Enclose the To URI and update its user portion if a request user has been specified.
[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="media_address">
212                                         <synopsis>IP address used in SDP for media handling</synopsis>
213                                         <description><para>
214                                                 At the time of SDP creation, the IP address defined here will be used as
215                                                 the media address for individual streams in the SDP.
216                                         </para>
217                                         <note><para>
218                                                 Be aware that the <literal>external_media_address</literal> option, set in Transport
219                                                 configuration, can also affect the final media address used in the SDP.
220                                         </para></note>
221                                         </description>
222                                 </configOption>
223                                 <configOption name="force_rport" default="yes">
224                                         <synopsis>Force use of return port</synopsis>
225                                 </configOption>
226                                 <configOption name="ice_support" default="no">
227                                         <synopsis>Enable the ICE mechanism to help traverse NAT</synopsis>
228                                 </configOption>
229                                 <configOption name="identify_by" default="username,location">
230                                         <synopsis>Way(s) for Endpoint to be identified</synopsis>
231                                         <description><para>
232                                                 An endpoint can be identified in multiple ways. Currently, the only supported
233                                                 option is <literal>username</literal>, which matches the endpoint based on the
234                                                 username in the From header.
235                                                 </para>
236                                                 <note><para>Endpoints can also be identified by IP address; however, that method
237                                                 of identification is not handled by this configuration option. See the documentation
238                                                 for the <literal>identify</literal> configuration section for more details on that
239                                                 method of endpoint identification. If this option is set to <literal>username</literal>
240                                                 and an <literal>identify</literal> configuration section exists for the endpoint, then
241                                                 the endpoint can be identified in multiple ways.</para></note>
242                                                 <enumlist>
243                                                         <enum name="username" />
244                                                 </enumlist>
245                                         </description>
246                                 </configOption>
247                                 <configOption name="mailboxes">
248                                         <synopsis>Mailbox(es) to be associated with</synopsis>
249                                 </configOption>
250                                 <configOption name="mohsuggest" default="default">
251                                         <synopsis>Default Music On Hold class</synopsis>
252                                 </configOption>
253                                 <configOption name="outbound_auth">
254                                         <synopsis>Authentication object used for outbound requests</synopsis>
255                                 </configOption>
256                                 <configOption name="outbound_proxy">
257                                         <synopsis>Proxy through which to send requests</synopsis>
258                                 </configOption>
259                                 <configOption name="rewrite_contact">
260                                         <synopsis>Allow Contact header to be rewritten with the source IP address-port</synopsis>
261                                         <description><para>
262                                                 On inbound SIP messages from this endpoint, the Contact header will be changed to have the
263                                                 source IP address and port. This option does not affect outbound messages send to this
264                                                 endpoint.
265                                         </para></description>
266                                 </configOption>
267                                 <configOption name="rtp_ipv6" default="no">
268                                         <synopsis>Allow use of IPv6 for RTP traffic</synopsis>
269                                 </configOption>
270                                 <configOption name="rtp_symmetric" default="no">
271                                         <synopsis>Enforce that RTP must be symmetric</synopsis>
272                                 </configOption>
273                                 <configOption name="send_diversion" default="yes">
274                                         <synopsis>Send the Diversion header, conveying the diversion
275                                         information to the called user agent</synopsis>
276                                 </configOption>
277                                 <configOption name="send_pai" default="no">
278                                         <synopsis>Send the P-Asserted-Identity header</synopsis>
279                                 </configOption>
280                                 <configOption name="send_rpid" default="no">
281                                         <synopsis>Send the Remote-Party-ID header</synopsis>
282                                 </configOption>
283                                 <configOption name="timers_min_se" default="90">
284                                         <synopsis>Minimum session timers expiration period</synopsis>
285                                         <description><para>
286                                                 Minimium session timer expiration period. Time in seconds.
287                                         </para></description>
288                                 </configOption>
289                                 <configOption name="timers" default="yes">
290                                         <synopsis>Session timers for SIP packets</synopsis>
291                                         <description>
292                                                 <enumlist>
293                                                         <enum name="forced" />
294                                                         <enum name="no" />
295                                                         <enum name="required" />
296                                                         <enum name="yes" />
297                                                 </enumlist>
298                                         </description>
299                                 </configOption>
300                                 <configOption name="timers_sess_expires" default="1800">
301                                         <synopsis>Maximum session timer expiration period</synopsis>
302                                         <description><para>
303                                                 Maximium session timer expiration period. Time in seconds.
304                                         </para></description>
305                                 </configOption>
306                                 <configOption name="transport">
307                                         <synopsis>Desired transport configuration</synopsis>
308                                         <description><para>
309                                                 This will set the desired transport configuration to send SIP data through.
310                                                 </para>
311                                                 <warning><para>Not specifying a transport will <emphasis>DEFAULT</emphasis>
312                                                 to the first configured transport in <filename>pjsip.conf</filename> which is
313                                                 valid for the URI we are trying to contact.
314                                                 </para></warning>
315                                                 <warning><para>Transport configuration is not affected by reloads. In order to
316                                                 change transports, a full Asterisk restart is required</para></warning>
317                                         </description>
318                                 </configOption>
319                                 <configOption name="trust_id_inbound" default="no">
320                                         <synopsis>Accept identification information received from this endpoint</synopsis>
321                                         <description><para>This option determines whether Asterisk will accept
322                                         identification from the endpoint from headers such as P-Asserted-Identity
323                                         or Remote-Party-ID header. This option applies both to calls originating from the
324                                         endpoint and calls originating from Asterisk. If <literal>no</literal>, the
325                                         configured Caller-ID from pjsip.conf will always be used as the identity for
326                                         the endpoint.</para></description>
327                                 </configOption>
328                                 <configOption name="trust_id_outbound" default="no">
329                                         <synopsis>Send private identification details to the endpoint.</synopsis>
330                                         <description><para>This option determines whether res_pjsip will send private
331                                         identification information to the endpoint. If <literal>no</literal>,
332                                         private Caller-ID information will not be forwarded to the endpoint.
333                                         "Private" in this case refers to any method of restricting identification.
334                                         Example: setting <replaceable>callerid_privacy</replaceable> to any
335                                         <literal>prohib</literal> variation.
336                                         Example: If <replaceable>trust_id_inbound</replaceable> is set to
337                                         <literal>yes</literal>, the presence of a <literal>Privacy: id</literal>
338                                         header in a SIP request or response would indicate the identification
339                                         provided in the request is private.</para></description>
340                                 </configOption>
341                                 <configOption name="type">
342                                         <synopsis>Must be of type 'endpoint'.</synopsis>
343                                 </configOption>
344                                 <configOption name="use_ptime" default="no">
345                                         <synopsis>Use Endpoint's requested packetisation interval</synopsis>
346                                 </configOption>
347                                 <configOption name="use_avpf" default="no">
348                                         <synopsis>Determines whether res_pjsip will use and enforce usage of AVPF for this
349                                         endpoint.</synopsis>
350                                         <description><para>
351                                                 If set to <literal>yes</literal>, res_pjsip will use use the AVPF or SAVPF RTP
352                                                 profile for all media offers on outbound calls and media updates and will
353                                                 decline media offers not using the AVPF or SAVPF profile.
354                                         </para><para>
355                                                 If set to <literal>no</literal>, res_pjsip will use use the AVP or SAVP RTP
356                                                 profile for all media offers on outbound calls and media updates and will
357                                                 decline media offers not using the AVP or SAVP profile.
358                                         </para></description>
359                                 </configOption>
360                                 <configOption name="media_encryption" default="no">
361                                         <synopsis>Determines whether res_pjsip will use and enforce usage of media encryption
362                                         for this endpoint.</synopsis>
363                                         <description>
364                                                 <enumlist>
365                                                         <enum name="no"><para>
366                                                                 res_pjsip will offer no encryption and allow no encryption to be setup.
367                                                         </para></enum>
368                                                         <enum name="sdes"><para>
369                                                                 res_pjsip will offer standard SRTP setup via in-SDP keys. Encrypted SIP
370                                                                 transport should be used in conjunction with this option to prevent
371                                                                 exposure of media encryption keys.
372                                                         </para></enum>
373                                                         <enum name="dtls"><para>
374                                                                 res_pjsip will offer DTLS-SRTP setup.
375                                                         </para></enum>
376                                                 </enumlist>
377                                         </description>
378                                 </configOption>
379                                 <configOption name="inband_progress" default="no">
380                                         <synopsis>Determines whether chan_pjsip will indicate ringing using inband
381                                             progress.</synopsis>
382                                         <description><para>
383                                                 If set to <literal>yes</literal>, chan_pjsip will send a 183 Session Progress
384                                                 when told to indicate ringing and will immediately start sending ringing
385                                                 as audio.
386                                         </para><para>
387                                                 If set to <literal>no</literal>, chan_pjsip will send a 180 Ringing when told
388                                                 to indicate ringing and will NOT send it as audio.
389                                         </para></description>
390                                 </configOption>
391                                 <configOption name="callgroup">
392                                         <synopsis>The numeric pickup groups for a channel.</synopsis>
393                                         <description><para>
394                                                 Can be set to a comma separated list of numbers or ranges between the values
395                                                 of 0-63 (maximum of 64 groups).
396                                         </para></description>
397                                 </configOption>
398                                 <configOption name="pickupgroup">
399                                         <synopsis>The numeric pickup groups that a channel can pickup.</synopsis>
400                                         <description><para>
401                                                 Can be set to a comma separated list of numbers or ranges between the values
402                                                 of 0-63 (maximum of 64 groups).
