 <configOption name="type">
893                                         <synopsis>Must be of type 'aor'.</synopsis>
894                                 </configOption>
895                                 <configOption name="qualify_frequency" default="0">
896                                         <synopsis>Interval at which to qualify an AoR</synopsis>
897                                         <description><para>
898                                                 Interval between attempts to qualify the AoR for reachability.
899                                                 If <literal>0</literal> never qualify. Time in seconds.
900                                         </para></description>
901                                 </configOption>
902                                 <configOption name="authenticate_qualify" default="no">
903                                         <synopsis>Authenticates a qualify request if needed</synopsis>
904                                         <description><para>
905                                                 If true and a qualify request receives a challenge or authenticate response
906                                                 authentication is attempted before declaring the contact available.
907                                         </para></description>
908                                 </configOption>
909                         </configObject>
910                         <configObject name="system">
911                                 <synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis>
912                                 <description><para>
913                                         The settings in this section are global. In addition to being global, the values will
914                                         not be re-evaluated when a reload is performed. This is because the values must be set
915                                         before the SIP stack is initialized. The only way to reset these values is to either 
916                                         restart Asterisk, or unload res_pjsip.so and then load it again.
917                                 </para></description>
918                                 <configOption name="timert1" default="500">
919                                         <synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
920                                         <description><para>
921                                                 Timer T1 is the base for determining how long to wait before retransmitting
922                                                 requests that receive no response when using an unreliable transport (e.g. UDP).
923                                                 For more information on this timer, see RFC 3261, Section 17.1.1.1.
924                                         </para></description>
925                                 </configOption>
926                                 <configOption name="timerb" default="32000">
927                                         <synopsis>Set transaction timer B value (milliseconds).</synopsis>
928                                         <description><para>
929                                                 Timer B determines the maximum amount of time to wait after sending an INVITE
930                                                 request before terminating the transaction. It is recommended that this be set
931                                                 to 64 * Timer T1, but it may be set higher if desired. For more information on
932                                                 this timer, see RFC 3261, Section 17.1.1.1.
933                                         </para></description>
934                                 </configOption>
935                                 <configOption name="compactheaders" default="no">
936                                         <synopsis>Use the short forms of common SIP header names.</synopsis>
937                                 </configOption>
938                                 <configOption name="threadpool_initial_size" default="0">
939                                         <synopsis>Initial number of threads in the res_pjsip threadpool.</synopsis>
940                                 </configOption>
941                                 <configOption name="threadpool_auto_increment" default="5">
942                                         <synopsis>The amount by which the number of threads is incremented when necessary.</synopsis>
943                                 </configOption>
944                                 <configOption name="threadpool_idle_timeout" default="60">
945                                         <synopsis>Number of seconds before an idle thread should be disposed of.</synopsis>
946                                 </configOption>
947                                 <configOption name="threadpool_max_size" default="0">
948                                         <synopsis>Maximum number of threads in the res_pjsip threadpool.
949                                         A value of 0 indicates no maximum.</synopsis>
950                                 </configOption>
951                         </configObject>
952                         <configObject name="global">
953                                 <synopsis>Options that apply globally to all SIP communications</synopsis>
954                                 <description><para>
955                                         The settings in this section are global. Unlike options in the <literal>system</literal>
956                                         section, these options can be refreshed by performing a reload.
957                                 </para></description>
958                                 <configOption name="maxforwards" default="70">
959                                         <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
960                                 </configOption>
961                                 <configOption name="useragent" default="Asterisk &lt;Asterisk Version&gt;">
962                                         <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
