645dcb54a0a5bfae2fdeb7695c062a5f1e2e0156
[asterisk/asterisk.git] / res / res_pjsip.c
1 /*
2  * Asterisk -- An open source telephony toolkit.
3  *
4  * Copyright (C) 2013, Digium, Inc.
5  *
6  * Mark Michelson <mmichelson@digium.com>
7  *
8  * See http://www.asterisk.org for more information about
9  * the Asterisk project. Please do not directly contact
10  * any of the maintainers of this project for assistance;
11  * the project provides a web site, mailing lists and IRC
12  * channels for your use.
13  *
14  * This program is free software, distributed under the terms of
15  * the GNU General Public License Version 2. See the LICENSE file
16  * at the top of the source tree.
17  */
18
19 #include "asterisk.h"
20
21 #include <pjsip.h>
22 /* Needed for SUBSCRIBE, NOTIFY, and PUBLISH method definitions */
23 #include <pjsip_simple.h>
24 #include <pjlib.h>
25
26 #include "asterisk/res_pjsip.h"
27 #include "res_pjsip/include/res_pjsip_private.h"
28 #include "asterisk/linkedlists.h"
29 #include "asterisk/logger.h"
30 #include "asterisk/lock.h"
31 #include "asterisk/utils.h"
32 #include "asterisk/astobj2.h"
33 #include "asterisk/module.h"
34 #include "asterisk/threadpool.h"
35 #include "asterisk/taskprocessor.h"
36 #include "asterisk/uuid.h"
37 #include "asterisk/sorcery.h"
38
39 /*** MODULEINFO
40         <depend>pjproject</depend>
41         <depend>res_sorcery_config</depend>
42         <support_level>core</support_level>
43  ***/
44
45 /*** DOCUMENTATION
46         <configInfo name="res_pjsip" language="en_US">
47                 <synopsis>SIP Resource using PJProject</synopsis>
48                 <configFile name="pjsip.conf">
49                         <configObject name="endpoint">
50                                 <synopsis>Endpoint</synopsis>
51                                 <description><para>
52                                         The <emphasis>Endpoint</emphasis> is the primary configuration object.
53                                         It contains the core SIP related options only, endpoints are <emphasis>NOT</emphasis>
54                                         dialable entries of their own. Communication with another SIP device is
55                                         accomplished via Addresses of Record (AoRs) which have one or more
56                                         contacts assicated with them. Endpoints <emphasis>NOT</emphasis> configured to
57                                         use a <literal>transport</literal> will default to first transport found
58                                         in <filename>pjsip.conf</filename> that matches its type.
59                                         </para>
60                                         <para>Example: An Endpoint has been configured with no transport.
61                                         When it comes time to call an AoR, PJSIP will find the
62                                         first transport that matches the type. A SIP URI of <literal>sip:5000@[11::33]</literal>
63                                         will use the first IPv6 transport and try to send the request.
64                                         </para>
65                                         <para>If the anonymous endpoint identifier is in use an endpoint with the name
66                                         "anonymous@domain" will be searched for as a last resort. If this is not found
67                                         it will fall back to searching for "anonymous". If neither endpoints are found
68                                         the anonymous endpoint identifier will not return an endpoint and anonymous
69                                         calling will not be possible.
70                                         </para>
71                                 </description>
72                                 <configOption name="100rel" default="yes">
73                                         <synopsis>Allow support for RFC3262 provisional ACK tags</synopsis>
74                                         <description>
75                                                 <enumlist>
76                                                         <enum name="no" />
77                                                         <enum name="required" />
78                                                         <enum name="yes" />
79                                                 </enumlist>
80                                         </description>
81                                 </configOption>
82                                 <configOption name="aggregate_mwi" default="yes">
83                                         <synopsis></synopsis>
84                                         <description><para>When enabled, <replaceable>aggregate_mwi</replaceable> condenses message
85                                         waiting notifications from multiple mailboxes into a single NOTIFY. If it is disabled,
86                                         individual NOTIFYs are sent for each mailbox.</para></description>
87                                 </configOption>
88                                 <configOption name="allow">
89                                         <synopsis>Media Codec(s) to allow</synopsis>
90                                 </configOption>
91                                 <configOption name="aors">
92                                         <synopsis>AoR(s) to be used with the endpoint</synopsis>
93                                         <description><para>
94                                                 List of comma separated AoRs that the endpoint should be associated with.
95                                         </para></description>
96                                 </configOption>
97                                 <configOption name="auth">
98                                         <synopsis>Authentication Object(s) associated with the endpoint</synopsis>
99                                         <description><para>
100                                                 This is a comma-delimited list of <replaceable>auth</replaceable> sections defined
101                                                 in <filename>pjsip.conf</filename> to be used to verify inbound connection attempts.
102                                                 </para><para>
103                                                 Endpoints without an <literal>authentication</literal> object
104                                                 configured will allow connections without vertification.
105                                         </para></description>
106                                 </configOption>
107                                 <configOption name="callerid">
108                                         <synopsis>CallerID information for the endpoint</synopsis>
109                                         <description><para>
110                                                 Must be in the format <literal>Name &lt;Number&gt;</literal>,
111                                                 or only <literal>&lt;Number&gt;</literal>.
112                                         </para></description>
113                                 </configOption>
114                                 <configOption name="callerid_privacy">
115                                         <synopsis>Default privacy level</synopsis>
116                                         <description>
117                                                 <enumlist>
118                                                         <enum name="allowed_not_screened" />
119                                                         <enum name="allowed_passed_screened" />
120                                                         <enum name="allowed_failed_screened" />
121                                                         <enum name="allowed" />
122                                                         <enum name="prohib_not_screened" />
123                                                         <enum name="prohib_passed_screened" />
124                                                         <enum name="prohib_failed_screened" />
125                                                         <enum name="prohib" />
126                                                         <enum name="unavailable" />
127                                                 </enumlist>
128                                         </description>
129                                 </configOption>
130                                 <configOption name="callerid_tag">
131                                         <synopsis>Internal id_tag for the endpoint</synopsis>
132                                 </configOption>
133                                 <configOption name="context">
134                                         <synopsis>Dialplan context for inbound sessions</synopsis>
135                                 </configOption>
136                                 <configOption name="direct_media_glare_mitigation" default="none">
137                                         <synopsis>Mitigation of direct media (re)INVITE glare</synopsis>
138                                         <description>
139                                                 <para>
140                                                 This setting attempts to avoid creating INVITE glare scenarios
141                                                 by disabling direct media reINVITEs in one direction thereby allowing
142                                                 designated servers (according to this option) to initiate direct
143                                                 media reINVITEs without contention and significantly reducing call
144                                                 setup time.
145                                                 </para>
146                                                 <para>
147                                                 A more detailed description of how this option functions can be found on
148                                                 the Asterisk wiki https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance
149                                                 </para>
150                                                 <enumlist>
151                                                         <enum name="none" />
152                                                         <enum name="outgoing" />
153                                                         <enum name="incoming" />
154                                                 </enumlist>
155                                         </description>
156                                 </configOption>
157                                 <configOption name="direct_media_method" default="invite">
158                                         <synopsis>Direct Media method type</synopsis>
159                                         <description>
160                                                 <para>Method for setting up Direct Media between endpoints.</para>
161                                                 <enumlist>
162                                                         <enum name="invite" />
163                                                         <enum name="reinvite">
164                                                                 <para>Alias for the <literal>invite</literal> value.</para>
165                                                         </enum>
166                                                         <enum name="update" />
167                                                 </enumlist>
168                                         </description>
169                                 </configOption>
170                                 <configOption name="connected_line_method" default="invite">
171                                         <synopsis>Connected line method type</synopsis>
172                                         <description>
173                                                 <para>Method used when updating connected line information.</para>
174                                                 <enumlist>
175                                                         <enum name="invite" />
176                                                         <enum name="reinvite">
177                                                                 <para>Alias for the <literal>invite</literal> value.</para>
178                                                         </enum>
179                                                         <enum name="update" />
180                                                 </enumlist>
181                                         </description>
182                                 </configOption>
183                                 <configOption name="direct_media" default="yes">
184                                         <synopsis>Determines whether media may flow directly between endpoints.</synopsis>
185                                 </configOption>
186                                 <configOption name="disable_direct_media_on_nat" default="no">
187                                         <synopsis>Disable direct media session refreshes when NAT obstructs the media session</synopsis>
188                                 </configOption>
189                                 <configOption name="disallow">
190                                         <synopsis>Media Codec(s) to disallow</synopsis>
191                                 </configOption>
192                                 <configOption name="dtmfmode" default="rfc4733">
193                                         <synopsis>DTMF mode</synopsis>
194                                         <description>
195                                                 <para>This setting allows to choose the DTMF mode for endpoint communication.</para>
196                                                 <enumlist>
197                                                         <enum name="rfc4733">
198                                                                 <para>DTMF is sent out of band of the main audio stream.This
199                                                                 supercedes the older <emphasis>RFC-2833</emphasis> used within
200                                                                 the older <literal>chan_sip</literal>.</para>
201                                                         </enum>
202                                                         <enum name="inband">
203                                                                 <para>DTMF is sent as part of audio stream.</para>
204                                                         </enum>
205                                                         <enum name="info">
206                                                                 <para>DTMF is sent as SIP INFO packets.</para>
207                                                         </enum>
208                                                 </enumlist>
209                                         </description>
210                                 </configOption>
211                                 <configOption name="external_media_address">
212                                         <synopsis>IP used for External Media handling</synopsis>
213                                 </configOption>
214                                 <configOption name="force_rport" default="yes">
215                                         <synopsis>Force use of return port</synopsis>
216                                 </configOption>
217                                 <configOption name="ice_support" default="no">
218                                         <synopsis>Enable the ICE mechanism to help traverse NAT</synopsis>
219                                 </configOption>
220                                 <configOption name="identify_by" default="username,location">
221                                         <synopsis>Way(s) for Endpoint to be identified</synopsis>
222                                         <description><para>
223                                                 An endpoint can be identified in multiple ways. Currently, the only supported
224                                                 option is <literal>username</literal>, which matches the endpoint based on the
225                                                 username in the From header.
