eab9b14037ede2e538f976832c520a74fbf674ff
[asterisk/asterisk.git] / res / res_pjsip.c
1 /*
2  * Asterisk -- An open source telephony toolkit.
3  *
4  * Copyright (C) 2013, Digium, Inc.
5  *
6  * Mark Michelson <mmichelson@digium.com>
7  *
8  * See http://www.asterisk.org for more information about
9  * the Asterisk project. Please do not directly contact
10  * any of the maintainers of this project for assistance;
11  * the project provides a web site, mailing lists and IRC
12  * channels for your use.
13  *
14  * This program is free software, distributed under the terms of
15  * the GNU General Public License Version 2. See the LICENSE file
16  * at the top of the source tree.
17  */
18
19 #include "asterisk.h"
20
21 #include <pjsip.h>
22 /* Needed for SUBSCRIBE, NOTIFY, and PUBLISH method definitions */
23 #include <pjsip_simple.h>
24 #include <pjlib.h>
25
26 #include "asterisk/res_pjsip.h"
27 #include "res_pjsip/include/res_pjsip_private.h"
28 #include "asterisk/linkedlists.h"
29 #include "asterisk/logger.h"
30 #include "asterisk/lock.h"
31 #include "asterisk/utils.h"
32 #include "asterisk/astobj2.h"
33 #include "asterisk/module.h"
34 #include "asterisk/threadpool.h"
35 #include "asterisk/taskprocessor.h"
36 #include "asterisk/uuid.h"
37 #include "asterisk/sorcery.h"
38
39 /*** MODULEINFO
40         <depend>pjproject</depend>
41         <depend>res_sorcery_config</depend>
42         <support_level>core</support_level>
43  ***/
44
45 /*** DOCUMENTATION
46         <configInfo name="res_pjsip" language="en_US">
47                 <synopsis>SIP Resource using PJProject</synopsis>
48                 <configFile name="pjsip.conf">
49                         <configObject name="endpoint">
50                                 <synopsis>Endpoint</synopsis>
51                                 <description><para>
52                                         The <emphasis>Endpoint</emphasis> is the primary configuration object.
53                                         It contains the core SIP related options only, endpoints are <emphasis>NOT</emphasis>
54                                         dialable entries of their own. Communication with another SIP device is
55                                         accomplished via Addresses of Record (AoRs) which have one or more
56                                         contacts assicated with them. Endpoints <emphasis>NOT</emphasis> configured to
57                                         use a <literal>transport</literal> will default to first transport found
58                                         in <filename>pjsip.conf</filename> that matches its type.
59                                         </para>
60                                         <para>Example: An Endpoint has been configured with no transport.
61                                         When it comes time to call an AoR, PJSIP will find the
62                                         first transport that matches the type. A SIP URI of <literal>sip:5000@[11::33]</literal>
63                                         will use the first IPv6 transport and try to send the request.
64                                         </para>
65                                         <para>If the anonymous endpoint identifier is in use an endpoint with the name
66                                         "anonymous@domain" will be searched for as a last resort. If this is not found
67                                         it will fall back to searching for "anonymous". If neither endpoints are found
68                                         the anonymous endpoint identifier will not return an endpoint and anonymous
69                                         calling will not be possible.
70                                         </para>
71                                 </description>
72                                 <configOption name="100rel" default="yes">
73                                         <synopsis>Allow support for RFC3262 provisional ACK tags</synopsis>
74                                         <description>
75                                                 <enumlist>
76                                                         <enum name="no" />
77                                                         <enum name="required" />
78                                                         <enum name="yes" />
79                                                 </enumlist>
80                                         </description>
81                                 </configOption>
82                                 <configOption name="aggregate_mwi" default="yes">
83                                         <synopsis></synopsis>
84                                         <description><para>When enabled, <replaceable>aggregate_mwi</replaceable> condenses message
85                                         waiting notifications from multiple mailboxes into a single NOTIFY. If it is disabled,
86                                         individual NOTIFYs are sent for each mailbox.</para></description>
87                                 </configOption>
88                                 <configOption name="allow">
89                                         <synopsis>Media Codec(s) to allow</synopsis>
90                                 </configOption>
91                                 <configOption name="aors">
92                                         <synopsis>AoR(s) to be used with the endpoint</synopsis>
93                                         <description><para>
94                                                 List of comma separated AoRs that the endpoint should be associated with.
95                                         </para></description>
96                                 </configOption>
97                                 <configOption name="auth">
98                                         <synopsis>Authentication Object(s) associated with the endpoint</synopsis>
99                                         <description><para>
100                                                 This is a comma-delimited list of <replaceable>auth</replaceable> sections defined
101                                                 in <filename>pjsip.conf</filename> to be used to verify inbound connection attempts.
102                                                 </para><para>
103                                                 Endpoints without an <literal>authentication</literal> object
104                                                 configured will allow connections without vertification.
105                                         </para></description>
106                                 </configOption>
107                                 <configOption name="callerid">
108                                         <synopsis>CallerID information for the endpoint</synopsis>
109                                         <description><para>
110                                                 Must be in the format <literal>Name &lt;Number&gt;</literal>,
111                                                 or only <literal>&lt;Number&gt;</literal>.
112                                         </para></description>
113                                 </configOption>
114                                 <configOption name="callerid_privacy">
115                                         <synopsis>Default privacy level</synopsis>
116                                         <description>
117                                                 <enumlist>
118                                                         <enum name="allowed_not_screened" />
119                                                         <enum name="allowed_passed_screened" />
120                                                         <enum name="allowed_failed_screened" />
121                                                         <enum name="allowed" />
122                                                         <enum name="prohib_not_screened" />
123                                                         <enum name="prohib_passed_screened" />
124                                                         <enum name="prohib_failed_screened" />
125                                                         <enum name="prohib" />
126                                                         <enum name="unavailable" />
127                                                 </enumlist>
128                                         </description>
129                                 </configOption>
130                                 <configOption name="callerid_tag">
131                                         <synopsis>Internal id_tag for the endpoint</synopsis>
132                                 </configOption>
133                                 <configOption name="context">
134                                         <synopsis>Dialplan context for inbound sessions</synopsis>
135                                 </configOption>
136                                 <configOption name="direct_media_glare_mitigation" default="none">
137                                         <synopsis>Mitigation of direct media (re)INVITE glare</synopsis>
138                                         <description>
139                                                 <para>
140                                                 This setting attempts to avoid creating INVITE glare scenarios
141                                                 by disabling direct media reINVITEs in one direction thereby allowing
142                                                 designated servers (according to this option) to initiate direct
143                                                 media reINVITEs without contention and significantly reducing call
144                                                 setup time.
145                                                 </para>
146                                                 <para>
147                                                 A more detailed description of how this option functions can be found on
148                                                 the Asterisk wiki https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance
149                                                 </para>
150                                                 <enumlist>
151                                                         <enum name="none" />
152                                                         <enum name="outgoing" />
153                                                         <enum name="incoming" />
154                                                 </enumlist>
155                                         </description>
156                                 </configOption>
157                                 <configOption name="direct_media_method" default="invite">
158                                         <synopsis>Direct Media method type</synopsis>
159                                         <description>
160                                                 <para>Method for setting up Direct Media between endpoints.</para>
161                                                 <enumlist>
162                                                         <enum name="invite" />
163                                                         <enum name="reinvite">
164                                                                 <para>Alias for the <literal>invite</literal> value.</para>
165                                                         </enum>
166                                                         <enum name="update" />
167                                                 </enumlist>
168                                         </description>
169                                 </configOption>
170                                 <configOption name="connected_line_method" default="invite">
171                                         <synopsis>Connected line method type</synopsis>
172                                         <description>
173                                                 <para>Method used when updating connected line information.</para>
174                                                 <enumlist>
175                                                         <enum name="invite" />
176                                                         <enum name="reinvite">
177                                                                 <para>Alias for the <literal>invite</literal> value.</para>
178                                                         </enum>
179                                                         <enum name="update" />
180                                                 </enumlist>
181                                         </description>
182                                 </configOption>
183                                 <configOption name="direct_media" default="yes">
184                                         <synopsis>Determines whether media may flow directly between endpoints.</synopsis>
185                                 </configOption>
186                                 <configOption name="disable_direct_media_on_nat" default="no">
187                                         <synopsis>Disable direct media session refreshes when NAT obstructs the media session</synopsis>
188                                 </configOption>
189                                 <configOption name="disallow">
190                                         <synopsis>Media Codec(s) to disallow</synopsis>
191                                 </configOption>
192                                 <configOption name="dtmfmode" default="rfc4733">
193                                         <synopsis>DTMF mode</synopsis>
194                                         <description>
195                                                 <para>This setting allows to choose the DTMF mode for endpoint communication.</para>
196                                                 <enumlist>
197                                                         <enum name="rfc4733">
198                                                                 <para>DTMF is sent out of band of the main audio stream.This
199                                                                 supercedes the older <emphasis>RFC-2833</emphasis> used within
200                                                                 the older <literal>chan_sip</literal>.</para>
201                                                         </enum>
202                                                         <enum name="inband">
203                                                                 <para>DTMF is sent as part of audio stream.</para>
204                                                         </enum>
205                                                         <enum name="info">
206                                                                 <para>DTMF is sent as SIP INFO packets.</para>
207                                                         </enum>
208                                                 </enumlist>
209                                         </description>
210                                 </configOption>
211                                 <configOption name="media_address">
212                                         <synopsis>IP address used in SDP for media handling</synopsis>
213                                         <description><para>
214                                                 At the time of SDP creation, the IP address defined here will be used as
215                                                 the media address for individual streams in the SDP.
216                                         </para>
217                                         <note><para>
218                                                 Be aware that the <literal>external_media_address</literal> option, set in Transport
219                                                 configuration, can also affect the final media address used in the SDP.
220                                         </para></note>
221                                         </description>
222                                 </configOption>
223                                 <configOption name="force_rport" default="yes">
224                                         <synopsis>Force use of return port</synopsis>
225                                 </configOption>
226                                 <configOption name="ice_support" default="no">
227                                         <synopsis>Enable the ICE mechanism to help traverse NAT</synopsis>
228                                 </configOption>
229                                 <configOption name="identify_by" default="username,location">
230                                         <synopsis>Way(s) for Endpoint to be identified</synopsis>
231                                         <description><para>
232                                                 An endpoint can be identified in multiple ways. Currently, the only supported
233                                                 option is <literal>username</literal>, which matches the endpoint based on the
234                                                 username in the From header.
