2 * Asterisk -- An open source telephony toolkit.
4 * Copyright (C) 2013, Digium, Inc.
6 * Mark Michelson <mmichelson@digium.com>
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.
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.
22 /* Needed for SUBSCRIBE, NOTIFY, and PUBLISH method definitions */
23 #include <pjsip_simple.h>
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"
40 <depend>pjproject</depend>
41 <depend>res_sorcery_config</depend>
42 <support_level>core</support_level>
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>
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.
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.
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.
72 <configOption name="100rel" default="yes">
73 <synopsis>Allow support for RFC3262 provisional ACK tags</synopsis>
77 <enum name="required" />
82 <configOption name="aggregate_mwi" default="yes">
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>
88 <configOption name="allow">
89 <synopsis>Media Codec(s) to allow</synopsis>
91 <configOption name="aors">
92 <synopsis>AoR(s) to be used with the endpoint</synopsis>
94 List of comma separated AoRs that the endpoint should be associated with.
97 <configOption name="auth">
98 <synopsis>Authentication Object(s) associated with the endpoint</synopsis>
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.
103 Endpoints without an <literal>authentication</literal> object
104 configured will allow connections without vertification.
105 </para></description>
107 <configOption name="callerid">
108 <synopsis>CallerID information for the endpoint</synopsis>
110 Must be in the format <literal>Name <Number></literal>,
111 or only <literal><Number></literal>.
112 </para></description>
114 <configOption name="callerid_privacy">
115 <synopsis>Default privacy level</synopsis>
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" />
130 <configOption name="callerid_tag">
131 <synopsis>Internal id_tag for the endpoint</synopsis>
133 <configOption name="context">
134 <synopsis>Dialplan context for inbound sessions</synopsis>
136 <configOption name="direct_media_glare_mitigation" default="none">
137 <synopsis>Mitigation of direct media (re)INVITE glare</synopsis>
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
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
152 <enum name="outgoing" />
153 <enum name="incoming" />
157 <configOption name="direct_media_method" default="invite">
158 <synopsis>Direct Media method type</synopsis>
160 <para>Method for setting up Direct Media between endpoints.</para>
162 <enum name="invite" />
163 <enum name="reinvite">
164 <para>Alias for the <literal>invite</literal> value.</para>
166 <enum name="update" />
170 <configOption name="connected_line_method" default="invite">
171 <synopsis>Connected line method type</synopsis>
173 <para>Method used when updating connected line information.</para>
175 <enum name="invite" />
176 <enum name="reinvite">
177 <para>Alias for the <literal>invite</literal> value.</para>
179 <enum name="update" />
183 <configOption name="direct_media" default="yes">
184 <synopsis>Determines whether media may flow directly between endpoints.</synopsis>
186 <configOption name="disable_direct_media_on_nat" default="no">
187 <synopsis>Disable direct media session refreshes when NAT obstructs the media session</synopsis>
189 <configOption name="disallow">
190 <synopsis>Media Codec(s) to disallow</synopsis>
192 <configOption name="dtmf_mode" default="rfc4733">
193 <synopsis>DTMF mode</synopsis>
195 <para>This setting allows to choose the DTMF mode for endpoint communication.</para>
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>
203 <para>DTMF is sent as part of audio stream.</para>
206 <para>DTMF is sent as SIP INFO packets.</para>
211 <configOption name="media_address">
212 <synopsis>IP address used in SDP for media handling</synopsis>
214 At the time of SDP creation, the IP address defined here will be used as
215 the media address for individual streams in the SDP.
218 Be aware that the <literal>external_media_address</literal> option, set in Transport
219 configuration, can also affect the final media address used in the SDP.
223 <configOption name="force_rport" default="yes">
224 <synopsis>Force use of return port</synopsis>
226 <configOption name="ice_support" default="no">
227 <synopsis>Enable the ICE mechanism to help traverse NAT</synopsis>
229 <configOption name="identify_by" default="username,location">
230 <synopsis>Way(s) for Endpoint to be identified</synopsis>
232 An endpoint can be identified in multiple ways. Currently, the only supported
233 option is <literal>username</literal>, which matches the endpoint based on the
234 username in the From header.