226                                                 </para>
227                                                 <note><para>Endpoints can also be identified by IP address; however, that method
228                                                 of identification is not handled by this configuration option. See the documentation
229                                                 for the <literal>identify</literal> configuration section for more details on that
230                                                 method of endpoint identification. If this option is set to <literal>username</literal>
231                                                 and an <literal>identify</literal> configuration section exists for the endpoint, then
232                                                 the endpoint can be identified in multiple ways.</para></note>
233                                                 <enumlist>
234                                                         <enum name="username" />
235                                                 </enumlist>
236                                         </description>
237                                 </configOption>
238                                 <configOption name="mailboxes">
239                                         <synopsis>Mailbox(es) to be associated with</synopsis>
240                                 </configOption>
241                                 <configOption name="mohsuggest" default="default">
242                                         <synopsis>Default Music On Hold class</synopsis>
243                                 </configOption>
244                                 <configOption name="outbound_auth">
245                                         <synopsis>Authentication object used for outbound requests</synopsis>
246                                 </configOption>
247                                 <configOption name="outbound_proxy">
248                                         <synopsis>Proxy through which to send requests</synopsis>
249                                 </configOption>
250                                 <configOption name="rewrite_contact">
251                                         <synopsis>Allow Contact header to be rewritten with the source IP address-port</synopsis>
252                                 </configOption>
253                                 <configOption name="rtp_ipv6" default="no">
254                                         <synopsis>Allow use of IPv6 for RTP traffic</synopsis>
255                                 </configOption>
256                                 <configOption name="rtp_symmetric" default="no">
257                                         <synopsis>Enforce that RTP must be symmetric</synopsis>
258                                 </configOption>
259                                 <configOption name="send_pai" default="no">
260                                         <synopsis>Send the P-Asserted-Identity header</synopsis>
261                                 </configOption>
262                                 <configOption name="send_rpid" default="no">
263                                         <synopsis>Send the Remote-Party-ID header</synopsis>
264                                 </configOption>
265                                 <configOption name="timers_min_se" default="90">
266                                         <synopsis>Minimum session timers expiration period</synopsis>
267                                         <description><para>
268                                                 Minimium session timer expiration period. Time in seconds.
269                                         </para></description>
270                                 </configOption>
271                                 <configOption name="timers" default="yes">
272                                         <synopsis>Session timers for SIP packets</synopsis>
273                                         <description>
274                                                 <enumlist>
275                                                         <enum name="forced" />
276                                                         <enum name="no" />
277                                                         <enum name="required" />
278                                                         <enum name="yes" />
279                                                 </enumlist>
280                                         </description>
281                                 </configOption>
282                                 <configOption name="timers_sess_expires" default="1800">
283                                         <synopsis>Maximum session timer expiration period</synopsis>
284                                         <description><para>
285                                                 Maximium session timer expiration period. Time in seconds.
286                                         </para></description>
287                                 </configOption>
288                                 <configOption name="transport">
289                                         <synopsis>Desired transport configuration</synopsis>
290                                         <description><para>
291                                                 This will set the desired transport configuration to send SIP data through.
292                                                 </para>
293                                                 <warning><para>Not specifying a transport will <emphasis>DEFAULT</emphasis>
294                                                 to the first configured transport in <filename>pjsip.conf</filename> which is
295                                                 valid for the URI we are trying to contact.
296                                                 </para></warning>
297                                                 <warning><para>Transport configuration is not affected by reloads. In order to
298                                                 change transports, a full Asterisk restart is required</para></warning>
299                                         </description>
300                                 </configOption>
301                                 <configOption name="trust_id_inbound" default="no">
302                                         <synopsis>Accept identification information received from this endpoint</synopsis>
303                                         <description><para>This option determines whether Asterisk will accept
304                                         identification from the endpoint from headers such as P-Asserted-Identity
305                                         or Remote-Party-ID header. This option applies both to calls originating from the
306                                         endpoint and calls originating from Asterisk. If <literal>no</literal>, the
307                                         configured Caller-ID from pjsip.conf will always be used as the identity for
308                                         the endpoint.</para></description>
309                                 </configOption>
310                                 <configOption name="trust_id_outbound" default="no">
311                                         <synopsis>Send private identification details to the endpoint.</synopsis>
312                                         <description><para>This option determines whether res_pjsip will send private
313                                         identification information to the endpoint. If <literal>no</literal>,
314                                         private Caller-ID information will not be forwarded to the endpoint.
315                                         "Private" in this case refers to any method of restricting identification.
316                                         Example: setting <replaceable>callerid_privacy</replaceable> to any
317                                         <literal>prohib</literal> variation.
318                                         Example: If <replaceable>trust_id_inbound</replaceable> is set to
319                                         <literal>yes</literal>, the presence of a <literal>Privacy: id</literal>
320                                         header in a SIP request or response would indicate the identification
321                                         provided in the request is private.</para></description>
322                                 </configOption>
323                                 <configOption name="type">
324                                         <synopsis>Must be of type 'endpoint'.</synopsis>
325                                 </configOption>
326                                 <configOption name="use_ptime" default="no">
327                                         <synopsis>Use Endpoint's requested packetisation interval</synopsis>
328                                 </configOption>
329                                 <configOption name="use_avpf" default="no">
330                                         <synopsis>Determines whether res_pjsip will use and enforce usage of AVPF for this
331                                         endpoint.</synopsis>
332                                         <description><para>
333                                                 If set to <literal>yes</literal>, res_pjsip will use use the AVPF or SAVPF RTP
334                                                 profile for all media offers on outbound calls and media updates and will
335                                                 decline media offers not using the AVPF or SAVPF profile.
336                                         </para><para>
337                                                 If set to <literal>no</literal>, res_pjsip will use use the AVP or SAVP RTP
338                                                 profile for all media offers on outbound calls and media updates and will
339                                                 decline media offers not using the AVP or SAVP profile.
340                                         </para></description>
341                                 </configOption>
342                                 <configOption name="media_encryption" default="no">
343                                         <synopsis>Determines whether res_pjsip will use and enforce usage of media encryption
344                                         for this endpoint.</synopsis>
345                                         <description>
346                                                 <enumlist>
347                                                         <enum name="no"><para>
348                                                                 res_pjsip will offer no encryption and allow no encryption to be setup.
349                                                         </para></enum>
350                                                         <enum name="sdes"><para>
351                                                                 res_pjsip will offer standard SRTP setup via in-SDP keys. Encrypted SIP
352                                                                 transport should be used in conjunction with this option to prevent
353                                                                 exposure of media encryption keys.
354                                                         </para></enum>
355                                                         <enum name="dtls"><para>
356                                                                 res_pjsip will offer DTLS-SRTP setup.
357                                                         </para></enum>
358                                                 </enumlist>
359                                         </description>
360                                 </configOption>
361                                 <configOption name="inband_progress" default="no">
362                                         <synopsis>Determines whether chan_pjsip will indicate ringing using inband
363                                             progress.</synopsis>
364                                         <description><para>
365                                                 If set to <literal>yes</literal>, chan_pjsip will send a 183 Session Progress
366                                                 when told to indicate ringing and will immediately start sending ringing
367                                                 as audio.
368                                         </para><para>
369                                                 If set to <literal>no</literal>, chan_pjsip will send a 180 Ringing when told
370                                                 to indicate ringing and will NOT send it as audio.
371                                         </para></description>
372                                 </configOption>
373                                 <configOption name="callgroup">
374                                         <synopsis>The numeric pickup groups for a channel.</synopsis>
375                                         <description><para>
376                                                 Can be set to a comma separated list of numbers or ranges between the values
377                                                 of 0-63 (maximum of 64 groups).
378                                         </para></description>
379                                 </configOption>
380                                 <configOption name="pickupgroup">
381                                         <synopsis>The numeric pickup groups that a channel can pickup.</synopsis>
382                                         <description><para>
383                                                 Can be set to a comma separated list of numbers or ranges between the values
384                                                 of 0-63 (maximum of 64 groups).
385                                         </para></description>
386                                 </configOption>
387                                 <configOption name="namedcallgroup">
388                                         <synopsis>The named pickup groups for a channel.</synopsis>
389                                         <description><para>
390                                                 Can be set to a comma separated list of case sensitive strings limited by
391                                                 supported line length.
392                                         </para></description>
393                                 </configOption>
394                                 <configOption name="namedpickupgroup">
395                                         <synopsis>The named pickup groups that a channel can pickup.</synopsis>
396                                         <description><para>
397                                                 Can be set to a comma separated list of case sensitive strings limited by
398                                                 supported line length.
399                                         </para></description>
400                                 </configOption>
401                                 <configOption name="devicestate_busy_at" default="0">
402                                         <synopsis>The number of in-use channels which will cause busy to be returned as device state</synopsis>
403                                         <description><para>
404                                                 When the number of in-use channels for the endpoint matches the devicestate_busy_at setting the
405                                                 PJSIP channel driver will return busy as the device state instead of in use.
406                                         </para></description>
407                                 </configOption>
408                                 <configOption name="t38udptl" default="no">
409                                         <synopsis>Whether T.38 UDPTL support is enabled or not</synopsis>
410                                         <description><para>
411                                                 If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted
412                                                 and relayed.
413                                         </para></description>
414                                 </configOption>
415                                 <configOption name="t38udptl_ec" default="none">
416                                         <synopsis>T.38 UDPTL error correction method</synopsis>
417                                         <description>
418                                                 <enumlist>
419                                                         <enum name="none"><para>
420                                                                 No error correction should be used.
421                                                         </para></enum>
422                                                         <enum name="fec"><para>
423                                                                 Forward error correction should be used.
424                                                         </para></enum>
425                                                         <enum name="redundancy"><para>
426                                                                 Redundacy error correction should be used.
427                                                         </para></enum>
428                                                 </enumlist>
429                                         </description>
430                                 </configOption>
431                                 <configOption name="t38udptl_maxdatagram" default="0">
432                                         <synopsis>T.38 UDPTL maximum datagram size</synopsis>
433                                         <description><para>
434                                                 This option can be set to override the maximum datagram of a remote endpoint for broken
435                                                 endpoints.
