'domain_alias' config object XML help doesn't make it clear that the name used for...
[asterisk/asterisk.git] / res / res_pjsip.c
1 /*
2  * Asterisk -- An open source telephony toolkit.
3  *
4  * Copyright (C) 2013, Digium, Inc.
5  *
6  * Mark Michelson <mmichelson@digium.com>
7  *
8  * See http://www.asterisk.org for more information about
9  * the Asterisk project. Please do not directly contact
10  * any of the maintainers of this project for assistance;
11  * the project provides a web site, mailing lists and IRC
12  * channels for your use.
13  *
14  * This program is free software, distributed under the terms of
15  * the GNU General Public License Version 2. See the LICENSE file
16  * at the top of the source tree.
17  */
18
19 #include "asterisk.h"
20
21 #include <pjsip.h>
22 /* Needed for SUBSCRIBE, NOTIFY, and PUBLISH method definitions */
23 #include <pjsip_simple.h>
24 #include <pjlib.h>
25
26 #include "asterisk/res_pjsip.h"
27 #include "res_pjsip/include/res_pjsip_private.h"
28 #include "asterisk/linkedlists.h"
29 #include "asterisk/logger.h"
30 #include "asterisk/lock.h"
31 #include "asterisk/utils.h"
32 #include "asterisk/astobj2.h"
33 #include "asterisk/module.h"
34 #include "asterisk/threadpool.h"
35 #include "asterisk/taskprocessor.h"
36 #include "asterisk/uuid.h"
37 #include "asterisk/sorcery.h"
38
39 /*** MODULEINFO
40         <depend>pjproject</depend>
41         <depend>res_sorcery_config</depend>
42         <support_level>core</support_level>
43  ***/
44
45 /*** DOCUMENTATION
46         <configInfo name="res_pjsip" language="en_US">
47                 <synopsis>SIP Resource using PJProject</synopsis>
48                 <configFile name="pjsip.conf">
49                         <configObject name="endpoint">
50                                 <synopsis>Endpoint</synopsis>
51                                 <description><para>
52                                         The <emphasis>Endpoint</emphasis> is the primary configuration object.
53                                         It contains the core SIP related options only, endpoints are <emphasis>NOT</emphasis>
54                                         dialable entries of their own. Communication with another SIP device is
55                                         accomplished via Addresses of Record (AoRs) which have one or more
56                                         contacts assicated with them. Endpoints <emphasis>NOT</emphasis> configured to
57                                         use a <literal>transport</literal> will default to first transport found
58                                         in <filename>pjsip.conf</filename> that matches its type.
59                                         </para>
60                                         <para>Example: An Endpoint has been configured with no transport.
61                                         When it comes time to call an AoR, PJSIP will find the
62                                         first transport that matches the type. A SIP URI of <literal>sip:5000@[11::33]</literal>
63                                         will use the first IPv6 transport and try to send the request.
64                                         </para>
65                                         <para>If the anonymous endpoint identifier is in use an endpoint with the name
66                                         "anonymous@domain" will be searched for as a last resort. If this is not found
67                                         it will fall back to searching for "anonymous". If neither endpoints are found
68                                         the anonymous endpoint identifier will not return an endpoint and anonymous
69                                         calling will not be possible.
70                                         </para>
71                                 </description>
72                                 <configOption name="100rel" default="yes">
73                                         <synopsis>Allow support for RFC3262 provisional ACK tags</synopsis>
74                                         <description>
75                                                 <enumlist>
76                                                         <enum name="no" />
77                                                         <enum name="required" />
78                                                         <enum name="yes" />
79                                                 </enumlist>
80                                         </description>
81                                 </configOption>
82                                 <configOption name="aggregate_mwi" default="yes">
83                                         <synopsis></synopsis>
84                                         <description><para>When enabled, <replaceable>aggregate_mwi</replaceable> condenses message
85                                         waiting notifications from multiple mailboxes into a single NOTIFY. If it is disabled,
86                                         individual NOTIFYs are sent for each mailbox.</para></description>
87                                 </configOption>
88                                 <configOption name="allow">
89                                         <synopsis>Media Codec(s) to allow</synopsis>
90                                 </configOption>
91                                 <configOption name="aors">
92                                         <synopsis>AoR(s) to be used with the endpoint</synopsis>
93                                         <description><para>
94                                                 List of comma separated AoRs that the endpoint should be associated with.
95                                         </para></description>
96                                 </configOption>
97                                 <configOption name="auth">
98                                         <synopsis>Authentication Object(s) associated with the endpoint</synopsis>
99                                         <description><para>
100                                                 This is a comma-delimited list of <replaceable>auth</replaceable> sections defined
101                                                 in <filename>pjsip.conf</filename> to be used to verify inbound connection attempts.
102                                                 </para><para>
103                                                 Endpoints without an <literal>authentication</literal> object
104                                                 configured will allow connections without vertification.
105                                         </para></description>
106                                 </configOption>
107                                 <configOption name="callerid">
108                                         <synopsis>CallerID information for the endpoint</synopsis>
109                                         <description><para>
110                                                 Must be in the format <literal>Name &lt;Number&gt;</literal>,
111                                                 or only <literal>&lt;Number&gt;</literal>.
112                                         </para></description>
113                                 </configOption>
114                                 <configOption name="callerid_privacy">
115                                         <synopsis>Default privacy level</synopsis>
116                                         <description>
117                                                 <enumlist>
118                                                         <enum name="allowed_not_screened" />
119                                                         <enum name="allowed_passed_screened" />
120                                                         <enum name="allowed_failed_screened" />
121                                                         <enum name="allowed" />
122                                                         <enum name="prohib_not_screened" />
123                                                         <enum name="prohib_passed_screened" />
124                                                         <enum name="prohib_failed_screened" />
125                                                         <enum name="prohib" />
126                                                         <enum name="unavailable" />
127                                                 </enumlist>
128                                         </description>
129                                 </configOption>
130                                 <configOption name="callerid_tag">
131                                         <synopsis>Internal id_tag for the endpoint</synopsis>
132                                 </configOption>
133                                 <configOption name="context">
134                                         <synopsis>Dialplan context for inbound sessions</synopsis>
135                                 </configOption>
136                                 <configOption name="direct_media_glare_mitigation" default="none">
137                                         <synopsis>Mitigation of direct media (re)INVITE glare</synopsis>
138                                         <description>
139                                                 <para>
140                                                 This setting attempts to avoid creating INVITE glare scenarios
141                                                 by disabling direct media reINVITEs in one direction thereby allowing
142                                                 designated servers (according to this option) to initiate direct
143                                                 media reINVITEs without contention and significantly reducing call
144                                                 setup time.
145                                                 </para>
146                                                 <para>
147                                                 A more detailed description of how this option functions can be found on
148                                                 the Asterisk wiki https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance
149                                                 </para>
150                                                 <enumlist>
151                                                         <enum name="none" />
152                                                         <enum name="outgoing" />
153                                                         <enum name="incoming" />
154                                                 </enumlist>
155                                         </description>
156                                 </configOption>
157                                 <configOption name="direct_media_method" default="invite">
158                                         <synopsis>Direct Media method type</synopsis>
159                                         <description>
160                                                 <para>Method for setting up Direct Media between endpoints.</para>
161                                                 <enumlist>
162                                                         <enum name="invite" />
163                                                         <enum name="reinvite">
164                                                                 <para>Alias for the <literal>invite</literal> value.</para>
165                                                         </enum>
166                                                         <enum name="update" />
167                                                 </enumlist>
168                                         </description>
169                                 </configOption>
170                                 <configOption name="connected_line_method" default="invite">
171                                         <synopsis>Connected line method type</synopsis>
172                                         <description>
173                                                 <para>Method used when updating connected line information.</para>
174                                                 <enumlist>
175                                                         <enum name="invite" />
176                                                         <enum name="reinvite">
177                                                                 <para>Alias for the <literal>invite</literal> value.</para>
178                                                         </enum>
179                                                         <enum name="update" />
180                                                 </enumlist>
181                                         </description>
182                                 </configOption>
183                                 <configOption name="direct_media" default="yes">
184                                         <synopsis>Determines whether media may flow directly between endpoints.</synopsis>
185                                 </configOption>
186                                 <configOption name="disable_direct_media_on_nat" default="no">
187                                         <synopsis>Disable direct media session refreshes when NAT obstructs the media session</synopsis>
188                                 </configOption>
189                                 <configOption name="disallow">
190                                         <synopsis>Media Codec(s) to disallow</synopsis>
191                                 </configOption>
192                                 <configOption name="dtmfmode" default="rfc4733">
193                                         <synopsis>DTMF mode</synopsis>
194                                         <description>
195                                                 <para>This setting allows to choose the DTMF mode for endpoint communication.</para>
196                                                 <enumlist>
197                                                         <enum name="rfc4733">
198                                                                 <para>DTMF is sent out of band of the main audio stream.This
199                                                                 supercedes the older <emphasis>RFC-2833</emphasis> used within
200                                                                 the older <literal>chan_sip</literal>.</para>
201                                                         </enum>
202                                                         <enum name="inband">
203                                                                 <para>DTMF is sent as part of audio stream.</para>
204                                                         </enum>
205                                                         <enum name="info">
206                                                                 <para>DTMF is sent as SIP INFO packets.</para>
207                                                         </enum>
208                                                 </enumlist>
209                                         </description>
210                                 </configOption>
211                                 <configOption name="external_media_address">
212                                         <synopsis>IP used for External Media handling</synopsis>
213                                 </configOption>
214                                 <configOption name="force_rport" default="yes">
215                                         <synopsis>Force use of return port</synopsis>
216                                 </configOption>
217                                 <configOption name="ice_support" default="no">
218                                         <synopsis>Enable the ICE mechanism to help traverse NAT</synopsis>
219                                 </configOption>
220                                 <configOption name="identify_by" default="username,location">
221                                         <synopsis>Way(s) for Endpoint to be identified</synopsis>
222                                         <description><para>
223                                                 There are currently two methods to identify an endpoint. By default
224                                                 both are used to identify an endpoint.