236 <note><para>Endpoints can also be identified by IP address; however, that method
237 of identification is not handled by this configuration option. See the documentation
238 for the <literal>identify</literal> configuration section for more details on that
239 method of endpoint identification. If this option is set to <literal>username</literal>
240 and an <literal>identify</literal> configuration section exists for the endpoint, then
241 the endpoint can be identified in multiple ways.</para></note>
243 <enum name="username" />
247 <configOption name="redirect_method">
248 <synopsis>How redirects received from an endpoint are handled</synopsis>
250 When a redirect is received from an endpoint there are multiple ways it can be handled.
251 If this option is set to <literal>user</literal> the user portion of the redirect target
252 is treated as an extension within the dialplan and dialed using a Local channel. If this option
253 is set to <literal>uri_core</literal> the target URI is returned to the dialing application
254 which dials it using the PJSIP channel driver and endpoint originally used. If this option is
255 set to <literal>uri_pjsip</literal> the redirect occurs within chan_pjsip itself and is not exposed
256 to the core at all. The <literal>uri_pjsip</literal> option has the benefit of being more efficient
257 and also supporting multiple potential redirect targets. The con is that since redirection occurs
258 within chan_pjsip redirecting information is not forwarded and redirection can not be
263 <enum name="uri_core" />
264 <enum name="uri_pjsip" />
268 <configOption name="mailboxes">
269 <synopsis>NOTIFY the endpoint when state changes for any of the specified mailboxes</synopsis>
271 Asterisk will send unsolicited MWI NOTIFY messages to the endpoint when state
272 changes happen for any of the specified mailboxes. More than one mailbox can be
273 specified with a comma-delimited string. Mailboxes must be specified as <mailbox>@<context>.
274 For endpoints that SUBSCRIBE for MWI, you can set the <literal>mailboxes</literal> option in your AOR
276 </para></description>
278 <configOption name="moh_suggest" default="default">
279 <synopsis>Default Music On Hold class</synopsis>
281 <configOption name="outbound_auth">
282 <synopsis>Authentication object used for outbound requests</synopsis>
284 <configOption name="outbound_proxy">
285 <synopsis>Proxy through which to send requests, a full SIP URI must be provided</synopsis>
287 <configOption name="rewrite_contact">
288 <synopsis>Allow Contact header to be rewritten with the source IP address-port</synopsis>
290 On inbound SIP messages from this endpoint, the Contact header will be changed to have the
291 source IP address and port. This option does not affect outbound messages send to this
293 </para></description>
295 <configOption name="rtp_ipv6" default="no">
296 <synopsis>Allow use of IPv6 for RTP traffic</synopsis>
298 <configOption name="rtp_symmetric" default="no">
299 <synopsis>Enforce that RTP must be symmetric</synopsis>
301 <configOption name="send_diversion" default="yes">
302 <synopsis>Send the Diversion header, conveying the diversion
303 information to the called user agent</synopsis>
305 <configOption name="send_pai" default="no">
306 <synopsis>Send the P-Asserted-Identity header</synopsis>
308 <configOption name="send_rpid" default="no">
309 <synopsis>Send the Remote-Party-ID header</synopsis>
311 <configOption name="timers_min_se" default="90">
312 <synopsis>Minimum session timers expiration period</synopsis>
314 Minimium session timer expiration period. Time in seconds.
315 </para></description>
317 <configOption name="timers" default="yes">
318 <synopsis>Session timers for SIP packets</synopsis>
321 <enum name="forced" />
323 <enum name="required" />
328 <configOption name="timers_sess_expires" default="1800">
329 <synopsis>Maximum session timer expiration period</synopsis>
331 Maximium session timer expiration period. Time in seconds.
332 </para></description>
334 <configOption name="transport">
335 <synopsis>Desired transport configuration</synopsis>
337 This will set the desired transport configuration to send SIP data through.
339 <warning><para>Not specifying a transport will <emphasis>DEFAULT</emphasis>
340 to the first configured transport in <filename>pjsip.conf</filename> which is
341 valid for the URI we are trying to contact.
343 <warning><para>Transport configuration is not affected by reloads. In order to
344 change transports, a full Asterisk restart is required</para></warning>
347 <configOption name="trust_id_inbound" default="no">
348 <synopsis>Accept identification information received from this endpoint</synopsis>
349 <description><para>This option determines whether Asterisk will accept
350 identification from the endpoint from headers such as P-Asserted-Identity
351 or Remote-Party-ID header. This option applies both to calls originating from the
352 endpoint and calls originating from Asterisk. If <literal>no</literal>, the
353 configured Caller-ID from pjsip.conf will always be used as the identity for
354 the endpoint.</para></description>
356 <configOption name="trust_id_outbound" default="no">
357 <synopsis>Send private identification details to the endpoint.</synopsis>
358 <description><para>This option determines whether res_pjsip will send private
359 identification information to the endpoint. If <literal>no</literal>,
360 private Caller-ID information will not be forwarded to the endpoint.
361 "Private" in this case refers to any method of restricting identification.
362 Example: setting <replaceable>callerid_privacy</replaceable> to any
363 <literal>prohib</literal> variation.