235                                                 </para>
236                                                 <note><para>Endpoints can also be identified by IP address; however, that method
237                                                 of identification is not handled by this configuration option. See the documentation
238                                                 for the <literal>identify</literal> configuration section for more details on that
239                                                 method of endpoint identification. If this option is set to <literal>username</literal>
240                                                 and an <literal>identify</literal> configuration section exists for the endpoint, then
241                                                 the endpoint can be identified in multiple ways.</para></note>
242                                                 <enumlist>
243                                                         <enum name="username" />
244                                                 </enumlist>
245                                         </description>
246                                 </configOption>
247                                 <configOption name="mailboxes">
248                                         <synopsis>Mailbox(es) to be associated with</synopsis>
249                                 </configOption>
250                                 <configOption name="mohsuggest" default="default">
251                                         <synopsis>Default Music On Hold class</synopsis>
252                                 </configOption>
253                                 <configOption name="outbound_auth">
254                                         <synopsis>Authentication object used for outbound requests</synopsis>
255                                 </configOption>
256                                 <configOption name="outbound_proxy">
257                                         <synopsis>Proxy through which to send requests</synopsis>
258                                 </configOption>
259                                 <configOption name="rewrite_contact">
260                                         <synopsis>Allow Contact header to be rewritten with the source IP address-port</synopsis>
261                                         <description><para>
262                                                 On inbound SIP messages from this endpoint, the Contact header will be changed to have the
263                                                 source IP address and port. This option does not affect outbound messages send to this
264                                                 endpoint.
265                                         </para></description>
266                                 </configOption>
267                                 <configOption name="rtp_ipv6" default="no">
268                                         <synopsis>Allow use of IPv6 for RTP traffic</synopsis>
269                                 </configOption>
270                                 <configOption name="rtp_symmetric" default="no">
271                                         <synopsis>Enforce that RTP must be symmetric</synopsis>
272                                 </configOption>
273                                 <configOption name="send_diversion" default="yes">
274                                         <synopsis>Send the Diversion header, conveying the diversion
275                                         information to the called user agent</synopsis>
276                                 </configOption>
277                                 <configOption name="send_pai" default="no">
278                                         <synopsis>Send the P-Asserted-Identity header</synopsis>
279                                 </configOption>
280                                 <configOption name="send_rpid" default="no">
281                                         <synopsis>Send the Remote-Party-ID header</synopsis>
282                                 </configOption>
283                                 <configOption name="timers_min_se" default="90">
284                                         <synopsis>Minimum session timers expiration period</synopsis>
285                                         <description><para>
286                                                 Minimium session timer expiration period. Time in seconds.
287                                         </para></description>
288                                 </configOption>
289                                 <configOption name="timers" default="yes">
290                                         <synopsis>Session timers for SIP packets</synopsis>
291                                         <description>
292                                                 <enumlist>
293                                                         <enum name="forced" />
294                                                         <enum name="no" />
295                                                         <enum name="required" />
296                                                         <enum name="yes" />
297                                                 </enumlist>
298                                         </description>
299                                 </configOption>
300                                 <configOption name="timers_sess_expires" default="1800">
301                                         <synopsis>Maximum session timer expiration period</synopsis>
302                                         <description><para>
303                                                 Maximium session timer expiration period. Time in seconds.
304                                         </para></description>
305                                 </configOption>
306                                 <configOption name="transport">
307                                         <synopsis>Desired transport configuration</synopsis>
308                                         <description><para>
309                                                 This will set the desired transport configuration to send SIP data through.
310                                                 </para>
311                                                 <warning><para>Not specifying a transport will <emphasis>DEFAULT</emphasis>
312                                                 to the first configured transport in <filename>pjsip.conf</filename> which is
313                                                 valid for the URI we are trying to contact.
314                                                 </para></warning>
315                                                 <warning><para>Transport configuration is not affected by reloads. In order to
316                                                 change transports, a full Asterisk restart is required</para></warning>
317                                         </description>
318                                 </configOption>
319                                 <configOption name="trust_id_inbound" default="no">
320                                         <synopsis>Accept identification information received from this endpoint</synopsis>
321                                         <description><para>This option determines whether Asterisk will accept
322                                         identification from the endpoint from headers such as P-Asserted-Identity
323                                         or Remote-Party-ID header. This option applies both to calls originating from the
324                                         endpoint and calls originating from Asterisk. If <literal>no</literal>, the
325                                         configured Caller-ID from pjsip.conf will always be used as the identity for
326                                         the endpoint.</para></description>
327                                 </configOption>
328                                 <configOption name="trust_id_outbound" default="no">
329                                         <synopsis>Send private identification details to the endpoint.</synopsis>
330                                         <description><para>This option determines whether res_pjsip will send private
331                                         identification information to the endpoint. If <literal>no</literal>,
332                                         private Caller-ID information will not be forwarded to the endpoint.
333                                         "Private" in this case refers to any method of restricting identification.
334                                         Example: setting <replaceable>callerid_privacy</replaceable> to any
335                                         <literal>prohib</literal> variation.
336                                         Example: If <replaceable>trust_id_inbound</replaceable> is set to
337                                         <literal>yes</literal>, the presence of a <literal>Privacy: id</literal>
338                                         header in a SIP request or response would indicate the identification
339                                         provided in the request is private.</para></description>
340                                 </configOption>
341                                 <configOption name="type">
342                                         <synopsis>Must be of type 'endpoint'.</synopsis>
343                                 </configOption>
344                                 <configOption name="use_ptime" default="no">
345                                         <synopsis>Use Endpoint's requested packetisation interval</synopsis>
346                                 </configOption>
347                                 <configOption name="use_avpf" default="no">
348                                         <synopsis>Determines whether res_pjsip will use and enforce usage of AVPF for this
349                                         endpoint.</synopsis>
350                                         <description><para>
351                                                 If set to <literal>yes</literal>, res_pjsip will use use the AVPF or SAVPF RTP
352                                                 profile for all media offers on outbound calls and media updates and will
353                                                 decline media offers not using the AVPF or SAVPF profile.
354                                         </para><para>
355                                                 If set to <literal>no</literal>, res_pjsip will use use the AVP or SAVP RTP
356                                                 profile for all media offers on outbound calls and media updates and will
357                                                 decline media offers not using the AVP or SAVP profile.
358                                         </para></description>
359                                 </configOption>
360                                 <configOption name="media_encryption" default="no">
361                                         <synopsis>Determines whether res_pjsip will use and enforce usage of media encryption
362                                         for this endpoint.</synopsis>
363                                         <description>
364                                                 <enumlist>
365                                                         <enum name="no"><para>
366                                                                 res_pjsip will offer no encryption and allow no encryption to be setup.
367                                                         </para></enum>
368                                                         <enum name="sdes"><para>
369                                                                 res_pjsip will offer standard SRTP setup via in-SDP keys. Encrypted SIP
370                                                                 transport should be used in conjunction with this option to prevent
371                                                                 exposure of media encryption keys.
372                                                         </para></enum>
373                                                         <enum name="dtls"><para>
374                                                                 res_pjsip will offer DTLS-SRTP setup.
375                                                         </para></enum>
376                                                 </enumlist>
377                                         </description>
378                                 </configOption>
379                                 <configOption name="inband_progress" default="no">
380                                         <synopsis>Determines whether chan_pjsip will indicate ringing using inband
381                                             progress.</synopsis>
382                                         <description><para>
383                                                 If set to <literal>yes</literal>, chan_pjsip will send a 183 Session Progress
384                                                 when told to indicate ringing and will immediately start sending ringing
385                                                 as audio.
386                                         </para><para>
387                                                 If set to <literal>no</literal>, chan_pjsip will send a 180 Ringing when told
388                                                 to indicate ringing and will NOT send it as audio.
389                                         </para></description>
390                                 </configOption>
391                                 <configOption name="callgroup">
392                                         <synopsis>The numeric pickup groups for a channel.</synopsis>
393                                         <description><para>
394                                                 Can be set to a comma separated list of numbers or ranges between the values
395                                                 of 0-63 (maximum of 64 groups).
396                                         </para></description>
397                                 </configOption>
398                                 <configOption name="pickupgroup">
399                                         <synopsis>The numeric pickup groups that a channel can pickup.</synopsis>
400                                         <description><para>
401                                                 Can be set to a comma separated list of numbers or ranges between the values
402                                                 of 0-63 (maximum of 64 groups).
403                                         </para></description>
404                                 </configOption>
405                                 <configOption name="namedcallgroup">
406                                         <synopsis>The named pickup groups for a channel.</synopsis>
407                                         <description><para>
408                                                 Can be set to a comma separated list of case sensitive strings limited by
409                                                 supported line length.
410                                         </para></description>
411                                 </configOption>
412                                 <configOption name="namedpickupgroup">
413                                         <synopsis>The named pickup groups that a channel can pickup.</synopsis>
414                                         <description><para>
415                                                 Can be set to a comma separated list of case sensitive strings limited by
416                                                 supported line length.
417                                         </para></description>
418                                 </configOption>
419                                 <configOption name="devicestate_busy_at" default="0">
420                                         <synopsis>The number of in-use channels which will cause busy to be returned as device state</synopsis>
421                                         <description><para>
422                                                 When the number of in-use channels for the endpoint matches the devicestate_busy_at setting the
423                                                 PJSIP channel driver will return busy as the device state instead of in use.
424                                         </para></description>
425                                 </configOption>
426                                 <configOption name="t38udptl" default="no">
427                                         <synopsis>Whether T.38 UDPTL support is enabled or not</synopsis>
428                                         <description><para>
429                                                 If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted
430                                                 and relayed.
431                                         </para></description>
432                                 </configOption>
433                                 <configOption name="t38udptl_ec" default="none">
434                                         <synopsis>T.38 UDPTL error correction method</synopsis>
435                                         <description>
436                                                 <enumlist>
437                                                         <enum name="none"><para>
438                                                                 No error correction should be used.
439                                                         </para></enum>
440                                                         <enum name="fec"><para>
441                                                                 Forward error correction should be used.
442                                                         </para></enum>
443                                                         <enum name="redundancy"><para>
444                                                                 Redundacy error correction should be used.
445                                                         </para></enum>
446                                                 </enumlist>
447                                         </description>
448                                 </configOption>
449                                 <configOption name="t38udptl_maxdatagram" default="0">
450                                         <synopsis>T.38 UDPTL maximum datagram size</synopsis>
451                                         <description><para>
452                                                 This option can be set to override the maximum datagram of a remote endpoint for broken
453                                                 endpoints.