225                                                 </para>
226                                                 <enumlist>
227                                                         <enum name="username" />
228                                                         <enum name="location" />
229                                                         <enum name="username,location" />
230                                                 </enumlist>
231                                         </description>
232                                 </configOption>
233                                 <configOption name="mailboxes">
234                                         <synopsis>Mailbox(es) to be associated with</synopsis>
235                                 </configOption>
236                                 <configOption name="mohsuggest" default="default">
237                                         <synopsis>Default Music On Hold class</synopsis>
238                                 </configOption>
239                                 <configOption name="outbound_auth">
240                                         <synopsis>Authentication object used for outbound requests</synopsis>
241                                 </configOption>
242                                 <configOption name="outbound_proxy">
243                                         <synopsis>Proxy through which to send requests</synopsis>
244                                 </configOption>
245                                 <configOption name="rewrite_contact">
246                                         <synopsis>Allow Contact header to be rewritten with the source IP address-port</synopsis>
247                                 </configOption>
248                                 <configOption name="rtp_ipv6" default="no">
249                                         <synopsis>Allow use of IPv6 for RTP traffic</synopsis>
250                                 </configOption>
251                                 <configOption name="rtp_symmetric" default="no">
252                                         <synopsis>Enforce that RTP must be symmetric</synopsis>
253                                 </configOption>
254                                 <configOption name="send_pai" default="no">
255                                         <synopsis>Send the P-Asserted-Identity header</synopsis>
256                                 </configOption>
257                                 <configOption name="send_rpid" default="no">
258                                         <synopsis>Send the Remote-Party-ID header</synopsis>
259                                 </configOption>
260                                 <configOption name="timers_min_se" default="90">
261                                         <synopsis>Minimum session timers expiration period</synopsis>
262                                         <description><para>
263                                                 Minimium session timer expiration period. Time in seconds.
264                                         </para></description>
265                                 </configOption>
266                                 <configOption name="timers" default="yes">
267                                         <synopsis>Session timers for SIP packets</synopsis>
268                                         <description>
269                                                 <enumlist>
270                                                         <enum name="forced" />
271                                                         <enum name="no" />
272                                                         <enum name="required" />
273                                                         <enum name="yes" />
274                                                 </enumlist>
275                                         </description>
276                                 </configOption>
277                                 <configOption name="timers_sess_expires" default="1800">
278                                         <synopsis>Maximum session timer expiration period</synopsis>
279                                         <description><para>
280                                                 Maximium session timer expiration period. Time in seconds.
281                                         </para></description>
282                                 </configOption>
283                                 <configOption name="transport">
284                                         <synopsis>Desired transport configuration</synopsis>
285                                         <description><para>
286                                                 This will set the desired transport configuration to send SIP data through.
287                                                 </para>
288                                                 <warning><para>Not specifying a transport will <emphasis>DEFAULT</emphasis>
289                                                 to the first configured transport in <filename>pjsip.conf</filename> which is
290                                                 valid for the URI we are trying to contact.
291                                                 </para></warning>
292                                         </description>
293                                 </configOption>
294                                 <configOption name="trust_id_inbound" default="no">
295                                         <synopsis>Accept identification information received from this endpoint</synopsis>
296                                         <description><para>This option determines whether Asterisk will accept
297                                         identification from the endpoint from headers such as P-Asserted-Identity
298                                         or Remote-Party-ID header. This option applies both to calls originating from the
299                                         endpoint and calls originating from Asterisk. If <literal>no</literal>, the
300                                         configured Caller-ID from pjsip.conf will always be used as the identity for
301                                         the endpoint.</para></description>
302                                 </configOption>
303                                 <configOption name="trust_id_outbound" default="no">
304                                         <synopsis>Send private identification details to the endpoint.</synopsis>
305                                         <description><para>This option determines whether res_pjsip will send private
306                                         identification information to the endpoint. If <literal>no</literal>,
307                                         private Caller-ID information will not be forwarded to the endpoint.
308                                         "Private" in this case refers to any method of restricting identification.
309                                         Example: setting <replaceable>callerid_privacy</replaceable> to any
310                                         <literal>prohib</literal> variation.
311                                         Example: If <replaceable>trust_id_inbound</replaceable> is set to
312                                         <literal>yes</literal>, the presence of a <literal>Privacy: id</literal>
313                                         header in a SIP request or response would indicate the identification
314                                         provided in the request is private.</para></description>
315                                 </configOption>
316                                 <configOption name="type">
317                                         <synopsis>Must be of type 'endpoint'.</synopsis>
318                                 </configOption>
319                                 <configOption name="use_ptime" default="no">
320                                         <synopsis>Use Endpoint's requested packetisation interval</synopsis>
321                                 </configOption>
322                                 <configOption name="use_avpf" default="no">
323                                         <synopsis>Determines whether res_pjsip will use and enforce usage of AVPF for this
324                                         endpoint.</synopsis>
325                                         <description><para>
326                                                 If set to <literal>yes</literal>, res_pjsip will use use the AVPF or SAVPF RTP
327                                                 profile for all media offers on outbound calls and media updates and will
328                                                 decline media offers not using the AVPF or SAVPF profile.
329                                         </para><para>
330                                                 If set to <literal>no</literal>, res_pjsip will use use the AVP or SAVP RTP
331                                                 profile for all media offers on outbound calls and media updates and will
332                                                 decline media offers not using the AVP or SAVP profile.
333                                         </para></description>
334                                 </configOption>
335                                 <configOption name="media_encryption" default="no">
336                                         <synopsis>Determines whether res_pjsip will use and enforce usage of media encryption
337                                         for this endpoint.</synopsis>
338                                         <description>
339                                                 <enumlist>
340                                                         <enum name="no"><para>
341                                                                 res_pjsip will offer no encryption and allow no encryption to be setup.
342                                                         </para></enum>
343                                                         <enum name="sdes"><para>
344                                                                 res_pjsip will offer standard SRTP setup via in-SDP keys. Encrypted SIP
345                                                                 transport should be used in conjunction with this option to prevent
346                                                                 exposure of media encryption keys.
347                                                         </para></enum>
348                                                         <enum name="dtls"><para>
349                                                                 res_pjsip will offer DTLS-SRTP setup.
350                                                         </para></enum>
351                                                 </enumlist>
352                                         </description>
353                                 </configOption>
354                                 <configOption name="inband_progress" default="no">
355                                         <synopsis>Determines whether chan_pjsip will indicate ringing using inband
356                                             progress.</synopsis>
357                                         <description><para>
358                                                 If set to <literal>yes</literal>, chan_pjsip will send a 183 Session Progress
359                                                 when told to indicate ringing and will immediately start sending ringing
360                                                 as audio.
361                                         </para><para>
362                                                 If set to <literal>no</literal>, chan_pjsip will send a 180 Ringing when told
363                                                 to indicate ringing and will NOT send it as audio.
364                                         </para></description>
365                                 </configOption>
366                                 <configOption name="callgroup">
367                                         <synopsis>The numeric pickup groups for a channel.</synopsis>
368                                         <description><para>
369                                                 Can be set to a comma separated list of numbers or ranges between the values
370                                                 of 0-63 (maximum of 64 groups).
371                                         </para></description>
372                                 </configOption>
373                                 <configOption name="pickupgroup">
374                                         <synopsis>The numeric pickup groups that a channel can pickup.</synopsis>
375                                         <description><para>
376                                                 Can be set to a comma separated list of numbers or ranges between the values
377                                                 of 0-63 (maximum of 64 groups).
378                                         </para></description>
379                                 </configOption>
380                                 <configOption name="namedcallgroup">
381                                         <synopsis>The named pickup groups for a channel.</synopsis>
382                                         <description><para>
383                                                 Can be set to a comma separated list of case sensitive strings limited by
384                                                 supported line length.
385                                         </para></description>
386                                 </configOption>
387                                 <configOption name="namedpickupgroup">
388                                         <synopsis>The named pickup groups that a channel can pickup.</synopsis>
389                                         <description><para>
390                                                 Can be set to a comma separated list of case sensitive strings limited by
391                                                 supported line length.
392                                         </para></description>
393                                 </configOption>
394                                 <configOption name="devicestate_busy_at" default="0">
395                                         <synopsis>The number of in-use channels which will cause busy to be returned as device state</synopsis>
396                                         <description><para>
397                                                 When the number of in-use channels for the endpoint matches the devicestate_busy_at setting the
398                                                 PJSIP channel driver will return busy as the device state instead of in use.
399                                         </para></description>
400                                 </configOption>
401                                 <configOption name="t38udptl" default="no">
402                                         <synopsis>Whether T.38 UDPTL support is enabled or not</synopsis>
403                                         <description><para>
404                                                 If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted
405                                                 and relayed.
406                                         </para></description>
407                                 </configOption>
408                                 <configOption name="t38udptl_ec" default="none">
409                                         <synopsis>T.38 UDPTL error correction method</synopsis>
410                                         <description>
411                                                 <enumlist>
412                                                         <enum name="none"><para>
413                                                                 No error correction should be used.
414                                                         </para></enum>
415                                                         <enum name="fec"><para>
416                                                                 Forward error correction should be used.
417                                                         </para></enum>
418                                                         <enum name="redundancy"><para>
419                                                                 Redundacy error correction should be used.
420                                                         </para></enum>
421                                                 </enumlist>
422                                         </description>
423                                 </configOption>
424                                 <configOption name="t38udptl_maxdatagram" default="0">
425                                         <synopsis>T.38 UDPTL maximum datagram size</synopsis>
426                                         <description><para>
427                                                 This option can be set to override the maximum datagram of a remote endpoint for broken
428                                                 endpoints.
429                                         </para></description>
430                                 </configOption>
431                                 <configOption name="faxdetect" default="no">
432                                         <synopsis>Whether CNG tone detection is enabled</synopsis>
433                                         <description><para>
434                                                 This option can be set to send the session to the fax extension when a CNG tone is
435                                                 detected.