963                                 </configOption>
964                         </configObject>
965                 </configFile>
966         </configInfo>
967         <manager name="PJSIPQualify" language="en_US">
968                 <synopsis>
969                         Qualify a chan_pjsip endpoint.
970                 </synopsis>
971                 <syntax>
972                         <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
973                         <parameter name="Endpoint" required="true">
974                                 <para>The endpoint you want to qualify.</para>
975                         </parameter>
976                 </syntax>
977                 <description>
978                         <para>Qualify a chan_pjsip endpoint.</para>
979                 </description>
980         </manager>
981  ***/
982
983
984 static pjsip_endpoint *ast_pjsip_endpoint;
985
986 static struct ast_threadpool *sip_threadpool;
987
988 static int register_service(void *data)
989 {
990         pjsip_module **module = data;
991         if (!ast_pjsip_endpoint) {
992                 ast_log(LOG_ERROR, "There is no PJSIP endpoint. Unable to register services\n");
993                 return -1;
994         }
995         if (pjsip_endpt_register_module(ast_pjsip_endpoint, *module) != PJ_SUCCESS) {
996                 ast_log(LOG_ERROR, "Unable to register module %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
997                 return -1;
998         }
999         ast_debug(1, "Registered SIP service %.*s (%p)\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name), *module);
1000         ast_module_ref(ast_module_info->self);
1001         return 0;
1002 }
1003
1004 int ast_sip_register_service(pjsip_module *module)
1005 {
1006         return ast_sip_push_task_synchronous(NULL, register_service, &module);
1007 }
1008
1009 static int unregister_service(void *data)
1010 {
1011         pjsip_module **module = data;
1012         ast_module_unref(ast_module_info->self);
1013         if (!ast_pjsip_endpoint) {
1014                 return -1;
1015         }
1016         pjsip_endpt_unregister_module(ast_pjsip_endpoint, *module);
1017         ast_debug(1, "Unregistered SIP service %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1018         return 0;
1019 }
1020
1021 void ast_sip_unregister_service(pjsip_module *module)
1022 {
1023         ast_sip_push_task_synchronous(NULL, unregister_service, &module);
1024 }
1025
1026 static struct ast_sip_authenticator *registered_authenticator;
1027
1028 int ast_sip_register_authenticator(struct ast_sip_authenticator *auth)
1029 {
1030         if (registered_authenticator) {
1031                 ast_log(LOG_WARNING, "Authenticator %p is already registered. Cannot register a new one\n", registered_authenticator);
1032                 return -1;
1033         }
1034         registered_authenticator = auth;
1035         ast_debug(1, "Registered SIP authenticator module %p\n", auth);
1036         ast_module_ref(ast_module_info->self);
1037         return 0;
1038 }
1039
1040 void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth)
1041 {
1042         if (registered_authenticator != auth) {
1043                 ast_log(LOG_WARNING, "Trying to unregister authenticator %p but authenticator %p registered\n",
1044                                 auth, registered_authenticator);
1045                 return;
1046         }
1047         registered_authenticator = NULL;
1048         ast_debug(1, "Unregistered SIP authenticator %p\n", auth);
1049         ast_module_unref(ast_module_info->self);
1050 }
1051
1052 int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1053 {
1054         if (!registered_authenticator) {
1055                 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is not required\n");
1056                 return 0;
1057         }
1058
1059         return registered_authenticator->requires_authentication(endpoint, rdata);
1060 }
1061
1062 enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint,
1063                 pjsip_rx_data *rdata, pjsip_tx_data *tdata)
1064 {
1065         if (!registered_authenticator) {
1066                 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is successful\n");
1067                 return 0;
1068         }
1069         return registered_authenticator->check_authentication(endpoint, rdata, tdata);
1070 }
1071
1072 static struct ast_sip_outbound_authenticator *registered_outbound_authenticator;
1073
1074 int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1075 {
1076         if (registered_outbound_authenticator) {
1077                 ast_log(LOG_WARNING, "Outbound authenticator %p is already registered. Cannot register a new one\n", registered_outbound_authenticator);
1078                 return -1;
1079         }
1080         registered_outbound_authenticator = auth;
1081         ast_debug(1, "Registered SIP outbound authenticator module %p\n", auth);
1082         ast_module_ref(ast_module_info->self);
1083         return 0;
1084 }
1085
1086 void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1087 {
1088         if (registered_outbound_authenticator != auth) {
1089                 ast_log(LOG_WARNING, "Trying to unregister outbound authenticator %p but outbound authenticator %p registered\n",
1090                                 auth, registered_outbound_authenticator);
1091                 return;
1092         }
1093         registered_outbound_authenticator = NULL;