364 Example: If <replaceable>trust_id_inbound</replaceable> is set to
365 <literal>yes</literal>, the presence of a <literal>Privacy: id</literal>
366 header in a SIP request or response would indicate the identification
367 provided in the request is private.</para></description>
369 <configOption name="type">
370 <synopsis>Must be of type 'endpoint'.</synopsis>
372 <configOption name="use_ptime" default="no">
373 <synopsis>Use Endpoint's requested packetisation interval</synopsis>
375 <configOption name="use_avpf" default="no">
376 <synopsis>Determines whether res_pjsip will use and enforce usage of AVPF for this
379 If set to <literal>yes</literal>, res_pjsip will use use the AVPF or SAVPF RTP
380 profile for all media offers on outbound calls and media updates and will
381 decline media offers not using the AVPF or SAVPF profile.
383 If set to <literal>no</literal>, res_pjsip will use use the AVP or SAVP RTP
384 profile for all media offers on outbound calls and media updates and will
385 decline media offers not using the AVP or SAVP profile.
386 </para></description>
388 <configOption name="media_encryption" default="no">
389 <synopsis>Determines whether res_pjsip will use and enforce usage of media encryption
390 for this endpoint.</synopsis>
393 <enum name="no"><para>
394 res_pjsip will offer no encryption and allow no encryption to be setup.
396 <enum name="sdes"><para>
397 res_pjsip will offer standard SRTP setup via in-SDP keys. Encrypted SIP
398 transport should be used in conjunction with this option to prevent
399 exposure of media encryption keys.
401 <enum name="dtls"><para>
402 res_pjsip will offer DTLS-SRTP setup.
407 <configOption name="inband_progress" default="no">
408 <synopsis>Determines whether chan_pjsip will indicate ringing using inband
411 If set to <literal>yes</literal>, chan_pjsip will send a 183 Session Progress
412 when told to indicate ringing and will immediately start sending ringing
415 If set to <literal>no</literal>, chan_pjsip will send a 180 Ringing when told
416 to indicate ringing and will NOT send it as audio.
417 </para></description>
419 <configOption name="call_group">
420 <synopsis>The numeric pickup groups for a channel.</synopsis>
422 Can be set to a comma separated list of numbers or ranges between the values
423 of 0-63 (maximum of 64 groups).
424 </para></description>
426 <configOption name="pickup_group">
427 <synopsis>The numeric pickup groups that a channel can pickup.</synopsis>
429 Can be set to a comma separated list of numbers or ranges between the values
430 of 0-63 (maximum of 64 groups).
431 </para></description>
433 <configOption name="named_call_group">
434 <synopsis>The named pickup groups for a channel.</synopsis>
436 Can be set to a comma separated list of case sensitive strings limited by
437 supported line length.
438 </para></description>
440 <configOption name="named_pickup_group">
441 <synopsis>The named pickup groups that a channel can pickup.</synopsis>
443 Can be set to a comma separated list of case sensitive strings limited by
444 supported line length.
445 </para></description>
447 <configOption name="device_state_busy_at" default="0">
448 <synopsis>The number of in-use channels which will cause busy to be returned as device state</synopsis>