454                                         </para></description>
455                                 </configOption>
456                                 <configOption name="faxdetect" default="no">
457                                         <synopsis>Whether CNG tone detection is enabled</synopsis>
458                                         <description><para>
459                                                 This option can be set to send the session to the fax extension when a CNG tone is
460                                                 detected.
461                                         </para></description>
462                                 </configOption>
463                                 <configOption name="t38udptl_nat" default="no">
464                                         <synopsis>Whether NAT support is enabled on UDPTL sessions</synopsis>
465                                         <description><para>
466                                                 When enabled the UDPTL stack will send UDPTL packets to the source address of
467                                                 received packets.
468                                         </para></description>
469                                 </configOption>
470                                 <configOption name="t38udptl_ipv6" default="no">
471                                         <synopsis>Whether IPv6 is used for UDPTL Sessions</synopsis>
472                                         <description><para>
473                                                 When enabled the UDPTL stack will use IPv6.
474                                         </para></description>
475                                 </configOption>
476                                 <configOption name="tonezone">
477                                         <synopsis>Set which country's indications to use for channels created for this endpoint.</synopsis>
478                                 </configOption>
479                                 <configOption name="language">
480                                         <synopsis>Set the default language to use for channels created for this endpoint.</synopsis>
481                                 </configOption>
482                                 <configOption name="one_touch_recording" default="no">
483                                         <synopsis>Determines whether one-touch recording is allowed for this endpoint.</synopsis>
484                                         <see-also>
485                                                 <ref type="configOption">recordonfeature</ref>
486                                                 <ref type="configOption">recordofffeature</ref>
487                                         </see-also>
488                                 </configOption>
489                                 <configOption name="recordonfeature" default="automixmon">
490                                         <synopsis>The feature to enact when one-touch recording is turned on.</synopsis>
491                                         <description>
492                                                 <para>When an INFO request for one-touch recording arrives with a Record header set to "on", this
493                                                 feature will be enabled for the channel. The feature designated here can be any built-in
494                                                 or dynamic feature defined in features.conf.</para>
495                                                 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
496                                         </description>
497                                         <see-also>
498                                                 <ref type="configOption">one_touch_recording</ref>
499                                                 <ref type="configOption">recordofffeature</ref>
500                                         </see-also>
501                                 </configOption>
502                                 <configOption name="recordofffeature" default="automixmon">
503                                         <synopsis>The feature to enact when one-touch recording is turned off.</synopsis>
504                                         <description>
505                                                 <para>When an INFO request for one-touch recording arrives with a Record header set to "off", this
506                                                 feature will be enabled for the channel. The feature designated here can be any built-in
507                                                 or dynamic feature defined in features.conf.</para>
508                                                 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
509                                         </description>
510                                         <see-also>
511                                                 <ref type="configOption">one_touch_recording</ref>
512                                                 <ref type="configOption">recordonfeature</ref>
513                                         </see-also>
514                                 </configOption>
515                                 <configOption name="rtpengine" default="asterisk">
516                                         <synopsis>Name of the RTP engine to use for channels created for this endpoint</synopsis>
517                                 </configOption>
518                                 <configOption name="allowtransfer" default="yes">
519                                         <synopsis>Determines whether SIP REFER transfers are allowed for this endpoint</synopsis>
520                                 </configOption>
521                                 <configOption name="sdpowner" default="-">
522                                         <synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis>
523                                 </configOption>
524                                 <configOption name="sdpsession" default="Asterisk">
525                                         <synopsis>String used for the SDP session (s=) line.</synopsis>
526                                 </configOption>
527                                 <configOption name="tos_audio">
528                                         <synopsis>DSCP TOS bits for audio streams</synopsis>
529                                         <description><para>
530                                                 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
531                                         </para></description>
532                                 </configOption>
533                                 <configOption name="tos_video">
534                                         <synopsis>DSCP TOS bits for video streams</synopsis>
535                                         <description><para>
536                                                 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
537                                         </para></description>
538                                 </configOption>
539                                 <configOption name="cos_audio">
540                                         <synopsis>Priority for audio streams</synopsis>
541                                         <description><para>
542                                                 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
543                                         </para></description>
544                                 </configOption>
545                                 <configOption name="cos_video">
546                                         <synopsis>Priority for video streams</synopsis>
547                                         <description><para>
548                                                 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
549                                         </para></description>
550                                 </configOption>
551                                 <configOption name="allowsubscribe" default="yes">
552                                         <synopsis>Determines if endpoint is allowed to initiate subscriptions with Asterisk.</synopsis>
553                                 </configOption>
554                                 <configOption name="subminexpiry" default="60">
555                                         <synopsis>The minimum allowed expiry time for subscriptions initiated by the endpoint.</synopsis>
556                                 </configOption>
557                                 <configOption name="fromuser">
558                                         <synopsis>Username to use in From header for requests to this endpoint.</synopsis>
559                                 </configOption>
560                                 <configOption name="mwifromuser">
561                                         <synopsis>Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.</synopsis>
562                                 </configOption>
563                                 <configOption name="fromdomain">
564                                         <synopsis>Domain to user in From header for requests to this endpoint.</synopsis>
565                                 </configOption>
566                                 <configOption name="dtlsverify">
567                                         <synopsis>Verify that the provided peer certificate is valid</synopsis>
568                                         <description><para>
569                                                 This option only applies if <replaceable>media_encryption</replaceable> is
570                                                 set to <literal>dtls</literal>.
571                                         </para></description>
572                                 </configOption>
573                                 <configOption name="dtlsrekey">
574                                         <synopsis>Interval at which to renegotiate the TLS session and rekey the SRTP session</synopsis>
575                                         <description><para>
576                                                 This option only applies if <replaceable>media_encryption</replaceable> is
577                                                 set to <literal>dtls</literal>.
578                                         </para><para>
579                                                 If this is not set or the value provided is 0 rekeying will be disabled.
580                                         </para></description>
581                                 </configOption>
582                                 <configOption name="dtlscertfile">
583                                         <synopsis>Path to certificate file to present to peer</synopsis>
584                                         <description><para>
585                                                 This option only applies if <replaceable>media_encryption</replaceable> is
586                                                 set to <literal>dtls</literal>.
587                                         </para></description>
588                                 </configOption>
589                                 <configOption name="dtlsprivatekey">
590                                         <synopsis>Path to private key for certificate file</synopsis>
591                                         <description><para>
592                                                 This option only applies if <replaceable>media_encryption</replaceable> is
593                                                 set to <literal>dtls</literal>.
594                                         </para></description>
595                                 </configOption>
596                                 <configOption name="dtlscipher">
597                                         <synopsis>Cipher to use for DTLS negotiation</synopsis>
598                                         <description><para>
599                                                 This option only applies if <replaceable>media_encryption</replaceable> is
600                                                 set to <literal>dtls</literal>.
601                                         </para><para>
602                                                 Many options for acceptable ciphers. See link for more:
603                                                 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
604                                         </para></description>
605                                 </configOption>
606                                 <configOption name="dtlscafile">
607                                         <synopsis>Path to certificate authority certificate</synopsis>
608                                         <description><para>
609                                                 This option only applies if <replaceable>media_encryption</replaceable> is
610                                                 set to <literal>dtls</literal>.
611                                         </para></description>
612                                 </configOption>
613                                 <configOption name="dtlscapath">
614                                         <synopsis>Path to a directory containing certificate authority certificates</synopsis>
615                                         <description><para>
616                                                 This option only applies if <replaceable>media_encryption</replaceable> is
617                                                 set to <literal>dtls</literal>.
618                                         </para></description>
619                                 </configOption>
620                                 <configOption name="dtlssetup">
621                                         <synopsis>Whether we are willing to accept connections, connect to the other party, or both.</synopsis>
622                                         <description>
623                                                 <para>
624                                                         This option only applies if <replaceable>media_encryption</replaceable> is
625                                                         set to <literal>dtls</literal>.
626                                                 </para>
627                                                 <enumlist>
628                                                         <enum name="active"><para>
629                                                                 res_pjsip will make a connection to the peer.
630                                                         </para></enum>
631                                                         <enum name="passive"><para>
632                                                                 res_pjsip will accept connections from the peer.
633                                                         </para></enum>
634                                                         <enum name="actpass"><para>
635                                                                 res_pjsip will offer and accept connections from the peer.
636                                                         </para></enum>
637                                                 </enumlist>
638                                         </description>
639                                 </configOption>
640                                 <configOption name="srtp_tag_32">
641                                         <synopsis>Determines whether 32 byte tags should be used instead of 80 byte tags.</synopsis>
642                                         <description><para>
643                                                 This option only applies if <replaceable>media_encryption</replaceable> is
644                                                 set to <literal>sdes</literal> or <literal>dtls</literal>.
645                                         </para></description>
646                                 </configOption>
647                         </configObject>
648                         <configObject name="auth">
649                                 <synopsis>Authentication type</synopsis>
650                                 <description><para>
651                                         Authentication objects hold the authentication information for use
652                                         by other objects such as <literal>endpoints</literal> or <literal>registrations</literal>.
653                                         This also allows for multiple objects to use a single auth object. See
654                                         the <literal>auth_type</literal> config option for password style choices.
655                                 </para></description>
656                                 <configOption name="auth_type" default="userpass">
657                                         <synopsis>Authentication type</synopsis>
658                                         <description><para>
659                                                 This option specifies which of the password style config options should be read
660                                                 when trying to authenticate an endpoint inbound request. If set to <literal>userpass</literal>
661                                                 then we'll read from the 'password' option. For <literal>md5</literal> we'll read
662                                                 from 'md5_cred'.
663                                                 </para>
664                                                 <enumlist>
665                                                         <enum name="md5"/>
666                                                         <enum name="userpass"/>
667                                                 </enumlist>
668                                         </description>
669                                 </configOption>
670                                 <configOption name="nonce_lifetime" default="32">
671                                         <synopsis>Lifetime of a nonce associated with this authentication config.</synopsis>
672                                 </configOption>
673                                 <configOption name="md5_cred">
674                                         <synopsis>MD5 Hash used for authentication.</synopsis>
675                                         <description><para>Only used when auth_type is <literal>md5</literal>.</para></description>
676                                 </configOption>
677                                 <configOption name="password">
678                                         <synopsis>PlainText password used for authentication.</synopsis>
679                                         <description><para>Only used when auth_type is <literal>userpass</literal>.</para></description>
680                                 </configOption>
681                                 <configOption name="realm" default="asterisk">
682                                         <synopsis>SIP realm for endpoint</synopsis>
683                                 </configOption>
684                                 <configOption name="type">
685                                         <synopsis>Must be 'auth'</synopsis>
686                                 </configOption>
687                                 <configOption name="username">
688                                         <synopsis>Username to use for account</synopsis>
689                                 </configOption>
690                         </configObject>
691                         <configObject name="domain_alias">
692                                 <synopsis>Domain Alias</synopsis>
693                                 <description><para>
694                                         Signifies that a domain is an alias. If the domain on a session is
695                                         not found to match an AoR then this object is used to see if we have
696                                         an alias for the AoR to which the endpoint is binding. This objects
697                                         name as defined in configuration should be the domain alias and a
698                                         config option is provided to specify the domain to be aliased.