1094         ast_debug(1, "Unregistered SIP outbound authenticator %p\n", auth);
1095         ast_module_unref(ast_module_info->self);
1096 }
1097
1098 int ast_sip_create_request_with_auth(const struct ast_sip_auth_array *auths, pjsip_rx_data *challenge,
1099                 pjsip_transaction *tsx, pjsip_tx_data **new_request)
1100 {
1101         if (!registered_outbound_authenticator) {
1102                 ast_log(LOG_WARNING, "No SIP outbound authenticator registered. Cannot respond to authentication challenge\n");
1103                 return -1;
1104         }
1105         return registered_outbound_authenticator->create_request_with_auth(auths, challenge, tsx, new_request);
1106 }
1107
1108 struct endpoint_identifier_list {
1109         struct ast_sip_endpoint_identifier *identifier;
1110         AST_RWLIST_ENTRY(endpoint_identifier_list) list;
1111 };
1112
1113 static AST_RWLIST_HEAD_STATIC(endpoint_identifiers, endpoint_identifier_list);
1114
1115 int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1116 {
1117         struct endpoint_identifier_list *id_list_item;
1118         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1119
1120         id_list_item = ast_calloc(1, sizeof(*id_list_item));
1121         if (!id_list_item) {
1122                 ast_log(LOG_ERROR, "Unabled to add endpoint identifier. Out of memory.\n");
1123                 return -1;
1124         }
1125         id_list_item->identifier = identifier;
1126
1127         AST_RWLIST_INSERT_TAIL(&endpoint_identifiers, id_list_item, list);
1128         ast_debug(1, "Registered endpoint identifier %p\n", identifier);
1129
1130         ast_module_ref(ast_module_info->self);
1131         return 0;
1132 }
1133
1134 void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1135 {
1136         struct endpoint_identifier_list *iter;
1137         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1138         AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_identifiers, iter, list) {
1139                 if (iter->identifier == identifier) {
1140                         AST_RWLIST_REMOVE_CURRENT(list);
1141                         ast_free(iter);
1142                         ast_debug(1, "Unregistered endpoint identifier %p\n", identifier);
1143                         ast_module_unref(ast_module_info->self);
1144                         break;
1145                 }
1146         }
1147         AST_RWLIST_TRAVERSE_SAFE_END;
1148 }
1149
1150 struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata)
1151 {
1152         struct endpoint_identifier_list *iter;
1153         struct ast_sip_endpoint *endpoint = NULL;
1154         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1155         AST_RWLIST_TRAVERSE(&endpoint_identifiers, iter, list) {
1156                 ast_assert(iter->identifier->identify_endpoint != NULL);
1157                 endpoint = iter->identifier->identify_endpoint(rdata);
1158                 if (endpoint) {
1159                         break;
1160                 }
1161         }
1162         return endpoint;
1163 }
1164
1165 pjsip_endpoint *ast_sip_get_pjsip_endpoint(void)
1166 {
1167         return ast_pjsip_endpoint;
1168 }
1169
1170 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)
1171 {
1172         pj_str_t tmp, local_addr;
1173         pjsip_uri *uri;
1174         pjsip_sip_uri *sip_uri;
1175         pjsip_transport_type_e type = PJSIP_TRANSPORT_UNSPECIFIED;
1176         int local_port;
1177         char uuid_str[AST_UUID_STR_LEN];
1178
1179         if (ast_strlen_zero(user)) {
1180                 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1181                 if (!uuid) {
1182                         return -1;
1183                 }
1184                 user = ast_uuid_to_str(uuid, uuid_str, sizeof(uuid_str));
1185         }
1186
1187         /* Parse the provided target URI so we can determine what transport it will end up using */
1188         pj_strdup_with_null(pool, &tmp, target);
1189
1190         if (!(uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0)) ||
1191             (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
1192                 return -1;
1193         }
1194
1195         sip_uri = pjsip_uri_get_uri(uri);
1196
1197         /* Determine the transport type to use */
1198         if (PJSIP_URI_SCHEME_IS_SIPS(sip_uri)) {
1199                 type = PJSIP_TRANSPORT_TLS;
1200         } else if (!sip_uri->transport_param.slen) {
1201                 type = PJSIP_TRANSPORT_UDP;
1202         } else {
1203                 type = pjsip_transport_get_type_from_name(&sip_uri->transport_param);
1204         }
1205
1206         if (type == PJSIP_TRANSPORT_UNSPECIFIED) {
1207                 return -1;
1208         }
1209
1210         /* If the host is IPv6 turn the transport into an IPv6 version */
1211         if (pj_strchr(&sip_uri->host, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1212                 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1213         }
1214
1215         if (!ast_strlen_zero(domain)) {
1216                 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1217                 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1218                                 "<%s:%s@%s%s%s>",
1219                                 (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1220                                 user,
1221                                 domain,