450 When the number of in-use channels for the endpoint matches the devicestate_busy_at setting the
451 PJSIP channel driver will return busy as the device state instead of in use.
452 </para></description>
454 <configOption name="t38_udptl" default="no">
455 <synopsis>Whether T.38 UDPTL support is enabled or not</synopsis>
457 If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted
459 </para></description>
461 <configOption name="t38_udptl_ec" default="none">
462 <synopsis>T.38 UDPTL error correction method</synopsis>
465 <enum name="none"><para>
466 No error correction should be used.
468 <enum name="fec"><para>
469 Forward error correction should be used.
471 <enum name="redundancy"><para>
472 Redundacy error correction should be used.
477 <configOption name="t38_udptl_maxdatagram" default="0">
478 <synopsis>T.38 UDPTL maximum datagram size</synopsis>
480 This option can be set to override the maximum datagram of a remote endpoint for broken
482 </para></description>
484 <configOption name="fax_detect" default="no">
485 <synopsis>Whether CNG tone detection is enabled</synopsis>
487 This option can be set to send the session to the fax extension when a CNG tone is
489 </para></description>
491 <configOption name="t38_udptl_nat" default="no">
492 <synopsis>Whether NAT support is enabled on UDPTL sessions</synopsis>
494 When enabled the UDPTL stack will send UDPTL packets to the source address of
496 </para></description>
498 <configOption name="t38_udptl_ipv6" default="no">
499 <synopsis>Whether IPv6 is used for UDPTL Sessions</synopsis>
501 When enabled the UDPTL stack will use IPv6.
502 </para></description>
504 <configOption name="tone_zone">
505 <synopsis>Set which country's indications to use for channels created for this endpoint.</synopsis>
507 <configOption name="language">
508 <synopsis>Set the default language to use for channels created for this endpoint.</synopsis>
510 <configOption name="one_touch_recording" default="no">
511 <synopsis>Determines whether one-touch recording is allowed for this endpoint.</synopsis>
513 <ref type="configOption">recordonfeature</ref>
514 <ref type="configOption">recordofffeature</ref>
517 <configOption name="record_on_feature" default="automixmon">
518 <synopsis>The feature to enact when one-touch recording is turned on.</synopsis>
520 <para>When an INFO request for one-touch recording arrives with a Record header set to "on", this
521 feature will be enabled for the channel. The feature designated here can be any built-in
522 or dynamic feature defined in features.conf.</para>
523 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
526 <ref type="configOption">one_touch_recording</ref>
527 <ref type="configOption">recordofffeature</ref>
530 <configOption name="record_off_feature" default="automixmon">
531 <synopsis>The feature to enact when one-touch recording is turned off.</synopsis>
533 <para>When an INFO request for one-touch recording arrives with a Record header set to "off", this
534 feature will be enabled for the channel. The feature designated here can be any built-in
535 or dynamic feature defined in features.conf.</para>
536 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
539 <ref type="configOption">one_touch_recording</ref>
540 <ref type="configOption">recordonfeature</ref>
543 <configOption name="rtp_engine" default="asterisk">
544 <synopsis>Name of the RTP engine to use for channels created for this endpoint</synopsis>
546 <configOption name="allow_transfer" default="yes">
547 <synopsis>Determines whether SIP REFER transfers are allowed for this endpoint</synopsis>
549 <configOption name="sdp_owner" default="-">
550 <synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis>
552 <configOption name="sdp_session" default="Asterisk">
553 <synopsis>String used for the SDP session (s=) line.</synopsis>
555 <configOption name="tos_audio">
556 <synopsis>DSCP TOS bits for audio streams</synopsis>
558 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
559 </para></description>
561 <configOption name="tos_video">
562 <synopsis>DSCP TOS bits for video streams</synopsis>
564 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
565 </para></description>
567 <configOption name="cos_audio">
568 <synopsis>Priority for audio streams</synopsis>
570 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
571 </para></description>
573 <configOption name="cos_video">
574 <synopsis>Priority for video streams</synopsis>
576 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
577 </para></description>
579 <configOption name="allow_subscribe" default="yes">
580 <synopsis>Determines if endpoint is allowed to initiate subscriptions with Asterisk.</synopsis>
582 <configOption name="sub_min_expiry" default="60">
583 <synopsis>The minimum allowed expiry time for subscriptions initiated by the endpoint.</synopsis>
585 <configOption name="from_user">
586 <synopsis>Username to use in From header for requests to this endpoint.</synopsis>
588 <configOption name="mwi_from_user">
589 <synopsis>Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.</synopsis>
591 <configOption name="from_domain">
592 <synopsis>Domain to user in From header for requests to this endpoint.</synopsis>
594 <configOption name="dtls_verify">
595 <synopsis>Verify that the provided peer certificate is valid</synopsis>
597 This option only applies if <replaceable>media_encryption</replaceable> is
598 set to <literal>dtls</literal>.
599 </para></description>
601 <configOption name="dtls_rekey">
602 <synopsis>Interval at which to renegotiate the TLS session and rekey the SRTP session</synopsis>
604 This option only applies if <replaceable>media_encryption</replaceable> is
605 set to <literal>dtls</literal>.
607 If this is not set or the value provided is 0 rekeying will be disabled.
608 </para></description>
610 <configOption name="dtls_cert_file">
611 <synopsis>Path to certificate file to present to peer</synopsis>
613 This option only applies if <replaceable>media_encryption</replaceable> is
614 set to <literal>dtls</literal>.
615 </para></description>
617 <configOption name="dtls_private_key">
618 <synopsis>Path to private key for certificate file</synopsis>
620 This option only applies if <replaceable>media_encryption</replaceable> is
621 set to <literal>dtls</literal>.
622 </para></description>
624 <configOption name="dtls_cipher">
625 <synopsis>Cipher to use for DTLS negotiation</synopsis>
627 This option only applies if <replaceable>media_encryption</replaceable> is
628 set to <literal>dtls</literal>.
630 Many options for acceptable ciphers. See link for more:
631 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
632 </para></description>
634 <configOption name="dtls_ca_file">
635 <synopsis>Path to certificate authority certificate</synopsis>
637 This option only applies if <replaceable>media_encryption</replaceable> is
638 set to <literal>dtls</literal>.
639 </para></description>
641 <configOption name="dtls_ca_path">
642 <synopsis>Path to a directory containing certificate authority certificates</synopsis>
644 This option only applies if <replaceable>media_encryption</replaceable> is
645 set to <literal>dtls</literal>.