699                                 </para></description>
700                                 <configOption name="type">
701                                         <synopsis>Must be of type 'domain_alias'.</synopsis>
702                                 </configOption>
703                                 <configOption name="domain">
704                                         <synopsis>Domain to be aliased</synopsis>
705                                 </configOption>
706                         </configObject>
707                         <configObject name="transport">
708                                 <synopsis>SIP Transport</synopsis>
709                                 <description><para>
710                                         <emphasis>Transports</emphasis>
711                                         </para>
712                                         <para>There are different transports and protocol derivatives
713                                                 supported by <literal>res_pjsip</literal>. They are in order of
714                                                 preference: UDP, TCP, and WebSocket (WS).</para>
715                                         <note><para>Changes to transport configuration in pjsip.conf will only be
716                                                 effected on a complete restart of Asterisk. A module reload
717                                                 will not suffice.</para></note>
718                                 </description>
719                                 <configOption name="async_operations" default="1">
720                                         <synopsis>Number of simultaneous Asynchronous Operations</synopsis>
721                                 </configOption>
722                                 <configOption name="bind">
723                                         <synopsis>IP Address and optional port to bind to for this transport</synopsis>
724                                 </configOption>
725                                 <configOption name="ca_list_file">
726                                         <synopsis>File containing a list of certificates to read (TLS ONLY)</synopsis>
727                                 </configOption>
728                                 <configOption name="cert_file">
729                                         <synopsis>Certificate file for endpoint (TLS ONLY)</synopsis>
730                                 </configOption>
731                                 <configOption name="cipher">
732                                         <synopsis>Preferred Cryptography Cipher (TLS ONLY)</synopsis>
733                                         <description><para>
734                                                 Many options for acceptable ciphers see link for more:
735                                                 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
736                                         </para></description>
737                                 </configOption>
738                                 <configOption name="domain">
739                                         <synopsis>Domain the transport comes from</synopsis>
740                                 </configOption>
741                                 <configOption name="external_media_address">
742                                         <synopsis>External IP address to use in RTP handling</synopsis>
743                                         <description><para>
744                                                 When a request or response is sent out, if the destination of the
745                                                 message is outside the IP network defined in the option <literal>localnet</literal>,
746                                                 and the media address in the SDP is within the localnet network, then the
747                                                 media address in the SDP will be rewritten to the value defined for
748                                                 <literal>external_media_address</literal>.
749                                         </para></description>
750                                 </configOption>
751                                 <configOption name="external_signaling_address">
752                                         <synopsis>External address for SIP signalling</synopsis>
753                                 </configOption>
754                                 <configOption name="external_signaling_port" default="0">
755                                         <synopsis>External port for SIP signalling</synopsis>
756                                 </configOption>
757                                 <configOption name="method">
758                                         <synopsis>Method of SSL transport (TLS ONLY)</synopsis>
759                                         <description>
760                                                 <enumlist>
761                                                         <enum name="default" />
762                                                         <enum name="unspecified" />
763                                                         <enum name="tlsv1" />
764                                                         <enum name="sslv2" />
765                                                         <enum name="sslv3" />
766                                                         <enum name="sslv23" />
767                                                 </enumlist>
768                                         </description>
769                                 </configOption>
770                                 <configOption name="localnet">
771                                         <synopsis>Network to consider local (used for NAT purposes).</synopsis>
772                                         <description><para>This must be in CIDR or dotted decimal format with the IP
773                                         and mask separated with a slash ('/').</para></description>
774                                 </configOption>
775                                 <configOption name="password">
776                                         <synopsis>Password required for transport</synopsis>
777                                 </configOption>
778                                 <configOption name="privkey_file">
779                                         <synopsis>Private key file (TLS ONLY)</synopsis>
780                                 </configOption>
781                                 <configOption name="protocol" default="udp">
782                                         <synopsis>Protocol to use for SIP traffic</synopsis>
783                                         <description>
784                                                 <enumlist>
785                                                         <enum name="udp" />
786                                                         <enum name="tcp" />
787                                                         <enum name="tls" />
788                                                         <enum name="ws" />
789                                                         <enum name="wss" />
790                                                 </enumlist>
791                                         </description>
792                                 </configOption>
793                                 <configOption name="require_client_cert" default="false">
794                                         <synopsis>Require client certificate (TLS ONLY)</synopsis>
795                                 </configOption>
796                                 <configOption name="type">
797                                         <synopsis>Must be of type 'transport'.</synopsis>
798                                 </configOption>
799                                 <configOption name="verify_client" default="false">
800                                         <synopsis>Require verification of client certificate (TLS ONLY)</synopsis>
801                                 </configOption>
802                                 <configOption name="verify_server" default="false">
803                                         <synopsis>Require verification of server certificate (TLS ONLY)</synopsis>
804                                 </configOption>
805                                 <configOption name="tos" default="false">
806                                         <synopsis>Enable TOS for the signalling sent over this transport</synopsis>
807                                         <description>
808                                         <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
809                                         for more information on this parameter.</para>
810                                         <note><para>This option does not apply to the <replaceable>ws</replaceable>
811                                         or the <replaceable>wss</replaceable> protocols.</para></note>
812                                         </description>
813                                 </configOption>
814                                 <configOption name="cos" default="false">
815                                         <synopsis>Enable COS for the signalling sent over this transport</synopsis>
816                                         <description>
817                                         <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
818                                         for more information on this parameter.</para>
819                                         <note><para>This option does not apply to the <replaceable>ws</replaceable>
820                                         or the <replaceable>wss</replaceable> protocols.</para></note>
821                                         </description>
822                                 </configOption>
823                         </configObject>
824                         <configObject name="contact">
825                                 <synopsis>A way of creating an aliased name to a SIP URI</synopsis>
826                                 <description><para>
827                                         Contacts are a way to hide SIP URIs from the dialplan directly.
828                                         They are also used to make a group of contactable parties when
829                                         in use with <literal>AoR</literal> lists.
830                                 </para></description>
831                                 <configOption name="type">
832                                         <synopsis>Must be of type 'contact'.</synopsis>
833                                 </configOption>
834                                 <configOption name="uri">
835                                         <synopsis>SIP URI to contact peer</synopsis>
836                                 </configOption>
837                                 <configOption name="expiration_time">
838                                         <synopsis>Time to keep alive a contact</synopsis>
839                                         <description><para>
840                                                 Time to keep alive a contact. String style specification.
841                                         </para></description>
842                                 </configOption>
843                                 <configOption name="qualify_frequency" default="0">
844                                         <synopsis>Interval at which to qualify a contact</synopsis>
845                                         <description><para>
846                                                 Interval between attempts to qualify the contact for reachability.
847                                                 If <literal>0</literal> never qualify. Time in seconds.
848                                         </para></description>
849                                 </configOption>
850                         </configObject>
851                         <configObject name="aor">
852                                 <synopsis>The configuration for a location of an endpoint</synopsis>
853                                 <description><para>
854                                         An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
855                                         AoRs are specified, an endpoint will not be reachable by Asterisk.
856                                         Beyond that, an AoR has other uses within Asterisk, such as inbound
857                                         registration.
858                                         </para><para>
859                                         An <literal>AoR</literal> is a way to allow dialing a group
860                                         of <literal>Contacts</literal> that all use the same
861                                         <literal>endpoint</literal> for calls.
862                                         </para><para>
863                                         This can be used as another way of grouping a list of contacts to dial
864                                         rather than specifing them each directly when dialing via the dialplan.
865                                         This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
866                                         </para><para>
867                                         Registrations: For Asterisk to match an inbound registration to an endpoint,
868                                         the AoR object name must match the user portion of the SIP URI in the "To:"
869                                         header of the inbound SIP registration. That will usually be equivalent
870                                         to the "user name" set in your hard or soft phones configuration.
871                                 </para></description>
872                                 <configOption name="contact">
873                                         <synopsis>Permanent contacts assigned to AoR</synopsis>
874                                         <description><para>
875                                                 Contacts specified will be called whenever referenced
876                                                 by <literal>chan_pjsip</literal>.
877                                                 </para><para>
878                                                 Use a separate "contact=" entry for each contact required. Contacts
879                                                 are specified using a SIP URI.
880                                         </para></description>
881                                 </configOption>
882                                 <configOption name="default_expiration" default="3600">
883                                         <synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis>
884                                 </configOption>
885                                 <configOption name="mailboxes">
886                                         <synopsis>Mailbox(es) to be associated with</synopsis>
887                                         <description><para>This option applies when an external entity subscribes to an AoR
888                                         for message waiting indications. The mailboxes specified will be subscribed to.
889                                         More than one mailbox can be specified with a comma-delimited string.</para></description>
890                                 </configOption>
891                                 <configOption name="maximum_expiration" default="7200">
892                                         <synopsis>Maximum time to keep an AoR</synopsis>
893                                         <description><para>
894                                                 Maximium time to keep a peer with explicit expiration. Time in seconds.
895                                         </para></description>
896                                 </configOption>
897                                 <configOption name="max_contacts" default="0">
898                                         <synopsis>Maximum number of contacts that can bind to an AoR</synopsis>
899                                         <description><para>
900                                                 Maximum number of contacts that can associate with this AoR. This value does
901                                                 not affect the number of contacts that can be added with the "contact" option.
902                                                 It only limits contacts added through external interaction, such as
903                                                 registration.
904                                                 </para>
905                                                 <note><para>This should be set to <literal>1</literal> and
906                                                 <replaceable>remove_existing</replaceable> set to <literal>yes</literal> if you
907                                                 wish to stick with the older <literal>chan_sip</literal> behaviour.
908                                                 </para></note>
909                                         </description>
910                                 </configOption>
911                                 <configOption name="minimum_expiration" default="60">
912                                         <synopsis>Minimum keep alive time for an AoR</synopsis>
913                                         <description><para>
914                                                 Minimum time to keep a peer with an explict expiration. Time in seconds.