436                                         </para></description>
437                                 </configOption>
438                                 <configOption name="t38udptl_nat" default="no">
439                                         <synopsis>Whether NAT support is enabled on UDPTL sessions</synopsis>
440                                         <description><para>
441                                                 When enabled the UDPTL stack will send UDPTL packets to the source address of
442                                                 received packets.
443                                         </para></description>
444                                 </configOption>
445                                 <configOption name="t38udptl_ipv6" default="no">
446                                         <synopsis>Whether IPv6 is used for UDPTL Sessions</synopsis>
447                                         <description><para>
448                                                 When enabled the UDPTL stack will use IPv6.
449                                         </para></description>
450                                 </configOption>
451                                 <configOption name="tonezone">
452                                         <synopsis>Set which country's indications to use for channels created for this endpoint.</synopsis>
453                                 </configOption>
454                                 <configOption name="language">
455                                         <synopsis>Set the default language to use for channels created for this endpoint.</synopsis>
456                                 </configOption>
457                                 <configOption name="one_touch_recording" default="no">
458                                         <synopsis>Determines whether one-touch recording is allowed for this endpoint.</synopsis>
459                                         <see-also>
460                                                 <ref type="configOption">recordonfeature</ref>
461                                                 <ref type="configOption">recordofffeature</ref>
462                                         </see-also>
463                                 </configOption>
464                                 <configOption name="recordonfeature" default="automixmon">
465                                         <synopsis>The feature to enact when one-touch recording is turned on.</synopsis>
466                                         <description>
467                                                 <para>When an INFO request for one-touch recording arrives with a Record header set to "on", this
468                                                 feature will be enabled for the channel. The feature designated here can be any built-in
469                                                 or dynamic feature defined in features.conf.</para>
470                                                 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
471                                         </description>
472                                         <see-also>
473                                                 <ref type="configOption">one_touch_recording</ref>
474                                                 <ref type="configOption">recordofffeature</ref>
475                                         </see-also>
476                                 </configOption>
477                                 <configOption name="recordofffeature" default="automixmon">
478                                         <synopsis>The feature to enact when one-touch recording is turned off.</synopsis>
479                                         <description>
480                                                 <para>When an INFO request for one-touch recording arrives with a Record header set to "off", this
481                                                 feature will be enabled for the channel. The feature designated here can be any built-in
482                                                 or dynamic feature defined in features.conf.</para>
483                                                 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
484                                         </description>
485                                         <see-also>
486                                                 <ref type="configOption">one_touch_recording</ref>
487                                                 <ref type="configOption">recordonfeature</ref>
488                                         </see-also>
489                                 </configOption>
490                                 <configOption name="rtpengine" default="asterisk">
491                                         <synopsis>Name of the RTP engine to use for channels created for this endpoint</synopsis>
492                                 </configOption>
493                                 <configOption name="allowtransfer" default="yes">
494                                         <synopsis>Determines whether SIP REFER transfers are allowed for this endpoint</synopsis>
495                                 </configOption>
496                                 <configOption name="sdpowner" default="-">
497                                         <synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis>
498                                 </configOption>
499                                 <configOption name="sdpsession" default="Asterisk">
500                                         <synopsis>String used for the SDP session (s=) line.</synopsis>
501                                 </configOption>
502                                 <configOption name="tos_audio">
503                                         <synopsis>DSCP TOS bits for audio streams</synopsis>
504                                         <description><para>
505                                                 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
506                                         </para></description>
507                                 </configOption>
508                                 <configOption name="tos_video">
509                                         <synopsis>DSCP TOS bits for video streams</synopsis>
510                                         <description><para>
511                                                 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
512                                         </para></description>
513                                 </configOption>
514                                 <configOption name="cos_audio">
515                                         <synopsis>Priority for audio streams</synopsis>
516                                         <description><para>
517                                                 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
518                                         </para></description>
519                                 </configOption>
520                                 <configOption name="cos_video">
521                                         <synopsis>Priority for video streams</synopsis>
522                                         <description><para>
523                                                 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
524                                         </para></description>
525                                 </configOption>
526                                 <configOption name="allowsubscribe" default="yes">
527                                         <synopsis>Determines if endpoint is allowed to initiate subscriptions with Asterisk.</synopsis>
528                                 </configOption>
529                                 <configOption name="subminexpiry" default="60">
530                                         <synopsis>The minimum allowed expiry time for subscriptions initiated by the endpoint.</synopsis>
531                                 </configOption>
532                                 <configOption name="fromuser">
533                                         <synopsis>Username to use in From header for requests to this endpoint.</synopsis>
534                                 </configOption>
535                                 <configOption name="mwifromuser">
536                                         <synopsis>Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.</synopsis>
537                                 </configOption>
538                                 <configOption name="fromdomain">
539                                         <synopsis>Domain to user in From header for requests to this endpoint.</synopsis>
540                                 </configOption>
541                                 <configOption name="dtlsverify">
542                                         <synopsis>Verify that the provided peer certificate is valid</synopsis>
543                                         <description><para>
544                                                 This option only applies if <replaceable>media_encryption</replaceable> is
545                                                 set to <literal>dtls</literal>.
546                                         </para></description>
547                                 </configOption>
548                                 <configOption name="dtlsrekey">
549                                         <synopsis>Interval at which to renegotiate the TLS session and rekey the SRTP session</synopsis>
550                                         <description><para>
551                                                 This option only applies if <replaceable>media_encryption</replaceable> is
552                                                 set to <literal>dtls</literal>.
553                                         </para><para>
554                                                 If this is not set or the value provided is 0 rekeying will be disabled.
555                                         </para></description>
556                                 </configOption>
557                                 <configOption name="dtlscertfile">
558                                         <synopsis>Path to certificate file to present to peer</synopsis>
559                                         <description><para>
560                                                 This option only applies if <replaceable>media_encryption</replaceable> is
561                                                 set to <literal>dtls</literal>.
562                                         </para></description>
563                                 </configOption>
564                                 <configOption name="dtlsprivatekey">
565                                         <synopsis>Path to private key for certificate file</synopsis>
566                                         <description><para>
567                                                 This option only applies if <replaceable>media_encryption</replaceable> is
568                                                 set to <literal>dtls</literal>.
569                                         </para></description>
570                                 </configOption>
571                                 <configOption name="dtlscipher">
572                                         <synopsis>Cipher to use for DTLS negotiation</synopsis>
573                                         <description><para>
574                                                 This option only applies if <replaceable>media_encryption</replaceable> is
575                                                 set to <literal>dtls</literal>.
576                                         </para><para>
577                                                 Many options for acceptable ciphers. See link for more:
578                                                 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
579                                         </para></description>
580                                 </configOption>
581                                 <configOption name="dtlscafile">
582                                         <synopsis>Path to certificate authority certificate</synopsis>
583                                         <description><para>
584                                                 This option only applies if <replaceable>media_encryption</replaceable> is
585                                                 set to <literal>dtls</literal>.
586                                         </para></description>
587                                 </configOption>
588                                 <configOption name="dtlscapath">
589                                         <synopsis>Path to a directory containing certificate authority certificates</synopsis>
590                                         <description><para>
591                                                 This option only applies if <replaceable>media_encryption</replaceable> is
592                                                 set to <literal>dtls</literal>.
593                                         </para></description>
594                                 </configOption>
595                                 <configOption name="dtlssetup">
596                                         <synopsis>Whether we are willing to accept connections, connect to the other party, or both.</synopsis>
597                                         <description>
598                                                 <para>
599                                                         This option only applies if <replaceable>media_encryption</replaceable> is
600                                                         set to <literal>dtls</literal>.
601                                                 </para>
602                                                 <enumlist>
603                                                         <enum name="active"><para>
604                                                                 res_pjsip will make a connection to the peer.
605                                                         </para></enum>
606                                                         <enum name="passive"><para>
607                                                                 res_pjsip will accept connections from the peer.
608                                                         </para></enum>
609                                                         <enum name="actpass"><para>
610                                                                 res_pjsip will offer and accept connections from the peer.
611                                                         </para></enum>
612                                                 </enumlist>
613                                         </description>
614                                 </configOption>
615                                 <configOption name="srtp_tag_32">
616                                         <synopsis>Determines whether 32 byte tags should be used instead of 80 byte tags.</synopsis>
617                                         <description><para>
618                                                 This option only applies if <replaceable>media_encryption</replaceable> is
619                                                 set to <literal>sdes</literal> or <literal>dtls</literal>.
620                                         </para></description>
621                                 </configOption>
622                         </configObject>
623                         <configObject name="auth">
624                                 <synopsis>Authentication type</synopsis>
625                                 <description><para>
626                                         Authentication objects hold the authentication information for use
627                                         by other objects such as <literal>endpoints</literal> or <literal>registrations</literal>.
628                                         This also allows for multiple objects to use a single auth object. See
629                                         the <literal>auth_type</literal> config option for password style choices.
630                                 </para></description>
631                                 <configOption name="auth_type" default="userpass">
632                                         <synopsis>Authentication type</synopsis>
633                                         <description><para>
634                                                 This option specifies which of the password style config options should be read
635                                                 when trying to authenticate an endpoint inbound request. If set to <literal>userpass</literal>
636                                                 then we'll read from the 'password' option. For <literal>md5</literal> we'll read
637                                                 from 'md5_cred'.