436                                         </para></description>
437                                 </configOption>
438                                 <configOption name="faxdetect" default="no">
439                                         <synopsis>Whether CNG tone detection is enabled</synopsis>
440                                         <description><para>
441                                                 This option can be set to send the session to the fax extension when a CNG tone is
442                                                 detected.
443                                         </para></description>
444                                 </configOption>
445                                 <configOption name="t38udptl_nat" default="no">
446                                         <synopsis>Whether NAT support is enabled on UDPTL sessions</synopsis>
447                                         <description><para>
448                                                 When enabled the UDPTL stack will send UDPTL packets to the source address of
449                                                 received packets.
450                                         </para></description>
451                                 </configOption>
452                                 <configOption name="t38udptl_ipv6" default="no">
453                                         <synopsis>Whether IPv6 is used for UDPTL Sessions</synopsis>
454                                         <description><para>
455                                                 When enabled the UDPTL stack will use IPv6.
456                                         </para></description>
457                                 </configOption>
458                                 <configOption name="tonezone">
459                                         <synopsis>Set which country's indications to use for channels created for this endpoint.</synopsis>
460                                 </configOption>
461                                 <configOption name="language">
462                                         <synopsis>Set the default language to use for channels created for this endpoint.</synopsis>
463                                 </configOption>
464                                 <configOption name="one_touch_recording" default="no">
465                                         <synopsis>Determines whether one-touch recording is allowed for this endpoint.</synopsis>
466                                         <see-also>
467                                                 <ref type="configOption">recordonfeature</ref>
468                                                 <ref type="configOption">recordofffeature</ref>
469                                         </see-also>
470                                 </configOption>
471                                 <configOption name="recordonfeature" default="automixmon">
472                                         <synopsis>The feature to enact when one-touch recording is turned on.</synopsis>
473                                         <description>
474                                                 <para>When an INFO request for one-touch recording arrives with a Record header set to "on", this
475                                                 feature will be enabled for the channel. The feature designated here can be any built-in
476                                                 or dynamic feature defined in features.conf.</para>
477                                                 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
478                                         </description>
479                                         <see-also>
480                                                 <ref type="configOption">one_touch_recording</ref>
481                                                 <ref type="configOption">recordofffeature</ref>
482                                         </see-also>
483                                 </configOption>
484                                 <configOption name="recordofffeature" default="automixmon">
485                                         <synopsis>The feature to enact when one-touch recording is turned off.</synopsis>
486                                         <description>
487                                                 <para>When an INFO request for one-touch recording arrives with a Record header set to "off", this
488                                                 feature will be enabled for the channel. The feature designated here can be any built-in
489                                                 or dynamic feature defined in features.conf.</para>
490                                                 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
491                                         </description>
492                                         <see-also>
493                                                 <ref type="configOption">one_touch_recording</ref>
494                                                 <ref type="configOption">recordonfeature</ref>
495                                         </see-also>
496                                 </configOption>
497                                 <configOption name="rtpengine" default="asterisk">
498                                         <synopsis>Name of the RTP engine to use for channels created for this endpoint</synopsis>
499                                 </configOption>
500                                 <configOption name="allowtransfer" default="yes">
501                                         <synopsis>Determines whether SIP REFER transfers are allowed for this endpoint</synopsis>
502                                 </configOption>
503                                 <configOption name="sdpowner" default="-">
504                                         <synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis>
505                                 </configOption>
506                                 <configOption name="sdpsession" default="Asterisk">
507                                         <synopsis>String used for the SDP session (s=) line.</synopsis>
508                                 </configOption>
509                                 <configOption name="tos_audio">
510                                         <synopsis>DSCP TOS bits for audio streams</synopsis>
511                                         <description><para>
512                                                 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
513                                         </para></description>
514                                 </configOption>
515                                 <configOption name="tos_video">
516                                         <synopsis>DSCP TOS bits for video streams</synopsis>
517                                         <description><para>
518                                                 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
519                                         </para></description>
520                                 </configOption>
521                                 <configOption name="cos_audio">
522                                         <synopsis>Priority for audio streams</synopsis>
523                                         <description><para>
524                                                 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
525                                         </para></description>
526                                 </configOption>
527                                 <configOption name="cos_video">
528                                         <synopsis>Priority for video 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="allowsubscribe" default="yes">
534                                         <synopsis>Determines if endpoint is allowed to initiate subscriptions with Asterisk.</synopsis>
535                                 </configOption>
536                                 <configOption name="subminexpiry" default="60">
537                                         <synopsis>The minimum allowed expiry time for subscriptions initiated by the endpoint.</synopsis>
538                                 </configOption>
539                                 <configOption name="fromuser">
540                                         <synopsis>Username to use in From header for requests to this endpoint.</synopsis>
541                                 </configOption>
542                                 <configOption name="mwifromuser">
543                                         <synopsis>Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.</synopsis>
544                                 </configOption>
545                                 <configOption name="fromdomain">
546                                         <synopsis>Domain to user in From header for requests to this endpoint.</synopsis>
547                                 </configOption>
548                                 <configOption name="dtlsverify">
549                                         <synopsis>Verify that the provided peer certificate is valid</synopsis>
550                                         <description><para>
551                                                 This option only applies if <replaceable>media_encryption</replaceable> is
552                                                 set to <literal>dtls</literal>.
553                                         </para></description>
554                                 </configOption>
555                                 <configOption name="dtlsrekey">
556                                         <synopsis>Interval at which to renegotiate the TLS session and rekey the SRTP session</synopsis>
557                                         <description><para>
558                                                 This option only applies if <replaceable>media_encryption</replaceable> is
559                                                 set to <literal>dtls</literal>.
560                                         </para><para>
561                                                 If this is not set or the value provided is 0 rekeying will be disabled.
562                                         </para></description>
563                                 </configOption>
564                                 <configOption name="dtlscertfile">
565                                         <synopsis>Path to certificate file to present to peer</synopsis>
566                                         <description><para>
567                                                 This option only applies if <replaceable>media_encryption</replaceable> is
568                                                 set to <literal>dtls</literal>.
569                                         </para></description>
570                                 </configOption>
571                                 <configOption name="dtlsprivatekey">
572                                         <synopsis>Path to private key for certificate file</synopsis>
573                                         <description><para>
574                                                 This option only applies if <replaceable>media_encryption</replaceable> is
575                                                 set to <literal>dtls</literal>.
576                                         </para></description>
577                                 </configOption>
578                                 <configOption name="dtlscipher">
579                                         <synopsis>Cipher to use for DTLS negotiation</synopsis>
580                                         <description><para>
581                                                 This option only applies if <replaceable>media_encryption</replaceable> is
582                                                 set to <literal>dtls</literal>.
583                                         </para><para>
584                                                 Many options for acceptable ciphers. See link for more:
585                                                 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
586                                         </para></description>
587                                 </configOption>
588                                 <configOption name="dtlscafile">
589                                         <synopsis>Path to certificate authority certificate</synopsis>
590                                         <description><para>
591                                                 This option only applies if <replaceable>media_encryption</replaceable> is
592                                                 set to <literal>dtls</literal>.
593                                         </para></description>
594                                 </configOption>
595                                 <configOption name="dtlscapath">
596                                         <synopsis>Path to a directory containing certificate authority certificates</synopsis>
597                                         <description><para>
598                                                 This option only applies if <replaceable>media_encryption</replaceable> is
599                                                 set to <literal>dtls</literal>.
600                                         </para></description>
601                                 </configOption>
602                                 <configOption name="dtlssetup">
603                                         <synopsis>Whether we are willing to accept connections, connect to the other party, or both.</synopsis>
604                                         <description>
605                                                 <para>
606                                                         This option only applies if <replaceable>media_encryption</replaceable> is
607                                                         set to <literal>dtls</literal>.
608                                                 </para>
609                                                 <enumlist>
610                                                         <enum name="active"><para>
611                                                                 res_pjsip will make a connection to the peer.
612                                                         </para></enum>
613                                                         <enum name="passive"><para>
614                                                                 res_pjsip will accept connections from the peer.
615                                                         </para></enum>
616                                                         <enum name="actpass"><para>
617                                                                 res_pjsip will offer and accept connections from the peer.
618                                                         </para></enum>
619                                                 </enumlist>
620                                         </description>
621                                 </configOption>
622                                 <configOption name="srtp_tag_32">
623                                         <synopsis>Determines whether 32 byte tags should be used instead of 80 byte tags.</synopsis>
624                                         <description><para>
625                                                 This option only applies if <replaceable>media_encryption</replaceable> is
626                                                 set to <literal>sdes</literal> or <literal>dtls</literal>.
627                                         </para></description>
628                                 </configOption>
629                         </configObject>
630                         <configObject name="auth">
631                                 <synopsis>Authentication type</synopsis>
632                                 <description><para>
633                                         Authentication objects hold the authentication information for use
634                                         by other objects such as <literal>endpoints</literal> or <literal>registrations</literal>.
635                                         This also allows for multiple objects to use a single auth object. See
636                                         the <literal>auth_type</literal> config option for password style choices.
637                                 </para></description>
638                                 <configOption name="auth_type" default="userpass">
639                                         <synopsis>Authentication type</synopsis>
640                                         <description><para>
641                                                 This option specifies which of the password style config options should be read
642                                                 when trying to authenticate an endpoint inbound request. If set to <literal>userpass</literal>
643                                                 then we'll read from the 'password' option. For <literal>md5</literal> we'll read
644                                                 from 'md5_cred'.