403                                         </para></description>
404                                 </configOption>
405                                 <configOption name="namedcallgroup">
406                                         <synopsis>The named pickup groups for a channel.</synopsis>
407                                         <description><para>
408                                                 Can be set to a comma separated list of case sensitive strings limited by
409                                                 supported line length.
410                                         </para></description>
411                                 </configOption>
412                                 <configOption name="namedpickupgroup">
413                                         <synopsis>The named pickup groups that a channel can pickup.</synopsis>
414                                         <description><para>
415                                                 Can be set to a comma separated list of case sensitive strings limited by
416                                                 supported line length.
417                                         </para></description>
418                                 </configOption>
419                                 <configOption name="devicestate_busy_at" default="0">
420                                         <synopsis>The number of in-use channels which will cause busy to be returned as device state</synopsis>
421                                         <description><para>
422                                                 When the number of in-use channels for the endpoint matches the devicestate_busy_at setting the
423                                                 PJSIP channel driver will return busy as the device state instead of in use.
424                                         </para></description>
425                                 </configOption>
426                                 <configOption name="t38udptl" default="no">
427                                         <synopsis>Whether T.38 UDPTL support is enabled or not</synopsis>
428                                         <description><para>
429                                                 If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted
430                                                 and relayed.
431                                         </para></description>
432                                 </configOption>
433                                 <configOption name="t38udptl_ec" default="none">
434                                         <synopsis>T.38 UDPTL error correction method</synopsis>
435                                         <description>
436                                                 <enumlist>
437                                                         <enum name="none"><para>
438                                                                 No error correction should be used.
439                                                         </para></enum>
440                                                         <enum name="fec"><para>
441                                                                 Forward error correction should be used.
442                                                         </para></enum>
443                                                         <enum name="redundancy"><para>
444                                                                 Redundacy error correction should be used.
445                                                         </para></enum>
446                                                 </enumlist>
447                                         </description>
448                                 </configOption>
449                                 <configOption name="t38udptl_maxdatagram" default="0">
450                                         <synopsis>T.38 UDPTL maximum datagram size</synopsis>
451                                         <description><para>
452                                                 This option can be set to override the maximum datagram of a remote endpoint for broken
453                                                 endpoints.
454                                         </para></description>
455                                 </configOption>
456                                 <configOption name="faxdetect" default="no">
457                                         <synopsis>Whether CNG tone detection is enabled</synopsis>
458                                         <description><para>
459                                                 This option can be set to send the session to the fax extension when a CNG tone is
460                                                 detected.
461                                         </para></description>
462                                 </configOption>
463                                 <configOption name="t38udptl_nat" default="no">
464                                         <synopsis>Whether NAT support is enabled on UDPTL sessions</synopsis>
465                                         <description><para>
466                                                 When enabled the UDPTL stack will send UDPTL packets to the source address of
467                                                 received packets.
468                                         </para></description>
469                                 </configOption>
470                                 <configOption name="t38udptl_ipv6" default="no">
471                                         <synopsis>Whether IPv6 is used for UDPTL Sessions</synopsis>
472                                         <description><para>
473                                                 When enabled the UDPTL stack will use IPv6.
474                                         </para></description>
475                                 </configOption>
476                                 <configOption name="tonezone">
477                                         <synopsis>Set which country's indications to use for channels created for this endpoint.</synopsis>
478                                 </configOption>
479                                 <configOption name="language">
480                                         <synopsis>Set the default language to use for channels created for this endpoint.</synopsis>
481                                 </configOption>
482                                 <configOption name="one_touch_recording" default="no">
483                                         <synopsis>Determines whether one-touch recording is allowed for this endpoint.</synopsis>
484                                         <see-also>
485                                                 <ref type="configOption">recordonfeature</ref>
486                                                 <ref type="configOption">recordofffeature</ref>
487                                         </see-also>
488                                 </configOption>
489                                 <configOption name="recordonfeature" default="automixmon">
490                                         <synopsis>The feature to enact when one-touch recording is turned on.</synopsis>
491                                         <description>
492                                                 <para>When an INFO request for one-touch recording arrives with a Record header set to "on", this
493                                                 feature will be enabled for the channel. The feature designated here can be any built-in
494                                                 or dynamic feature defined in features.conf.</para>
495                                                 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
496                                         </description>
497                                         <see-also>
498                                                 <ref type="configOption">one_touch_recording</ref>
499                                                 <ref type="configOption">recordofffeature</ref>
500                                         </see-also>
501                                 </configOption>
502                                 <configOption name="recordofffeature" default="automixmon">
503                                         <synopsis>The feature to enact when one-touch recording is turned off.</synopsis>
504                                         <description>
505                                                 <para>When an INFO request for one-touch recording arrives with a Record header set to "off", this
506                                                 feature will be enabled for the channel. The feature designated here can be any built-in
507                                                 or dynamic feature defined in features.conf.</para>
508                                                 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
509                                         </description>
510                                         <see-also>
511                                                 <ref type="configOption">one_touch_recording</ref>
512                                                 <ref type="configOption">recordonfeature</ref>
513                                         </see-also>
514                                 </configOption>
515                                 <configOption name="rtpengine" default="asterisk">
516                                         <synopsis>Name of the RTP engine to use for channels created for this endpoint</synopsis>
517                                 </configOption>
518                                 <configOption name="allowtransfer" default="yes">
519                                         <synopsis>Determines whether SIP REFER transfers are allowed for this endpoint</synopsis>
520                                 </configOption>
521                                 <configOption name="sdpowner" default="-">
522                                         <synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis>
523                                 </configOption>
524                                 <configOption name="sdpsession" default="Asterisk">
525                                         <synopsis>String used for the SDP session (s=) line.</synopsis>
526                                 </configOption>
527                                 <configOption name="tos_audio">
528                                         <synopsis>DSCP TOS bits for audio streams</synopsis>
529                                         <description><para>
530                                                 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
531                                         </para></description>
532                                 </configOption>
533                                 <configOption name="tos_video">
534                                         <synopsis>DSCP TOS bits for video streams</synopsis>
535                                         <description><para>
536                                                 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
537                                         </para></description>
538                                 </configOption>
539                                 <configOption name="cos_audio">
540                                         <synopsis>Priority for audio streams</synopsis>
541                                         <description><para>
542                                                 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
543                                         </para></description>
544                                 </configOption>
545                                 <configOption name="cos_video">
546                                         <synopsis>Priority for video streams</synopsis>
547                                         <description><para>
548                                                 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
549                                         </para></description>
550                                 </configOption>
551                                 <configOption name="allowsubscribe" default="yes">
552                                         <synopsis>Determines if endpoint is allowed to initiate subscriptions with Asterisk.</synopsis>
553                                 </configOption>
554                                 <configOption name="subminexpiry" default="60">
555                                         <synopsis>The minimum allowed expiry time for subscriptions initiated by the endpoint.</synopsis>
556                                 </configOption>
557                                 <configOption name="fromuser">
558                                         <synopsis>Username to use in From header for requests to this endpoint.</synopsis>
559                                 </configOption>
560                                 <configOption name="mwifromuser">
561                                         <synopsis>Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.</synopsis>
562                                 </configOption>
563                                 <configOption name="fromdomain">
564                                         <synopsis>Domain to user in From header for requests to this endpoint.</synopsis>
565                                 </configOption>
566                                 <configOption name="dtlsverify">
567                                         <synopsis>Verify that the provided peer certificate is valid</synopsis>
568                                         <description><para>
569                                                 This option only applies if <replaceable>media_encryption</replaceable> is
570                                                 set to <literal>dtls</literal>.
571                                         </para></description>
572                                 </configOption>
573                                 <configOption name="dtlsrekey">
574                                         <synopsis>Interval at which to renegotiate the TLS session and rekey the SRTP session</synopsis>
575                                         <description><para>
576                                                 This option only applies if <replaceable>media_encryption</replaceable> is
577                                                 set to <literal>dtls</literal>.
578                                         </para><para>
579                                                 If this is not set or the value provided is 0 rekeying will be disabled.
580                                         </para></description>
581                                 </configOption>
582                                 <configOption name="dtlscertfile">
583                                         <synopsis>Path to certificate file to present to peer</synopsis>
584                                         <description><para>
585                                                 This option only applies if <replaceable>media_encryption</replaceable> is
586                                                 set to <literal>dtls</literal>.
587                                         </para></description>
588                                 </configOption>
589                                 <configOption name="dtlsprivatekey">
590                                         <synopsis>Path to private key for certificate file</synopsis>
591                                         <description><para>
592                                                 This option only applies if <replaceable>media_encryption</replaceable> is
593                                                 set to <literal>dtls</literal>.
594                                         </para></description>
595                                 </configOption>
596                                 <configOption name="dtlscipher">
597                                         <synopsis>Cipher to use for DTLS negotiation</synopsis>
598                                         <description><para>
599                                                 This option only applies if <replaceable>media_encryption</replaceable> is
600                                                 set to <literal>dtls</literal>.
601                                         </para><para>
602                                                 Many options for acceptable ciphers. See link for more:
603                                                 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
604                                         </para></description>
605                                 </configOption>
606                                 <configOption name="dtlscafile">
607                                         <synopsis>Path to certificate authority certificate</synopsis>
608                                         <description><para>
609                                                 This option only applies if <replaceable>media_encryption</replaceable> is
610                                                 set to <literal>dtls</literal>.