638                                                 </para>
639                                                 <enumlist>
640                                                         <enum name="md5"/>
641                                                         <enum name="userpass"/>
642                                                 </enumlist>
643                                         </description>
644                                 </configOption>
645                                 <configOption name="nonce_lifetime" default="32">
646                                         <synopsis>Lifetime of a nonce associated with this authentication config.</synopsis>
647                                 </configOption>
648                                 <configOption name="md5_cred">
649                                         <synopsis>MD5 Hash used for authentication.</synopsis>
650                                         <description><para>Only used when auth_type is <literal>md5</literal>.</para></description>
651                                 </configOption>
652                                 <configOption name="password">
653                                         <synopsis>PlainText password used for authentication.</synopsis>
654                                         <description><para>Only used when auth_type is <literal>userpass</literal>.</para></description>
655                                 </configOption>
656                                 <configOption name="realm" default="asterisk">
657                                         <synopsis>SIP realm for endpoint</synopsis>
658                                 </configOption>
659                                 <configOption name="type">
660                                         <synopsis>Must be 'auth'</synopsis>
661                                 </configOption>
662                                 <configOption name="username">
663                                         <synopsis>Username to use for account</synopsis>
664                                 </configOption>
665                         </configObject>
666                         <configObject name="nat_hook">
667                                 <synopsis>XXX This exists only to prevent XML documentation errors.</synopsis>
668                                 <configOption name="external_media_address">
669                                         <synopsis>I should be undocumented or hidden</synopsis>
670                                 </configOption>
671                                 <configOption name="method">
672                                         <synopsis>I should be undocumented or hidden</synopsis>
673                                 </configOption>
674                         </configObject>
675                         <configObject name="domain_alias">
676                                 <synopsis>Domain Alias</synopsis>
677                                 <description><para>
678                                         Signifies that a domain is an alias. If the domain on a session is
679                                         not found to match an AoR then this object is used to see if we have
680                                         an alias for the AoR to which the endpoint is binding. This objects
681                                         name as defined in configuration should be the domain alias and a 
682                                         config option is provided to specify the domain to be aliased.
683                                 </para></description>
684                                 <configOption name="type">
685                                         <synopsis>Must be of type 'domain_alias'.</synopsis>
686                                 </configOption>
687                                 <configOption name="domain">
688                                         <synopsis>Domain to be aliased</synopsis>
689                                 </configOption>
690                         </configObject>
691                         <configObject name="transport">
692                                 <synopsis>SIP Transport</synopsis>
693                                 <description><para>
694                                         <emphasis>Transports</emphasis>
695                                         </para>
696                                         <para>There are different transports and protocol derivatives
697                                                 supported by <literal>res_pjsip</literal>. They are in order of
698                                                 preference: UDP, TCP, and WebSocket (WS).</para>
699                                         <note><para>Changes to transport configuration in pjsip.conf will only be
700                                                 effected on a complete restart of Asterisk. A module reload
701                                                 will not suffice.</para></note>
702                                 </description>
703                                 <configOption name="async_operations" default="1">
704                                         <synopsis>Number of simultaneous Asynchronous Operations</synopsis>
705                                 </configOption>
706                                 <configOption name="bind">
707                                         <synopsis>IP Address and optional port to bind to for this transport</synopsis>
708                                 </configOption>
709                                 <configOption name="ca_list_file">
710                                         <synopsis>File containing a list of certificates to read (TLS ONLY)</synopsis>
711                                 </configOption>
712                                 <configOption name="cert_file">
713                                         <synopsis>Certificate file for endpoint (TLS ONLY)</synopsis>
714                                 </configOption>
715                                 <configOption name="cipher">
716                                         <synopsis>Preferred Cryptography Cipher (TLS ONLY)</synopsis>
717                                         <description><para>
718                                                 Many options for acceptable ciphers see link for more:
719                                                 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
720                                         </para></description>
721                                 </configOption>
722                                 <configOption name="domain">
723                                         <synopsis>Domain the transport comes from</synopsis>
724                                 </configOption>
725                                 <configOption name="external_media_address">
726                                         <synopsis>External Address to use in RTP handling</synopsis>
727                                 </configOption>
728                                 <configOption name="external_signaling_address">
729                                         <synopsis>External address for SIP signalling</synopsis>
730                                 </configOption>
731                                 <configOption name="external_signaling_port" default="0">
732                                         <synopsis>External port for SIP signalling</synopsis>
733                                 </configOption>
734                                 <configOption name="method">
735                                         <synopsis>Method of SSL transport (TLS ONLY)</synopsis>
736                                         <description>
737                                                 <enumlist>
738                                                         <enum name="default" />
739                                                         <enum name="unspecified" />
740                                                         <enum name="tlsv1" />
741                                                         <enum name="sslv2" />
742                                                         <enum name="sslv3" />
743                                                         <enum name="sslv23" />
744                                                 </enumlist>
745                                         </description>
746                                 </configOption>
747                                 <configOption name="localnet">
748                                         <synopsis>Network to consider local (used for NAT purposes).</synopsis>
749                                         <description><para>This must be in CIDR or dotted decimal format with the IP
750                                         and mask separated with a slash ('/').</para></description>
751                                 </configOption>
752                                 <configOption name="password">
753                                         <synopsis>Password required for transport</synopsis>
754                                 </configOption>
755                                 <configOption name="privkey_file">
756                                         <synopsis>Private key file (TLS ONLY)</synopsis>
757                                 </configOption>
758                                 <configOption name="protocol" default="udp">
759                                         <synopsis>Protocol to use for SIP traffic</synopsis>
760                                         <description>
761                                                 <enumlist>
762                                                         <enum name="udp" />
763                                                         <enum name="tcp" />
764                                                         <enum name="tls" />
765                                                 </enumlist>
766                                         </description>
767                                 </configOption>
768                                 <configOption name="require_client_cert" default="false">
769                                         <synopsis>Require client certificate (TLS ONLY)</synopsis>
770                                 </configOption>
771                                 <configOption name="type">
772                                         <synopsis>Must be of type 'transport'.</synopsis>
773                                 </configOption>
774                                 <configOption name="verify_client" default="false">
775                                         <synopsis>Require verification of client certificate (TLS ONLY)</synopsis>
776                                 </configOption>
777                                 <configOption name="verify_server" default="false">
778                                         <synopsis>Require verification of server certificate (TLS ONLY)</synopsis>
779                                 </configOption>
780                         </configObject>
781                         <configObject name="contact">
782                                 <synopsis>A way of creating an aliased name to a SIP URI</synopsis>
783                                 <description><para>
784                                         Contacts are a way to hide SIP URIs from the dialplan directly.
785                                         They are also used to make a group of contactable parties when
786                                         in use with <literal>AoR</literal> lists.
787                                 </para></description>
788                                 <configOption name="type">
789                                         <synopsis>Must be of type 'contact'.</synopsis>
790                                 </configOption>
791                                 <configOption name="uri">
792                                         <synopsis>SIP URI to contact peer</synopsis>
793                                 </configOption>
794                                 <configOption name="expiration_time">
795                                         <synopsis>Time to keep alive a contact</synopsis>
796                                         <description><para>
797                                                 Time to keep alive a contact. String style specification.
798                                         </para></description>
799                                 </configOption>
800                                 <configOption name="qualify_frequency" default="0">
801                                         <synopsis>Interval at which to qualify a contact</synopsis>
802                                         <description><para>
803                                                 Interval between attempts to qualify the contact for reachability.
804                                                 If <literal>0</literal> never qualify. Time in seconds.
805                                         </para></description>
806                                 </configOption>
807                         </configObject>
808                         <configObject name="contact_status">
809                                 <synopsis>Status for a contact</synopsis>
810                                 <description><para>
811                                         The contact status keeps track of whether or not a contact is reachable
812                                         and how long it took to qualify the contact (round trip time).
813                                 </para></description>
814                                 <configOption name="status">
815                                         <synopsis>A contact's status</synopsis>
816                                         <description>
817                                                 <enumlist>
818                                                         <enum name="AVAILABLE" />
819                                                         <enum name="UNAVAILABLE" />
820                                                 </enumlist>
821                                         </description>
822                                 </configOption>
823                                 <configOption name="rtt">
824                                         <synopsis>Round trip time</synopsis>
825                                         <description><para>
826                                                 The time, in microseconds, it took to qualify the contact.
827                                         </para></description>
828                                 </configOption>
829                         </configObject>
830                         <configObject name="aor">
831                                 <synopsis>The configuration for a location of an endpoint</synopsis>
832                                 <description><para>
833                                         An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
834                                         AoRs are specified, an endpoint will not be reachable by Asterisk.
835                                         Beyond that, an AoR has other uses within Asterisk.
836                                         </para><para>
837                                         An <literal>AoR</literal> is a way to allow dialing a group
838                                         of <literal>Contacts</literal> that all use the same
839                                         <literal>endpoint</literal> for calls.
840                                         </para><para>
841                                         This can be used as another way of grouping a list of contacts to dial
842                                         rather than specifing them each directly when dialing via the dialplan.
843                                         This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
844                                 </para></description>
845                                 <configOption name="contact">
846                                         <synopsis>Permanent contacts assigned to AoR</synopsis>
847                                         <description><para>
848                                                 Contacts included in this list will be called whenever referenced
849                                                 by <literal>chan_pjsip</literal>.
850                                         </para></description>
851                                 </configOption>
852                                 <configOption name="default_expiration" default="3600">
853                                         <synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis>
854                                 </configOption>
855                                 <configOption name="mailboxes">
856                                         <synopsis>Mailbox(es) to be associated with</synopsis>
857                                         <description><para>This option applies when an external entity subscribes to an AoR
858                                         for message waiting indications. The mailboxes specified here will be
859                                         subscribed to.</para></description>
860                                 </configOption>
861                                 <configOption name="maximum_expiration" default="7200">
862                                         <synopsis>Maximum time to keep an AoR</synopsis>
863                                         <description><para>
864                                                 Maximium time to keep a peer with explicit expiration. Time in seconds.
865                                         </para></description>
866                                 </configOption>
867                                 <configOption name="max_contacts" default="0">
868                                         <synopsis>Maximum number of contacts that can bind to an AoR</synopsis>
869                                         <description><para>
870                                                 Maximum number of contacts that can associate with this AoR.
871                                                 </para>
872                                                 <note><para>This should be set to <literal>1</literal> and
873                                                 <replaceable>remove_existing</replaceable> set to <literal>yes</literal> if you
874                                                 wish to stick with the older <literal>chan_sip</literal> behaviour.