645                                                 </para>
646                                                 <enumlist>
647                                                         <enum name="md5"/>
648                                                         <enum name="userpass"/>
649                                                 </enumlist>
650                                         </description>
651                                 </configOption>
652                                 <configOption name="nonce_lifetime" default="32">
653                                         <synopsis>Lifetime of a nonce associated with this authentication config.</synopsis>
654                                 </configOption>
655                                 <configOption name="md5_cred">
656                                         <synopsis>MD5 Hash used for authentication.</synopsis>
657                                         <description><para>Only used when auth_type is <literal>md5</literal>.</para></description>
658                                 </configOption>
659                                 <configOption name="password">
660                                         <synopsis>PlainText password used for authentication.</synopsis>
661                                         <description><para>Only used when auth_type is <literal>userpass</literal>.</para></description>
662                                 </configOption>
663                                 <configOption name="realm" default="asterisk">
664                                         <synopsis>SIP realm for endpoint</synopsis>
665                                 </configOption>
666                                 <configOption name="type">
667                                         <synopsis>Must be 'auth'</synopsis>
668                                 </configOption>
669                                 <configOption name="username">
670                                         <synopsis>Username to use for account</synopsis>
671                                 </configOption>
672                         </configObject>
673                         <configObject name="nat_hook">
674                                 <synopsis>XXX This exists only to prevent XML documentation errors.</synopsis>
675                                 <configOption name="external_media_address">
676                                         <synopsis>I should be undocumented or hidden</synopsis>
677                                 </configOption>
678                                 <configOption name="method">
679                                         <synopsis>I should be undocumented or hidden</synopsis>
680                                 </configOption>
681                         </configObject>
682                         <configObject name="domain_alias">
683                                 <synopsis>Domain Alias</synopsis>
684                                 <description><para>
685                                         Signifies that a domain is an alias. If the domain on a session is
686                                         not found to match an AoR then this object is used to see if we have
687                                         an alias for the AoR to which the endpoint is binding. This objects
688                                         name as defined in configuration should be the domain alias and a 
689                                         config option is provided to specify the domain to be aliased.
690                                 </para></description>
691                                 <configOption name="type">
692                                         <synopsis>Must be of type 'domain_alias'.</synopsis>
693                                 </configOption>
694                                 <configOption name="domain">
695                                         <synopsis>Domain to be aliased</synopsis>
696                                 </configOption>
697                         </configObject>
698                         <configObject name="transport">
699                                 <synopsis>SIP Transport</synopsis>
700                                 <description><para>
701                                         <emphasis>Transports</emphasis>
702                                         </para>
703                                         <para>There are different transports and protocol derivatives
704                                                 supported by <literal>res_pjsip</literal>. They are in order of
705                                                 preference: UDP, TCP, and WebSocket (WS).</para>
706                                         <note><para>Changes to transport configuration in pjsip.conf will only be
707                                                 effected on a complete restart of Asterisk. A module reload
708                                                 will not suffice.</para></note>
709                                 </description>
710                                 <configOption name="async_operations" default="1">
711                                         <synopsis>Number of simultaneous Asynchronous Operations</synopsis>
712                                 </configOption>
713                                 <configOption name="bind">
714                                         <synopsis>IP Address and optional port to bind to for this transport</synopsis>
715                                 </configOption>
716                                 <configOption name="ca_list_file">
717                                         <synopsis>File containing a list of certificates to read (TLS ONLY)</synopsis>
718                                 </configOption>
719                                 <configOption name="cert_file">
720                                         <synopsis>Certificate file for endpoint (TLS ONLY)</synopsis>
721                                 </configOption>
722                                 <configOption name="cipher">
723                                         <synopsis>Preferred Cryptography Cipher (TLS ONLY)</synopsis>
724                                         <description><para>
725                                                 Many options for acceptable ciphers see link for more:
726                                                 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
727                                         </para></description>
728                                 </configOption>
729                                 <configOption name="domain">
730                                         <synopsis>Domain the transport comes from</synopsis>
731                                 </configOption>
732                                 <configOption name="external_media_address">
733                                         <synopsis>External Address to use in RTP handling</synopsis>
734                                 </configOption>
735                                 <configOption name="external_signaling_address">
736                                         <synopsis>External address for SIP signalling</synopsis>
737                                 </configOption>
738                                 <configOption name="external_signaling_port" default="0">
739                                         <synopsis>External port for SIP signalling</synopsis>
740                                 </configOption>
741                                 <configOption name="method">
742                                         <synopsis>Method of SSL transport (TLS ONLY)</synopsis>
743                                         <description>
744                                                 <enumlist>
745                                                         <enum name="default" />
746                                                         <enum name="unspecified" />
747                                                         <enum name="tlsv1" />
748                                                         <enum name="sslv2" />
749                                                         <enum name="sslv3" />
750                                                         <enum name="sslv23" />
751                                                 </enumlist>
752                                         </description>
753                                 </configOption>
754                                 <configOption name="localnet">
755                                         <synopsis>Network to consider local (used for NAT purposes).</synopsis>
756                                         <description><para>This must be in CIDR or dotted decimal format with the IP
757                                         and mask separated with a slash ('/').</para></description>
758                                 </configOption>
759                                 <configOption name="password">
760                                         <synopsis>Password required for transport</synopsis>
761                                 </configOption>
762                                 <configOption name="privkey_file">
763                                         <synopsis>Private key file (TLS ONLY)</synopsis>
764                                 </configOption>
765                                 <configOption name="protocol" default="udp">
766                                         <synopsis>Protocol to use for SIP traffic</synopsis>
767                                         <description>
768                                                 <enumlist>
769                                                         <enum name="udp" />
770                                                         <enum name="tcp" />
771                                                         <enum name="tls" />
772                                                 </enumlist>
773                                         </description>
774                                 </configOption>
775                                 <configOption name="require_client_cert" default="false">
776                                         <synopsis>Require client certificate (TLS ONLY)</synopsis>
777                                 </configOption>
778                                 <configOption name="type">
779                                         <synopsis>Must be of type 'transport'.</synopsis>
780                                 </configOption>
781                                 <configOption name="verify_client" default="false">
782                                         <synopsis>Require verification of client certificate (TLS ONLY)</synopsis>
783                                 </configOption>
784                                 <configOption name="verify_server" default="false">
785                                         <synopsis>Require verification of server certificate (TLS ONLY)</synopsis>
786                                 </configOption>
787                         </configObject>
788                         <configObject name="contact">
789                                 <synopsis>A way of creating an aliased name to a SIP URI</synopsis>
790                                 <description><para>
791                                         Contacts are a way to hide SIP URIs from the dialplan directly.
792                                         They are also used to make a group of contactable parties when
793                                         in use with <literal>AoR</literal> lists.
794                                 </para></description>
795                                 <configOption name="type">
796                                         <synopsis>Must be of type 'contact'.</synopsis>
797                                 </configOption>
798                                 <configOption name="uri">
799                                         <synopsis>SIP URI to contact peer</synopsis>
800                                 </configOption>
801                                 <configOption name="expiration_time">
802                                         <synopsis>Time to keep alive a contact</synopsis>
803                                         <description><para>
804                                                 Time to keep alive a contact. String style specification.
805                                         </para></description>
806                                 </configOption>
807                                 <configOption name="qualify_frequency" default="0">
808                                         <synopsis>Interval at which to qualify a contact</synopsis>
809                                         <description><para>
810                                                 Interval between attempts to qualify the contact for reachability.
811                                                 If <literal>0</literal> never qualify. Time in seconds.
812                                         </para></description>
813                                 </configOption>
814                         </configObject>
815                         <configObject name="contact_status">
816                                 <synopsis>Status for a contact</synopsis>
817                                 <description><para>
818                                         The contact status keeps track of whether or not a contact is reachable
819                                         and how long it took to qualify the contact (round trip time).
820                                 </para></description>
821                                 <configOption name="status">
822                                         <synopsis>A contact's status</synopsis>
823                                         <description>
824                                                 <enumlist>
825                                                         <enum name="AVAILABLE" />
826                                                         <enum name="UNAVAILABLE" />
827                                                 </enumlist>
828                                         </description>
829                                 </configOption>
830                                 <configOption name="rtt">
831                                         <synopsis>Round trip time</synopsis>
832                                         <description><para>
833                                                 The time, in microseconds, it took to qualify the contact.
834                                         </para></description>
835                                 </configOption>
836                         </configObject>
837                         <configObject name="aor">
838                                 <synopsis>The configuration for a location of an endpoint</synopsis>
839                                 <description><para>
840                                         An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
841                                         AoRs are specified, an endpoint will not be reachable by Asterisk.
842                                         Beyond that, an AoR has other uses within Asterisk, such as inbound
843                                         registration.
844                                         </para><para>
845                                         An <literal>AoR</literal> is a way to allow dialing a group
846                                         of <literal>Contacts</literal> that all use the same
847                                         <literal>endpoint</literal> for calls.
848                                         </para><para>
849                                         This can be used as another way of grouping a list of contacts to dial
850                                         rather than specifing them each directly when dialing via the dialplan.
851                                         This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
852                                         </para><para>
853                                         Registrations: For Asterisk to match an inbound registration to an endpoint,
854                                         the AoR object name must match the user portion of the SIP URI in the "To:" 
855                                         header of the inbound SIP registration. That will usually be equivalent
856                                         to the "user name" set in your hard or soft phones configuration.
857                                 </para></description>
858                                 <configOption name="contact">
859                                         <synopsis>Permanent contacts assigned to AoR</synopsis>
860                                         <description><para>
861                                                 Contacts specified will be called whenever referenced
862                                                 by <literal>chan_pjsip</literal>.
863                                                 </para><para>
864                                                 Use a separate "contact=" entry for each contact required. Contacts
865                                                 are specified using a SIP URI.
866                                         </para></description>
867                                 </configOption>
868                                 <configOption name="default_expiration" default="3600">
869                                         <synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis>
870                                 </configOption>
871                                 <configOption name="mailboxes">
872                                         <synopsis>Mailbox(es) to be associated with</synopsis>
873                                         <description><para>This option applies when an external entity subscribes to an AoR
874                                         for message waiting indications. The mailboxes specified will be subscribed to.
875                                         More than one mailbox can be specified with a comma-delimited string.</para></description>
876                                 </configOption>
877                                 <configOption name="maximum_expiration" default="7200">
878                                         <synopsis>Maximum time to keep an AoR</synopsis>
879                                         <description><para>
880                                                 Maximium time to keep a peer with explicit expiration. Time in seconds.
881                                         </para></description>
882                                 </configOption>
883                                 <configOption name="max_contacts" default="0">
884                                         <synopsis>Maximum number of contacts that can bind to an AoR</synopsis>
885                                         <description><para>
886                                                 Maximum number of contacts that can associate with this AoR. This value does
887                                                 not affect the number of contacts that can be added with the "contact" option.