1222                                 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1223                                 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1224                 return 0;
1225         }
1226
1227         /* Get the local bound address for the transport that will be used when communicating with the provided URI */
1228         if (pjsip_tpmgr_find_local_addr(pjsip_endpt_get_tpmgr(ast_sip_get_pjsip_endpoint()), pool, type, selector,
1229                                                               &local_addr, &local_port) != PJ_SUCCESS) {
1230                 return -1;
1231         }
1232
1233         /* If IPv6 was specified in the transport, set the proper type */
1234         if (pj_strchr(&local_addr, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1235                 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1236         }
1237
1238         from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1239         from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1240                                       "<%s:%s@%s%.*s%s:%d%s%s>",
1241                                       (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1242                                       user,
1243                                       (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1244                                       (int)local_addr.slen,
1245                                       local_addr.ptr,
1246                                       (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1247                                       local_port,
1248                                       (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1249                                       (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1250
1251         return 0;
1252 }
1253
1254 static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpoint, pjsip_tpselector *selector)
1255 {
1256         RAII_VAR(struct ast_sip_transport *, transport, NULL, ao2_cleanup);
1257         const char *transport_name = endpoint->transport;
1258
1259         if (ast_strlen_zero(transport_name)) {
1260                 return 0;
1261         }
1262
1263         transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_name);
1264
1265         if (!transport || !transport->state) {
1266                 return -1;
1267         }
1268
1269         if (transport->state->transport) {
1270                 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1271                 selector->u.transport = transport->state->transport;
1272         } else if (transport->state->factory) {
1273                 selector->type = PJSIP_TPSELECTOR_LISTENER;
1274                 selector->u.listener = transport->state->factory;
1275         } else {
1276                 return -1;
1277         }
1278
1279         return 0;
1280 }
1281
1282 static int sip_get_tpselector_from_uri(const char *uri, pjsip_tpselector *selector)
1283 {
1284         RAII_VAR(struct ast_sip_contact_transport *, contact_transport, NULL, ao2_cleanup);
1285
1286         contact_transport = ast_sip_location_retrieve_contact_transport_by_uri(uri);
1287
1288         if (!contact_transport) {
1289                 return -1;
1290         }
1291
1292         selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1293         selector->u.transport = contact_transport->transport;
1294
1295         return 0;
1296 }
1297
1298 pjsip_dialog *ast_sip_create_dialog(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user)
1299 {
1300         pj_str_t local_uri = { "sip:temp@temp", 13 }, remote_uri;
1301         pjsip_dialog *dlg = NULL;
1302         const char *outbound_proxy = endpoint->outbound_proxy;
1303         pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1304         static const pj_str_t HCONTACT = { "Contact", 7 };
1305
1306         pj_cstr(&remote_uri, uri);
1307
1308         if (pjsip_dlg_create_uac(pjsip_ua_instance(), &local_uri, NULL, &remote_uri, NULL, &dlg) != PJ_SUCCESS) {
1309                 return NULL;
1310         }
1311
1312         if (sip_get_tpselector_from_uri(uri, &selector) && sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1313                 pjsip_dlg_terminate(dlg);
1314                 return NULL;
1315         }
1316
1317         if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
1318                 pjsip_dlg_terminate(dlg);
1319                 return NULL;
1320         }
1321
1322         /* Update the dialog with the new local URI, we do it afterwards so we can use the dialog pool for construction */
1323         pj_strdup_with_null(dlg->pool, &dlg->local.info_str, &local_uri);
1324         dlg->local.info->uri = pjsip_parse_uri(dlg->pool, dlg->local.info_str.ptr, dlg->local.info_str.slen, 0);
1325         dlg->local.contact = pjsip_parse_hdr(dlg->pool, &HCONTACT, local_uri.ptr, local_uri.slen, NULL);
1326
1327         /* If a request user has been specified and we are permitted to change it, do so */
1328         if (!ast_strlen_zero(request_user) && (PJSIP_URI_SCHEME_IS_SIP(dlg->target) || PJSIP_URI_SCHEME_IS_SIPS(dlg->target))) {
1329                 pjsip_sip_uri *target = pjsip_uri_get_uri(dlg->target);
1330                 pj_strdup2(dlg->pool, &target->user, request_user);
1331         }
1332
1333         /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
1334         dlg->sess_count++;
1335
1336         pjsip_dlg_set_transport(dlg, &selector);
1337
1338         if (!ast_strlen_zero(outbound_proxy)) {