875                                                 </para></note>
876                                         </description>
877                                 </configOption>
878                                 <configOption name="minimum_expiration" default="60">
879                                         <synopsis>Minimum keep alive time for an AoR</synopsis>
880                                         <description><para>
881                                                 Minimum time to keep a peer with an explict expiration. Time in seconds.
882                                         </para></description>
883                                 </configOption>
884                                 <configOption name="remove_existing" default="no">
885                                         <synopsis>Determines whether new contacts replace existing ones.</synopsis>
886                                         <description><para>
887                                                 On receiving a new registration to the AoR should it remove
888                                                 the existing contact that was registered against it?
889                                                 </para>
890                                                 <note><para>This should be set to <literal>yes</literal> and
891                                                 <replaceable>max_contacts</replaceable> set to <literal>1</literal> if you
892                                                 wish to stick with the older <literal>chan_sip</literal> behaviour.
893                                                 </para></note>
894                                         </description>
895                                 </configOption>
896                                 <configOption name="type">
897                                         <synopsis>Must be of type 'aor'.</synopsis>
898                                 </configOption>
899                                 <configOption name="qualify_frequency" default="0">
900                                         <synopsis>Interval at which to qualify an AoR</synopsis>
901                                         <description><para>
902                                                 Interval between attempts to qualify the AoR for reachability.
903                                                 If <literal>0</literal> never qualify. Time in seconds.
904                                         </para></description>
905                                 </configOption>
906                                 <configOption name="authenticate_qualify" default="no">
907                                         <synopsis>Authenticates a qualify request if needed</synopsis>
908                                         <description><para>
909                                                 If true and a qualify request receives a challenge or authenticate response
910                                                 authentication is attempted before declaring the contact available.
911                                         </para></description>
912                                 </configOption>
913                         </configObject>
914                         <configObject name="system">
915                                 <synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis>
916                                 <description><para>
917                                         The settings in this section are global. In addition to being global, the values will
918                                         not be re-evaluated when a reload is performed. This is because the values must be set
919                                         before the SIP stack is initialized. The only way to reset these values is to either 
920                                         restart Asterisk, or unload res_pjsip.so and then load it again.
921                                 </para></description>
922                                 <configOption name="timert1" default="500">
923                                         <synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
924                                         <description><para>
925                                                 Timer T1 is the base for determining how long to wait before retransmitting
926                                                 requests that receive no response when using an unreliable transport (e.g. UDP).
927                                                 For more information on this timer, see RFC 3261, Section 17.1.1.1.
928                                         </para></description>
929                                 </configOption>
930                                 <configOption name="timerb" default="32000">
931                                         <synopsis>Set transaction timer B value (milliseconds).</synopsis>
932                                         <description><para>
933                                                 Timer B determines the maximum amount of time to wait after sending an INVITE
934                                                 request before terminating the transaction. It is recommended that this be set
935                                                 to 64 * Timer T1, but it may be set higher if desired. For more information on
936                                                 this timer, see RFC 3261, Section 17.1.1.1.
937                                         </para></description>
938                                 </configOption>
939                                 <configOption name="compactheaders" default="no">
940                                         <synopsis>Use the short forms of common SIP header names.</synopsis>
941                                 </configOption>
942                                 <configOption name="threadpool_initial_size" default="0">
943                                         <synopsis>Initial number of threads in the res_pjsip threadpool.</synopsis>
944                                 </configOption>
945                                 <configOption name="threadpool_auto_increment" default="5">
946                                         <synopsis>The amount by which the number of threads is incremented when necessary.</synopsis>
947                                 </configOption>
948                                 <configOption name="threadpool_idle_timeout" default="60">
949                                         <synopsis>Number of seconds before an idle thread should be disposed of.</synopsis>
950                                 </configOption>
951                                 <configOption name="threadpool_max_size" default="0">
952                                         <synopsis>Maximum number of threads in the res_pjsip threadpool.
953                                         A value of 0 indicates no maximum.</synopsis>
954                                 </configOption>
955                         </configObject>
956                         <configObject name="global">
957                                 <synopsis>Options that apply globally to all SIP communications</synopsis>
958                                 <description><para>
959                                         The settings in this section are global. Unlike options in the <literal>system</literal>
960                                         section, these options can be refreshed by performing a reload.
961                                 </para></description>
962                                 <configOption name="maxforwards" default="70">
963                                         <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
964                                 </configOption>
965                                 <configOption name="useragent" default="Asterisk &lt;Asterisk Version&gt;">
966                                         <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
967                                 </configOption>
968                         </configObject>
969                 </configFile>
970         </configInfo>
971         <manager name="PJSIPQualify" language="en_US">
972                 <synopsis>
973                         Qualify a chan_pjsip endpoint.
974                 </synopsis>
975                 <syntax>
976                         <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
977                         <parameter name="Endpoint" required="true">
978                                 <para>The endpoint you want to qualify.</para>
979                         </parameter>
980                 </syntax>
981                 <description>
982                         <para>Qualify a chan_pjsip endpoint.</para>
983                 </description>
984         </manager>
985  ***/
986
987
988 static pjsip_endpoint *ast_pjsip_endpoint;
989
990 static struct ast_threadpool *sip_threadpool;
991
992 static int register_service(void *data)
993 {
994         pjsip_module **module = data;
995         if (!ast_pjsip_endpoint) {
996                 ast_log(LOG_ERROR, "There is no PJSIP endpoint. Unable to register services\n");
997                 return -1;
998         }
999         if (pjsip_endpt_register_module(ast_pjsip_endpoint, *module) != PJ_SUCCESS) {
1000                 ast_log(LOG_ERROR, "Unable to register module %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1001                 return -1;
1002         }
1003         ast_debug(1, "Registered SIP service %.*s (%p)\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name), *module);
1004         ast_module_ref(ast_module_info->self);
1005         return 0;
1006 }
1007
1008 int ast_sip_register_service(pjsip_module *module)
1009 {
1010         return ast_sip_push_task_synchronous(NULL, register_service, &module);
1011 }
1012
1013 static int unregister_service(void *data)
1014 {
1015         pjsip_module **module = data;
1016         ast_module_unref(ast_module_info->self);
1017         if (!ast_pjsip_endpoint) {
1018                 return -1;
1019         }
1020         pjsip_endpt_unregister_module(ast_pjsip_endpoint, *module);
1021         ast_debug(1, "Unregistered SIP service %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1022         return 0;
1023 }
1024
1025 void ast_sip_unregister_service(pjsip_module *module)
1026 {
1027         ast_sip_push_task_synchronous(NULL, unregister_service, &module);
1028 }
1029
1030 static struct ast_sip_authenticator *registered_authenticator;
1031
1032 int ast_sip_register_authenticator(struct ast_sip_authenticator *auth)
1033 {
1034         if (registered_authenticator) {
1035                 ast_log(LOG_WARNING, "Authenticator %p is already registered. Cannot register a new one\n", registered_authenticator);
1036                 return -1;
1037         }
1038         registered_authenticator = auth;
1039         ast_debug(1, "Registered SIP authenticator module %p\n", auth);
1040         ast_module_ref(ast_module_info->self);
1041         return 0;
1042 }
1043
1044 void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth)
1045 {
1046         if (registered_authenticator != auth) {
1047                 ast_log(LOG_WARNING, "Trying to unregister authenticator %p but authenticator %p registered\n",
1048                                 auth, registered_authenticator);
1049                 return;
1050         }
1051         registered_authenticator = NULL;
1052         ast_debug(1, "Unregistered SIP authenticator %p\n", auth);
1053         ast_module_unref(ast_module_info->self);