611                                         </para></description>
612                                 </configOption>
613                                 <configOption name="dtlscapath">
614                                         <synopsis>Path to a directory containing certificate authority certificates</synopsis>
615                                         <description><para>
616                                                 This option only applies if <replaceable>media_encryption</replaceable> is
617                                                 set to <literal>dtls</literal>.
618                                         </para></description>
619                                 </configOption>
620                                 <configOption name="dtlssetup">
621                                         <synopsis>Whether we are willing to accept connections, connect to the other party, or both.</synopsis>
622                                         <description>
623                                                 <para>
624                                                         This option only applies if <replaceable>media_encryption</replaceable> is
625                                                         set to <literal>dtls</literal>.
626                                                 </para>
627                                                 <enumlist>
628                                                         <enum name="active"><para>
629                                                                 res_pjsip will make a connection to the peer.
630                                                         </para></enum>
631                                                         <enum name="passive"><para>
632                                                                 res_pjsip will accept connections from the peer.
633                                                         </para></enum>
634                                                         <enum name="actpass"><para>
635                                                                 res_pjsip will offer and accept connections from the peer.
636                                                         </para></enum>
637                                                 </enumlist>
638                                         </description>
639                                 </configOption>
640                                 <configOption name="srtp_tag_32">
641                                         <synopsis>Determines whether 32 byte tags should be used instead of 80 byte tags.</synopsis>
642                                         <description><para>
643                                                 This option only applies if <replaceable>media_encryption</replaceable> is
644                                                 set to <literal>sdes</literal> or <literal>dtls</literal>.
645                                         </para></description>
646                                 </configOption>
647                         </configObject>
648                         <configObject name="auth">
649                                 <synopsis>Authentication type</synopsis>
650                                 <description><para>
651                                         Authentication objects hold the authentication information for use
652                                         by other objects such as <literal>endpoints</literal> or <literal>registrations</literal>.
653                                         This also allows for multiple objects to use a single auth object. See
654                                         the <literal>auth_type</literal> config option for password style choices.
655                                 </para></description>
656                                 <configOption name="auth_type" default="userpass">
657                                         <synopsis>Authentication type</synopsis>
658                                         <description><para>
659                                                 This option specifies which of the password style config options should be read
660                                                 when trying to authenticate an endpoint inbound request. If set to <literal>userpass</literal>
661                                                 then we'll read from the 'password' option. For <literal>md5</literal> we'll read
662                                                 from 'md5_cred'.
663                                                 </para>
664                                                 <enumlist>
665                                                         <enum name="md5"/>
666                                                         <enum name="userpass"/>
667                                                 </enumlist>
668                                         </description>
669                                 </configOption>
670                                 <configOption name="nonce_lifetime" default="32">
671                                         <synopsis>Lifetime of a nonce associated with this authentication config.</synopsis>
672                                 </configOption>
673                                 <configOption name="md5_cred">
674                                         <synopsis>MD5 Hash used for authentication.</synopsis>
675                                         <description><para>Only used when auth_type is <literal>md5</literal>.</para></description>
676                                 </configOption>
677                                 <configOption name="password">
678                                         <synopsis>PlainText password used for authentication.</synopsis>
679                                         <description><para>Only used when auth_type is <literal>userpass</literal>.</para></description>
680                                 </configOption>
681                                 <configOption name="realm" default="asterisk">
682                                         <synopsis>SIP realm for endpoint</synopsis>
683                                 </configOption>
684                                 <configOption name="type">
685                                         <synopsis>Must be 'auth'</synopsis>
686                                 </configOption>
687                                 <configOption name="username">
688                                         <synopsis>Username to use for account</synopsis>
689                                 </configOption>
690                         </configObject>
691                         <configObject name="domain_alias">
692                                 <synopsis>Domain Alias</synopsis>
693                                 <description><para>
694                                         Signifies that a domain is an alias. If the domain on a session is
695                                         not found to match an AoR then this object is used to see if we have
696                                         an alias for the AoR to which the endpoint is binding. This objects
697                                         name as defined in configuration should be the domain alias and a
698                                         config option is provided to specify the domain to be aliased.
699                                 </para></description>
700                                 <configOption name="type">
701                                         <synopsis>Must be of type 'domain_alias'.</synopsis>
702                                 </configOption>
703                                 <configOption name="domain">
704                                         <synopsis>Domain to be aliased</synopsis>
705                                 </configOption>
706                         </configObject>
707                         <configObject name="transport">
708                                 <synopsis>SIP Transport</synopsis>
709                                 <description><para>
710                                         <emphasis>Transports</emphasis>
711                                         </para>
712                                         <para>There are different transports and protocol derivatives
713                                                 supported by <literal>res_pjsip</literal>. They are in order of
714                                                 preference: UDP, TCP, and WebSocket (WS).</para>
715                                         <note><para>Changes to transport configuration in pjsip.conf will only be
716                                                 effected on a complete restart of Asterisk. A module reload
717                                                 will not suffice.</para></note>
718                                 </description>
719                                 <configOption name="async_operations" default="1">
720                                         <synopsis>Number of simultaneous Asynchronous Operations</synopsis>
721                                 </configOption>
722                                 <configOption name="bind">
723                                         <synopsis>IP Address and optional port to bind to for this transport</synopsis>
724                                 </configOption>
725                                 <configOption name="ca_list_file">
726                                         <synopsis>File containing a list of certificates to read (TLS ONLY)</synopsis>
727                                 </configOption>
728                                 <configOption name="cert_file">
729                                         <synopsis>Certificate file for endpoint (TLS ONLY)</synopsis>
730                                 </configOption>
731                                 <configOption name="cipher">
732                                         <synopsis>Preferred Cryptography Cipher (TLS ONLY)</synopsis>
733                                         <description><para>
734                                                 Many options for acceptable ciphers see link for more:
735                                                 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
736                                         </para></description>
737                                 </configOption>
738                                 <configOption name="domain">
739                                         <synopsis>Domain the transport comes from</synopsis>
740                                 </configOption>
741                                 <configOption name="external_media_address">
742                                         <synopsis>External IP address to use in RTP handling</synopsis>
743                                         <description><para>
744                                                 When a request or response is sent out, if the destination of the
745                                                 message is outside the IP network defined in the option <literal>localnet</literal>,
746                                                 and the media address in the SDP is within the localnet network, then the
747                                                 media address in the SDP will be rewritten to the value defined for
748                                                 <literal>external_media_address</literal>.
749                                         </para></description>
750                                 </configOption>
751                                 <configOption name="external_signaling_address">
752                                         <synopsis>External address for SIP signalling</synopsis>
753                                 </configOption>
754                                 <configOption name="external_signaling_port" default="0">
755                                         <synopsis>External port for SIP signalling</synopsis>
756                                 </configOption>
757                                 <configOption name="method">
758                                         <synopsis>Method of SSL transport (TLS ONLY)</synopsis>
759                                         <description>
760                                                 <enumlist>
761                                                         <enum name="default" />
762                                                         <enum name="unspecified" />
763                                                         <enum name="tlsv1" />
764                                                         <enum name="sslv2" />
765                                                         <enum name="sslv3" />
766                                                         <enum name="sslv23" />
767                                                 </enumlist>
768                                         </description>
769                                 </configOption>
770                                 <configOption name="localnet">
771                                         <synopsis>Network to consider local (used for NAT purposes).</synopsis>
772                                         <description><para>This must be in CIDR or dotted decimal format with the IP
773                                         and mask separated with a slash ('/').</para></description>
774                                 </configOption>
775                                 <configOption name="password">
776                                         <synopsis>Password required for transport</synopsis>
777                                 </configOption>
778                                 <configOption name="privkey_file">
779                                         <synopsis>Private key file (TLS ONLY)</synopsis>
780                                 </configOption>
781                                 <configOption name="protocol" default="udp">
782                                         <synopsis>Protocol to use for SIP traffic</synopsis>
783                                         <description>
784                                                 <enumlist>
785                                                         <enum name="udp" />
786                                                         <enum name="tcp" />
787                                                         <enum name="tls" />
788                                                         <enum name="ws" />
789                                                         <enum name="wss" />
790                                                 </enumlist>
791                                         </description>
792                                 </configOption>
793                                 <configOption name="require_client_cert" default="false">
794                                         <synopsis>Require client certificate (TLS ONLY)</synopsis>
795                                 </configOption>
796                                 <configOption name="type">
797                                         <synopsis>Must be of type 'transport'.</synopsis>
798                                 </configOption>
799                                 <configOption name="verify_client" default="false">
800                                         <synopsis>Require verification of client certificate (TLS ONLY)</synopsis>
801                                 </configOption>
802                                 <configOption name="verify_server" default="false">
803                                         <synopsis>Require verification of server certificate (TLS ONLY)</synopsis>
804                                 </configOption>
805                                 <configOption name="tos" default="false">
806                                         <synopsis>Enable TOS for the signalling sent over this transport</synopsis>
807                                         <description>
808                                         <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
809                                         for more information on this parameter.</para>
810                                         <note><para>This option does not apply to the <replaceable>ws</replaceable>
811                                         or the <replaceable>wss</replaceable> protocols.</para></note>
812                                         </description>
813                                 </configOption>
814                                 <configOption name="cos" default="false">
815                                         <synopsis>Enable COS for the signalling sent over this transport</synopsis>
816                                         <description>
817                                         <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
818                                         for more information on this parameter.</para>
819                                         <note><para>This option does not apply to the <replaceable>ws</replaceable>
820                                         or the <replaceable>wss</replaceable> protocols.</para></note>
821                                         </description>
822                                 </configOption>
823                         </configObject>
824                         <configObject name="contact">
825                                 <synopsis>A way of creating an aliased name to a SIP URI</synopsis>
826                                 <description><para>
827                                         Contacts are a way to hide SIP URIs from the dialplan directly.
828                                         They are also used to make a group of contactable parties when
829                                         in use with <literal>AoR</literal> lists.