1339                 pjsip_route_hdr route_set, *route;
1340                 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1341                 pj_str_t tmp;
1342
1343                 pj_list_init(&route_set);
1344
1345                 pj_strdup2_with_null(dlg->pool, &tmp, outbound_proxy);
1346                 if (!(route = pjsip_parse_hdr(dlg->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1347                         pjsip_dlg_terminate(dlg);
1348                         return NULL;
1349                 }
1350                 pj_list_push_back(&route_set, route);
1351
1352                 pjsip_dlg_set_route_set(dlg, &route_set);
1353         }
1354
1355         dlg->sess_count--;
1356
1357         return dlg;
1358 }
1359
1360 /* PJSIP doesn't know about the INFO method, so we have to define it ourselves */
1361 const pjsip_method pjsip_info_method = {PJSIP_OTHER_METHOD, {"INFO", 4} };
1362 const pjsip_method pjsip_message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
1363
1364 static struct {
1365         const char *method;
1366         const pjsip_method *pmethod;
1367 } methods [] = {
1368         { "INVITE", &pjsip_invite_method },
1369         { "CANCEL", &pjsip_cancel_method },
1370         { "ACK", &pjsip_ack_method },
1371         { "BYE", &pjsip_bye_method },
1372         { "REGISTER", &pjsip_register_method },
1373         { "OPTIONS", &pjsip_options_method },
1374         { "SUBSCRIBE", &pjsip_subscribe_method },
1375         { "NOTIFY", &pjsip_notify_method },
1376         { "PUBLISH", &pjsip_publish_method },
1377         { "INFO", &pjsip_info_method },
1378         { "MESSAGE", &pjsip_message_method },
1379 };
1380
1381 static const pjsip_method *get_pjsip_method(const char *method)
1382 {
1383         int i;
1384         for (i = 0; i < ARRAY_LEN(methods); ++i) {
1385                 if (!strcmp(method, methods[i].method)) {
1386                         return methods[i].pmethod;
1387                 }
1388         }
1389         return NULL;
1390 }
1391
1392 static int create_in_dialog_request(const pjsip_method *method, struct pjsip_dialog *dlg, pjsip_tx_data **tdata)
1393 {
1394         if (pjsip_dlg_create_request(dlg, method, -1, tdata) != PJ_SUCCESS) {
1395                 ast_log(LOG_WARNING, "Unable to create in-dialog request.\n");
1396                 return -1;
1397         }
1398
1399         return 0;
1400 }
1401
1402 static int create_out_of_dialog_request(const pjsip_method *method, struct ast_sip_endpoint *endpoint,
1403                 const char *uri, pjsip_tx_data **tdata)
1404 {
1405         RAII_VAR(struct ast_sip_contact *, contact, NULL, ao2_cleanup);
1406         pj_str_t remote_uri;
1407         pj_str_t from;
1408         pj_pool_t *pool;
1409         pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1410
1411         if (ast_strlen_zero(uri)) {
1412                 if (!endpoint) {
1413                         ast_log(LOG_ERROR, "An endpoint and/or uri must be specified\n");
1414                         return -1;
1415                 }
1416
1417                 contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
1418                 if (!contact || ast_strlen_zero(contact->uri)) {
1419                         ast_log(LOG_ERROR, "Unable to retrieve contact for endpoint %s\n",
1420                                         ast_sorcery_object_get_id(endpoint));
1421                         return -1;
1422                 }
1423
1424                 pj_cstr(&remote_uri, contact->uri);
1425         } else {
1426                 pj_cstr(&remote_uri, uri);
1427         }
1428
1429         if (endpoint) {
1430                 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1431                         ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport selector for endpoint %s\n",
1432                                 ast_sorcery_object_get_id(endpoint));
1433                         return -1;
1434                 }
1435         }
1436
1437         pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "Outbound request", 256, 256);
1438
1439         if (!pool) {
1440                 ast_log(LOG_ERROR, "Unable to create PJLIB memory pool\n");
1441                 return -1;
1442         }
1443
1444         if (sip_dialog_create_from(pool, &from, endpoint ? endpoint->fromuser : NULL,
1445                                 endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) {
1446                 ast_log(LOG_ERROR, "Unable to create From header for %.*s request to endpoint %s\n",
1447                                 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1448                 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1449                 return -1;
1450         }
1451
1452         if (pjsip_endpt_create_request(ast_sip_get_pjsip_endpoint(), method, &remote_uri,
1453                         &from, &remote_uri, &from, NULL, -1, NULL, tdata) != PJ_SUCCESS) {
1454                 ast_log(LOG_ERROR, "Unable to create outbound %.*s request to endpoint %s\n",
1455                                 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1456                 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1457                 return -1;
1458         }
1459
1460         /* We can release this pool since request creation copied all the necessary
1461          * data into the outbound request's pool
1462          */
1463         pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1464         return 0;