1054 }
1055
1056 int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1057 {
1058         if (!registered_authenticator) {
1059                 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is not required\n");
1060                 return 0;
1061         }
1062
1063         return registered_authenticator->requires_authentication(endpoint, rdata);
1064 }
1065
1066 enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint,
1067                 pjsip_rx_data *rdata, pjsip_tx_data *tdata)
1068 {
1069         if (!registered_authenticator) {
1070                 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is successful\n");
1071                 return 0;
1072         }
1073         return registered_authenticator->check_authentication(endpoint, rdata, tdata);
1074 }
1075
1076 static struct ast_sip_outbound_authenticator *registered_outbound_authenticator;
1077
1078 int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1079 {
1080         if (registered_outbound_authenticator) {
1081                 ast_log(LOG_WARNING, "Outbound authenticator %p is already registered. Cannot register a new one\n", registered_outbound_authenticator);
1082                 return -1;
1083         }
1084         registered_outbound_authenticator = auth;
1085         ast_debug(1, "Registered SIP outbound authenticator module %p\n", auth);
1086         ast_module_ref(ast_module_info->self);
1087         return 0;
1088 }
1089
1090 void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1091 {
1092         if (registered_outbound_authenticator != auth) {
1093                 ast_log(LOG_WARNING, "Trying to unregister outbound authenticator %p but outbound authenticator %p registered\n",
1094                                 auth, registered_outbound_authenticator);
1095                 return;
1096         }
1097         registered_outbound_authenticator = NULL;
1098         ast_debug(1, "Unregistered SIP outbound authenticator %p\n", auth);
1099         ast_module_unref(ast_module_info->self);
1100 }
1101
1102 int ast_sip_create_request_with_auth(const struct ast_sip_auth_array *auths, pjsip_rx_data *challenge,
1103                 pjsip_transaction *tsx, pjsip_tx_data **new_request)
1104 {
1105         if (!registered_outbound_authenticator) {
1106                 ast_log(LOG_WARNING, "No SIP outbound authenticator registered. Cannot respond to authentication challenge\n");
1107                 return -1;
1108         }
1109         return registered_outbound_authenticator->create_request_with_auth(auths, challenge, tsx, new_request);
1110 }
1111
1112 struct endpoint_identifier_list {
1113         struct ast_sip_endpoint_identifier *identifier;
1114         AST_RWLIST_ENTRY(endpoint_identifier_list) list;
1115 };
1116
1117 static AST_RWLIST_HEAD_STATIC(endpoint_identifiers, endpoint_identifier_list);
1118
1119 int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1120 {
1121         struct endpoint_identifier_list *id_list_item;
1122         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1123
1124         id_list_item = ast_calloc(1, sizeof(*id_list_item));
1125         if (!id_list_item) {
1126                 ast_log(LOG_ERROR, "Unabled to add endpoint identifier. Out of memory.\n");
1127                 return -1;
1128         }
1129         id_list_item->identifier = identifier;
1130
1131         AST_RWLIST_INSERT_TAIL(&endpoint_identifiers, id_list_item, list);
1132         ast_debug(1, "Registered endpoint identifier %p\n", identifier);
1133
1134         ast_module_ref(ast_module_info->self);
1135         return 0;
1136 }
1137
1138 void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1139 {
1140         struct endpoint_identifier_list *iter;
1141         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1142         AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_identifiers, iter, list) {
1143                 if (iter->identifier == identifier) {
1144                         AST_RWLIST_REMOVE_CURRENT(list);
1145                         ast_free(iter);
1146                         ast_debug(1, "Unregistered endpoint identifier %p\n", identifier);
1147                         ast_module_unref(ast_module_info->self);
1148                         break;
1149                 }
1150         }
1151         AST_RWLIST_TRAVERSE_SAFE_END;
1152 }
1153
1154 struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata)
1155 {
1156         struct endpoint_identifier_list *iter;
1157         struct ast_sip_endpoint *endpoint = NULL;
1158         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1159         AST_RWLIST_TRAVERSE(&endpoint_identifiers, iter, list) {
1160                 ast_assert(iter->identifier->identify_endpoint != NULL);
1161                 endpoint = iter->identifier->identify_endpoint(rdata);
1162                 if (endpoint) {
1163                         break;
1164                 }
1165         }
1166         return endpoint;
1167 }
1168
1169 pjsip_endpoint *ast_sip_get_pjsip_endpoint(void)
1170 {
1171         return ast_pjsip_endpoint;
1172 }
1173
1174 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)
1175 {
1176         pj_str_t tmp, local_addr;
1177         pjsip_uri *uri;
1178         pjsip_sip_uri *sip_uri;
1179         pjsip_transport_type_e type = PJSIP_TRANSPORT_UNSPECIFIED;
1180         int local_port;
1181         char uuid_str[AST_UUID_STR_LEN];
1182
1183         if (ast_strlen_zero(user)) {
1184                 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1185                 if (!uuid) {
1186                         return -1;
1187                 }
1188                 user = ast_uuid_to_str(uuid, uuid_str, sizeof(uuid_str));
1189         }
1190
1191         /* Parse the provided target URI so we can determine what transport it will end up using */
1192         pj_strdup_with_null(pool, &tmp, target);
1193
1194         if (!(uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0)) ||
1195             (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
1196                 return -1;
1197         }
1198
1199         sip_uri = pjsip_uri_get_uri(uri);
1200
1201         /* Determine the transport type to use */
1202         if (PJSIP_URI_SCHEME_IS_SIPS(sip_uri)) {
1203                 type = PJSIP_TRANSPORT_TLS;
1204         } else if (!sip_uri->transport_param.slen) {
1205                 type = PJSIP_TRANSPORT_UDP;
1206         } else {
1207                 type = pjsip_transport_get_type_from_name(&sip_uri->transport_param);
1208         }
1209
1210         if (type == PJSIP_TRANSPORT_UNSPECIFIED) {
1211                 return -1;
1212         }
1213
1214         /* If the host is IPv6 turn the transport into an IPv6 version */
1215         if (pj_strchr(&sip_uri->host, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1216                 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1217         }
1218
1219         if (!ast_strlen_zero(domain)) {
1220                 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1221                 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1222                                 "<%s:%s@%s%s%s>",
1223                                 (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1224                                 user,
1225                                 domain,
1226                                 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1227                                 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1228                 return 0;
1229         }
1230
1231         /* Get the local bound address for the transport that will be used when communicating with the provided URI */
1232         if (pjsip_tpmgr_find_local_addr(pjsip_endpt_get_tpmgr(ast_sip_get_pjsip_endpoint()), pool, type, selector,
1233                                                               &local_addr, &local_port) != PJ_SUCCESS) {
1234                 return -1;
1235         }
1236
1237         /* If IPv6 was specified in the transport, set the proper type */
1238         if (pj_strchr(&local_addr, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1239                 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1240         }
1241
1242         from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1243         from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1244                                       "<%s:%s@%s%.*s%s:%d%s%s>",
1245                                       (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1246                                       user,
1247                                       (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1248                                       (int)local_addr.slen,
1249                                       local_addr.ptr,
1250                                       (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1251                                       local_port,
1252                                       (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1253                                       (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1254
1255         return 0;
1256 }
1257
1258 static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpoint, pjsip_tpselector *selector)
1259 {
1260         RAII_VAR(struct ast_sip_transport *, transport, NULL, ao2_cleanup);
1261         const char *transport_name = endpoint->transport;
1262
1263         if (ast_strlen_zero(transport_name)) {
1264                 return 0;
1265         }
1266
1267         transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_name);
1268
1269         if (!transport || !transport->state) {
1270                 return -1;
1271         }
1272
1273         if (transport->state->transport) {
1274                 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1275                 selector->u.transport = transport->state->transport;
1276         } else if (transport->state->factory) {
1277                 selector->type = PJSIP_TPSELECTOR_LISTENER;
1278                 selector->u.listener = transport->state->factory;
1279         } else {
1280                 return -1;
1281         }
1282
1283         return 0;
1284 }
1285
1286 static int sip_get_tpselector_from_uri(const char *uri, pjsip_tpselector *selector)
1287 {
1288         RAII_VAR(struct ast_sip_contact_transport *, contact_transport, NULL, ao2_cleanup);
1289
1290         contact_transport = ast_sip_location_retrieve_contact_transport_by_uri(uri);
1291
1292         if (!contact_transport) {
1293                 return -1;
1294         }
1295
1296         selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1297         selector->u.transport = contact_transport->transport;
1298
1299         return 0;
1300 }
1301
1302 pjsip_dialog *ast_sip_create_dialog(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user)
1303 {
1304         pj_str_t local_uri = { "sip:temp@temp", 13 }, remote_uri;
1305         pjsip_dialog *dlg = NULL;
1306         const char *outbound_proxy = endpoint->outbound_proxy;
1307         pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1308         static const pj_str_t HCONTACT = { "Contact", 7 };
1309
1310         pj_cstr(&remote_uri, uri);
1311
1312         if (pjsip_dlg_create_uac(pjsip_ua_instance(), &local_uri, NULL, &remote_uri, NULL, &dlg) != PJ_SUCCESS) {
1313                 return NULL;
1314         }
1315
1316         if (sip_get_tpselector_from_uri(uri, &selector) && sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1317                 pjsip_dlg_terminate(dlg);