915                                         </para></description>
916                                 </configOption>
917                                 <configOption name="remove_existing" default="no">
918                                         <synopsis>Determines whether new contacts replace existing ones.</synopsis>
919                                         <description><para>
920                                                 On receiving a new registration to the AoR should it remove
921                                                 the existing contact that was registered against it?
922                                                 </para>
923                                                 <note><para>This should be set to <literal>yes</literal> and
924                                                 <replaceable>max_contacts</replaceable> set to <literal>1</literal> if you
925                                                 wish to stick with the older <literal>chan_sip</literal> behaviour.
926                                                 </para></note>
927                                         </description>
928                                 </configOption>
929                                 <configOption name="type">
930                                         <synopsis>Must be of type 'aor'.</synopsis>
931                                 </configOption>
932                                 <configOption name="qualify_frequency" default="0">
933                                         <synopsis>Interval at which to qualify an AoR</synopsis>
934                                         <description><para>
935                                                 Interval between attempts to qualify the AoR for reachability.
936                                                 If <literal>0</literal> never qualify. Time in seconds.
937                                         </para></description>
938                                 </configOption>
939                                 <configOption name="authenticate_qualify" default="no">
940                                         <synopsis>Authenticates a qualify request if needed</synopsis>
941                                         <description><para>
942                                                 If true and a qualify request receives a challenge or authenticate response
943                                                 authentication is attempted before declaring the contact available.
944                                         </para></description>
945                                 </configOption>
946                         </configObject>
947                         <configObject name="system">
948                                 <synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis>
949                                 <description><para>
950                                         The settings in this section are global. In addition to being global, the values will
951                                         not be re-evaluated when a reload is performed. This is because the values must be set
952                                         before the SIP stack is initialized. The only way to reset these values is to either
953                                         restart Asterisk, or unload res_pjsip.so and then load it again.
954                                 </para></description>
955                                 <configOption name="timert1" default="500">
956                                         <synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
957                                         <description><para>
958                                                 Timer T1 is the base for determining how long to wait before retransmitting
959                                                 requests that receive no response when using an unreliable transport (e.g. UDP).
960                                                 For more information on this timer, see RFC 3261, Section 17.1.1.1.
961                                         </para></description>
962                                 </configOption>
963                                 <configOption name="timerb" default="32000">
964                                         <synopsis>Set transaction timer B value (milliseconds).</synopsis>
965                                         <description><para>
966                                                 Timer B determines the maximum amount of time to wait after sending an INVITE
967                                                 request before terminating the transaction. It is recommended that this be set
968                                                 to 64 * Timer T1, but it may be set higher if desired. For more information on
969                                                 this timer, see RFC 3261, Section 17.1.1.1.
970                                         </para></description>
971                                 </configOption>
972                                 <configOption name="compactheaders" default="no">
973                                         <synopsis>Use the short forms of common SIP header names.</synopsis>
974                                 </configOption>
975                                 <configOption name="threadpool_initial_size" default="0">
976                                         <synopsis>Initial number of threads in the res_pjsip threadpool.</synopsis>
977                                 </configOption>
978                                 <configOption name="threadpool_auto_increment" default="5">
979                                         <synopsis>The amount by which the number of threads is incremented when necessary.</synopsis>
980                                 </configOption>
981                                 <configOption name="threadpool_idle_timeout" default="60">
982                                         <synopsis>Number of seconds before an idle thread should be disposed of.</synopsis>
983                                 </configOption>
984                                 <configOption name="threadpool_max_size" default="0">
985                                         <synopsis>Maximum number of threads in the res_pjsip threadpool.
986                                         A value of 0 indicates no maximum.</synopsis>
987                                 </configOption>
988                                 <configOption name="type">
989                                         <synopsis>Must be of type 'system'.</synopsis>
990                                 </configOption>
991                         </configObject>
992                         <configObject name="global">
993                                 <synopsis>Options that apply globally to all SIP communications</synopsis>
994                                 <description><para>
995                                         The settings in this section are global. Unlike options in the <literal>system</literal>
996                                         section, these options can be refreshed by performing a reload.
997                                 </para></description>
998                                 <configOption name="maxforwards" default="70">
999                                         <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
1000                                 </configOption>
1001                                 <configOption name="type">
1002                                         <synopsis>Must be of type 'global'.</synopsis>
1003                                 </configOption>
1004                                 <configOption name="useragent" default="Asterisk &lt;Asterisk Version&gt;">
1005                                         <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
1006                                 </configOption>
1007                         </configObject>
1008                 </configFile>
1009         </configInfo>
1010         <manager name="PJSIPQualify" language="en_US">
1011                 <synopsis>
1012                         Qualify a chan_pjsip endpoint.
1013                 </synopsis>
1014                 <syntax>
1015                         <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
1016                         <parameter name="Endpoint" required="true">
1017                                 <para>The endpoint you want to qualify.</para>
1018                         </parameter>
1019                 </syntax>
1020                 <description>
1021                         <para>Qualify a chan_pjsip endpoint.</para>
1022                 </description>
1023         </manager>
1024  ***/
1025
1026
1027 static pjsip_endpoint *ast_pjsip_endpoint;
1028
1029 static struct ast_threadpool *sip_threadpool;
1030
1031 static int register_service(void *data)
1032 {
1033         pjsip_module **module = data;
1034         if (!ast_pjsip_endpoint) {
1035                 ast_log(LOG_ERROR, "There is no PJSIP endpoint. Unable to register services\n");
1036                 return -1;
1037         }
1038         if (pjsip_endpt_register_module(ast_pjsip_endpoint, *module) != PJ_SUCCESS) {
1039                 ast_log(LOG_ERROR, "Unable to register module %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1040                 return -1;
1041         }
1042         ast_debug(1, "Registered SIP service %.*s (%p)\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name), *module);
1043         ast_module_ref(ast_module_info->self);
1044         return 0;
1045 }
1046
1047 int ast_sip_register_service(pjsip_module *module)
1048 {
1049         return ast_sip_push_task_synchronous(NULL, register_service, &module);
1050 }
1051
1052 static int unregister_service(void *data)
1053 {
1054         pjsip_module **module = data;
1055         ast_module_unref(ast_module_info->self);
1056         if (!ast_pjsip_endpoint) {
1057                 return -1;
1058         }
1059         pjsip_endpt_unregister_module(ast_pjsip_endpoint, *module);
1060         ast_debug(1, "Unregistered SIP service %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1061         return 0;
1062 }
1063
1064 void ast_sip_unregister_service(pjsip_module *module)
1065 {
1066         ast_sip_push_task_synchronous(NULL, unregister_service, &module);
1067 }
1068
1069 static struct ast_sip_authenticator *registered_authenticator;
1070
1071 int ast_sip_register_authenticator(struct ast_sip_authenticator *auth)
1072 {
1073         if (registered_authenticator) {
1074                 ast_log(LOG_WARNING, "Authenticator %p is already registered. Cannot register a new one\n", registered_authenticator);
1075                 return -1;
1076         }
1077         registered_authenticator = auth;
1078         ast_debug(1, "Registered SIP authenticator module %p\n", auth);
1079         ast_module_ref(ast_module_info->self);
1080         return 0;
1081 }
1082
1083 void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth)
1084 {
1085         if (registered_authenticator != auth) {
1086                 ast_log(LOG_WARNING, "Trying to unregister authenticator %p but authenticator %p registered\n",
1087                                 auth, registered_authenticator);
1088                 return;
1089         }
1090         registered_authenticator = NULL;
1091         ast_debug(1, "Unregistered SIP authenticator %p\n", auth);
1092         ast_module_unref(ast_module_info->self);
1093 }
1094
1095 int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1096 {
1097         if (!registered_authenticator) {
1098                 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is not required\n");
1099                 return 0;
1100         }
1101
1102         return registered_authenticator->requires_authentication(endpoint, rdata);
1103 }
1104
1105 enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint,
1106                 pjsip_rx_data *rdata, pjsip_tx_data *tdata)
1107 {
1108         if (!registered_authenticator) {
1109                 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is successful\n");
1110                 return 0;
1111         }
1112         return registered_authenticator->check_authentication(endpoint, rdata, tdata);