646 </para></description>
648 <configOption name="dtls_setup">
649 <synopsis>Whether we are willing to accept connections, connect to the other party, or both.</synopsis>
652 This option only applies if <replaceable>media_encryption</replaceable> is
653 set to <literal>dtls</literal>.
656 <enum name="active"><para>
657 res_pjsip will make a connection to the peer.
659 <enum name="passive"><para>
660 res_pjsip will accept connections from the peer.
662 <enum name="actpass"><para>
663 res_pjsip will offer and accept connections from the peer.
668 <configOption name="srtp_tag_32">
669 <synopsis>Determines whether 32 byte tags should be used instead of 80 byte tags.</synopsis>
671 This option only applies if <replaceable>media_encryption</replaceable> is
672 set to <literal>sdes</literal> or <literal>dtls</literal>.
673 </para></description>
675 <configOption name="set_var">
676 <synopsis>Variable set on a channel involving the endpoint.</synopsis>
678 When a new channel is created using the endpoint set the specified
679 variable(s) on that channel. For multiple channel variables specify
680 multiple 'set_var'(s).
681 </para></description>
684 <configObject name="auth">
685 <synopsis>Authentication type</synopsis>
687 Authentication objects hold the authentication information for use
688 by other objects such as <literal>endpoints</literal> or <literal>registrations</literal>.
689 This also allows for multiple objects to use a single auth object. See
690 the <literal>auth_type</literal> config option for password style choices.
691 </para></description>
692 <configOption name="auth_type" default="userpass">
693 <synopsis>Authentication type</synopsis>
695 This option specifies which of the password style config options should be read
696 when trying to authenticate an endpoint inbound request. If set to <literal>userpass</literal>
697 then we'll read from the 'password' option. For <literal>md5</literal> we'll read
702 <enum name="userpass"/>
706 <configOption name="nonce_lifetime" default="32">
707 <synopsis>Lifetime of a nonce associated with this authentication config.</synopsis>
709 <configOption name="md5_cred">
710 <synopsis>MD5 Hash used for authentication.</synopsis>
711 <description><para>Only used when auth_type is <literal>md5</literal>.</para></description>
713 <configOption name="password">
714 <synopsis>PlainText password used for authentication.</synopsis>
715 <description><para>Only used when auth_type is <literal>userpass</literal>.</para></description>
717 <configOption name="realm" default="asterisk">
718 <synopsis>SIP realm for endpoint</synopsis>
720 <configOption name="type">
721 <synopsis>Must be 'auth'</synopsis>
723 <configOption name="username">
724 <synopsis>Username to use for account</synopsis>
727 <configObject name="domain_alias">
728 <synopsis>Domain Alias</synopsis>
730 Signifies that a domain is an alias. If the domain on a session is
731 not found to match an AoR then this object is used to see if we have
732 an alias for the AoR to which the endpoint is binding. This objects
733 name as defined in configuration should be the domain alias and a
734 config option is provided to specify the domain to be aliased.
735 </para></description>
736 <configOption name="type">
737 <synopsis>Must be of type 'domain_alias'.</synopsis>
739 <configOption name="domain">
740 <synopsis>Domain to be aliased</synopsis>
743 <configObject name="transport">
744 <synopsis>SIP Transport</synopsis>
746 <emphasis>Transports</emphasis>
748 <para>There are different transports and protocol derivatives
749 supported by <literal>res_pjsip</literal>. They are in order of
750 preference: UDP, TCP, and WebSocket (WS).</para>
751 <note><para>Changes to transport configuration in pjsip.conf will only be
752 effected on a complete restart of Asterisk. A module reload
753 will not suffice.</para></note>
755 <configOption name="async_operations" default="1">
756 <synopsis>Number of simultaneous Asynchronous Operations</synopsis>
758 <configOption name="bind">
759 <synopsis>IP Address and optional port to bind to for this transport</synopsis>
761 <configOption name="ca_list_file">
762 <synopsis>File containing a list of certificates to read (TLS ONLY)</synopsis>
764 <configOption name="cert_file">
765 <synopsis>Certificate file for endpoint (TLS ONLY)</synopsis>
767 <configOption name="cipher">
768 <synopsis>Preferred Cryptography Cipher (TLS ONLY)</synopsis>
770 Many options for acceptable ciphers see link for more:
771 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
772 </para></description>
774 <configOption name="domain">
775 <synopsis>Domain the transport comes from</synopsis>
777 <configOption name="external_media_address">
778 <synopsis>External IP address to use in RTP handling</synopsis>
780 When a request or response is sent out, if the destination of the
781 message is outside the IP network defined in the option <literal>localnet</literal>,
782 and the media address in the SDP is within the localnet network, then the
783 media address in the SDP will be rewritten to the value defined for
784 <literal>external_media_address</literal>.