1465 }
1466
1467 int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
1468                 struct ast_sip_endpoint *endpoint, const char *uri,
1469                 pjsip_tx_data **tdata)
1470 {
1471         const pjsip_method *pmethod = get_pjsip_method(method);
1472
1473         if (!pmethod) {
1474                 ast_log(LOG_WARNING, "Unknown method '%s'. Cannot send request\n", method);
1475                 return -1;
1476         }
1477
1478         if (dlg) {
1479                 return create_in_dialog_request(pmethod, dlg, tdata);
1480         } else {
1481                 return create_out_of_dialog_request(pmethod, endpoint, uri, tdata);
1482         }
1483 }
1484
1485 static int send_in_dialog_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg)
1486 {
1487         if (pjsip_dlg_send_request(dlg, tdata, -1, NULL) != PJ_SUCCESS) {
1488                 ast_log(LOG_WARNING, "Unable to send in-dialog request.\n");
1489                 return -1;
1490         }
1491         return 0;
1492 }
1493
1494 static void send_request_cb(void *token, pjsip_event *e)
1495 {
1496         RAII_VAR(struct ast_sip_endpoint *, endpoint, token, ao2_cleanup);
1497         pjsip_transaction *tsx = e->body.tsx_state.tsx;
1498         pjsip_rx_data *challenge = e->body.tsx_state.src.rdata;
1499         pjsip_tx_data *tdata;
1500
1501         if (tsx->status_code != 401 && tsx->status_code != 407) {
1502                 return;
1503         }
1504
1505         if (!ast_sip_create_request_with_auth(&endpoint->outbound_auths, challenge, tsx, &tdata)) {
1506                 pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, NULL, NULL);
1507         }
1508 }
1509
1510 static int send_out_of_dialog_request(pjsip_tx_data *tdata, struct ast_sip_endpoint *endpoint)
1511 {
1512         ao2_ref(endpoint, +1);
1513         if (pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, endpoint, send_request_cb) != PJ_SUCCESS) {
1514                 ast_log(LOG_ERROR, "Error attempting to send outbound %.*s request to endpoint %s\n",
1515                                 (int) pj_strlen(&tdata->msg->line.req.method.name),
1516                                 pj_strbuf(&tdata->msg->line.req.method.name),
1517                                 ast_sorcery_object_get_id(endpoint));
1518                 ao2_ref(endpoint, -1);
1519                 return -1;
1520         }
1521
1522         return 0;
1523 }
1524
1525 int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg, struct ast_sip_endpoint *endpoint)
1526 {
1527         ast_assert(tdata->msg->type == PJSIP_REQUEST_MSG);
1528
1529         if (dlg) {
1530                 return send_in_dialog_request(tdata, dlg);
1531         } else {
1532                 return send_out_of_dialog_request(tdata, endpoint);
1533         }
1534 }
1535
1536 int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value)
1537 {
1538         pj_str_t hdr_name;
1539         pj_str_t hdr_value;
1540         pjsip_generic_string_hdr *hdr;
1541
1542         pj_cstr(&hdr_name, name);
1543         pj_cstr(&hdr_value, value);
1544
1545         hdr = pjsip_generic_string_hdr_create(tdata->pool, &hdr_name, &hdr_value);
1546
1547         pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *) hdr);
1548         return 0;
1549 }
1550
1551 static pjsip_msg_body *ast_body_to_pjsip_body(pj_pool_t *pool, const struct ast_sip_body *body)
1552 {
1553         pj_str_t type;
1554         pj_str_t subtype;
1555         pj_str_t body_text;
1556
1557         pj_cstr(&type, body->type);
1558         pj_cstr(&subtype, body->subtype);
1559         pj_cstr(&body_text, body->body_text);
1560
1561         return pjsip_msg_body_create(pool, &type, &subtype, &body_text);
1562 }
1563
1564 int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body)
1565 {
1566         pjsip_msg_body *pjsip_body = ast_body_to_pjsip_body(tdata->pool, body);
1567         tdata->msg->body = pjsip_body;
1568         return 0;
1569 }
1570
1571 int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies)
1572 {
1573         int i;
1574         /* NULL for type and subtype automatically creates "multipart/mixed" */
1575         pjsip_msg_body *body = pjsip_multipart_create(tdata->pool, NULL, NULL);
1576
1577         for (i = 0; i < num_bodies; ++i) {
1578                 pjsip_multipart_part *part = pjsip_multipart_create_part(tdata->pool);
1579                 part->body = ast_body_to_pjsip_body(tdata->pool, bodies[i]);
1580                 pjsip_multipart_add_part(tdata->pool, body, part);
1581         }
1582
1583         tdata->msg->body = body;
1584         return 0;
1585 }
1586
1587 int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text)
1588 {
1589         size_t combined_size = strlen(body_text) + tdata->msg->body->len;
1590         struct ast_str *body_buffer = ast_str_alloca(combined_size);
1591
1592         ast_str_set(&body_buffer, 0, "%.*s%s", (int) tdata->msg->body->len, (char *) tdata->msg->body->data, body_text);
1593
1594         tdata->msg->body->data = pj_pool_alloc(tdata->pool, combined_size);
1595         pj_memcpy(tdata->msg->body->data, ast_str_buffer(body_buffer), combined_size);
1596         tdata->msg->body->len = combined_size;
1597
1598         return 0;
1599 }
1600
1601 struct ast_taskprocessor *ast_sip_create_serializer(void)
1602 {
1603         struct ast_taskprocessor *serializer;
1604         RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1605         char name[AST_UUID_STR_LEN];