888                                                 It only limits contacts added through external interaction, such as
889                                                 registration.
890                                                 </para>
891                                                 <note><para>This should be set to <literal>1</literal> and
892                                                 <replaceable>remove_existing</replaceable> set to <literal>yes</literal> if you
893                                                 wish to stick with the older <literal>chan_sip</literal> behaviour.
894                                                 </para></note>
895                                         </description>
896                                 </configOption>
897                                 <configOption name="minimum_expiration" default="60">
898                                         <synopsis>Minimum keep alive time for an AoR</synopsis>
899                                         <description><para>
900                                                 Minimum time to keep a peer with an explict expiration. Time in seconds.
901                                         </para></description>
902                                 </configOption>
903                                 <configOption name="remove_existing" default="no">
904                                         <synopsis>Determines whether new contacts replace existing ones.</synopsis>
905                                         <description><para>
906                                                 On receiving a new registration to the AoR should it remove
907                                                 the existing contact that was registered against it?
908                                                 </para>
909                                                 <note><para>This should be set to <literal>yes</literal> and
910                                                 <replaceable>max_contacts</replaceable> set to <literal>1</literal> if you
911                                                 wish to stick with the older <literal>chan_sip</literal> behaviour.
912                                                 </para></note>
913                                         </description>
914                                 </configOption>
915                                 <configOption name="type">
916                                         <synopsis>Must be of type 'aor'.</synopsis>
917                                 </configOption>
918                                 <configOption name="qualify_frequency" default="0">
919                                         <synopsis>Interval at which to qualify an AoR</synopsis>
920                                         <description><para>
921                                                 Interval between attempts to qualify the AoR for reachability.
922                                                 If <literal>0</literal> never qualify. Time in seconds.
923                                         </para></description>
924                                 </configOption>
925                                 <configOption name="authenticate_qualify" default="no">
926                                         <synopsis>Authenticates a qualify request if needed</synopsis>
927                                         <description><para>
928                                                 If true and a qualify request receives a challenge or authenticate response
929                                                 authentication is attempted before declaring the contact available.
930                                         </para></description>
931                                 </configOption>
932                         </configObject>
933                         <configObject name="system">
934                                 <synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis>
935                                 <description><para>
936                                         The settings in this section are global. In addition to being global, the values will
937                                         not be re-evaluated when a reload is performed. This is because the values must be set
938                                         before the SIP stack is initialized. The only way to reset these values is to either 
939                                         restart Asterisk, or unload res_pjsip.so and then load it again.
940                                 </para></description>
941                                 <configOption name="timert1" default="500">
942                                         <synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
943                                         <description><para>
944                                                 Timer T1 is the base for determining how long to wait before retransmitting
945                                                 requests that receive no response when using an unreliable transport (e.g. UDP).
946                                                 For more information on this timer, see RFC 3261, Section 17.1.1.1.
947                                         </para></description>
948                                 </configOption>
949                                 <configOption name="timerb" default="32000">
950                                         <synopsis>Set transaction timer B value (milliseconds).</synopsis>
951                                         <description><para>
952                                                 Timer B determines the maximum amount of time to wait after sending an INVITE
953                                                 request before terminating the transaction. It is recommended that this be set
954                                                 to 64 * Timer T1, but it may be set higher if desired. For more information on
955                                                 this timer, see RFC 3261, Section 17.1.1.1.
956                                         </para></description>
957                                 </configOption>
958                                 <configOption name="compactheaders" default="no">
959                                         <synopsis>Use the short forms of common SIP header names.</synopsis>
960                                 </configOption>
961                                 <configOption name="threadpool_initial_size" default="0">
962                                         <synopsis>Initial number of threads in the res_pjsip threadpool.</synopsis>
963                                 </configOption>
964                                 <configOption name="threadpool_auto_increment" default="5">
965                                         <synopsis>The amount by which the number of threads is incremented when necessary.</synopsis>
966                                 </configOption>
967                                 <configOption name="threadpool_idle_timeout" default="60">
968                                         <synopsis>Number of seconds before an idle thread should be disposed of.</synopsis>
969                                 </configOption>
970                                 <configOption name="threadpool_max_size" default="0">
971                                         <synopsis>Maximum number of threads in the res_pjsip threadpool.
972                                         A value of 0 indicates no maximum.</synopsis>
973                                 </configOption>
974                                 <configOption name="type">
975                                         <synopsis>Must be of type 'system'.</synopsis>
976                                 </configOption>
977                         </configObject>
978                         <configObject name="global">
979                                 <synopsis>Options that apply globally to all SIP communications</synopsis>
980                                 <description><para>
981                                         The settings in this section are global. Unlike options in the <literal>system</literal>
982                                         section, these options can be refreshed by performing a reload.
983                                 </para></description>
984                                 <configOption name="maxforwards" default="70">
985                                         <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
986                                 </configOption>
987                                 <configOption name="useragent" default="Asterisk &lt;Asterisk Version&gt;">
988                                         <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
989                                 </configOption>
990                                 <configOption name="type">
991                                         <synopsis>Must be of type 'global'.</synopsis>
992                                 </configOption>
993                         </configObject>
994                 </configFile>
995         </configInfo>
996         <manager name="PJSIPQualify" language="en_US">
997                 <synopsis>
998                         Qualify a chan_pjsip endpoint.
999                 </synopsis>
1000                 <syntax>
1001                         <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
1002                         <parameter name="Endpoint" required="true">
1003                                 <para>The endpoint you want to qualify.</para>
1004                         </parameter>
1005                 </syntax>
1006                 <description>
1007                         <para>Qualify a chan_pjsip endpoint.</para>
1008                 </description>
1009         </manager>
1010  ***/
1011
1012
1013 static pjsip_endpoint *ast_pjsip_endpoint;
1014
1015 static struct ast_threadpool *sip_threadpool;
1016
1017 static int register_service(void *data)
1018 {
1019         pjsip_module **module = data;
1020         if (!ast_pjsip_endpoint) {
1021                 ast_log(LOG_ERROR, "There is no PJSIP endpoint. Unable to register services\n");
1022                 return -1;
1023         }
1024         if (pjsip_endpt_register_module(ast_pjsip_endpoint, *module) != PJ_SUCCESS) {
1025                 ast_log(LOG_ERROR, "Unable to register module %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1026                 return -1;
1027         }
1028         ast_debug(1, "Registered SIP service %.*s (%p)\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name), *module);
1029         ast_module_ref(ast_module_info->self);
1030         return 0;
1031 }
1032
1033 int ast_sip_register_service(pjsip_module *module)
1034 {
1035         return ast_sip_push_task_synchronous(NULL, register_service, &module);
1036 }
1037
1038 static int unregister_service(void *data)
1039 {
1040         pjsip_module **module = data;
1041         ast_module_unref(ast_module_info->self);
1042         if (!ast_pjsip_endpoint) {
1043                 return -1;
1044         }
1045         pjsip_endpt_unregister_module(ast_pjsip_endpoint, *module);
1046         ast_debug(1, "Unregistered SIP service %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1047         return 0;
1048 }
1049
1050 void ast_sip_unregister_service(pjsip_module *module)
1051 {
1052         ast_sip_push_task_synchronous(NULL, unregister_service, &module);
1053 }
1054
1055 static struct ast_sip_authenticator *registered_authenticator;
1056
1057 int ast_sip_register_authenticator(struct ast_sip_authenticator *auth)
1058 {
1059         if (registered_authenticator) {
1060                 ast_log(LOG_WARNING, "Authenticator %p is already registered. Cannot register a new one\n", registered_authenticator);