785 </para></description>
787 <configOption name="external_signaling_address">
788 <synopsis>External address for SIP signalling</synopsis>
790 <configOption name="external_signaling_port" default="0">
791 <synopsis>External port for SIP signalling</synopsis>
793 <configOption name="method">
794 <synopsis>Method of SSL transport (TLS ONLY)</synopsis>
797 <enum name="default" />
798 <enum name="unspecified" />
799 <enum name="tlsv1" />
800 <enum name="sslv2" />
801 <enum name="sslv3" />
802 <enum name="sslv23" />
806 <configOption name="local_net">
807 <synopsis>Network to consider local (used for NAT purposes).</synopsis>
808 <description><para>This must be in CIDR or dotted decimal format with the IP
809 and mask separated with a slash ('/').</para></description>
811 <configOption name="password">
812 <synopsis>Password required for transport</synopsis>
814 <configOption name="priv_key_file">
815 <synopsis>Private key file (TLS ONLY)</synopsis>
817 <configOption name="protocol" default="udp">
818 <synopsis>Protocol to use for SIP traffic</synopsis>
829 <configOption name="require_client_cert" default="false">
830 <synopsis>Require client certificate (TLS ONLY)</synopsis>
832 <configOption name="type">
833 <synopsis>Must be of type 'transport'.</synopsis>
835 <configOption name="verify_client" default="false">
836 <synopsis>Require verification of client certificate (TLS ONLY)</synopsis>
838 <configOption name="verify_server" default="false">
839 <synopsis>Require verification of server certificate (TLS ONLY)</synopsis>
841 <configOption name="tos" default="false">
842 <synopsis>Enable TOS for the signalling sent over this transport</synopsis>
844 <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
845 for more information on this parameter.</para>
846 <note><para>This option does not apply to the <replaceable>ws</replaceable>
847 or the <replaceable>wss</replaceable> protocols.</para></note>
850 <configOption name="cos" default="false">
851 <synopsis>Enable COS for the signalling sent over this transport</synopsis>
853 <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
854 for more information on this parameter.</para>
855 <note><para>This option does not apply to the <replaceable>ws</replaceable>
856 or the <replaceable>wss</replaceable> protocols.</para></note>
860 <configObject name="contact">
861 <synopsis>A way of creating an aliased name to a SIP URI</synopsis>
863 Contacts are a way to hide SIP URIs from the dialplan directly.
864 They are also used to make a group of contactable parties when
865 in use with <literal>AoR</literal> lists.
866 </para></description>
867 <configOption name="type">
868 <synopsis>Must be of type 'contact'.</synopsis>
870 <configOption name="uri">
871 <synopsis>SIP URI to contact peer</synopsis>
873 <configOption name="expiration_time">
874 <synopsis>Time to keep alive a contact</synopsis>
876 Time to keep alive a contact. String style specification.
877 </para></description>
879 <configOption name="qualify_frequency" default="0">
880 <synopsis>Interval at which to qualify a contact</synopsis>
882 Interval between attempts to qualify the contact for reachability.
883 If <literal>0</literal> never qualify. Time in seconds.
884 </para></description>
886 <configOption name="outbound_proxy">
887 <synopsis>Outbound proxy used when sending OPTIONS request</synopsis>
889 If set the provided URI will be used as the outbound proxy when an
890 OPTIONS request is sent to a contact for qualify purposes.
891 </para></description>
893 <configOption name="path">
894 <synopsis>Stored Path vector for use in Route headers on outgoing requests.</synopsis>
897 <configObject name="aor">
898 <synopsis>The configuration for a location of an endpoint</synopsis>
900 An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
901 AoRs are specified, an endpoint will not be reachable by Asterisk.
902 Beyond that, an AoR has other uses within Asterisk, such as inbound
905 An <literal>AoR</literal> is a way to allow dialing a group
906 of <literal>Contacts</literal> that all use the same
907 <literal>endpoint</literal> for calls.
909 This can be used as another way of grouping a list of contacts to dial
910 rather than specifing them each directly when dialing via the dialplan.
911 This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
913 Registrations: For Asterisk to match an inbound registration to an endpoint,
914 the AoR object name must match the user portion of the SIP URI in the "To:"
915 header of the inbound SIP registration. That will usually be equivalent
916 to the "user name" set in your hard or soft phones configuration.
917 </para></description>
918 <configOption name="contact">
919 <synopsis>Permanent contacts assigned to AoR</synopsis>
921 Contacts specified will be called whenever referenced
922 by <literal>chan_pjsip</literal>.
924 Use a separate "contact=" entry for each contact required. Contacts
925 are specified using a SIP URI.
926 </para></description>
928 <configOption name="default_expiration" default="3600">
929 <synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis>
931 <configOption name="mailboxes">
932 <synopsis>Mailbox(es) to be associated with</synopsis>
933 <description><para>This option applies when an external entity subscribes to an AoR
934 for message waiting indications. The mailboxes specified will be subscribed to.