830                                 </para></description>
831                                 <configOption name="type">
832                                         <synopsis>Must be of type 'contact'.</synopsis>
833                                 </configOption>
834                                 <configOption name="uri">
835                                         <synopsis>SIP URI to contact peer</synopsis>
836                                 </configOption>
837                                 <configOption name="expiration_time">
838                                         <synopsis>Time to keep alive a contact</synopsis>
839                                         <description><para>
840                                                 Time to keep alive a contact. String style specification.
841                                         </para></description>
842                                 </configOption>
843                                 <configOption name="qualify_frequency" default="0">
844                                         <synopsis>Interval at which to qualify a contact</synopsis>
845                                         <description><para>
846                                                 Interval between attempts to qualify the contact for reachability.
847                                                 If <literal>0</literal> never qualify. Time in seconds.
848                                         </para></description>
849                                 </configOption>
850                         </configObject>
851                         <configObject name="aor">
852                                 <synopsis>The configuration for a location of an endpoint</synopsis>
853                                 <description><para>
854                                         An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
855                                         AoRs are specified, an endpoint will not be reachable by Asterisk.
856                                         Beyond that, an AoR has other uses within Asterisk, such as inbound
857                                         registration.
858                                         </para><para>
859                                         An <literal>AoR</literal> is a way to allow dialing a group
860                                         of <literal>Contacts</literal> that all use the same
861                                         <literal>endpoint</literal> for calls.
862                                         </para><para>
863                                         This can be used as another way of grouping a list of contacts to dial
864                                         rather than specifing them each directly when dialing via the dialplan.
865                                         This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
866                                         </para><para>
867                                         Registrations: For Asterisk to match an inbound registration to an endpoint,
868                                         the AoR object name must match the user portion of the SIP URI in the "To:"
869                                         header of the inbound SIP registration. That will usually be equivalent
870                                         to the "user name" set in your hard or soft phones configuration.
871                                 </para></description>
872                                 <configOption name="contact">
873                                         <synopsis>Permanent contacts assigned to AoR</synopsis>
874                                         <description><para>
875                                                 Contacts specified will be called whenever referenced
876                                                 by <literal>chan_pjsip</literal>.
877                                                 </para><para>
878                                                 Use a separate "contact=" entry for each contact required. Contacts
879                                                 are specified using a SIP URI.
880                                         </para></description>
881                                 </configOption>
882                                 <configOption name="default_expiration" default="3600">
883                                         <synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis>
884                                 </configOption>
885                                 <configOption name="mailboxes">
886                                         <synopsis>Mailbox(es) to be associated with</synopsis>
887                                         <description><para>This option applies when an external entity subscribes to an AoR
888                                         for message waiting indications. The mailboxes specified will be subscribed to.
889                                         More than one mailbox can be specified with a comma-delimited string.</para></description>
890                                 </configOption>
891                                 <configOption name="maximum_expiration" default="7200">
892                                         <synopsis>Maximum time to keep an AoR</synopsis>
893                                         <description><para>
894                                                 Maximium time to keep a peer with explicit expiration. Time in seconds.
895                                         </para></description>
896                                 </configOption>
897                                 <configOption name="max_contacts" default="0">
898                                         <synopsis>Maximum number of contacts that can bind to an AoR</synopsis>
899                                         <description><para>
900                                                 Maximum number of contacts that can associate with this AoR. This value does
901                                                 not affect the number of contacts that can be added with the "contact" option.
902                                                 It only limits contacts added through external interaction, such as
903                                                 registration.
904                                                 </para>
905                                                 <note><para>This should be set to <literal>1</literal> and
906                                                 <replaceable>remove_existing</replaceable> set to <literal>yes</literal> if you
907                                                 wish to stick with the older <literal>chan_sip</literal> behaviour.
908                                                 </para></note>
909                                         </description>
910                                 </configOption>
911                                 <configOption name="minimum_expiration" default="60">
912                                         <synopsis>Minimum keep alive time for an AoR</synopsis>
913                                         <description><para>
914                                                 Minimum time to keep a peer with an explict expiration. Time in seconds.
915                                         </para></description>
916                                 </configOption>
917                                 <configOption name="remove_existing" default="no">
918                                         <synopsis>Determines whether new contacts replace existing ones.</synopsis>
919                                         <description><para>
920                                                 On receiving a new registration to the AoR should it remove
921                                                 the existing contact that was registered against it?
922                                                 </para>
923                                                 <note><para>This should be set to <literal>yes</literal> and
924                                                 <replaceable>max_contacts</replaceable> set to <literal>1</literal> if you
925                                                 wish to stick with the older <literal>chan_sip</literal> behaviour.
926                                                 </para></note>
927                                         </description>
928                                 </configOption>
929                                 <configOption name="type">
930                                         <synopsis>Must be of type 'aor'.</synopsis>
931                                 </configOption>
932                                 <configOption name="qualify_frequency" default="0">
933                                         <synopsis>Interval at which to qualify an AoR</synopsis>
934                                         <description><para>
935                                                 Interval between attempts to qualify the AoR for reachability.
936                                                 If <literal>0</literal> never qualify. Time in seconds.
937                                         </para></description>
938                                 </configOption>
939                                 <configOption name="authenticate_qualify" default="no">
940                                         <synopsis>Authenticates a qualify request if needed</synopsis>
941                                         <description><para>
942                                                 If true and a qualify request receives a challenge or authenticate response
943                                                 authentication is attempted before declaring the contact available.
944                                         </para></description>
945                                 </configOption>
946                         </configObject>
947                         <configObject name="system">
948                                 <synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis>
949                                 <description><para>
950                                         The settings in this section are global. In addition to being global, the values will
951                                         not be re-evaluated when a reload is performed. This is because the values must be set
952                                         before the SIP stack is initialized. The only way to reset these values is to either
953                                         restart Asterisk, or unload res_pjsip.so and then load it again.
954                                 </para></description>
955                                 <configOption name="timert1" default="500">
956                                         <synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
957                                         <description><para>
958                                                 Timer T1 is the base for determining how long to wait before retransmitting
959                                                 requests that receive no response when using an unreliable transport (e.g. UDP).
960                                                 For more information on this timer, see RFC 3261, Section 17.1.1.1.
961                                         </para></description>
962                                 </configOption>
963                                 <configOption name="timerb" default="32000">
964                                         <synopsis>Set transaction timer B value (milliseconds).</synopsis>
965                                         <description><para>
966                                                 Timer B determines the maximum amount of time to wait after sending an INVITE
967                                                 request before terminating the transaction. It is recommended that this be set
968                                                 to 64 * Timer T1, but it may be set higher if desired. For more information on
969                                                 this timer, see RFC 3261, Section 17.1.1.1.
970                                         </para></description>
971                                 </configOption>
972                                 <configOption name="compactheaders" default="no">
973                                         <synopsis>Use the short forms of common SIP header names.</synopsis>
974                                 </configOption>
975                                 <configOption name="threadpool_initial_size" default="0">
976                                         <synopsis>Initial number of threads in the res_pjsip threadpool.</synopsis>
977                                 </configOption>
978                                 <configOption name="threadpool_auto_increment" default="5">
979                                         <synopsis>The amount by which the number of threads is incremented when necessary.</synopsis>
980                                 </configOption>
981                                 <configOption name="threadpool_idle_timeout" default="60">
982                                         <synopsis>Number of seconds before an idle thread should be disposed of.</synopsis>
983                                 </configOption>
984                                 <configOption name="threadpool_max_size" default="0">
985                                         <synopsis>Maximum number of threads in the res_pjsip threadpool.
986                                         A value of 0 indicates no maximum.</synopsis>
987                                 </configOption>
988                                 <configOption name="type">
989                                         <synopsis>Must be of type 'system'.</synopsis>
990                                 </configOption>
991                         </configObject>
992                         <configObject name="global">
993                                 <synopsis>Options that apply globally to all SIP communications</synopsis>
994                                 <description><para>
995                                         The settings in this section are global. Unlike options in the <literal>system</literal>
996                                         section, these options can be refreshed by performing a reload.