1606
1607         if (!uuid) {
1608                 return NULL;
1609         }
1610
1611         ast_uuid_to_str(uuid, name, sizeof(name));
1612
1613         serializer = ast_threadpool_serializer(name, sip_threadpool);
1614         if (!serializer) {
1615                 return NULL;
1616         }
1617         return serializer;
1618 }
1619
1620 int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1621 {
1622         if (serializer) {
1623                 return ast_taskprocessor_push(serializer, sip_task, task_data);
1624         } else {
1625                 return ast_threadpool_push(sip_threadpool, sip_task, task_data);
1626         }
1627 }
1628
1629 struct sync_task_data {
1630         ast_mutex_t lock;
1631         ast_cond_t cond;
1632         int complete;
1633         int fail;
1634         int (*task)(void *);
1635         void *task_data;
1636 };
1637
1638 static int sync_task(void *data)
1639 {
1640         struct sync_task_data *std = data;
1641         std->fail = std->task(std->task_data);
1642
1643         ast_mutex_lock(&std->lock);
1644         std->complete = 1;
1645         ast_cond_signal(&std->cond);
1646         ast_mutex_unlock(&std->lock);
1647         return std->fail;
1648 }
1649
1650 int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1651 {
1652         /* This method is an onion */
1653         struct sync_task_data std;
1654         ast_mutex_init(&std.lock);
1655         ast_cond_init(&std.cond, NULL);
1656         std.fail = std.complete = 0;
1657         std.task = sip_task;
1658         std.task_data = task_data;
1659
1660         if (serializer) {
1661                 if (ast_taskprocessor_push(serializer, sync_task, &std)) {
1662                         return -1;
1663                 }
1664         } else {
1665                 if (ast_threadpool_push(sip_threadpool, sync_task, &std)) {
1666                         return -1;
1667                 }
1668         }
1669
1670         ast_mutex_lock(&std.lock);
1671         while (!std.complete) {
1672                 ast_cond_wait(&std.cond, &std.lock);
1673         }
1674         ast_mutex_unlock(&std.lock);
1675
1676         ast_mutex_destroy(&std.lock);
1677         ast_cond_destroy(&std.cond);
1678         return std.fail;
1679 }
1680
1681 void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size)
1682 {
1683         size_t chars_to_copy = MIN(size - 1, pj_strlen(src));
1684         memcpy(dest, pj_strbuf(src), chars_to_copy);
1685         dest[chars_to_copy] = '\0';
1686 }
1687
1688 int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype)
1689 {
1690         pjsip_media_type compare;
1691
1692         if (!content_type) {
1693                 return 0;
1694         }
1695
1696         pjsip_media_type_init2(&compare, type, subtype);
1697
1698         return pjsip_media_type_cmp(content_type, &compare, 0) ? -1 : 0;
1699 }
1700
1701 pj_caching_pool caching_pool;
1702 pj_pool_t *memory_pool;
1703 pj_thread_t *monitor_thread;
1704 static int monitor_continue;
1705
1706 static void *monitor_thread_exec(void *endpt)
1707 {
1708         while (monitor_continue) {
1709                 const pj_time_val delay = {0, 10};
1710                 pjsip_endpt_handle_events(ast_pjsip_endpoint, &delay);
1711         }
1712         return NULL;
1713 }
1714
1715 static void stop_monitor_thread(void)
1716 {
1717         monitor_continue = 0;
1718         pj_thread_join(monitor_thread);
1719 }
1720
1721 AST_THREADSTORAGE(pj_thread_storage);
1722 AST_THREADSTORAGE(servant_id_storage);
1723 #define SIP_SERVANT_ID 0x5E2F1D
1724
1725 static void sip_thread_start(void)
1726 {
1727         pj_thread_desc *desc;
1728         pj_thread_t *thread;
1729         uint32_t *servant_id;
1730
1731         servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1732         if (!servant_id) {
1733                 ast_log(LOG_ERROR, "Could not set SIP servant ID in thread-local storage.\n");
1734                 return;
1735         }
1736         *servant_id = SIP_SERVANT_ID;
1737
1738         desc = ast_threadstorage_get(&pj_thread_storage, sizeof(pj_thread_desc));
1739         if (!desc) {
1740                 ast_log(LOG_ERROR, "Could not get thread desc from thread-local storage. Expect awful things to occur\n");
1741                 return;
1742         }
1743         pj_bzero(*desc, sizeof(*desc));
1744
1745         if (pj_thread_register("Asterisk Thread", *desc, &thread) != PJ_SUCCESS) {
1746                 ast_log(LOG_ERROR, "Couldn't register thread with PJLIB.\n");
1747         }
1748 }
1749
1750 int ast_sip_thread_is_servant(void)
1751 {
1752         uint32_t *servant_id;
1753
1754         servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1755         if (!servant_id) {
1756                 return 0;
1757         }
1758
1759         return *servant_id == SIP_SERVANT_ID;
1760 }
1761
1762 static void remove_request_headers(pjsip_endpoint *endpt)
1763 {
1764         const pjsip_hdr *request_headers = pjsip_endpt_get_request_headers(endpt);
1765         pjsip_hdr *iter = request_headers->next;
1766
1767         while (iter != request_headers) {
1768                 pjsip_hdr *to_erase = iter;
1769                 iter = iter->next;
1770                 pj_list_erase(to_erase);
1771         }
1772 }
1773
1774 static int load_module(void)
1775 {
1776         /* The third parameter is just copied from
1777          * example code from PJLIB. This can be adjusted
1778          * if necessary.