935 More than one mailbox can be specified with a comma-delimited string.
936 For endpoints that cannot SUBSCRIBE for MWI, you can set the <literal>mailboxes</literal> option in your
937 Endpoint configuration section.
938 </para></description>
940 <configOption name="maximum_expiration" default="7200">
941 <synopsis>Maximum time to keep an AoR</synopsis>
943 Maximium time to keep a peer with explicit expiration. Time in seconds.
944 </para></description>
946 <configOption name="max_contacts" default="0">
947 <synopsis>Maximum number of contacts that can bind to an AoR</synopsis>
949 Maximum number of contacts that can associate with this AoR. This value does
950 not affect the number of contacts that can be added with the "contact" option.
951 It only limits contacts added through external interaction, such as
954 <note><para>This should be set to <literal>1</literal> and
955 <replaceable>remove_existing</replaceable> set to <literal>yes</literal> if you
956 wish to stick with the older <literal>chan_sip</literal> behaviour.
960 <configOption name="minimum_expiration" default="60">
961 <synopsis>Minimum keep alive time for an AoR</synopsis>
963 Minimum time to keep a peer with an explict expiration. Time in seconds.
964 </para></description>
966 <configOption name="remove_existing" default="no">
967 <synopsis>Determines whether new contacts replace existing ones.</synopsis>
969 On receiving a new registration to the AoR should it remove
970 the existing contact that was registered against it?
972 <note><para>This should be set to <literal>yes</literal> and
973 <replaceable>max_contacts</replaceable> set to <literal>1</literal> if you
974 wish to stick with the older <literal>chan_sip</literal> behaviour.
978 <configOption name="type">
979 <synopsis>Must be of type 'aor'.</synopsis>
981 <configOption name="qualify_frequency" default="0">
982 <synopsis>Interval at which to qualify an AoR</synopsis>
984 Interval between attempts to qualify the AoR for reachability.
985 If <literal>0</literal> never qualify. Time in seconds.
986 </para></description>
988 <configOption name="authenticate_qualify" default="no">
989 <synopsis>Authenticates a qualify request if needed</synopsis>
991 If true and a qualify request receives a challenge or authenticate response
992 authentication is attempted before declaring the contact available.
993 </para></description>
995 <configOption name="outbound_proxy">
996 <synopsis>Outbound proxy used when sending OPTIONS request</synopsis>
998 If set the provided URI will be used as the outbound proxy when an
999 OPTIONS request is sent to a contact for qualify purposes.
1000 </para></description>
1002 <configOption name="support_path">
1003 <synopsis>Enables Path support for REGISTER requests and Route support for other requests.</synopsis>
1005 When this option is enabled, the Path headers in register requests will be saved
1006 and its contents will be used in Route headers for outbound out-of-dialog requests
1007 and in Path headers for outbound 200 responses. Path support will also be indicated
1008 in the Supported header.
1009 </para></description>
1012 <configObject name="system">
1013 <synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis>
1015 The settings in this section are global. In addition to being global, the values will
1016 not be re-evaluated when a reload is performed. This is because the values must be set
1017 before the SIP stack is initialized. The only way to reset these values is to either
1018 restart Asterisk, or unload res_pjsip.so and then load it again.
1019 </para></description>
1020 <configOption name="timer_t1" default="500">
1021 <synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
1023 Timer T1 is the base for determining how long to wait before retransmitting
1024 requests that receive no response when using an unreliable transport (e.g. UDP).
1025 For more information on this timer, see RFC 3261, Section 17.1.1.1.
1026 </para></description>
1028 <configOption name="timer_b" default="32000">
1029 <synopsis>Set transaction timer B value (milliseconds).</synopsis>
1031 Timer B determines the maximum amount of time to wait after sending an INVITE
1032 request before terminating the transaction. It is recommended that this be set
1033 to 64 * Timer T1, but it may be set higher if desired. For more information on
1034 this timer, see RFC 3261, Section 17.1.1.1.
1035 </para></description>
1037 <configOption name="compact_headers" default="no">
1038 <synopsis>Use the short forms of common SIP header names.</synopsis>
1040 <configOption name="threadpool_initial_size" default="0">
1041 <synopsis>Initial number of threads in the res_pjsip threadpool.</synopsis>
1043 <configOption name="threadpool_auto_increment" default="5">
1044 <synopsis>The amount by which the number of threads is incremented when necessary.</synopsis>
1046 <configOption name="threadpool_idle_timeout" default="60">
1047 <synopsis>Number of seconds before an idle thread should be disposed of.</synopsis>
1049 <configOption name="threadpool_max_size" default="0">
1050 <synopsis>Maximum number of threads in the res_pjsip threadpool.