1113 }
1114
1115 static struct ast_sip_outbound_authenticator *registered_outbound_authenticator;
1116
1117 int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1118 {
1119         if (registered_outbound_authenticator) {
1120                 ast_log(LOG_WARNING, "Outbound authenticator %p is already registered. Cannot register a new one\n", registered_outbound_authenticator);
1121                 return -1;
1122         }
1123         registered_outbound_authenticator = auth;
1124         ast_debug(1, "Registered SIP outbound authenticator module %p\n", auth);
1125         ast_module_ref(ast_module_info->self);
1126         return 0;
1127 }
1128
1129 void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1130 {
1131         if (registered_outbound_authenticator != auth) {
1132                 ast_log(LOG_WARNING, "Trying to unregister outbound authenticator %p but outbound authenticator %p registered\n",
1133                                 auth, registered_outbound_authenticator);
1134                 return;
1135         }
1136         registered_outbound_authenticator = NULL;
1137         ast_debug(1, "Unregistered SIP outbound authenticator %p\n", auth);
1138         ast_module_unref(ast_module_info->self);
1139 }
1140
1141 int ast_sip_create_request_with_auth(const struct ast_sip_auth_array *auths, pjsip_rx_data *challenge,
1142                 pjsip_transaction *tsx, pjsip_tx_data **new_request)
1143 {
1144         if (!registered_outbound_authenticator) {
1145                 ast_log(LOG_WARNING, "No SIP outbound authenticator registered. Cannot respond to authentication challenge\n");
1146                 return -1;
1147         }
1148         return registered_outbound_authenticator->create_request_with_auth(auths, challenge, tsx, new_request);
1149 }
1150
1151 struct endpoint_identifier_list {
1152         struct ast_sip_endpoint_identifier *identifier;
1153         AST_RWLIST_ENTRY(endpoint_identifier_list) list;
1154 };
1155
1156 static AST_RWLIST_HEAD_STATIC(endpoint_identifiers, endpoint_identifier_list);
1157
1158 int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1159 {
1160         struct endpoint_identifier_list *id_list_item;
1161         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1162
1163         id_list_item = ast_calloc(1, sizeof(*id_list_item));
1164         if (!id_list_item) {
1165                 ast_log(LOG_ERROR, "Unabled to add endpoint identifier. Out of memory.\n");
1166                 return -1;
1167         }
1168         id_list_item->identifier = identifier;
1169
1170         AST_RWLIST_INSERT_TAIL(&endpoint_identifiers, id_list_item, list);
1171         ast_debug(1, "Registered endpoint identifier %p\n", identifier);
1172
1173         ast_module_ref(ast_module_info->self);
1174         return 0;
1175 }
1176
1177 void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1178 {
1179         struct endpoint_identifier_list *iter;
1180         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1181         AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_identifiers, iter, list) {
1182                 if (iter->identifier == identifier) {
1183                         AST_RWLIST_REMOVE_CURRENT(list);
1184                         ast_free(iter);
1185                         ast_debug(1, "Unregistered endpoint identifier %p\n", identifier);
1186                         ast_module_unref(ast_module_info->self);
1187                         break;
1188                 }
1189         }
1190         AST_RWLIST_TRAVERSE_SAFE_END;
1191 }
1192
1193 struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata)
1194 {
1195         struct endpoint_identifier_list *iter;
1196         struct ast_sip_endpoint *endpoint = NULL;
1197         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1198         AST_RWLIST_TRAVERSE(&endpoint_identifiers, iter, list) {
1199                 ast_assert(iter->identifier->identify_endpoint != NULL);
1200                 endpoint = iter->identifier->identify_endpoint(rdata);
1201                 if (endpoint) {
1202                         break;
1203                 }
1204         }
1205         return endpoint;
1206 }
1207
1208 pjsip_endpoint *ast_sip_get_pjsip_endpoint(void)
1209 {
1210         return ast_pjsip_endpoint;
1211 }
1212
1213 static int sip_dialog_create_from(pj_pool_t *pool, pj_str_t *from, const char *user, const char *domain, const pj_str_t *target, pjsip_tpselector *selector)
1214 {
1215         pj_str_t tmp, local_addr;
1216         pjsip_uri *uri;
1217         pjsip_sip_uri *sip_uri;
1218         pjsip_transport_type_e type = PJSIP_TRANSPORT_UNSPECIFIED;
1219         int local_port;
1220         char uuid_str[AST_UUID_STR_LEN];
1221
1222         if (ast_strlen_zero(user)) {
1223                 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1224                 if (!uuid) {
1225                         return -1;
1226                 }
1227                 user = ast_uuid_to_str(uuid, uuid_str, sizeof(uuid_str));
1228         }
1229
1230         /* Parse the provided target URI so we can determine what transport it will end up using */
1231         pj_strdup_with_null(pool, &tmp, target);
1232
1233         if (!(uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0)) ||
1234             (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
1235                 return -1;
1236         }
1237
1238         sip_uri = pjsip_uri_get_uri(uri);
1239
1240         /* Determine the transport type to use */
1241         if (PJSIP_URI_SCHEME_IS_SIPS(sip_uri)) {
1242                 type = PJSIP_TRANSPORT_TLS;
1243         } else if (!sip_uri->transport_param.slen) {
1244                 type = PJSIP_TRANSPORT_UDP;
1245         } else {
1246                 type = pjsip_transport_get_type_from_name(&sip_uri->transport_param);
1247         }
1248
1249         if (type == PJSIP_TRANSPORT_UNSPECIFIED) {
1250                 return -1;
1251         }
1252
1253         /* If the host is IPv6 turn the transport into an IPv6 version */
1254         if (pj_strchr(&sip_uri->host, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1255                 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1256         }
1257
1258         if (!ast_strlen_zero(domain)) {
1259                 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1260                 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1261                                 "<%s:%s@%s%s%s>",
1262                                 (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1263                                 user,
1264                                 domain,
1265                                 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1266                                 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1267                 return 0;
1268         }
1269
1270         /* Get the local bound address for the transport that will be used when communicating with the provided URI */
1271         if (pjsip_tpmgr_find_local_addr(pjsip_endpt_get_tpmgr(ast_sip_get_pjsip_endpoint()), pool, type, selector,
1272                                                               &local_addr, &local_port) != PJ_SUCCESS) {
1273                 return -1;
1274         }
1275
1276         /* If IPv6 was specified in the transport, set the proper type */
1277         if (pj_strchr(&local_addr, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1278                 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1279         }
1280
1281         from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1282         from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1283                                       "<%s:%s@%s%.*s%s:%d%s%s>",
1284                                       (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1285                                       user,
1286                                       (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1287                                       (int)local_addr.slen,
1288                                       local_addr.ptr,
1289                                       (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1290                                       local_port,
1291                                       (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1292                                       (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1293
1294         return 0;
1295 }
1296
1297 static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpoint, pjsip_tpselector *selector)
1298 {
1299         RAII_VAR(struct ast_sip_transport *, transport, NULL, ao2_cleanup);
1300         const char *transport_name = endpoint->transport;
1301
1302         if (ast_strlen_zero(transport_name)) {
1303                 return 0;
1304         }
1305
1306         transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_name);
1307
1308         if (!transport || !transport->state) {
1309                 ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport '%s' for endpoint '%s'\n",
1310                         transport_name, ast_sorcery_object_get_id(endpoint));
1311                 return -1;
1312         }
1313
1314         if (transport->state->transport) {
1315                 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1316                 selector->u.transport = transport->state->transport;
1317         } else if (transport->state->factory) {
1318                 selector->type = PJSIP_TPSELECTOR_LISTENER;
1319                 selector->u.listener = transport->state->factory;
1320         } else {
1321                 return -1;
1322         }
1323
1324         return 0;
1325 }
1326
1327 static int sip_get_tpselector_from_uri(const char *uri, pjsip_tpselector *selector)
1328 {
1329         RAII_VAR(struct ast_sip_contact_transport *, contact_transport, NULL, ao2_cleanup);
1330
1331         contact_transport = ast_sip_location_retrieve_contact_transport_by_uri(uri);
1332
1333         if (!contact_transport) {
1334                 return -1;
1335         }
1336
1337         selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1338         selector->u.transport = contact_transport->transport;
1339
1340         return 0;
1341 }
1342
1343 pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user)
1344 {
1345         pj_str_t local_uri = { "sip:temp@temp", 13 }, remote_uri;
1346         pjsip_dialog *dlg = NULL;
1347         const char *outbound_proxy = endpoint->outbound_proxy;
1348         pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1349         static const pj_str_t HCONTACT = { "Contact", 7 };
1350
1351         pj_cstr(&remote_uri, uri);
1352
1353         if (pjsip_dlg_create_uac(pjsip_ua_instance(), &local_uri, NULL, &remote_uri, NULL, &dlg) != PJ_SUCCESS) {
1354                 return NULL;
1355         }
1356
1357         if (sip_get_tpselector_from_uri(uri, &selector) && sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1358                 pjsip_dlg_terminate(dlg);
1359                 return NULL;
1360         }
1361
1362         if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
1363                 pjsip_dlg_terminate(dlg);
1364                 return NULL;
1365         }
1366
1367         /* Update the dialog with the new local URI, we do it afterwards so we can use the dialog pool for construction */
1368         pj_strdup_with_null(dlg->pool, &dlg->local.info_str, &local_uri);
1369         dlg->local.info->uri = pjsip_parse_uri(dlg->pool, dlg->local.info_str.ptr, dlg->local.info_str.slen, 0);
1370         dlg->local.contact = pjsip_parse_hdr(dlg->pool, &HCONTACT, local_uri.ptr, local_uri.slen, NULL);
1371
1372         /* If a request user has been specified and we are permitted to change it, do so */
1373         if (!ast_strlen_zero(request_user) && (PJSIP_URI_SCHEME_IS_SIP(dlg->target) || PJSIP_URI_SCHEME_IS_SIPS(dlg->target))) {
1374                 pjsip_sip_uri *target = pjsip_uri_get_uri(dlg->target);
1375                 pj_strdup2(dlg->pool, &target->user, request_user);
1376         }
1377
1378         /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
1379         dlg->sess_count++;
1380