997                                 </para></description>
998                                 <configOption name="maxforwards" default="70">
999                                         <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
1000                                 </configOption>
1001                                 <configOption name="type">
1002                                         <synopsis>Must be of type 'global'.</synopsis>
1003                                 </configOption>
1004                                 <configOption name="useragent" default="Asterisk &lt;Asterisk Version&gt;">
1005                                         <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
1006                                 </configOption>
1007                         </configObject>
1008                 </configFile>
1009         </configInfo>
1010         <manager name="PJSIPQualify" language="en_US">
1011                 <synopsis>
1012                         Qualify a chan_pjsip endpoint.
1013                 </synopsis>
1014                 <syntax>
1015                         <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
1016                         <parameter name="Endpoint" required="true">
1017                                 <para>The endpoint you want to qualify.</para>
1018                         </parameter>
1019                 </syntax>
1020                 <description>
1021                         <para>Qualify a chan_pjsip endpoint.</para>
1022                 </description>
1023         </manager>
1024  ***/
1025
1026
1027 static pjsip_endpoint *ast_pjsip_endpoint;
1028
1029 static struct ast_threadpool *sip_threadpool;
1030
1031 static int register_service(void *data)
1032 {
1033         pjsip_module **module = data;
1034         if (!ast_pjsip_endpoint) {
1035                 ast_log(LOG_ERROR, "There is no PJSIP endpoint. Unable to register services\n");
1036                 return -1;
1037         }
1038         if (pjsip_endpt_register_module(ast_pjsip_endpoint, *module) != PJ_SUCCESS) {
1039                 ast_log(LOG_ERROR, "Unable to register module %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1040                 return -1;
1041         }
1042         ast_debug(1, "Registered SIP service %.*s (%p)\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name), *module);
1043         ast_module_ref(ast_module_info->self);
1044         return 0;
1045 }
1046
1047 int ast_sip_register_service(pjsip_module *module)
1048 {
1049         return ast_sip_push_task_synchronous(NULL, register_service, &module);
1050 }
1051
1052 static int unregister_service(void *data)
1053 {
1054         pjsip_module **module = data;
1055         ast_module_unref(ast_module_info->self);
1056         if (!ast_pjsip_endpoint) {
1057                 return -1;
1058         }
1059         pjsip_endpt_unregister_module(ast_pjsip_endpoint, *module);
1060         ast_debug(1, "Unregistered SIP service %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1061         return 0;
1062 }
1063
1064 void ast_sip_unregister_service(pjsip_module *module)
1065 {
1066         ast_sip_push_task_synchronous(NULL, unregister_service, &module);
1067 }
1068
1069 static struct ast_sip_authenticator *registered_authenticator;
1070
1071 int ast_sip_register_authenticator(struct ast_sip_authenticator *auth)
1072 {
1073         if (registered_authenticator) {
1074                 ast_log(LOG_WARNING, "Authenticator %p is already registered. Cannot register a new one\n", registered_authenticator);
1075                 return -1;
1076         }
1077         registered_authenticator = auth;
1078         ast_debug(1, "Registered SIP authenticator module %p\n", auth);
1079         ast_module_ref(ast_module_info->self);
1080         return 0;
1081 }
1082
1083 void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth)
1084 {
1085         if (registered_authenticator != auth) {
1086                 ast_log(LOG_WARNING, "Trying to unregister authenticator %p but authenticator %p registered\n",
1087                                 auth, registered_authenticator);
1088                 return;
1089         }
1090         registered_authenticator = NULL;
1091         ast_debug(1, "Unregistered SIP authenticator %p\n", auth);
1092         ast_module_unref(ast_module_info->self);
1093 }
1094
1095 int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1096 {
1097         if (!registered_authenticator) {
1098                 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is not required\n");
1099                 return 0;
1100         }
1101
1102         return registered_authenticator->requires_authentication(endpoint, rdata);
1103 }
1104
1105 enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint,
1106                 pjsip_rx_data *rdata, pjsip_tx_data *tdata)
1107 {
1108         if (!registered_authenticator) {
1109                 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is successful\n");
1110                 return 0;
1111         }
1112         return registered_authenticator->check_authentication(endpoint, rdata, tdata);
1113 }
1114
1115 static struct ast_sip_outbound_authenticator *registered_outbound_authenticator;
1116
1117 int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1118 {
1119         if (registered_outbound_authenticator) {
1120                 ast_log(LOG_WARNING, "Outbound authenticator %p is already registered. Cannot register a new one\n", registered_outbound_authenticator);
1121                 return -1;
1122         }
1123         registered_outbound_authenticator = auth;
1124         ast_debug(1, "Registered SIP outbound authenticator module %p\n", auth);
1125         ast_module_ref(ast_module_info->self);
1126         return 0;
1127 }
1128
1129 void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1130 {
1131         if (registered_outbound_authenticator != auth) {
1132                 ast_log(LOG_WARNING, "Trying to unregister outbound authenticator %p but outbound authenticator %p registered\n",
1133                                 auth, registered_outbound_authenticator);
1134                 return;
1135         }
1136         registered_outbound_authenticator = NULL;
1137         ast_debug(1, "Unregistered SIP outbound authenticator %p\n", auth);
1138         ast_module_unref(ast_module_info->self);
1139 }
1140
1141 int ast_sip_create_request_with_auth(const struct ast_sip_auth_array *auths, pjsip_rx_data *challenge,
1142                 pjsip_transaction *tsx, pjsip_tx_data **new_request)
1143 {
1144         if (!registered_outbound_authenticator) {
1145                 ast_log(LOG_WARNING, "No SIP outbound authenticator registered. Cannot respond to authentication challenge\n");
1146                 return -1;
1147         }
1148         return registered_outbound_authenticator->create_request_with_auth(auths, challenge, tsx, new_request);
1149 }
1150
1151 struct endpoint_identifier_list {
1152         struct ast_sip_endpoint_identifier *identifier;
1153         AST_RWLIST_ENTRY(endpoint_identifier_list) list;
1154 };
1155
1156 static AST_RWLIST_HEAD_STATIC(endpoint_identifiers, endpoint_identifier_list);
1157
1158 int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1159 {
1160         struct endpoint_identifier_list *id_list_item;
1161         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1162
1163         id_list_item = ast_calloc(1, sizeof(*id_list_item));
1164         if (!id_list_item) {
1165                 ast_log(LOG_ERROR, "Unabled to add endpoint identifier. Out of memory.\n");
1166                 return -1;
1167         }
1168         id_list_item->identifier = identifier;
1169
1170         AST_RWLIST_INSERT_TAIL(&endpoint_identifiers, id_list_item, list);
1171         ast_debug(1, "Registered endpoint identifier %p\n", identifier);
1172
1173         ast_module_ref(ast_module_info->self);
1174         return 0;
1175 }
1176
1177 void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1178 {
1179         struct endpoint_identifier_list *iter;
1180         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1181         AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_identifiers, iter, list) {
1182                 if (iter->identifier == identifier) {
1183                         AST_RWLIST_REMOVE_CURRENT(list);
1184                         ast_free(iter);
1185                         ast_debug(1, "Unregistered endpoint identifier %p\n", identifier);
1186                         ast_module_unref(ast_module_info->self);
1187                         break;
1188                 }
1189         }
1190         AST_RWLIST_TRAVERSE_SAFE_END;
1191 }
1192
1193 struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata)
1194 {
1195         struct endpoint_identifier_list *iter;
1196         struct ast_sip_endpoint *endpoint = NULL;
1197         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1198         AST_RWLIST_TRAVERSE(&endpoint_identifiers, iter, list) {
1199                 ast_assert(iter->identifier->identify_endpoint != NULL);
1200                 endpoint = iter->identifier->identify_endpoint(rdata);
1201                 if (endpoint) {
1202                         break;
1203                 }
1204         }
1205         return endpoint;
1206 }
1207
1208 pjsip_endpoint *ast_sip_get_pjsip_endpoint(void)
1209 {
1210         return ast_pjsip_endpoint;
1211 }
1212
1213 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)
1214 {
1215         pj_str_t tmp, local_addr;
1216         pjsip_uri *uri;
1217         pjsip_sip_uri *sip_uri;
1218         pjsip_transport_type_e type = PJSIP_TRANSPORT_UNSPECIFIED;
1219         int local_port;
1220         char uuid_str[AST_UUID_STR_LEN];
1221
1222         if (ast_strlen_zero(user)) {
1223                 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1224                 if (!uuid) {
1225                         return -1;
1226                 }
1227                 user = ast_uuid_to_str(uuid, uuid_str, sizeof(uuid_str));
1228         }
1229
1230         /* Parse the provided target URI so we can determine what transport it will end up using */
1231         pj_strdup_with_null(pool, &tmp, target);
1232
1233         if (!(uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0)) ||
1234             (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
1235                 return -1;
1236         }
1237
1238         sip_uri = pjsip_uri_get_uri(uri);
1239
1240         /* Determine the transport type to use */
1241         if (PJSIP_URI_SCHEME_IS_SIPS(sip_uri)) {
1242                 type = PJSIP_TRANSPORT_TLS;
1243         } else if (!sip_uri->transport_param.slen) {
1244                 type = PJSIP_TRANSPORT_UDP;
1245         } else {
1246                 type = pjsip_transport_get_type_from_name(&sip_uri->transport_param);
1247         }
1248
1249         if (type == PJSIP_TRANSPORT_UNSPECIFIED) {
1250                 return -1;
1251         }
1252
1253         /* If the host is IPv6 turn the transport into an IPv6 version */
1254         if (pj_strchr(&sip_uri->host, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1255                 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1256         }
1257
1258         if (!ast_strlen_zero(domain)) {
1259                 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1260                 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1261                                 "<%s:%s@%s%s%s>",
1262                                 (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1263                                 user,
1264                                 domain,
1265                                 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1266                                 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1267                 return 0;
1268         }
1269
1270         /* Get the local bound address for the transport that will be used when communicating with the provided URI */
1271         if (pjsip_tpmgr_find_local_addr(pjsip_endpt_get_tpmgr(ast_sip_get_pjsip_endpoint()), pool, type, selector,