1779          */
1780         pj_status_t status;
1781         struct ast_threadpool_options options;
1782
1783         if (pj_init() != PJ_SUCCESS) {
1784                 return AST_MODULE_LOAD_DECLINE;
1785         }
1786
1787         if (pjlib_util_init() != PJ_SUCCESS) {
1788                 pj_shutdown();
1789                 return AST_MODULE_LOAD_DECLINE;
1790         }
1791
1792         pj_caching_pool_init(&caching_pool, NULL, 1024 * 1024);
1793         if (pjsip_endpt_create(&caching_pool.factory, "SIP", &ast_pjsip_endpoint) != PJ_SUCCESS) {
1794                 ast_log(LOG_ERROR, "Failed to create PJSIP endpoint structure. Aborting load\n");
1795                 goto error;
1796         }
1797
1798         /* PJSIP will automatically try to add a Max-Forwards header. Since we want to control that,
1799          * we need to stop PJSIP from doing it automatically
1800          */
1801         remove_request_headers(ast_pjsip_endpoint);
1802
1803         memory_pool = pj_pool_create(&caching_pool.factory, "SIP", 1024, 1024, NULL);
1804         if (!memory_pool) {
1805                 ast_log(LOG_ERROR, "Failed to create memory pool for SIP. Aborting load\n");
1806                 goto error;
1807         }
1808
1809         if (ast_sip_initialize_system()) {
1810                 ast_log(LOG_ERROR, "Failed to initialize SIP system configuration. Aborting load\n");
1811                 goto error;
1812         }
1813
1814         sip_get_threadpool_options(&options);
1815         options.thread_start = sip_thread_start;
1816         sip_threadpool = ast_threadpool_create("SIP", NULL, &options);
1817         if (!sip_threadpool) {
1818                 ast_log(LOG_ERROR, "Failed to create SIP threadpool. Aborting load\n");
1819                 goto error;
1820         }
1821
1822         pjsip_tsx_layer_init_module(ast_pjsip_endpoint);
1823         pjsip_ua_init_module(ast_pjsip_endpoint, NULL);
1824
1825         monitor_continue = 1;
1826         status = pj_thread_create(memory_pool, "SIP", (pj_thread_proc *) &monitor_thread_exec,
1827                         NULL, PJ_THREAD_DEFAULT_STACK_SIZE * 2, 0, &monitor_thread);
1828         if (status != PJ_SUCCESS) {
1829                 ast_log(LOG_ERROR, "Failed to start SIP monitor thread. Aborting load\n");
1830                 goto error;
1831         }
1832
1833         ast_sip_initialize_global_headers();
1834
1835         if (ast_res_pjsip_initialize_configuration()) {
1836                 ast_log(LOG_ERROR, "Failed to initialize SIP configuration. Aborting load\n");
1837                 goto error;
1838         }
1839
1840         if (ast_sip_initialize_distributor()) {
1841                 ast_log(LOG_ERROR, "Failed to register distributor module. Aborting load\n");
1842                 goto error;
1843         }
1844
1845         if (ast_sip_initialize_outbound_authentication()) {
1846                 ast_log(LOG_ERROR, "Failed to initialize outbound authentication. Aborting load\n");
1847                 goto error;
1848         }
1849
1850         ast_res_pjsip_init_options_handling(0);
1851
1852         ast_res_pjsip_init_contact_transports();
1853
1854 return AST_MODULE_LOAD_SUCCESS;
1855
1856 error:
1857         ast_sip_destroy_distributor();
1858         ast_res_pjsip_destroy_configuration();
1859         ast_sip_destroy_global_headers();
1860         if (monitor_thread) {
1861                 stop_monitor_thread();
1862         }
1863         if (memory_pool) {
1864                 pj_pool_release(memory_pool);
1865                 memory_pool = NULL;
1866         }
1867         if (ast_pjsip_endpoint) {
1868                 pjsip_endpt_destroy(ast_pjsip_endpoint);
1869                 ast_pjsip_endpoint = NULL;
1870         }
1871         pj_caching_pool_destroy(&caching_pool);
1872         return AST_MODULE_LOAD_DECLINE;
1873 }
1874
1875 static int reload_module(void)
1876 {
1877         if (ast_res_pjsip_reload_configuration()) {
1878                 return AST_MODULE_LOAD_DECLINE;
1879         }
1880         ast_res_pjsip_init_options_handling(1);
1881         return 0;
1882 }
1883
1884 static int unload_pjsip(void *data)
1885 {
1886         if (memory_pool) {
1887                 pj_pool_release(memory_pool);
1888                 memory_pool = NULL;
1889         }
1890         if (ast_pjsip_endpoint) {
1891                 pjsip_endpt_destroy(ast_pjsip_endpoint);
1892                 ast_pjsip_endpoint = NULL;
1893         }
1894         pj_caching_pool_destroy(&caching_pool);
1895         return 0;
1896 }
1897
1898 static int unload_module(void)
1899 {
1900         ast_res_pjsip_cleanup_options_handling();
1901         ast_sip_destroy_distributor();
1902         ast_res_pjsip_destroy_configuration();
1903         ast_sip_destroy_global_headers();
1904         if (monitor_thread) {
1905                 stop_monitor_thread();
1906         }
1907         /* The thread this is called from cannot call PJSIP/PJLIB functions,
1908          * so we have to push the work to the threadpool to handle
1909          */
1910         ast_sip_push_task_synchronous(NULL, unload_pjsip, NULL);
1911
1912         ast_threadpool_shutdown(sip_threadpool);
1913
1914         return 0;
1915 }
1916
1917 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Basic SIP resource",
1918                 .load = load_module,
1919                 .unload = unload_module,
1920                 .reload = reload_module,
1921                 .load_pri = AST_MODPRI_CHANNEL_DEPEND - 5,
1922 );