1381         pjsip_dlg_set_transport(dlg, &selector);
1382
1383         if (!ast_strlen_zero(outbound_proxy)) {
1384                 pjsip_route_hdr route_set, *route;
1385                 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1386                 pj_str_t tmp;
1387
1388                 pj_list_init(&route_set);
1389
1390                 pj_strdup2_with_null(dlg->pool, &tmp, outbound_proxy);
1391                 if (!(route = pjsip_parse_hdr(dlg->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1392                         pjsip_dlg_terminate(dlg);
1393                         return NULL;
1394                 }
1395                 pj_list_push_back(&route_set, route);
1396
1397                 pjsip_dlg_set_route_set(dlg, &route_set);
1398         }
1399
1400         dlg->sess_count--;
1401
1402         return dlg;
1403 }
1404
1405 pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1406 {
1407         pjsip_dialog *dlg;
1408         pj_str_t contact;
1409         pjsip_transport_type_e type = rdata->tp_info.transport->key.type;
1410         pj_status_t status;
1411
1412         contact.ptr = pj_pool_alloc(rdata->tp_info.pool, PJSIP_MAX_URL_SIZE);
1413         contact.slen = pj_ansi_snprintf(contact.ptr, PJSIP_MAX_URL_SIZE,
1414                         "<%s:%s%.*s%s:%d%s%s>",
1415                         (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1416                         (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1417                         (int)rdata->tp_info.transport->local_name.host.slen,
1418                         rdata->tp_info.transport->local_name.host.ptr,
1419                         (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1420                         rdata->tp_info.transport->local_name.port,
1421                         (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1422                         (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1423
1424         status = pjsip_dlg_create_uas(pjsip_ua_instance(), rdata, &contact, &dlg);
1425         if (status != PJ_SUCCESS) {
1426                 char err[PJ_ERR_MSG_SIZE];
1427
1428                 pjsip_strerror(status, err, sizeof(err));
1429                 ast_log(LOG_ERROR, "Could not create dialog with endpoint %s. %s\n",
1430                                 ast_sorcery_object_get_id(endpoint), err);
1431                 return NULL;
1432         }
1433
1434         return dlg;
1435 }
1436
1437 /* PJSIP doesn't know about the INFO method, so we have to define it ourselves */
1438 static const pjsip_method info_method = {PJSIP_OTHER_METHOD, {"INFO", 4} };
1439 static const pjsip_method message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
1440
1441 static struct {
1442         const char *method;
1443         const pjsip_method *pmethod;
1444 } methods [] = {
1445         { "INVITE", &pjsip_invite_method },
1446         { "CANCEL", &pjsip_cancel_method },
1447         { "ACK", &pjsip_ack_method },
1448         { "BYE", &pjsip_bye_method },
1449         { "REGISTER", &pjsip_register_method },
1450         { "OPTIONS", &pjsip_options_method },
1451         { "SUBSCRIBE", &pjsip_subscribe_method },
1452         { "NOTIFY", &pjsip_notify_method },
1453         { "PUBLISH", &pjsip_publish_method },
1454         { "INFO", &info_method },
1455         { "MESSAGE", &message_method },
1456 };
1457
1458 static const pjsip_method *get_pjsip_method(const char *method)
1459 {
1460         int i;
1461         for (i = 0; i < ARRAY_LEN(methods); ++i) {
1462                 if (!strcmp(method, methods[i].method)) {
1463                         return methods[i].pmethod;
1464                 }
1465         }
1466         return NULL;
1467 }
1468
1469 static int create_in_dialog_request(const pjsip_method *method, struct pjsip_dialog *dlg, pjsip_tx_data **tdata)
1470 {
1471         if (pjsip_dlg_create_request(dlg, method, -1, tdata) != PJ_SUCCESS) {
1472                 ast_log(LOG_WARNING, "Unable to create in-dialog request.\n");
1473                 return -1;
1474         }
1475
1476         return 0;
1477 }
1478
1479 static int create_out_of_dialog_request(const pjsip_method *method, struct ast_sip_endpoint *endpoint,
1480                 const char *uri, pjsip_tx_data **tdata)
1481 {
1482         RAII_VAR(struct ast_sip_contact *, contact, NULL, ao2_cleanup);
1483         pj_str_t remote_uri;
1484         pj_str_t from;
1485         pj_pool_t *pool;
1486         pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1487
1488         if (ast_strlen_zero(uri)) {
1489                 if (!endpoint) {
1490                         ast_log(LOG_ERROR, "An endpoint and/or uri must be specified\n");
1491                         return -1;
1492                 }
1493
1494                 contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
1495                 if (!contact || ast_strlen_zero(contact->uri)) {
1496                         ast_log(LOG_ERROR, "Unable to retrieve contact for endpoint %s\n",
1497                                         ast_sorcery_object_get_id(endpoint));
1498                         return -1;
1499                 }
1500
1501                 pj_cstr(&remote_uri, contact->uri);
1502         } else {
1503                 pj_cstr(&remote_uri, uri);
1504         }
1505
1506         if (endpoint) {
1507                 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1508                         ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport selector for endpoint %s\n",
1509                                 ast_sorcery_object_get_id(endpoint));
1510                         return -1;
1511                 }
1512         }
1513
1514         pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "Outbound request", 256, 256);
1515
1516         if (!pool) {
1517                 ast_log(LOG_ERROR, "Unable to create PJLIB memory pool\n");
1518                 return -1;
1519         }
1520
1521         if (sip_dialog_create_from(pool, &from, endpoint ? endpoint->fromuser : NULL,
1522                                 endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) {
1523                 ast_log(LOG_ERROR, "Unable to create From header for %.*s request to endpoint %s\n",
1524                                 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1525                 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1526                 return -1;
1527         }
1528
1529         if (pjsip_endpt_create_request(ast_sip_get_pjsip_endpoint(), method, &remote_uri,
1530                         &from, &remote_uri, &from, NULL, -1, NULL, tdata) != PJ_SUCCESS) {
1531                 ast_log(LOG_ERROR, "Unable to create outbound %.*s request to endpoint %s\n",
1532                                 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1533                 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1534                 return -1;
1535         }
1536
1537         /* We can release this pool since request creation copied all the necessary
1538          * data into the outbound request's pool
1539          */
1540         pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1541         return 0;
1542 }
1543
1544 int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
1545                 struct ast_sip_endpoint *endpoint, const char *uri,
1546                 pjsip_tx_data **tdata)
1547 {
1548         const pjsip_method *pmethod = get_pjsip_method(method);
1549
1550         if (!pmethod) {
1551                 ast_log(LOG_WARNING, "Unknown method '%s'. Cannot send request\n", method);
1552                 return -1;
1553         }
1554
1555         if (dlg) {
1556                 return create_in_dialog_request(pmethod, dlg, tdata);
1557         } else {
1558                 return create_out_of_dialog_request(pmethod, endpoint, uri, tdata);
1559         }
1560 }
1561
1562 static int send_in_dialog_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg)
1563 {
1564         if (pjsip_dlg_send_request(dlg, tdata, -1, NULL) != PJ_SUCCESS) {
1565                 ast_log(LOG_WARNING, "Unable to send in-dialog request.\n");
1566                 return -1;
1567         }
1568         return 0;
1569 }
1570
1571 static void send_request_cb(void *token, pjsip_event *e)
1572 {
1573         RAII_VAR(struct ast_sip_endpoint *, endpoint, token, ao2_cleanup);
1574         pjsip_transaction *tsx = e->body.tsx_state.tsx;
1575         pjsip_rx_data *challenge = e->body.tsx_state.src.rdata;
1576         pjsip_tx_data *tdata;
1577
1578         if (tsx->status_code != 401 && tsx->status_code != 407) {
1579                 return;
1580         }
1581
1582         if (!ast_sip_create_request_with_auth(&endpoint->outbound_auths, challenge, tsx, &tdata)) {
1583                 pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, NULL, NULL);
1584         }
1585 }
1586
1587 static int send_out_of_dialog_request(pjsip_tx_data *tdata, struct ast_sip_endpoint *endpoint)
1588 {
1589         ao2_ref(endpoint, +1);
1590         if (pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, endpoint, send_request_cb) != PJ_SUCCESS) {
1591                 ast_log(LOG_ERROR, "Error attempting to send outbound %.*s request to endpoint %s\n",
1592                                 (int) pj_strlen(&tdata->msg->line.req.method.name),
1593                                 pj_strbuf(&tdata->msg->line.req.method.name),
1594                                 ast_sorcery_object_get_id(endpoint));
1595                 ao2_ref(endpoint, -1);
1596                 return -1;
1597         }
1598
1599         return 0;
1600 }
1601
1602 int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg, struct ast_sip_endpoint *endpoint)
1603 {
1604         ast_assert(tdata->msg->type == PJSIP_REQUEST_MSG);
1605
1606         if (dlg) {
1607                 return send_in_dialog_request(tdata, dlg);
1608         } else {
1609                 return send_out_of_dialog_request(tdata, endpoint);
1610         }
1611 }
1612
1613 int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value)
1614 {
1615         pj_str_t hdr_name;
1616         pj_str_t hdr_value;
1617         pjsip_generic_string_hdr *hdr;
1618
1619         pj_cstr(&hdr_name, name);
1620         pj_cstr(&hdr_value, value);
1621
1622         hdr = pjsip_generic_string_hdr_create(tdata->pool, &hdr_name, &hdr_value);
1623
1624         pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *) hdr);
1625         return 0;
1626 }
1627
1628 static pjsip_msg_body *ast_body_to_pjsip_body(pj_pool_t *pool, const struct ast_sip_body *body)
1629 {
1630         pj_str_t type;
1631         pj_str_t subtype;
1632         pj_str_t body_text;
1633
1634         pj_cstr(&type, body->type);
1635         pj_cstr(&subtype, body->subtype);
1636         pj_cstr(&body_text, body->body_text);
1637
1638         return pjsip_msg_body_create(pool, &type, &subtype, &body_text);
1639 }
1640
1641 int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body)
1642 {
1643         pjsip_msg_body *pjsip_body = ast_body_to_pjsip_body(tdata->pool, body);
1644         tdata->msg->body = pjsip_body;
1645         return 0;
1646 }
1647
1648 int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies)
1649 {
1650         int i;
1651         /* NULL for type and subtype automatically creates "multipart/mixed" */
1652         pjsip_msg_body *body = pjsip_multipart_create(tdata->pool, NULL, NULL);
1653
1654         for (i = 0; i < num_bodies; ++i) {
1655                 pjsip_multipart_part *part = pjsip_multipart_create_part(tdata->pool);
1656                 part->body = ast_body_to_pjsip_body(tdata->pool, bodies[i]);
1657                 pjsip_multipart_add_part(tdata->pool, body, part);
1658         }
1659
1660         tdata->msg->body = body;
1661         return 0;
1662 }
1663
1664 int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text)
1665 {
1666         size_t combined_size = strlen(body_text) + tdata->msg->body->len;
1667         struct ast_str *body_buffer = ast_str_alloca(combined_size);
1668
1669         ast_str_set(&body_buffer, 0, "%.*s%s", (int) tdata->msg->body->len, (char *) tdata->msg->body->data, body_text);
1670
1671         tdata->msg->body->data = pj_pool_alloc(tdata->pool, combined_size);