1272                                                               &local_addr, &local_port) != PJ_SUCCESS) {
1273                 return -1;
1274         }
1275
1276         /* If IPv6 was specified in the transport, set the proper type */
1277         if (pj_strchr(&local_addr, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1278                 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1279         }
1280
1281         from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1282         from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1283                                       "<%s:%s@%s%.*s%s:%d%s%s>",
1284                                       (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1285                                       user,
1286                                       (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1287                                       (int)local_addr.slen,
1288                                       local_addr.ptr,
1289                                       (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1290                                       local_port,
1291                                       (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1292                                       (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1293
1294         return 0;
1295 }
1296
1297 static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpoint, pjsip_tpselector *selector)
1298 {
1299         RAII_VAR(struct ast_sip_transport *, transport, NULL, ao2_cleanup);
1300         const char *transport_name = endpoint->transport;
1301
1302         if (ast_strlen_zero(transport_name)) {
1303                 return 0;
1304         }
1305
1306         transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_name);
1307
1308         if (!transport || !transport->state) {
1309                 ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport '%s' for endpoint '%s'\n",
1310                         transport_name, ast_sorcery_object_get_id(endpoint));
1311                 return -1;
1312         }
1313
1314         if (transport->state->transport) {
1315                 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1316                 selector->u.transport = transport->state->transport;
1317         } else if (transport->state->factory) {
1318                 selector->type = PJSIP_TPSELECTOR_LISTENER;
1319                 selector->u.listener = transport->state->factory;
1320         } else {
1321                 return -1;
1322         }
1323
1324         return 0;
1325 }
1326
1327 static int sip_get_tpselector_from_uri(const char *uri, pjsip_tpselector *selector)
1328 {
1329         RAII_VAR(struct ast_sip_contact_transport *, contact_transport, NULL, ao2_cleanup);
1330
1331         contact_transport = ast_sip_location_retrieve_contact_transport_by_uri(uri);
1332
1333         if (!contact_transport) {
1334                 return -1;
1335         }
1336
1337         selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1338         selector->u.transport = contact_transport->transport;
1339
1340         return 0;
1341 }
1342
1343 pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user)
1344 {
1345         char enclosed_uri[PJSIP_MAX_URL_SIZE];
1346         pj_str_t local_uri = { "sip:temp@temp", 13 }, remote_uri, target_uri;
1347         pjsip_dialog *dlg = NULL;
1348         const char *outbound_proxy = endpoint->outbound_proxy;
1349         pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1350         static const pj_str_t HCONTACT = { "Contact", 7 };
1351
1352         snprintf(enclosed_uri, sizeof(enclosed_uri), "<%s>", uri);
1353         pj_cstr(&remote_uri, enclosed_uri);
1354
1355         pj_cstr(&target_uri, uri);
1356
1357         if (pjsip_dlg_create_uac(pjsip_ua_instance(), &local_uri, NULL, &remote_uri, &target_uri, &dlg) != PJ_SUCCESS) {
1358                 return NULL;
1359         }
1360
1361         if (sip_get_tpselector_from_uri(uri, &selector) && sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1362                 pjsip_dlg_terminate(dlg);
1363                 return NULL;
1364         }
1365
1366         if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
1367                 pjsip_dlg_terminate(dlg);
1368                 return NULL;
1369         }
1370
1371         /* Update the dialog with the new local URI, we do it afterwards so we can use the dialog pool for construction */
1372         pj_strdup_with_null(dlg->pool, &dlg->local.info_str, &local_uri);
1373         dlg->local.info->uri = pjsip_parse_uri(dlg->pool, dlg->local.info_str.ptr, dlg->local.info_str.slen, 0);
1374         dlg->local.contact = pjsip_parse_hdr(dlg->pool, &HCONTACT, local_uri.ptr, local_uri.slen, NULL);
1375
1376         /* If a request user has been specified and we are permitted to change it, do so */
1377         if (!ast_strlen_zero(request_user)) {
1378                 pjsip_sip_uri *sip_uri;
1379
1380                 if (PJSIP_URI_SCHEME_IS_SIP(dlg->target) || PJSIP_URI_SCHEME_IS_SIPS(dlg->target)) {
1381                         sip_uri = pjsip_uri_get_uri(dlg->target);
1382                         pj_strdup2(dlg->pool, &sip_uri->user, request_user);
1383                 }
1384                 if (PJSIP_URI_SCHEME_IS_SIP(dlg->remote.info->uri) || PJSIP_URI_SCHEME_IS_SIPS(dlg->remote.info->uri)) {
1385                         sip_uri = pjsip_uri_get_uri(dlg->remote.info->uri);
1386                         pj_strdup2(dlg->pool, &sip_uri->user, request_user);
1387                 }
1388         }
1389
1390         /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
1391         dlg->sess_count++;
1392
1393         pjsip_dlg_set_transport(dlg, &selector);
1394
1395         if (!ast_strlen_zero(outbound_proxy)) {
1396                 pjsip_route_hdr route_set, *route;
1397                 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1398                 pj_str_t tmp;
1399
1400                 pj_list_init(&route_set);
1401
1402                 pj_strdup2_with_null(dlg->pool, &tmp, outbound_proxy);
1403                 if (!(route = pjsip_parse_hdr(dlg->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1404                         pjsip_dlg_terminate(dlg);
1405                         return NULL;
1406                 }
1407                 pj_list_push_back(&route_set, route);
1408
1409                 pjsip_dlg_set_route_set(dlg, &route_set);
1410         }
1411
1412         dlg->sess_count--;
1413
1414         return dlg;
1415 }
1416
1417 pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1418 {
1419         pjsip_dialog *dlg;
1420         pj_str_t contact;
1421         pjsip_transport_type_e type = rdata->tp_info.transport->key.type;
1422         pj_status_t status;
1423
1424         contact.ptr = pj_pool_alloc(rdata->tp_info.pool, PJSIP_MAX_URL_SIZE);
1425         contact.slen = pj_ansi_snprintf(contact.ptr, PJSIP_MAX_URL_SIZE,
1426                         "<%s:%s%.*s%s:%d%s%s>",
1427                         (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1428                         (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1429                         (int)rdata->tp_info.transport->local_name.host.slen,
1430                         rdata->tp_info.transport->local_name.host.ptr,
1431                         (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1432                         rdata->tp_info.transport->local_name.port,
1433                         (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1434                         (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1435
1436         status = pjsip_dlg_create_uas(pjsip_ua_instance(), rdata, &contact, &dlg);
1437         if (status != PJ_SUCCESS) {
1438                 char err[PJ_ERR_MSG_SIZE];
1439
1440                 pjsip_strerror(status, err, sizeof(err));
1441                 ast_log(LOG_ERROR, "Could not create dialog with endpoint %s. %s\n",
1442                                 ast_sorcery_object_get_id(endpoint), err);
1443                 return NULL;
1444         }
1445
1446         return dlg;
1447 }
1448
1449 /* PJSIP doesn't know about the INFO method, so we have to define it ourselves */
1450 static const pjsip_method info_method = {PJSIP_OTHER_METHOD, {"INFO", 4} };
1451 static const pjsip_method message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
1452
1453 static struct {
1454         const char *method;
1455         const pjsip_method *pmethod;
1456 } methods [] = {
1457         { "INVITE", &pjsip_invite_method },
1458         { "CANCEL", &pjsip_cancel_method },
1459         { "ACK", &pjsip_ack_method },
1460         { "BYE", &pjsip_bye_method },
1461         { "REGISTER", &pjsip_register_method },
1462         { "OPTIONS", &pjsip_options_method },
1463         { "SUBSCRIBE", &pjsip_subscribe_method },
1464         { "NOTIFY", &pjsip_notify_method },
1465         { "PUBLISH", &pjsip_publish_method },
1466         { "INFO", &info_method },
1467         { "MESSAGE", &message_method },
1468 };
1469
1470 static const pjsip_method *get_pjsip_method(const char *method)
1471 {
1472         int i;
1473         for (i = 0; i < ARRAY_LEN(methods); ++i) {
1474                 if (!strcmp(method, methods[i].method)) {
1475                         return methods[i].pmethod;
1476                 }
1477         }
1478         return NULL;
1479 }
1480
1481 static int create_in_dialog_request(const pjsip_method *method, struct pjsip_dialog *dlg, pjsip_tx_data **tdata)
1482 {
1483         if (pjsip_dlg_create_request(dlg, method, -1, tdata) != PJ_SUCCESS) {
1484                 ast_log(LOG_WARNING, "Unable to create in-dialog request.\n");
1485                 return -1;
1486         }
1487
1488         return 0;
1489 }
1490
1491 static int create_out_of_dialog_request(const pjsip_method *method, struct ast_sip_endpoint *endpoint,
1492                 const char *uri, pjsip_tx_data **tdata)
1493 {
1494         RAII_VAR(struct ast_sip_contact *, contact, NULL, ao2_cleanup);
1495         pj_str_t remote_uri;
1496         pj_str_t from;
1497         pj_pool_t *pool;
1498         pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1499
1500         if (ast_strlen_zero(uri)) {
1501                 if (!endpoint) {
1502                         ast_log(LOG_ERROR, "An endpoint and/or uri must be specified\n");
1503                         return -1;
1504                 }
1505
1506                 contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
1507                 if (!contact || ast_strlen_zero(contact->uri)) {
1508                         ast_log(LOG_ERROR, "Unable to retrieve contact for endpoint %s\n",
1509                                         ast_sorcery_object_get_id(endpoint));
1510                         return -1;
1511                 }
1512
1513                 pj_cstr(&remote_uri, contact->uri);
1514         } else {
1515                 pj_cstr(&remote_uri, uri);
1516         }
1517
1518         if (endpoint) {
1519                 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1520                         ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport selector for endpoint %s\n",
1521                                 ast_sorcery_object_get_id(endpoint));
1522                         return -1;
1523                 }
1524         }
1525
1526         pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "Outbound request", 256, 256);
1527
1528         if (!pool) {
1529                 ast_log(LOG_ERROR, "Unable to create PJLIB memory pool\n");
1530                 return -1;
1531         }
1532
1533         if (sip_dialog_create_from(pool, &from, endpoint ? endpoint->fromuser : NULL,
1534                                 endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) {
1535                 ast_log(LOG_ERROR, "Unable to create From header for %.*s request to endpoint %s\n",
1536                                 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1537                 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1538                 return -1;
1539         }
1540
1541         if (pjsip_endpt_create_request(ast_sip_get_pjsip_endpoint(), method, &remote_uri,