1061                 return -1;
1062         }
1063         registered_authenticator = auth;
1064         ast_debug(1, "Registered SIP authenticator module %p\n", auth);
1065         ast_module_ref(ast_module_info->self);
1066         return 0;
1067 }
1068
1069 void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth)
1070 {
1071         if (registered_authenticator != auth) {
1072                 ast_log(LOG_WARNING, "Trying to unregister authenticator %p but authenticator %p registered\n",
1073                                 auth, registered_authenticator);
1074                 return;
1075         }
1076         registered_authenticator = NULL;
1077         ast_debug(1, "Unregistered SIP authenticator %p\n", auth);
1078         ast_module_unref(ast_module_info->self);
1079 }
1080
1081 int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1082 {
1083         if (!registered_authenticator) {
1084                 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is not required\n");
1085                 return 0;
1086         }
1087
1088         return registered_authenticator->requires_authentication(endpoint, rdata);
1089 }
1090
1091 enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint,
1092                 pjsip_rx_data *rdata, pjsip_tx_data *tdata)
1093 {
1094         if (!registered_authenticator) {
1095                 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is successful\n");
1096                 return 0;
1097         }
1098         return registered_authenticator->check_authentication(endpoint, rdata, tdata);
1099 }
1100
1101 static struct ast_sip_outbound_authenticator *registered_outbound_authenticator;
1102
1103 int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1104 {
1105         if (registered_outbound_authenticator) {
1106                 ast_log(LOG_WARNING, "Outbound authenticator %p is already registered. Cannot register a new one\n", registered_outbound_authenticator);
1107                 return -1;
1108         }
1109         registered_outbound_authenticator = auth;
1110         ast_debug(1, "Registered SIP outbound authenticator module %p\n", auth);
1111         ast_module_ref(ast_module_info->self);
1112         return 0;
1113 }
1114
1115 void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1116 {
1117         if (registered_outbound_authenticator != auth) {
1118                 ast_log(LOG_WARNING, "Trying to unregister outbound authenticator %p but outbound authenticator %p registered\n",
1119                                 auth, registered_outbound_authenticator);
1120                 return;
1121         }
1122         registered_outbound_authenticator = NULL;
1123         ast_debug(1, "Unregistered SIP outbound authenticator %p\n", auth);
1124         ast_module_unref(ast_module_info->self);
1125 }
1126
1127 int ast_sip_create_request_with_auth(const struct ast_sip_auth_array *auths, pjsip_rx_data *challenge,
1128                 pjsip_transaction *tsx, pjsip_tx_data **new_request)
1129 {
1130         if (!registered_outbound_authenticator) {
1131                 ast_log(LOG_WARNING, "No SIP outbound authenticator registered. Cannot respond to authentication challenge\n");
1132                 return -1;
1133         }
1134         return registered_outbound_authenticator->create_request_with_auth(auths, challenge, tsx, new_request);
1135 }
1136
1137 struct endpoint_identifier_list {
1138         struct ast_sip_endpoint_identifier *identifier;
1139         AST_RWLIST_ENTRY(endpoint_identifier_list) list;
1140 };
1141
1142 static AST_RWLIST_HEAD_STATIC(endpoint_identifiers, endpoint_identifier_list);
1143
1144 int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1145 {
1146         struct endpoint_identifier_list *id_list_item;
1147         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1148
1149         id_list_item = ast_calloc(1, sizeof(*id_list_item));
1150         if (!id_list_item) {
1151                 ast_log(LOG_ERROR, "Unabled to add endpoint identifier. Out of memory.\n");
1152                 return -1;
1153         }
1154         id_list_item->identifier = identifier;
1155
1156         AST_RWLIST_INSERT_TAIL(&endpoint_identifiers, id_list_item, list);
1157         ast_debug(1, "Registered endpoint identifier %p\n", identifier);
1158
1159         ast_module_ref(ast_module_info->self);
1160         return 0;
1161 }
1162
1163 void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1164 {
1165         struct endpoint_identifier_list *iter;
1166         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1167         AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_identifiers, iter, list) {
1168                 if (iter->identifier == identifier) {
1169                         AST_RWLIST_REMOVE_CURRENT(list);
1170                         ast_free(iter);
1171                         ast_debug(1, "Unregistered endpoint identifier %p\n", identifier);
1172                         ast_module_unref(ast_module_info->self);
1173                         break;
1174                 }
1175         }
1176         AST_RWLIST_TRAVERSE_SAFE_END;
1177 }
1178
1179 struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata)
1180 {
1181         struct endpoint_identifier_list *iter;
1182         struct ast_sip_endpoint *endpoint = NULL;
1183         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1184         AST_RWLIST_TRAVERSE(&endpoint_identifiers, iter, list) {
1185                 ast_assert(iter->identifier->identify_endpoint != NULL);
1186                 endpoint = iter->identifier->identify_endpoint(rdata);
1187                 if (endpoint) {
1188                         break;
1189                 }
1190         }
1191         return endpoint;
1192 }
1193
1194 pjsip_endpoint *ast_sip_get_pjsip_endpoint(void)
1195 {
1196         return ast_pjsip_endpoint;
1197 }
1198
1199 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)
1200 {
1201         pj_str_t tmp, local_addr;
1202         pjsip_uri *uri;
1203         pjsip_sip_uri *sip_uri;
1204         pjsip_transport_type_e type = PJSIP_TRANSPORT_UNSPECIFIED;
1205         int local_port;
1206         char uuid_str[AST_UUID_STR_LEN];
1207
1208         if (ast_strlen_zero(user)) {
1209                 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1210                 if (!uuid) {
1211                         return -1;
1212                 }
1213                 user = ast_uuid_to_str(uuid, uuid_str, sizeof(uuid_str));
1214         }
1215
1216         /* Parse the provided target URI so we can determine what transport it will end up using */
1217         pj_strdup_with_null(pool, &tmp, target);
1218
1219         if (!(uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0)) ||
1220             (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
1221                 return -1;
1222         }
1223
1224         sip_uri = pjsip_uri_get_uri(uri);
1225
1226         /* Determine the transport type to use */
1227         if (PJSIP_URI_SCHEME_IS_SIPS(sip_uri)) {
1228                 type = PJSIP_TRANSPORT_TLS;
1229         } else if (!sip_uri->transport_param.slen) {
1230                 type = PJSIP_TRANSPORT_UDP;
1231         } else {
1232                 type = pjsip_transport_get_type_from_name(&sip_uri->transport_param);
1233         }
1234
1235         if (type == PJSIP_TRANSPORT_UNSPECIFIED) {
1236                 return -1;
1237         }
1238
1239         /* If the host is IPv6 turn the transport into an IPv6 version */
1240         if (pj_strchr(&sip_uri->host, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1241                 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1242         }
1243
1244         if (!ast_strlen_zero(domain)) {
1245                 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1246                 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1247                                 "<%s:%s@%s%s%s>",
1248                                 (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1249                                 user,
1250                                 domain,
1251                                 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1252                                 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1253                 return 0;
1254         }
1255
1256         /* Get the local bound address for the transport that will be used when communicating with the provided URI */
1257         if (pjsip_tpmgr_find_local_addr(pjsip_endpt_get_tpmgr(ast_sip_get_pjsip_endpoint()), pool, type, selector,
1258                                                               &local_addr, &local_port) != PJ_SUCCESS) {
1259                 return -1;
1260         }
1261
1262         /* If IPv6 was specified in the transport, set the proper type */
1263         if (pj_strchr(&local_addr, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1264                 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1265         }
1266
1267         from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1268         from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1269                                       "<%s:%s@%s%.*s%s:%d%s%s>",
1270                                       (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1271                                       user,
1272                                       (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1273                                       (int)local_addr.slen,
1274                                       local_addr.ptr,
1275                                       (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1276                                       local_port,
1277                                       (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1278                                       (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1279
1280         return 0;
1281 }
1282
1283 static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpoint, pjsip_tpselector *selector)
1284 {
1285         RAII_VAR(struct ast_sip_transport *, transport, NULL, ao2_cleanup);
1286         const char *transport_name = endpoint->transport;
1287
1288         if (ast_strlen_zero(transport_name)) {
1289                 return 0;
1290         }
1291
1292         transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_name);
1293
1294         if (!transport || !transport->state) {
1295                 return -1;
1296         }
1297
1298         if (transport->state->transport) {
1299                 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1300                 selector->u.transport = transport->state->transport;
1301         } else if (transport->state->factory) {
1302                 selector->type = PJSIP_TPSELECTOR_LISTENER;
1303                 selector->u.listener = transport->state->factory;
1304         } else {
1305                 return -1;
1306         }
1307
1308         return 0;
1309 }
1310
1311 static int sip_get_tpselector_from_uri(const char *uri, pjsip_tpselector *selector)
1312 {
1313         RAII_VAR(struct ast_sip_contact_transport *, contact_transport, NULL, ao2_cleanup);
1314
1315         contact_transport = ast_sip_location_retrieve_contact_transport_by_uri(uri);
1316
1317         if (!contact_transport) {
1318                 return -1;
1319         }
1320
1321         selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1322         selector->u.transport = contact_transport->transport;
1323
1324         return 0;
1325 }
1326
1327 pjsip_dialog *ast_sip_create_dialog(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user)
1328 {
1329         pj_str_t local_uri = { "sip:temp@temp", 13 }, remote_uri;
1330         pjsip_dialog *dlg = NULL;
1331         const char *outbound_proxy = endpoint->outbound_proxy;