1318                 return NULL;
1319         }
1320
1321         if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
1322                 pjsip_dlg_terminate(dlg);
1323                 return NULL;
1324         }
1325
1326         /* Update the dialog with the new local URI, we do it afterwards so we can use the dialog pool for construction */
1327         pj_strdup_with_null(dlg->pool, &dlg->local.info_str, &local_uri);
1328         dlg->local.info->uri = pjsip_parse_uri(dlg->pool, dlg->local.info_str.ptr, dlg->local.info_str.slen, 0);
1329         dlg->local.contact = pjsip_parse_hdr(dlg->pool, &HCONTACT, local_uri.ptr, local_uri.slen, NULL);
1330
1331         /* If a request user has been specified and we are permitted to change it, do so */
1332         if (!ast_strlen_zero(request_user) && (PJSIP_URI_SCHEME_IS_SIP(dlg->target) || PJSIP_URI_SCHEME_IS_SIPS(dlg->target))) {
1333                 pjsip_sip_uri *target = pjsip_uri_get_uri(dlg->target);
1334                 pj_strdup2(dlg->pool, &target->user, request_user);
1335         }
1336
1337         /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
1338         dlg->sess_count++;
1339
1340         pjsip_dlg_set_transport(dlg, &selector);
1341
1342         if (!ast_strlen_zero(outbound_proxy)) {
1343                 pjsip_route_hdr route_set, *route;
1344                 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1345                 pj_str_t tmp;
1346
1347                 pj_list_init(&route_set);
1348
1349                 pj_strdup2_with_null(dlg->pool, &tmp, outbound_proxy);
1350                 if (!(route = pjsip_parse_hdr(dlg->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1351                         pjsip_dlg_terminate(dlg);
1352                         return NULL;
1353                 }
1354                 pj_list_push_back(&route_set, route);
1355
1356                 pjsip_dlg_set_route_set(dlg, &route_set);
1357         }
1358
1359         dlg->sess_count--;
1360
1361         return dlg;
1362 }
1363
1364 /* PJSIP doesn't know about the INFO method, so we have to define it ourselves */
1365 const pjsip_method pjsip_info_method = {PJSIP_OTHER_METHOD, {"INFO", 4} };
1366 const pjsip_method pjsip_message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
1367
1368 static struct {
1369         const char *method;
1370         const pjsip_method *pmethod;
1371 } methods [] = {
1372         { "INVITE", &pjsip_invite_method },
1373         { "CANCEL", &pjsip_cancel_method },
1374         { "ACK", &pjsip_ack_method },
1375         { "BYE", &pjsip_bye_method },
1376         { "REGISTER", &pjsip_register_method },
1377         { "OPTIONS", &pjsip_options_method },
1378         { "SUBSCRIBE", &pjsip_subscribe_method },
1379         { "NOTIFY", &pjsip_notify_method },
1380         { "PUBLISH", &pjsip_publish_method },
1381         { "INFO", &pjsip_info_method },
1382         { "MESSAGE", &pjsip_message_method },
1383 };
1384
1385 static const pjsip_method *get_pjsip_method(const char *method)
1386 {
1387         int i;
1388         for (i = 0; i < ARRAY_LEN(methods); ++i) {
1389                 if (!strcmp(method, methods[i].method)) {
1390                         return methods[i].pmethod;
1391                 }
1392         }
1393         return NULL;
1394 }
1395
1396 static int create_in_dialog_request(const pjsip_method *method, struct pjsip_dialog *dlg, pjsip_tx_data **tdata)
1397 {
1398         if (pjsip_dlg_create_request(dlg, method, -1, tdata) != PJ_SUCCESS) {
1399                 ast_log(LOG_WARNING, "Unable to create in-dialog request.\n");
1400                 return -1;
1401         }
1402
1403         return 0;
1404 }
1405
1406 static int create_out_of_dialog_request(const pjsip_method *method, struct ast_sip_endpoint *endpoint,
1407                 const char *uri, pjsip_tx_data **tdata)
1408 {
1409         RAII_VAR(struct ast_sip_contact *, contact, NULL, ao2_cleanup);
1410         pj_str_t remote_uri;
1411         pj_str_t from;
1412         pj_pool_t *pool;
1413         pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1414
1415         if (ast_strlen_zero(uri)) {
1416                 if (!endpoint) {
1417                         ast_log(LOG_ERROR, "An endpoint and/or uri must be specified\n");
1418                         return -1;
1419                 }
1420
1421                 contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
1422                 if (!contact || ast_strlen_zero(contact->uri)) {
1423                         ast_log(LOG_ERROR, "Unable to retrieve contact for endpoint %s\n",
1424                                         ast_sorcery_object_get_id(endpoint));
1425                         return -1;
1426                 }
1427
1428                 pj_cstr(&remote_uri, contact->uri);
1429         } else {
1430                 pj_cstr(&remote_uri, uri);
1431         }
1432
1433         if (endpoint) {
1434                 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1435                         ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport selector for endpoint %s\n",
1436                                 ast_sorcery_object_get_id(endpoint));
1437                         return -1;
1438                 }
1439         }
1440
1441         pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "Outbound request", 256, 256);
1442
1443         if (!pool) {
1444                 ast_log(LOG_ERROR, "Unable to create PJLIB memory pool\n");
1445                 return -1;
1446         }
1447
1448         if (sip_dialog_create_from(pool, &from, endpoint ? endpoint->fromuser : NULL,
1449                                 endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) {
1450                 ast_log(LOG_ERROR, "Unable to create From header for %.*s request to endpoint %s\n",
1451                                 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1452                 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1453                 return -1;
1454         }
1455
1456         if (pjsip_endpt_create_request(ast_sip_get_pjsip_endpoint(), method, &remote_uri,
1457                         &from, &remote_uri, &from, NULL, -1, NULL, tdata) != PJ_SUCCESS) {
1458                 ast_log(LOG_ERROR, "Unable to create outbound %.*s request to endpoint %s\n",
1459                                 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1460                 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1461                 return -1;
1462         }
1463
1464         /* We can release this pool since request creation copied all the necessary
1465          * data into the outbound request's pool
1466          */
1467         pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1468         return 0;
1469 }
1470
1471 int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
1472                 struct ast_sip_endpoint *endpoint, const char *uri,
1473                 pjsip_tx_data **tdata)
1474 {
1475         const pjsip_method *pmethod = get_pjsip_method(method);
1476
1477         if (!pmethod) {
1478                 ast_log(LOG_WARNING, "Unknown method '%s'. Cannot send request\n", method);
1479                 return -1;
1480         }
1481
1482         if (dlg) {
1483                 return create_in_dialog_request(pmethod, dlg, tdata);
1484         } else {
1485                 return create_out_of_dialog_request(pmethod, endpoint, uri, tdata);
1486         }
1487 }
1488
1489 static int send_in_dialog_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg)
1490 {
1491         if (pjsip_dlg_send_request(dlg, tdata, -1, NULL) != PJ_SUCCESS) {
1492                 ast_log(LOG_WARNING, "Unable to send in-dialog request.\n");
1493                 return -1;
1494         }
1495         return 0;
1496 }
1497
1498 static void send_request_cb(void *token, pjsip_event *e)
1499 {
1500         RAII_VAR(struct ast_sip_endpoint *, endpoint, token, ao2_cleanup);
1501         pjsip_transaction *tsx = e->body.tsx_state.tsx;
1502         pjsip_rx_data *challenge = e->body.tsx_state.src.rdata;
1503         pjsip_tx_data *tdata;
1504
1505         if (tsx->status_code != 401 && tsx->status_code != 407) {
1506                 return;
1507         }
1508
1509         if (!ast_sip_create_request_with_auth(&endpoint->outbound_auths, challenge, tsx, &tdata)) {
1510                 pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, NULL, NULL);
1511         }
1512 }
1513
1514 static int send_out_of_dialog_request(pjsip_tx_data *tdata, struct ast_sip_endpoint *endpoint)
1515 {
1516         ao2_ref(endpoint, +1);
1517         if (pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, endpoint, send_request_cb) != PJ_SUCCESS) {
1518                 ast_log(LOG_ERROR, "Error attempting to send outbound %.*s request to endpoint %s\n",
1519                                 (int) pj_strlen(&tdata->msg->line.req.method.name),
1520                                 pj_strbuf(&tdata->msg->line.req.method.name),
1521                                 ast_sorcery_object_get_id(endpoint));
1522                 ao2_ref(endpoint, -1);
1523                 return -1;
1524         }
1525
1526         return 0;
1527 }
1528
1529 int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg, struct ast_sip_endpoint *endpoint)
1530 {
1531         ast_assert(tdata->msg->type == PJSIP_REQUEST_MSG);
1532
1533         if (dlg) {
1534                 return send_in_dialog_request(tdata, dlg);
1535         } else {
1536                 return send_out_of_dialog_request(tdata, endpoint);
1537         }
1538 }
1539
1540 int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value)
1541 {
1542         pj_str_t hdr_name;
1543         pj_str_t hdr_value;
1544         pjsip_generic_string_hdr *hdr;
1545
1546         pj_cstr(&hdr_name, name);
1547         pj_cstr(&hdr_value, value);
1548
1549         hdr = pjsip_generic_string_hdr_create(tdata->pool, &hdr_name, &hdr_value);
1550
1551         pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *) hdr);
1552         return 0;
1553 }
1554
1555 static pjsip_msg_body *ast_body_to_pjsip_body(pj_pool_t *pool, const struct ast_sip_body *body)
1556 {
1557         pj_str_t type;
1558         pj_str_t subtype;
1559         pj_str_t body_text;
1560
1561         pj_cstr(&type, body->type);
1562         pj_cstr(&subtype, body->subtype);
1563         pj_cstr(&body_text, body->body_text);
1564
1565         return pjsip_msg_body_create(pool, &type, &subtype, &body_text);
1566 }
1567
1568 int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body)
1569 {
1570         pjsip_msg_body *pjsip_body = ast_body_to_pjsip_body(tdata->pool, body);
1571         tdata->msg->body = pjsip_body;
1572         return 0;
1573 }
1574
1575 int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies)
1576 {
1577         int i;
1578         /* NULL for type and subtype automatically creates "multipart/mixed" */
1579         pjsip_msg_body *body = pjsip_multipart_create(tdata->pool, NULL, NULL);
1580
1581         for (i = 0; i < num_bodies; ++i) {
1582                 pjsip_multipart_part *part = pjsip_multipart_create_part(tdata->pool);
1583                 part->body = ast_body_to_pjsip_body(tdata->pool, bodies[i]);
1584                 pjsip_multipart_add_part(tdata->pool, body, part);
1585         }
1586
1587         tdata->msg->body = body;
1588         return 0;
1589 }
1590
1591 int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text)
1592 {
1593         size_t combined_size = strlen(body_text) + tdata->msg->body->len;
1594         struct ast_str *body_buffer = ast_str_alloca(combined_size);