1542                         &from, &remote_uri, &from, NULL, -1, NULL, tdata) != PJ_SUCCESS) {
1543                 ast_log(LOG_ERROR, "Unable to create outbound %.*s request to endpoint %s\n",
1544                                 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1545                 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1546                 return -1;
1547         }
1548
1549         /* We can release this pool since request creation copied all the necessary
1550          * data into the outbound request's pool
1551          */
1552         pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1553         return 0;
1554 }
1555
1556 int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
1557                 struct ast_sip_endpoint *endpoint, const char *uri,
1558                 pjsip_tx_data **tdata)
1559 {
1560         const pjsip_method *pmethod = get_pjsip_method(method);
1561
1562         if (!pmethod) {
1563                 ast_log(LOG_WARNING, "Unknown method '%s'. Cannot send request\n", method);
1564                 return -1;
1565         }
1566
1567         if (dlg) {
1568                 return create_in_dialog_request(pmethod, dlg, tdata);
1569         } else {
1570                 return create_out_of_dialog_request(pmethod, endpoint, uri, tdata);
1571         }
1572 }
1573
1574 static int send_in_dialog_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg)
1575 {
1576         if (pjsip_dlg_send_request(dlg, tdata, -1, NULL) != PJ_SUCCESS) {
1577                 ast_log(LOG_WARNING, "Unable to send in-dialog request.\n");
1578                 return -1;
1579         }
1580         return 0;
1581 }
1582
1583 static void send_request_cb(void *token, pjsip_event *e)
1584 {
1585         RAII_VAR(struct ast_sip_endpoint *, endpoint, token, ao2_cleanup);
1586         pjsip_transaction *tsx = e->body.tsx_state.tsx;
1587         pjsip_rx_data *challenge = e->body.tsx_state.src.rdata;
1588         pjsip_tx_data *tdata;
1589
1590         if (tsx->status_code != 401 && tsx->status_code != 407) {
1591                 return;
1592         }
1593
1594         if (!ast_sip_create_request_with_auth(&endpoint->outbound_auths, challenge, tsx, &tdata)) {
1595                 pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, NULL, NULL);
1596         }
1597 }
1598
1599 static int send_out_of_dialog_request(pjsip_tx_data *tdata, struct ast_sip_endpoint *endpoint)
1600 {
1601         ao2_ref(endpoint, +1);
1602         if (pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, endpoint, send_request_cb) != PJ_SUCCESS) {
1603                 ast_log(LOG_ERROR, "Error attempting to send outbound %.*s request to endpoint %s\n",
1604                                 (int) pj_strlen(&tdata->msg->line.req.method.name),
1605                                 pj_strbuf(&tdata->msg->line.req.method.name),
1606                                 ast_sorcery_object_get_id(endpoint));
1607                 ao2_ref(endpoint, -1);
1608                 return -1;
1609         }
1610
1611         return 0;
1612 }
1613
1614 int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg, struct ast_sip_endpoint *endpoint)
1615 {
1616         ast_assert(tdata->msg->type == PJSIP_REQUEST_MSG);
1617
1618         if (dlg) {
1619                 return send_in_dialog_request(tdata, dlg);
1620         } else {
1621                 return send_out_of_dialog_request(tdata, endpoint);
1622         }
1623 }
1624
1625 int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value)
1626 {
1627         pj_str_t hdr_name;
1628         pj_str_t hdr_value;
1629         pjsip_generic_string_hdr *hdr;
1630
1631         pj_cstr(&hdr_name, name);
1632         pj_cstr(&hdr_value, value);
1633
1634         hdr = pjsip_generic_string_hdr_create(tdata->pool, &hdr_name, &hdr_value);
1635
1636         pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *) hdr);
1637         return 0;
1638 }
1639
1640 static pjsip_msg_body *ast_body_to_pjsip_body(pj_pool_t *pool, const struct ast_sip_body *body)
1641 {
1642         pj_str_t type;
1643         pj_str_t subtype;
1644         pj_str_t body_text;
1645
1646         pj_cstr(&type, body->type);
1647         pj_cstr(&subtype, body->subtype);
1648         pj_cstr(&body_text, body->body_text);
1649
1650         return pjsip_msg_body_create(pool, &type, &subtype, &body_text);
1651 }
1652
1653 int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body)
1654 {
1655         pjsip_msg_body *pjsip_body = ast_body_to_pjsip_body(tdata->pool, body);
1656         tdata->msg->body = pjsip_body;
1657         return 0;
1658 }
1659
1660 int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies)
1661 {
1662         int i;
1663         /* NULL for type and subtype automatically creates "multipart/mixed" */
1664         pjsip_msg_body *body = pjsip_multipart_create(tdata->pool, NULL, NULL);
1665
1666         for (i = 0; i < num_bodies; ++i) {
1667                 pjsip_multipart_part *part = pjsip_multipart_create_part(tdata->pool);
1668                 part->body = ast_body_to_pjsip_body(tdata->pool, bodies[i]);
1669                 pjsip_multipart_add_part(tdata->pool, body, part);
1670         }
1671
1672         tdata->msg->body = body;
1673         return 0;
1674 }
1675
1676 int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text)
1677 {
1678         size_t combined_size = strlen(body_text) + tdata->msg->body->len;
1679         struct ast_str *body_buffer = ast_str_alloca(combined_size);
1680
1681         ast_str_set(&body_buffer, 0, "%.*s%s", (int) tdata->msg->body->len, (char *) tdata->msg->body->data, body_text);
1682
1683         tdata->msg->body->data = pj_pool_alloc(tdata->pool, combined_size);
1684         pj_memcpy(tdata->msg->body->data, ast_str_buffer(body_buffer), combined_size);
1685         tdata->msg->body->len = combined_size;
1686
1687         return 0;
1688 }
1689
1690 struct ast_taskprocessor *ast_sip_create_serializer(void)
1691 {
1692         struct ast_taskprocessor *serializer;
1693         RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1694         char name[AST_UUID_STR_LEN];
1695
1696         if (!uuid) {
1697                 return NULL;
1698         }
1699
1700         ast_uuid_to_str(uuid, name, sizeof(name));
1701
1702         serializer = ast_threadpool_serializer(name, sip_threadpool);
1703         if (!serializer) {
1704                 return NULL;
1705         }
1706         return serializer;
1707 }
1708
1709 int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1710 {
1711         if (serializer) {
1712                 return ast_taskprocessor_push(serializer, sip_task, task_data);
1713         } else {
1714                 return ast_threadpool_push(sip_threadpool, sip_task, task_data);
1715         }
1716 }
1717
1718 struct sync_task_data {
1719         ast_mutex_t lock;
1720         ast_cond_t cond;
1721         int complete;
1722         int fail;
1723         int (*task)(void *);
1724         void *task_data;
1725 };
1726
1727 static int sync_task(void *data)
1728 {
1729         struct sync_task_data *std = data;
1730         std->fail = std->task(std->task_data);
1731
1732         ast_mutex_lock(&std->lock);
1733         std->complete = 1;
1734         ast_cond_signal(&std->cond);
1735         ast_mutex_unlock(&std->lock);
1736         return std->fail;
1737 }
1738
1739 int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1740 {
1741         /* This method is an onion */
1742         struct sync_task_data std;
1743         ast_mutex_init(&std.lock);
1744         ast_cond_init(&std.cond, NULL);
1745         std.fail = std.complete = 0;
1746         std.task = sip_task;
1747         std.task_data = task_data;
1748
1749         if (serializer) {
1750                 if (ast_taskprocessor_push(serializer, sync_task, &std)) {
1751                         return -1;
1752                 }
1753         } else {
1754                 if (ast_threadpool_push(sip_threadpool, sync_task, &std)) {
1755                         return -1;
1756                 }
1757         }
1758
1759         ast_mutex_lock(&std.lock);
1760         while (!std.complete) {
1761                 ast_cond_wait(&std.cond, &std.lock);
1762         }
1763         ast_mutex_unlock(&std.lock);
1764
1765         ast_mutex_destroy(&std.lock);
1766         ast_cond_destroy(&std.cond);
1767         return std.fail;
1768 }
1769
1770 void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size)
1771 {
1772         size_t chars_to_copy = MIN(size - 1, pj_strlen(src));
1773         memcpy(dest, pj_strbuf(src), chars_to_copy);
1774         dest[chars_to_copy] = '\0';
1775 }
1776
1777 int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype)
1778 {
1779         pjsip_media_type compare;
1780
1781         if (!content_type) {
1782                 return 0;
1783         }
1784
1785         pjsip_media_type_init2(&compare, type, subtype);
1786
1787         return pjsip_media_type_cmp(content_type, &compare, 0) ? 0 : -1;
1788 }
1789
1790 pj_caching_pool caching_pool;
1791 pj_pool_t *memory_pool;
1792 pj_thread_t *monitor_thread;
1793 static int monitor_continue;
1794
1795 static void *monitor_thread_exec(void *endpt)
1796 {
1797         while (monitor_continue) {
1798                 const pj_time_val delay = {0, 10};
1799                 pjsip_endpt_handle_events(ast_pjsip_endpoint, &delay);
1800         }
1801         return NULL;
1802 }
1803
1804 static void stop_monitor_thread(void)
1805 {
1806         monitor_continue = 0;
1807         pj_thread_join(monitor_thread);
1808 }
1809
1810 AST_THREADSTORAGE(pj_thread_storage);
1811 AST_THREADSTORAGE(servant_id_storage);
1812 #define SIP_SERVANT_ID 0x5E2F1D
1813
1814 static void sip_thread_start(void)
1815 {
1816         pj_thread_desc *desc;
1817         pj_thread_t *thread;
1818         uint32_t *servant_id;
1819
1820         servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1821         if (!servant_id) {
1822                 ast_log(LOG_ERROR, "Could not set SIP servant ID in thread-local storage.\n");
1823                 return;
1824         }
1825         *servant_id = SIP_SERVANT_ID;
1826
1827         desc = ast_threadstorage_get(&pj_thread_storage, sizeof(pj_thread_desc));
1828         if (!desc) {
1829                 ast_log(LOG_ERROR, "Could not get thread desc from thread-local storage. Expect awful things to occur\n");
1830                 return;
1831         }
1832         pj_bzero(*desc, sizeof(*desc));
1833
1834         if (pj_thread_register("Asterisk Thread", *desc, &thread) != PJ_SUCCESS) {
1835                 ast_log(LOG_ERROR, "Couldn't register thread with PJLIB.\n");
1836         }
1837 }
1838
1839 int ast_sip_thread_is_servant(void)
1840 {
1841         uint32_t *servant_id;
1842
1843         servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1844         if (!servant_id) {
1845                 return 0;
1846         }
1847
1848         return *servant_id == SIP_SERVANT_ID;
1849 }
1850
1851 void *ast_sip_dict_get(void *ht, const char *key)
1852 {
1853         unsigned int hval;
1854
1855         if (!ht) {
1856                 return NULL;
1857         }
1858
1859         return pj_hash_get(ht, key, PJ_HASH_KEY_STRING, &hval);
1860 }
1861
1862 void *ast_sip_dict_set(pj_pool_t* pool, void *ht,
1863                        const char *key, void *val)
1864 {
1865         if (!ht) {
1866                 ht = pj_hash_create(pool, 11);
1867         }
1868
1869         pj_hash_set(pool, ht, key, PJ_HASH_KEY_STRING, 0, val);
1870
1871         return ht;
1872 }
1873
1874 static void remove_request_headers(pjsip_endpoint *endpt)
1875 {
1876         const pjsip_hdr *request_headers = pjsip_endpt_get_request_headers(endpt);
1877         pjsip_hdr *iter = request_headers->next;
1878
1879         while (iter != request_headers) {
1880                 pjsip_hdr *to_erase = iter;
1881                 iter = iter->next;
1882                 pj_list_erase(to_erase);
1883         }
1884 }
1885
1886 static int load_module(void)
1887 {
1888         /* The third parameter is just copied from
1889          * example code from PJLIB. This can be adjusted
1890          * if necessary.