1672         pj_memcpy(tdata->msg->body->data, ast_str_buffer(body_buffer), combined_size);
1673         tdata->msg->body->len = combined_size;
1674
1675         return 0;
1676 }
1677
1678 struct ast_taskprocessor *ast_sip_create_serializer(void)
1679 {
1680         struct ast_taskprocessor *serializer;
1681         RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1682         char name[AST_UUID_STR_LEN];
1683
1684         if (!uuid) {
1685                 return NULL;
1686         }
1687
1688         ast_uuid_to_str(uuid, name, sizeof(name));
1689
1690         serializer = ast_threadpool_serializer(name, sip_threadpool);
1691         if (!serializer) {
1692                 return NULL;
1693         }
1694         return serializer;
1695 }
1696
1697 int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1698 {
1699         if (serializer) {
1700                 return ast_taskprocessor_push(serializer, sip_task, task_data);
1701         } else {
1702                 return ast_threadpool_push(sip_threadpool, sip_task, task_data);
1703         }
1704 }
1705
1706 struct sync_task_data {
1707         ast_mutex_t lock;
1708         ast_cond_t cond;
1709         int complete;
1710         int fail;
1711         int (*task)(void *);
1712         void *task_data;
1713 };
1714
1715 static int sync_task(void *data)
1716 {
1717         struct sync_task_data *std = data;
1718         std->fail = std->task(std->task_data);
1719
1720         ast_mutex_lock(&std->lock);
1721         std->complete = 1;
1722         ast_cond_signal(&std->cond);
1723         ast_mutex_unlock(&std->lock);
1724         return std->fail;
1725 }
1726
1727 int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1728 {
1729         /* This method is an onion */
1730         struct sync_task_data std;
1731         ast_mutex_init(&std.lock);
1732         ast_cond_init(&std.cond, NULL);
1733         std.fail = std.complete = 0;
1734         std.task = sip_task;
1735         std.task_data = task_data;
1736
1737         if (serializer) {
1738                 if (ast_taskprocessor_push(serializer, sync_task, &std)) {
1739                         return -1;
1740                 }
1741         } else {
1742                 if (ast_threadpool_push(sip_threadpool, sync_task, &std)) {
1743                         return -1;
1744                 }
1745         }
1746
1747         ast_mutex_lock(&std.lock);
1748         while (!std.complete) {
1749                 ast_cond_wait(&std.cond, &std.lock);
1750         }
1751         ast_mutex_unlock(&std.lock);
1752
1753         ast_mutex_destroy(&std.lock);
1754         ast_cond_destroy(&std.cond);
1755         return std.fail;
1756 }
1757
1758 void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size)
1759 {
1760         size_t chars_to_copy = MIN(size - 1, pj_strlen(src));
1761         memcpy(dest, pj_strbuf(src), chars_to_copy);
1762         dest[chars_to_copy] = '\0';
1763 }
1764
1765 int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype)
1766 {
1767         pjsip_media_type compare;
1768
1769         if (!content_type) {
1770                 return 0;
1771         }
1772
1773         pjsip_media_type_init2(&compare, type, subtype);
1774
1775         return pjsip_media_type_cmp(content_type, &compare, 0) ? 0 : -1;
1776 }
1777
1778 pj_caching_pool caching_pool;
1779 pj_pool_t *memory_pool;
1780 pj_thread_t *monitor_thread;
1781 static int monitor_continue;
1782
1783 static void *monitor_thread_exec(void *endpt)
1784 {
1785         while (monitor_continue) {
1786                 const pj_time_val delay = {0, 10};
1787                 pjsip_endpt_handle_events(ast_pjsip_endpoint, &delay);
1788         }
1789         return NULL;
1790 }
1791
1792 static void stop_monitor_thread(void)
1793 {
1794         monitor_continue = 0;
1795         pj_thread_join(monitor_thread);
1796 }
1797
1798 AST_THREADSTORAGE(pj_thread_storage);
1799 AST_THREADSTORAGE(servant_id_storage);
1800 #define SIP_SERVANT_ID 0x5E2F1D
1801
1802 static void sip_thread_start(void)
1803 {
1804         pj_thread_desc *desc;
1805         pj_thread_t *thread;
1806         uint32_t *servant_id;
1807
1808         servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1809         if (!servant_id) {
1810                 ast_log(LOG_ERROR, "Could not set SIP servant ID in thread-local storage.\n");
1811                 return;
1812         }
1813         *servant_id = SIP_SERVANT_ID;
1814
1815         desc = ast_threadstorage_get(&pj_thread_storage, sizeof(pj_thread_desc));
1816         if (!desc) {
1817                 ast_log(LOG_ERROR, "Could not get thread desc from thread-local storage. Expect awful things to occur\n");
1818                 return;
1819         }
1820         pj_bzero(*desc, sizeof(*desc));
1821
1822         if (pj_thread_register("Asterisk Thread", *desc, &thread) != PJ_SUCCESS) {
1823                 ast_log(LOG_ERROR, "Couldn't register thread with PJLIB.\n");
1824         }
1825 }
1826
1827 int ast_sip_thread_is_servant(void)
1828 {
1829         uint32_t *servant_id;
1830
1831         servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1832         if (!servant_id) {
1833                 return 0;
1834         }
1835
1836         return *servant_id == SIP_SERVANT_ID;
1837 }
1838
1839 static void remove_request_headers(pjsip_endpoint *endpt)
1840 {
1841         const pjsip_hdr *request_headers = pjsip_endpt_get_request_headers(endpt);
1842         pjsip_hdr *iter = request_headers->next;
1843
1844         while (iter != request_headers) {
1845                 pjsip_hdr *to_erase = iter;
1846                 iter = iter->next;
1847                 pj_list_erase(to_erase);
1848         }
1849 }
1850
1851 static int load_module(void)
1852 {
1853         /* The third parameter is just copied from
1854          * example code from PJLIB. This can be adjusted
1855          * if necessary.
1856          */
1857         pj_status_t status;
1858         struct ast_threadpool_options options;
1859
1860         if (pj_init() != PJ_SUCCESS) {
1861                 return AST_MODULE_LOAD_DECLINE;
1862         }
1863
1864         if (pjlib_util_init() != PJ_SUCCESS) {
1865                 pj_shutdown();
1866                 return AST_MODULE_LOAD_DECLINE;
1867         }
1868
1869         pj_caching_pool_init(&caching_pool, NULL, 1024 * 1024);
1870         if (pjsip_endpt_create(&caching_pool.factory, "SIP", &ast_pjsip_endpoint) != PJ_SUCCESS) {
1871                 ast_log(LOG_ERROR, "Failed to create PJSIP endpoint structure. Aborting load\n");
1872                 pj_caching_pool_destroy(&caching_pool);
1873                 return AST_MODULE_LOAD_DECLINE;
1874         }
1875
1876         /* PJSIP will automatically try to add a Max-Forwards header. Since we want to control that,
1877          * we need to stop PJSIP from doing it automatically
1878          */
1879         remove_request_headers(ast_pjsip_endpoint);
1880
1881         memory_pool = pj_pool_create(&caching_pool.factory, "SIP", 1024, 1024, NULL);
1882         if (!memory_pool) {
1883                 ast_log(LOG_ERROR, "Failed to create memory pool for SIP. Aborting load\n");
1884                 pjsip_endpt_destroy(ast_pjsip_endpoint);
1885                 ast_pjsip_endpoint = NULL;
1886                 pj_caching_pool_destroy(&caching_pool);
1887                 return AST_MODULE_LOAD_DECLINE;
1888         }
1889
1890         if (ast_sip_initialize_system()) {
1891                 ast_log(LOG_ERROR, "Failed to initialize SIP system configuration. Aborting load\n");
1892                 pj_pool_release(memory_pool);
1893                 memory_pool = NULL;
1894                 pjsip_endpt_destroy(ast_pjsip_endpoint);
1895                 ast_pjsip_endpoint = NULL;
1896                 pj_caching_pool_destroy(&caching_pool);
1897                 return AST_MODULE_LOAD_DECLINE;
1898         }
1899
1900         sip_get_threadpool_options(&options);
1901         options.thread_start = sip_thread_start;
1902         sip_threadpool = ast_threadpool_create("SIP", NULL, &options);
1903         if (!sip_threadpool) {
1904                 ast_log(LOG_ERROR, "Failed to create SIP threadpool. Aborting load\n");
1905                 pj_pool_release(memory_pool);
1906                 memory_pool = NULL;
1907                 pjsip_endpt_destroy(ast_pjsip_endpoint);
1908                 ast_pjsip_endpoint = NULL;
1909                 pj_caching_pool_destroy(&caching_pool);
1910                 return AST_MODULE_LOAD_DECLINE;
1911         }
1912
1913         pjsip_tsx_layer_init_module(ast_pjsip_endpoint);
1914         pjsip_ua_init_module(ast_pjsip_endpoint, NULL);
1915
1916         monitor_continue = 1;
1917         status = pj_thread_create(memory_pool, "SIP", (pj_thread_proc *) &monitor_thread_exec,
1918                         NULL, PJ_THREAD_DEFAULT_STACK_SIZE * 2, 0, &monitor_thread);
1919         if (status != PJ_SUCCESS) {
1920                 ast_log(LOG_ERROR, "Failed to start SIP monitor thread. Aborting load\n");
1921                 pj_pool_release(memory_pool);
1922                 memory_pool = NULL;
1923                 pjsip_endpt_destroy(ast_pjsip_endpoint);
1924                 ast_pjsip_endpoint = NULL;
1925                 pj_caching_pool_destroy(&caching_pool);
1926                 return AST_MODULE_LOAD_DECLINE;
1927         }
1928
1929         ast_sip_initialize_global_headers();
1930
1931         if (ast_res_pjsip_initialize_configuration()) {
1932                 ast_log(LOG_ERROR, "Failed to initialize SIP configuration. Aborting load\n");
1933                 ast_sip_destroy_global_headers();
1934                 stop_monitor_thread();
1935                 pj_pool_release(memory_pool);
1936                 memory_pool = NULL;
1937                 pjsip_endpt_destroy(ast_pjsip_endpoint);
1938                 ast_pjsip_endpoint = NULL;
1939                 pj_caching_pool_destroy(&caching_pool);
1940                 return AST_MODULE_LOAD_DECLINE;
1941         }
1942
1943         if (ast_sip_initialize_distributor()) {
1944                 ast_log(LOG_ERROR, "Failed to register distributor module. Aborting load\n");
1945                 ast_res_pjsip_destroy_configuration();
1946                 ast_sip_destroy_global_headers();
1947                 stop_monitor_thread();
1948                 pj_pool_release(memory_pool);
1949                 memory_pool = NULL;
1950                 pjsip_endpt_destroy(ast_pjsip_endpoint);
1951                 ast_pjsip_endpoint = NULL;
1952                 pj_caching_pool_destroy(&caching_pool);
1953                 return AST_MODULE_LOAD_DECLINE;
1954         }
1955
1956         if (ast_sip_initialize_outbound_authentication()) {
1957                 ast_log(LOG_ERROR, "Failed to initialize outbound authentication. Aborting load\n");
1958                 ast_sip_destroy_distributor();
1959                 ast_res_pjsip_destroy_configuration();
1960                 ast_sip_destroy_global_headers();
1961                 stop_monitor_thread();
1962                 pj_pool_release(memory_pool);
1963                 memory_pool = NULL;
1964                 pjsip_endpt_destroy(ast_pjsip_endpoint);
1965                 ast_pjsip_endpoint = NULL;
1966                 pj_caching_pool_destroy(&caching_pool);
1967                 return AST_MODULE_LOAD_DECLINE;
1968         }
1969
1970         ast_res_pjsip_init_options_handling(0);
1971
1972         ast_res_pjsip_init_contact_transports();
1973
1974         ast_module_ref(ast_module_info->self);
1975
1976         return AST_MODULE_LOAD_SUCCESS;
1977 }
1978
1979 static int reload_module(void)
1980 {
1981         if (ast_res_pjsip_reload_configuration()) {
1982                 return AST_MODULE_LOAD_DECLINE;
1983         }
1984         ast_res_pjsip_init_options_handling(1);
1985         return 0;
1986 }
1987
1988 static int unload_module(void)
1989 {
1990         /* This will never get called as this module can't be unloaded */
1991         return 0;
1992 }
1993
1994 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Basic SIP resource",
1995                 .load = load_module,
1996                 .unload = unload_module,
1997                 .reload = reload_module,
1998                 .load_pri = AST_MODPRI_CHANNEL_DEPEND - 5,
1999 );