1332         pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1333         static const pj_str_t HCONTACT = { "Contact", 7 };
1334
1335         pj_cstr(&remote_uri, uri);
1336
1337         if (pjsip_dlg_create_uac(pjsip_ua_instance(), &local_uri, NULL, &remote_uri, NULL, &dlg) != PJ_SUCCESS) {
1338                 return NULL;
1339         }
1340
1341         if (sip_get_tpselector_from_uri(uri, &selector) && sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1342                 pjsip_dlg_terminate(dlg);
1343                 return NULL;
1344         }
1345
1346         if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
1347                 pjsip_dlg_terminate(dlg);
1348                 return NULL;
1349         }
1350
1351         /* Update the dialog with the new local URI, we do it afterwards so we can use the dialog pool for construction */
1352         pj_strdup_with_null(dlg->pool, &dlg->local.info_str, &local_uri);
1353         dlg->local.info->uri = pjsip_parse_uri(dlg->pool, dlg->local.info_str.ptr, dlg->local.info_str.slen, 0);
1354         dlg->local.contact = pjsip_parse_hdr(dlg->pool, &HCONTACT, local_uri.ptr, local_uri.slen, NULL);
1355
1356         /* If a request user has been specified and we are permitted to change it, do so */
1357         if (!ast_strlen_zero(request_user) && (PJSIP_URI_SCHEME_IS_SIP(dlg->target) || PJSIP_URI_SCHEME_IS_SIPS(dlg->target))) {
1358                 pjsip_sip_uri *target = pjsip_uri_get_uri(dlg->target);
1359                 pj_strdup2(dlg->pool, &target->user, request_user);
1360         }
1361
1362         /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
1363         dlg->sess_count++;
1364
1365         pjsip_dlg_set_transport(dlg, &selector);
1366
1367         if (!ast_strlen_zero(outbound_proxy)) {
1368                 pjsip_route_hdr route_set, *route;
1369                 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1370                 pj_str_t tmp;
1371
1372                 pj_list_init(&route_set);
1373
1374                 pj_strdup2_with_null(dlg->pool, &tmp, outbound_proxy);
1375                 if (!(route = pjsip_parse_hdr(dlg->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1376                         pjsip_dlg_terminate(dlg);
1377                         return NULL;
1378                 }
1379                 pj_list_push_back(&route_set, route);
1380
1381                 pjsip_dlg_set_route_set(dlg, &route_set);
1382         }
1383
1384         dlg->sess_count--;
1385
1386         return dlg;
1387 }
1388
1389 /* PJSIP doesn't know about the INFO method, so we have to define it ourselves */
1390 const pjsip_method pjsip_info_method = {PJSIP_OTHER_METHOD, {"INFO", 4} };
1391 const pjsip_method pjsip_message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
1392
1393 static struct {
1394         const char *method;
1395         const pjsip_method *pmethod;
1396 } methods [] = {
1397         { "INVITE", &pjsip_invite_method },
1398         { "CANCEL", &pjsip_cancel_method },
1399         { "ACK", &pjsip_ack_method },
1400         { "BYE", &pjsip_bye_method },
1401         { "REGISTER", &pjsip_register_method },
1402         { "OPTIONS", &pjsip_options_method },
1403         { "SUBSCRIBE", &pjsip_subscribe_method },
1404         { "NOTIFY", &pjsip_notify_method },
1405         { "PUBLISH", &pjsip_publish_method },
1406         { "INFO", &pjsip_info_method },
1407         { "MESSAGE", &pjsip_message_method },
1408 };
1409
1410 static const pjsip_method *get_pjsip_method(const char *method)
1411 {
1412         int i;
1413         for (i = 0; i < ARRAY_LEN(methods); ++i) {
1414                 if (!strcmp(method, methods[i].method)) {
1415                         return methods[i].pmethod;
1416                 }
1417         }
1418         return NULL;
1419 }
1420
1421 static int create_in_dialog_request(const pjsip_method *method, struct pjsip_dialog *dlg, pjsip_tx_data **tdata)
1422 {
1423         if (pjsip_dlg_create_request(dlg, method, -1, tdata) != PJ_SUCCESS) {
1424                 ast_log(LOG_WARNING, "Unable to create in-dialog request.\n");
1425                 return -1;
1426         }
1427
1428         return 0;
1429 }
1430
1431 static int create_out_of_dialog_request(const pjsip_method *method, struct ast_sip_endpoint *endpoint,
1432                 const char *uri, pjsip_tx_data **tdata)
1433 {
1434         RAII_VAR(struct ast_sip_contact *, contact, NULL, ao2_cleanup);
1435         pj_str_t remote_uri;
1436         pj_str_t from;
1437         pj_pool_t *pool;
1438         pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1439
1440         if (ast_strlen_zero(uri)) {
1441                 if (!endpoint) {
1442                         ast_log(LOG_ERROR, "An endpoint and/or uri must be specified\n");
1443                         return -1;
1444                 }
1445
1446                 contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
1447                 if (!contact || ast_strlen_zero(contact->uri)) {
1448                         ast_log(LOG_ERROR, "Unable to retrieve contact for endpoint %s\n",
1449                                         ast_sorcery_object_get_id(endpoint));
1450                         return -1;
1451                 }
1452
1453                 pj_cstr(&remote_uri, contact->uri);
1454         } else {
1455                 pj_cstr(&remote_uri, uri);
1456         }
1457
1458         if (endpoint) {
1459                 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1460                         ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport selector for endpoint %s\n",
1461                                 ast_sorcery_object_get_id(endpoint));
1462                         return -1;
1463                 }
1464         }
1465
1466         pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "Outbound request", 256, 256);
1467
1468         if (!pool) {
1469                 ast_log(LOG_ERROR, "Unable to create PJLIB memory pool\n");
1470                 return -1;
1471         }
1472
1473         if (sip_dialog_create_from(pool, &from, endpoint ? endpoint->fromuser : NULL,
1474                                 endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) {
1475                 ast_log(LOG_ERROR, "Unable to create From header for %.*s request to endpoint %s\n",
1476                                 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1477                 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1478                 return -1;
1479         }
1480
1481         if (pjsip_endpt_create_request(ast_sip_get_pjsip_endpoint(), method, &remote_uri,
1482                         &from, &remote_uri, &from, NULL, -1, NULL, tdata) != PJ_SUCCESS) {
1483                 ast_log(LOG_ERROR, "Unable to create outbound %.*s request to endpoint %s\n",
1484                                 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1485                 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1486                 return -1;
1487         }
1488
1489         /* We can release this pool since request creation copied all the necessary
1490          * data into the outbound request's pool
1491          */
1492         pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1493         return 0;
1494 }
1495
1496 int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
1497                 struct ast_sip_endpoint *endpoint, const char *uri,
1498                 pjsip_tx_data **tdata)
1499 {
1500         const pjsip_method *pmethod = get_pjsip_method(method);
1501
1502         if (!pmethod) {
1503                 ast_log(LOG_WARNING, "Unknown method '%s'. Cannot send request\n", method);
1504                 return -1;
1505         }
1506
1507         if (dlg) {
1508                 return create_in_dialog_request(pmethod, dlg, tdata);
1509         } else {
1510                 return create_out_of_dialog_request(pmethod, endpoint, uri, tdata);
1511         }
1512 }
1513
1514 static int send_in_dialog_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg)
1515 {
1516         if (pjsip_dlg_send_request(dlg, tdata, -1, NULL) != PJ_SUCCESS) {
1517                 ast_log(LOG_WARNING, "Unable to send in-dialog request.\n");
1518                 return -1;
1519         }
1520         return 0;
1521 }
1522
1523 static void send_request_cb(void *token, pjsip_event *e)
1524 {
1525         RAII_VAR(struct ast_sip_endpoint *, endpoint, token, ao2_cleanup);
1526         pjsip_transaction *tsx = e->body.tsx_state.tsx;
1527         pjsip_rx_data *challenge = e->body.tsx_state.src.rdata;
1528         pjsip_tx_data *tdata;
1529
1530         if (tsx->status_code != 401 && tsx->status_code != 407) {
1531                 return;
1532         }
1533
1534         if (!ast_sip_create_request_with_auth(&endpoint->outbound_auths, challenge, tsx, &tdata)) {
1535                 pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, NULL, NULL);
1536         }
1537 }
1538
1539 static int send_out_of_dialog_request(pjsip_tx_data *tdata, struct ast_sip_endpoint *endpoint)
1540 {
1541         ao2_ref(endpoint, +1);
1542         if (pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, endpoint, send_request_cb) != PJ_SUCCESS) {
1543                 ast_log(LOG_ERROR, "Error attempting to send outbound %.*s request to endpoint %s\n",
1544                                 (int) pj_strlen(&tdata->msg->line.req.method.name),
1545                                 pj_strbuf(&tdata->msg->line.req.method.name),
1546                                 ast_sorcery_object_get_id(endpoint));
1547                 ao2_ref(endpoint, -1);
1548                 return -1;
1549         }
1550
1551         return 0;
1552 }
1553
1554 int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg, struct ast_sip_endpoint *endpoint)
1555 {
1556         ast_assert(tdata->msg->type == PJSIP_REQUEST_MSG);
1557
1558         if (dlg) {
1559                 return send_in_dialog_request(tdata, dlg);
1560         } else {
1561                 return send_out_of_dialog_request(tdata, endpoint);
1562         }
1563 }
1564
1565 int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value)
1566 {
1567         pj_str_t hdr_name;
1568         pj_str_t hdr_value;
1569         pjsip_generic_string_hdr *hdr;
1570
1571         pj_cstr(&hdr_name, name);
1572         pj_cstr(&hdr_value, value);
1573
1574         hdr = pjsip_generic_string_hdr_create(tdata->pool, &hdr_name, &hdr_value);
1575
1576         pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *) hdr);
1577         return 0;
1578 }
1579
1580 static pjsip_msg_body *ast_body_to_pjsip_body(pj_pool_t *pool, const struct ast_sip_body *body)
1581 {
1582         pj_str_t type;
1583         pj_str_t subtype;
1584         pj_str_t body_text;
1585
1586         pj_cstr(&type, body->type);
1587         pj_cstr(&subtype, body->subtype);
1588         pj_cstr(&body_text, body->body_text);
1589
1590         return pjsip_msg_body_create(pool, &type, &subtype, &body_text);
1591 }
1592
1593 int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body)
1594 {
1595         pjsip_msg_body *pjsip_body = ast_body_to_pjsip_body(tdata->pool, body);
1596         tdata->msg->body = pjsip_body;
1597         return 0;
1598 }
1599
1600 int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies)
1601 {
1602         int i;
1603         /* NULL for type and subtype automatically creates "multipart/mixed" */
1604         pjsip_msg_body *body = pjsip_multipart_create(tdata->pool, NULL, NULL);
1605
1606         for (i = 0; i < num_bodies; ++i) {
1607                 pjsip_multipart_part *part = pjsip_multipart_create_part(tdata->pool);
1608                 part->body = ast_body_to_pjsip_body(tdata->pool, bodies[i]);
1609                 pjsip_multipart_add_part(tdata->pool, body, part);
1610         }
1611
1612         tdata->msg->body = body;