1595
1596         ast_str_set(&body_buffer, 0, "%.*s%s", (int) tdata->msg->body->len, (char *) tdata->msg->body->data, body_text);
1597
1598         tdata->msg->body->data = pj_pool_alloc(tdata->pool, combined_size);
1599         pj_memcpy(tdata->msg->body->data, ast_str_buffer(body_buffer), combined_size);
1600         tdata->msg->body->len = combined_size;
1601
1602         return 0;
1603 }
1604
1605 struct ast_taskprocessor *ast_sip_create_serializer(void)
1606 {
1607         struct ast_taskprocessor *serializer;
1608         RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1609         char name[AST_UUID_STR_LEN];
1610
1611         if (!uuid) {
1612                 return NULL;
1613         }
1614
1615         ast_uuid_to_str(uuid, name, sizeof(name));
1616
1617         serializer = ast_threadpool_serializer(name, sip_threadpool);
1618         if (!serializer) {
1619                 return NULL;
1620         }
1621         return serializer;
1622 }
1623
1624 int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1625 {
1626         if (serializer) {
1627                 return ast_taskprocessor_push(serializer, sip_task, task_data);
1628         } else {
1629                 return ast_threadpool_push(sip_threadpool, sip_task, task_data);
1630         }
1631 }
1632
1633 struct sync_task_data {
1634         ast_mutex_t lock;
1635         ast_cond_t cond;
1636         int complete;
1637         int fail;
1638         int (*task)(void *);
1639         void *task_data;
1640 };
1641
1642 static int sync_task(void *data)
1643 {
1644         struct sync_task_data *std = data;
1645         std->fail = std->task(std->task_data);
1646
1647         ast_mutex_lock(&std->lock);
1648         std->complete = 1;
1649         ast_cond_signal(&std->cond);
1650         ast_mutex_unlock(&std->lock);
1651         return std->fail;
1652 }
1653
1654 int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1655 {
1656         /* This method is an onion */
1657         struct sync_task_data std;
1658         ast_mutex_init(&std.lock);
1659         ast_cond_init(&std.cond, NULL);
1660         std.fail = std.complete = 0;
1661         std.task = sip_task;
1662         std.task_data = task_data;
1663
1664         if (serializer) {
1665                 if (ast_taskprocessor_push(serializer, sync_task, &std)) {
1666                         return -1;
1667                 }
1668         } else {
1669                 if (ast_threadpool_push(sip_threadpool, sync_task, &std)) {
1670                         return -1;
1671                 }
1672         }
1673
1674         ast_mutex_lock(&std.lock);
1675         while (!std.complete) {
1676                 ast_cond_wait(&std.cond, &std.lock);
1677         }
1678         ast_mutex_unlock(&std.lock);
1679
1680         ast_mutex_destroy(&std.lock);
1681         ast_cond_destroy(&std.cond);
1682         return std.fail;
1683 }
1684
1685 void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size)
1686 {
1687         size_t chars_to_copy = MIN(size - 1, pj_strlen(src));
1688         memcpy(dest, pj_strbuf(src), chars_to_copy);
1689         dest[chars_to_copy] = '\0';
1690 }
1691
1692 int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype)
1693 {
1694         pjsip_media_type compare;
1695
1696         if (!content_type) {
1697                 return 0;
1698         }
1699
1700         pjsip_media_type_init2(&compare, type, subtype);
1701
1702         return pjsip_media_type_cmp(content_type, &compare, 0) ? -1 : 0;
1703 }
1704
1705 pj_caching_pool caching_pool;
1706 pj_pool_t *memory_pool;
1707 pj_thread_t *monitor_thread;
1708 static int monitor_continue;
1709
1710 static void *monitor_thread_exec(void *endpt)
1711 {
1712         while (monitor_continue) {
1713                 const pj_time_val delay = {0, 10};
1714                 pjsip_endpt_handle_events(ast_pjsip_endpoint, &delay);
1715         }
1716         return NULL;
1717 }
1718
1719 static void stop_monitor_thread(void)
1720 {
1721         monitor_continue = 0;
1722         pj_thread_join(monitor_thread);
1723 }
1724
1725 AST_THREADSTORAGE(pj_thread_storage);
1726 AST_THREADSTORAGE(servant_id_storage);
1727 #define SIP_SERVANT_ID 0x5E2F1D
1728
1729 static void sip_thread_start(void)
1730 {
1731         pj_thread_desc *desc;
1732         pj_thread_t *thread;
1733         uint32_t *servant_id;
1734
1735         servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1736         if (!servant_id) {
1737                 ast_log(LOG_ERROR, "Could not set SIP servant ID in thread-local storage.\n");
1738                 return;
1739         }
1740         *servant_id = SIP_SERVANT_ID;
1741
1742         desc = ast_threadstorage_get(&pj_thread_storage, sizeof(pj_thread_desc));
1743         if (!desc) {
1744                 ast_log(LOG_ERROR, "Could not get thread desc from thread-local storage. Expect awful things to occur\n");
1745                 return;
1746         }
1747         pj_bzero(*desc, sizeof(*desc));
1748
1749         if (pj_thread_register("Asterisk Thread", *desc, &thread) != PJ_SUCCESS) {
1750                 ast_log(LOG_ERROR, "Couldn't register thread with PJLIB.\n");
1751         }
1752 }
1753
1754 int ast_sip_thread_is_servant(void)
1755 {
1756         uint32_t *servant_id;
1757
1758         servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1759         if (!servant_id) {
1760                 return 0;
1761         }
1762
1763         return *servant_id == SIP_SERVANT_ID;
1764 }
1765
1766 static void remove_request_headers(pjsip_endpoint *endpt)
1767 {
1768         const pjsip_hdr *request_headers = pjsip_endpt_get_request_headers(endpt);
1769         pjsip_hdr *iter = request_headers->next;
1770
1771         while (iter != request_headers) {
1772                 pjsip_hdr *to_erase = iter;
1773                 iter = iter->next;
1774                 pj_list_erase(to_erase);
1775         }
1776 }
1777
1778 static int load_module(void)
1779 {
1780         /* The third parameter is just copied from
1781          * example code from PJLIB. This can be adjusted
1782          * if necessary.
1783          */
1784         pj_status_t status;
1785         struct ast_threadpool_options options;
1786
1787         if (pj_init() != PJ_SUCCESS) {
1788                 return AST_MODULE_LOAD_DECLINE;
1789         }
1790
1791         if (pjlib_util_init() != PJ_SUCCESS) {
1792                 pj_shutdown();
1793                 return AST_MODULE_LOAD_DECLINE;
1794         }
1795
1796         pj_caching_pool_init(&caching_pool, NULL, 1024 * 1024);
1797         if (pjsip_endpt_create(&caching_pool.factory, "SIP", &ast_pjsip_endpoint) != PJ_SUCCESS) {
1798                 ast_log(LOG_ERROR, "Failed to create PJSIP endpoint structure. Aborting load\n");
1799                 goto error;
1800         }
1801
1802         /* PJSIP will automatically try to add a Max-Forwards header. Since we want to control that,
1803          * we need to stop PJSIP from doing it automatically
1804          */
1805         remove_request_headers(ast_pjsip_endpoint);
1806
1807         memory_pool = pj_pool_create(&caching_pool.factory, "SIP", 1024, 1024, NULL);
1808         if (!memory_pool) {
1809                 ast_log(LOG_ERROR, "Failed to create memory pool for SIP. Aborting load\n");
1810                 goto error;
1811         }
1812
1813         if (ast_sip_initialize_system()) {
1814                 ast_log(LOG_ERROR, "Failed to initialize SIP system configuration. Aborting load\n");
1815                 goto error;
1816         }
1817
1818         sip_get_threadpool_options(&options);
1819         options.thread_start = sip_thread_start;
1820         sip_threadpool = ast_threadpool_create("SIP", NULL, &options);
1821         if (!sip_threadpool) {
1822                 ast_log(LOG_ERROR, "Failed to create SIP threadpool. Aborting load\n");
1823                 goto error;
1824         }
1825
1826         pjsip_tsx_layer_init_module(ast_pjsip_endpoint);
1827         pjsip_ua_init_module(ast_pjsip_endpoint, NULL);
1828
1829         monitor_continue = 1;
1830         status = pj_thread_create(memory_pool, "SIP", (pj_thread_proc *) &monitor_thread_exec,
1831                         NULL, PJ_THREAD_DEFAULT_STACK_SIZE * 2, 0, &monitor_thread);
1832         if (status != PJ_SUCCESS) {
1833                 ast_log(LOG_ERROR, "Failed to start SIP monitor thread. Aborting load\n");
1834                 goto error;
1835         }
1836
1837         ast_sip_initialize_global_headers();
1838
1839         if (ast_res_pjsip_initialize_configuration()) {
1840                 ast_log(LOG_ERROR, "Failed to initialize SIP configuration. Aborting load\n");
1841                 goto error;
1842         }
1843
1844         if (ast_sip_initialize_distributor()) {
1845                 ast_log(LOG_ERROR, "Failed to register distributor module. Aborting load\n");
1846                 goto error;
1847         }
1848
1849         if (ast_sip_initialize_outbound_authentication()) {
1850                 ast_log(LOG_ERROR, "Failed to initialize outbound authentication. Aborting load\n");
1851                 goto error;
1852         }
1853
1854         ast_res_pjsip_init_options_handling(0);
1855
1856         ast_res_pjsip_init_contact_transports();
1857
1858 return AST_MODULE_LOAD_SUCCESS;
1859
1860 error:
1861         ast_sip_destroy_distributor();
1862         ast_res_pjsip_destroy_configuration();
1863         ast_sip_destroy_global_headers();
1864         if (monitor_thread) {
1865                 stop_monitor_thread();
1866         }
1867         if (memory_pool) {
1868                 pj_pool_release(memory_pool);
1869                 memory_pool = NULL;
1870         }
1871         if (ast_pjsip_endpoint) {
1872                 pjsip_endpt_destroy(ast_pjsip_endpoint);
1873                 ast_pjsip_endpoint = NULL;
1874         }
1875         pj_caching_pool_destroy(&caching_pool);
1876         return AST_MODULE_LOAD_DECLINE;
1877 }
1878
1879 static int reload_module(void)
1880 {
1881         if (ast_res_pjsip_reload_configuration()) {
1882                 return AST_MODULE_LOAD_DECLINE;
1883         }
1884         ast_res_pjsip_init_options_handling(1);
1885         return 0;
1886 }
1887
1888 static int unload_pjsip(void *data)
1889 {
1890         if (memory_pool) {
1891                 pj_pool_release(memory_pool);
1892                 memory_pool = NULL;
1893         }
1894         if (ast_pjsip_endpoint) {
1895                 pjsip_endpt_destroy(ast_pjsip_endpoint);
1896                 ast_pjsip_endpoint = NULL;
1897         }
1898         pj_caching_pool_destroy(&caching_pool);
1899         return 0;
1900 }
1901
1902 static int unload_module(void)
1903 {
1904         ast_res_pjsip_cleanup_options_handling();
1905         ast_sip_destroy_distributor();
1906         ast_res_pjsip_destroy_configuration();
1907         ast_sip_destroy_global_headers();
1908         if (monitor_thread) {
1909                 stop_monitor_thread();
1910         }
1911         /* The thread this is called from cannot call PJSIP/PJLIB functions,
1912          * so we have to push the work to the threadpool to handle
1913          */
1914         ast_sip_push_task_synchronous(NULL, unload_pjsip, NULL);
1915
1916         ast_threadpool_shutdown(sip_threadpool);
1917
1918         return 0;
1919 }
1920
1921 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Basic SIP resource",
1922                 .load = load_module,
1923                 .unload = unload_module,
1924                 .reload = reload_module,
1925                 .load_pri = AST_MODPRI_CHANNEL_DEPEND - 5,
1926 );