1891          */
1892         pj_status_t status;
1893         struct ast_threadpool_options options;
1894
1895         if (pj_init() != PJ_SUCCESS) {
1896                 return AST_MODULE_LOAD_DECLINE;
1897         }
1898
1899         if (pjlib_util_init() != PJ_SUCCESS) {
1900                 pj_shutdown();
1901                 return AST_MODULE_LOAD_DECLINE;
1902         }
1903
1904         pj_caching_pool_init(&caching_pool, NULL, 1024 * 1024);
1905         if (pjsip_endpt_create(&caching_pool.factory, "SIP", &ast_pjsip_endpoint) != PJ_SUCCESS) {
1906                 ast_log(LOG_ERROR, "Failed to create PJSIP endpoint structure. Aborting load\n");
1907                 pj_caching_pool_destroy(&caching_pool);
1908                 return AST_MODULE_LOAD_DECLINE;
1909         }
1910
1911         /* PJSIP will automatically try to add a Max-Forwards header. Since we want to control that,
1912          * we need to stop PJSIP from doing it automatically
1913          */
1914         remove_request_headers(ast_pjsip_endpoint);
1915
1916         memory_pool = pj_pool_create(&caching_pool.factory, "SIP", 1024, 1024, NULL);
1917         if (!memory_pool) {
1918                 ast_log(LOG_ERROR, "Failed to create memory pool for SIP. Aborting load\n");
1919                 pjsip_endpt_destroy(ast_pjsip_endpoint);
1920                 ast_pjsip_endpoint = NULL;
1921                 pj_caching_pool_destroy(&caching_pool);
1922                 return AST_MODULE_LOAD_DECLINE;
1923         }
1924
1925         if (ast_sip_initialize_system()) {
1926                 ast_log(LOG_ERROR, "Failed to initialize SIP system configuration. Aborting load\n");
1927                 pj_pool_release(memory_pool);
1928                 memory_pool = NULL;
1929                 pjsip_endpt_destroy(ast_pjsip_endpoint);
1930                 ast_pjsip_endpoint = NULL;
1931                 pj_caching_pool_destroy(&caching_pool);
1932                 return AST_MODULE_LOAD_DECLINE;
1933         }
1934
1935         sip_get_threadpool_options(&options);
1936         options.thread_start = sip_thread_start;
1937         sip_threadpool = ast_threadpool_create("SIP", NULL, &options);
1938         if (!sip_threadpool) {
1939                 ast_log(LOG_ERROR, "Failed to create SIP threadpool. Aborting load\n");
1940                 pj_pool_release(memory_pool);
1941                 memory_pool = NULL;
1942                 pjsip_endpt_destroy(ast_pjsip_endpoint);
1943                 ast_pjsip_endpoint = NULL;
1944                 pj_caching_pool_destroy(&caching_pool);
1945                 return AST_MODULE_LOAD_DECLINE;
1946         }
1947
1948         pjsip_tsx_layer_init_module(ast_pjsip_endpoint);
1949         pjsip_ua_init_module(ast_pjsip_endpoint, NULL);
1950
1951         monitor_continue = 1;
1952         status = pj_thread_create(memory_pool, "SIP", (pj_thread_proc *) &monitor_thread_exec,
1953                         NULL, PJ_THREAD_DEFAULT_STACK_SIZE * 2, 0, &monitor_thread);
1954         if (status != PJ_SUCCESS) {
1955                 ast_log(LOG_ERROR, "Failed to start SIP monitor thread. Aborting load\n");
1956                 pj_pool_release(memory_pool);
1957                 memory_pool = NULL;
1958                 pjsip_endpt_destroy(ast_pjsip_endpoint);
1959                 ast_pjsip_endpoint = NULL;
1960                 pj_caching_pool_destroy(&caching_pool);
1961                 return AST_MODULE_LOAD_DECLINE;
1962         }
1963
1964         ast_sip_initialize_global_headers();
1965
1966         if (ast_res_pjsip_initialize_configuration()) {
1967                 ast_log(LOG_ERROR, "Failed to initialize SIP configuration. Aborting load\n");
1968                 ast_sip_destroy_global_headers();
1969                 stop_monitor_thread();
1970                 pj_pool_release(memory_pool);
1971                 memory_pool = NULL;
1972                 pjsip_endpt_destroy(ast_pjsip_endpoint);
1973                 ast_pjsip_endpoint = NULL;
1974                 pj_caching_pool_destroy(&caching_pool);
1975                 return AST_MODULE_LOAD_DECLINE;
1976         }
1977
1978         if (ast_sip_initialize_distributor()) {
1979                 ast_log(LOG_ERROR, "Failed to register distributor module. Aborting load\n");
1980                 ast_res_pjsip_destroy_configuration();
1981                 ast_sip_destroy_global_headers();
1982                 stop_monitor_thread();
1983                 pj_pool_release(memory_pool);
1984                 memory_pool = NULL;
1985                 pjsip_endpt_destroy(ast_pjsip_endpoint);
1986                 ast_pjsip_endpoint = NULL;
1987                 pj_caching_pool_destroy(&caching_pool);
1988                 return AST_MODULE_LOAD_DECLINE;
1989         }
1990
1991         if (ast_sip_initialize_outbound_authentication()) {
1992                 ast_log(LOG_ERROR, "Failed to initialize outbound authentication. Aborting load\n");
1993                 ast_sip_destroy_distributor();
1994                 ast_res_pjsip_destroy_configuration();
1995                 ast_sip_destroy_global_headers();
1996                 stop_monitor_thread();
1997                 pj_pool_release(memory_pool);
1998                 memory_pool = NULL;
1999                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2000                 ast_pjsip_endpoint = NULL;
2001                 pj_caching_pool_destroy(&caching_pool);
2002                 return AST_MODULE_LOAD_DECLINE;
2003         }
2004
2005         ast_res_pjsip_init_options_handling(0);
2006
2007         ast_res_pjsip_init_contact_transports();
2008
2009         ast_module_ref(ast_module_info->self);
2010
2011         return AST_MODULE_LOAD_SUCCESS;
2012 }
2013
2014 static int reload_module(void)
2015 {
2016         if (ast_res_pjsip_reload_configuration()) {
2017                 return AST_MODULE_LOAD_DECLINE;
2018         }
2019         ast_res_pjsip_init_options_handling(1);
2020         return 0;
2021 }
2022
2023 static int unload_module(void)
2024 {
2025         /* This will never get called as this module can't be unloaded */
2026         return 0;
2027 }
2028
2029 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Basic SIP resource",
2030                 .load = load_module,
2031                 .unload = unload_module,
2032                 .reload = reload_module,
2033                 .load_pri = AST_MODPRI_CHANNEL_DEPEND - 5,
2034 );