1613         return 0;
1614 }
1615
1616 int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text)
1617 {
1618         size_t combined_size = strlen(body_text) + tdata->msg->body->len;
1619         struct ast_str *body_buffer = ast_str_alloca(combined_size);
1620
1621         ast_str_set(&body_buffer, 0, "%.*s%s", (int) tdata->msg->body->len, (char *) tdata->msg->body->data, body_text);
1622
1623         tdata->msg->body->data = pj_pool_alloc(tdata->pool, combined_size);
1624         pj_memcpy(tdata->msg->body->data, ast_str_buffer(body_buffer), combined_size);
1625         tdata->msg->body->len = combined_size;
1626
1627         return 0;
1628 }
1629
1630 struct ast_taskprocessor *ast_sip_create_serializer(void)
1631 {
1632         struct ast_taskprocessor *serializer;
1633         RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1634         char name[AST_UUID_STR_LEN];
1635
1636         if (!uuid) {
1637                 return NULL;
1638         }
1639
1640         ast_uuid_to_str(uuid, name, sizeof(name));
1641
1642         serializer = ast_threadpool_serializer(name, sip_threadpool);
1643         if (!serializer) {
1644                 return NULL;
1645         }
1646         return serializer;
1647 }
1648
1649 int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1650 {
1651         if (serializer) {
1652                 return ast_taskprocessor_push(serializer, sip_task, task_data);
1653         } else {
1654                 return ast_threadpool_push(sip_threadpool, sip_task, task_data);
1655         }
1656 }
1657
1658 struct sync_task_data {
1659         ast_mutex_t lock;
1660         ast_cond_t cond;
1661         int complete;
1662         int fail;
1663         int (*task)(void *);
1664         void *task_data;
1665 };
1666
1667 static int sync_task(void *data)
1668 {
1669         struct sync_task_data *std = data;
1670         std->fail = std->task(std->task_data);
1671
1672         ast_mutex_lock(&std->lock);
1673         std->complete = 1;
1674         ast_cond_signal(&std->cond);
1675         ast_mutex_unlock(&std->lock);
1676         return std->fail;
1677 }
1678
1679 int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1680 {
1681         /* This method is an onion */
1682         struct sync_task_data std;
1683         ast_mutex_init(&std.lock);
1684         ast_cond_init(&std.cond, NULL);
1685         std.fail = std.complete = 0;
1686         std.task = sip_task;
1687         std.task_data = task_data;
1688
1689         if (serializer) {
1690                 if (ast_taskprocessor_push(serializer, sync_task, &std)) {
1691                         return -1;
1692                 }
1693         } else {
1694                 if (ast_threadpool_push(sip_threadpool, sync_task, &std)) {
1695                         return -1;
1696                 }
1697         }
1698
1699         ast_mutex_lock(&std.lock);
1700         while (!std.complete) {
1701                 ast_cond_wait(&std.cond, &std.lock);
1702         }
1703         ast_mutex_unlock(&std.lock);
1704
1705         ast_mutex_destroy(&std.lock);
1706         ast_cond_destroy(&std.cond);
1707         return std.fail;
1708 }
1709
1710 void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size)
1711 {
1712         size_t chars_to_copy = MIN(size - 1, pj_strlen(src));
1713         memcpy(dest, pj_strbuf(src), chars_to_copy);
1714         dest[chars_to_copy] = '\0';
1715 }
1716
1717 int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype)
1718 {
1719         pjsip_media_type compare;
1720
1721         if (!content_type) {
1722                 return 0;
1723         }
1724
1725         pjsip_media_type_init2(&compare, type, subtype);
1726
1727         return pjsip_media_type_cmp(content_type, &compare, 0) ? -1 : 0;
1728 }
1729
1730 pj_caching_pool caching_pool;
1731 pj_pool_t *memory_pool;
1732 pj_thread_t *monitor_thread;
1733 static int monitor_continue;
1734
1735 static void *monitor_thread_exec(void *endpt)
1736 {
1737         while (monitor_continue) {
1738                 const pj_time_val delay = {0, 10};
1739                 pjsip_endpt_handle_events(ast_pjsip_endpoint, &delay);
1740         }
1741         return NULL;
1742 }
1743
1744 static void stop_monitor_thread(void)
1745 {
1746         monitor_continue = 0;
1747         pj_thread_join(monitor_thread);
1748 }
1749
1750 AST_THREADSTORAGE(pj_thread_storage);
1751 AST_THREADSTORAGE(servant_id_storage);
1752 #define SIP_SERVANT_ID 0x5E2F1D
1753
1754 static void sip_thread_start(void)
1755 {
1756         pj_thread_desc *desc;
1757         pj_thread_t *thread;
1758         uint32_t *servant_id;
1759
1760         servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1761         if (!servant_id) {
1762                 ast_log(LOG_ERROR, "Could not set SIP servant ID in thread-local storage.\n");
1763                 return;
1764         }
1765         *servant_id = SIP_SERVANT_ID;
1766
1767         desc = ast_threadstorage_get(&pj_thread_storage, sizeof(pj_thread_desc));
1768         if (!desc) {
1769                 ast_log(LOG_ERROR, "Could not get thread desc from thread-local storage. Expect awful things to occur\n");
1770                 return;
1771         }
1772         pj_bzero(*desc, sizeof(*desc));
1773
1774         if (pj_thread_register("Asterisk Thread", *desc, &thread) != PJ_SUCCESS) {
1775                 ast_log(LOG_ERROR, "Couldn't register thread with PJLIB.\n");
1776         }
1777 }
1778
1779 int ast_sip_thread_is_servant(void)
1780 {
1781         uint32_t *servant_id;
1782
1783         servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1784         if (!servant_id) {
1785                 return 0;
1786         }
1787
1788         return *servant_id == SIP_SERVANT_ID;
1789 }
1790
1791 static void remove_request_headers(pjsip_endpoint *endpt)
1792 {
1793         const pjsip_hdr *request_headers = pjsip_endpt_get_request_headers(endpt);
1794         pjsip_hdr *iter = request_headers->next;
1795
1796         while (iter != request_headers) {
1797                 pjsip_hdr *to_erase = iter;
1798                 iter = iter->next;
1799                 pj_list_erase(to_erase);
1800         }
1801 }
1802
1803 static int load_module(void)
1804 {
1805         /* The third parameter is just copied from
1806          * example code from PJLIB. This can be adjusted
1807          * if necessary.
1808          */
1809         pj_status_t status;
1810         struct ast_threadpool_options options;
1811
1812         if (pj_init() != PJ_SUCCESS) {
1813                 return AST_MODULE_LOAD_DECLINE;
1814         }
1815
1816         if (pjlib_util_init() != PJ_SUCCESS) {
1817                 pj_shutdown();
1818                 return AST_MODULE_LOAD_DECLINE;
1819         }
1820
1821         pj_caching_pool_init(&caching_pool, NULL, 1024 * 1024);
1822         if (pjsip_endpt_create(&caching_pool.factory, "SIP", &ast_pjsip_endpoint) != PJ_SUCCESS) {
1823                 ast_log(LOG_ERROR, "Failed to create PJSIP endpoint structure. Aborting load\n");
1824                 goto error;
1825         }
1826
1827         /* PJSIP will automatically try to add a Max-Forwards header. Since we want to control that,
1828          * we need to stop PJSIP from doing it automatically
1829          */
1830         remove_request_headers(ast_pjsip_endpoint);
1831
1832         memory_pool = pj_pool_create(&caching_pool.factory, "SIP", 1024, 1024, NULL);
1833         if (!memory_pool) {
1834                 ast_log(LOG_ERROR, "Failed to create memory pool for SIP. Aborting load\n");
1835                 goto error;
1836         }
1837
1838         if (ast_sip_initialize_system()) {
1839                 ast_log(LOG_ERROR, "Failed to initialize SIP system configuration. Aborting load\n");
1840                 goto error;
1841         }
1842
1843         sip_get_threadpool_options(&options);
1844         options.thread_start = sip_thread_start;
1845         sip_threadpool = ast_threadpool_create("SIP", NULL, &options);
1846         if (!sip_threadpool) {
1847                 ast_log(LOG_ERROR, "Failed to create SIP threadpool. Aborting load\n");
1848                 goto error;
1849         }
1850
1851         pjsip_tsx_layer_init_module(ast_pjsip_endpoint);
1852         pjsip_ua_init_module(ast_pjsip_endpoint, NULL);
1853
1854         monitor_continue = 1;
1855         status = pj_thread_create(memory_pool, "SIP", (pj_thread_proc *) &monitor_thread_exec,
1856                         NULL, PJ_THREAD_DEFAULT_STACK_SIZE * 2, 0, &monitor_thread);
1857         if (status != PJ_SUCCESS) {
1858                 ast_log(LOG_ERROR, "Failed to start SIP monitor thread. Aborting load\n");
1859                 goto error;
1860         }
1861
1862         ast_sip_initialize_global_headers();
1863
1864         if (ast_res_pjsip_initialize_configuration()) {
1865                 ast_log(LOG_ERROR, "Failed to initialize SIP configuration. Aborting load\n");
1866                 goto error;
1867         }
1868
1869         if (ast_sip_initialize_distributor()) {
1870                 ast_log(LOG_ERROR, "Failed to register distributor module. Aborting load\n");
1871                 goto error;
1872         }
1873
1874         if (ast_sip_initialize_outbound_authentication()) {
1875                 ast_log(LOG_ERROR, "Failed to initialize outbound authentication. Aborting load\n");
1876                 goto error;
1877         }
1878
1879         ast_res_pjsip_init_options_handling(0);
1880
1881         ast_res_pjsip_init_contact_transports();
1882
1883 return AST_MODULE_LOAD_SUCCESS;
1884
1885 error:
1886         ast_sip_destroy_distributor();
1887         ast_res_pjsip_destroy_configuration();
1888         ast_sip_destroy_global_headers();
1889         if (monitor_thread) {
1890                 stop_monitor_thread();
1891         }
1892         if (memory_pool) {
1893                 pj_pool_release(memory_pool);
1894                 memory_pool = NULL;
1895         }
1896         if (ast_pjsip_endpoint) {
1897                 pjsip_endpt_destroy(ast_pjsip_endpoint);
1898                 ast_pjsip_endpoint = NULL;
1899         }
1900         pj_caching_pool_destroy(&caching_pool);
1901         return AST_MODULE_LOAD_DECLINE;
1902 }
1903
1904 static int reload_module(void)
1905 {
1906         if (ast_res_pjsip_reload_configuration()) {
1907                 return AST_MODULE_LOAD_DECLINE;
1908         }
1909         ast_res_pjsip_init_options_handling(1);
1910         return 0;
1911 }
1912
1913 static int unload_pjsip(void *data)
1914 {
1915         if (memory_pool) {
1916                 pj_pool_release(memory_pool);
1917                 memory_pool = NULL;
1918         }
1919         if (ast_pjsip_endpoint) {
1920                 pjsip_endpt_destroy(ast_pjsip_endpoint);
1921                 ast_pjsip_endpoint = NULL;
1922         }
1923         pj_caching_pool_destroy(&caching_pool);
1924         return 0;
1925 }
1926
1927 static int unload_module(void)
1928 {
1929         ast_res_pjsip_cleanup_options_handling();
1930         ast_sip_destroy_distributor();
1931         ast_res_pjsip_destroy_configuration();
1932         ast_sip_destroy_global_headers();
1933         if (monitor_thread) {
1934                 stop_monitor_thread();
1935         }
1936         /* The thread this is called from cannot call PJSIP/PJLIB functions,
1937          * so we have to push the work to the threadpool to handle
1938          */
1939         ast_sip_push_task_synchronous(NULL, unload_pjsip, NULL);
1940
1941         ast_threadpool_shutdown(sip_threadpool);
1942
1943         return 0;
1944 }
1945
1946 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Basic SIP resource",
1947                 .load = load_module,
1948                 .unload = unload_module,
1949                 .reload = reload_module,
1950                 .load_pri = AST_MODPRI_CHANNEL_DEPEND - 5,
1951 );