1051 A value of 0 indicates no maximum.</synopsis>
1053 <configOption name="type">
1054 <synopsis>Must be of type 'system'.</synopsis>
1057 <configObject name="global">
1058 <synopsis>Options that apply globally to all SIP communications</synopsis>
1060 The settings in this section are global. Unlike options in the <literal>system</literal>
1061 section, these options can be refreshed by performing a reload.
1062 </para></description>
1063 <configOption name="max_forwards" default="70">
1064 <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
1066 <configOption name="type">
1067 <synopsis>Must be of type 'global'.</synopsis>
1069 <configOption name="user_agent" default="Asterisk <Asterisk Version>">
1070 <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
1072 <configOption name="default_outbound_endpoint" default="default_outbound_endpoint">
1073 <synopsis>Endpoint to use when sending an outbound request to a URI without a specified endpoint.</synopsis>
1079 <manager name="PJSIPQualify" language="en_US">
1081 Qualify a chan_pjsip endpoint.
1084 <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
1085 <parameter name="Endpoint" required="true">
1086 <para>The endpoint you want to qualify.</para>
1090 <para>Qualify a chan_pjsip endpoint.</para>
1093 <manager name="PJSIPShowEndpoints" language="en_US">
1095 Lists PJSIP endpoints.
1100 Provides a listing of all endpoints. For each endpoint an <literal>EndpointList</literal> event
1101 is raised that contains relevant attributes and status information. Once all
1102 endpoints have been listed an <literal>EndpointListComplete</literal> event is issued.
1106 <manager name="PJSIPShowEndpoint" language="en_US">
1108 Detail listing of an endpoint and its objects.
1111 <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
1112 <parameter name="Endpoint" required="true">
1113 <para>The endpoint to list.</para>
1118 Provides a detailed listing of options for a given endpoint. Events are issued
1119 showing the configuration and status of the endpoint and associated objects. These
1120 events include <literal>EndpointDetail</literal>, <literal>AorDetail</literal>,
1121 <literal>AuthDetail</literal>, <literal>TransportDetail</literal>, and
1122 <literal>IdentifyDetail</literal>. Some events may be listed multiple times if multiple objects are
1123 associated (for instance AoRs). Once all detail events have been raised a final
1124 <literal>EndpointDetailComplete</literal> event is issued.
1130 #define MOD_DATA_CONTACT "contact"
1132 static pjsip_endpoint *ast_pjsip_endpoint;
1134 static struct ast_threadpool *sip_threadpool;
1136 static int register_service(void *data)
1138 pjsip_module **module = data;
1139 if (!ast_pjsip_endpoint) {
1140 ast_log(LOG_ERROR, "There is no PJSIP endpoint. Unable to register services\n");
1143 if (pjsip_endpt_register_module(ast_pjsip_endpoint, *module) != PJ_SUCCESS) {
1144 ast_log(LOG_ERROR, "Unable to register module %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1147 ast_debug(1, "Registered SIP service %.*s (%p)\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name), *module);
1148 ast_module_ref(ast_module_info->self);
1152 int ast_sip_register_service(pjsip_module *module)
1154 return ast_sip_push_task_synchronous(NULL, register_service, &module);
1157 static int unregister_service(void *data)
1159 pjsip_module **module = data;
1160 ast_module_unref(ast_module_info->self);
1161 if (!ast_pjsip_endpoint) {
1164 pjsip_endpt_unregister_module(ast_pjsip_endpoint, *module);
1165 ast_debug(1, "Unregistered SIP service %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1169 void ast_sip_unregister_service(pjsip_module *module)
1171 ast_sip_push_task_synchronous(NULL, unregister_service, &module);
1174 static struct ast_sip_authenticator *registered_authenticator;
1176 int ast_sip_register_authenticator(struct ast_sip_authenticator *auth)
1178 if (registered_authenticator) {
1179 ast_log(LOG_WARNING, "Authenticator %p is already registered. Cannot register a new one\n", registered_authenticator);
1182 registered_authenticator = auth;
1183 ast_debug(1, "Registered SIP authenticator module %p\n", auth);
1184 ast_module_ref(ast_module_info->self);
1188 void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth)
1190 if (registered_authenticator != auth) {
1191 ast_log(LOG_WARNING, "Trying to unregister authenticator %p but authenticator %p registered\n",
1192 auth, registered_authenticator);
1195 registered_authenticator = NULL;
1196 ast_debug(1, "Unregistered SIP authenticator %p\n", auth);
1197 ast_module_unref(ast_module_info->self);
1200 int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1202 if (!registered_authenticator) {
1203 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is not required\n");
1207 return registered_authenticator->requires_authentication(endpoint, rdata);
1210 enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint,
1211 pjsip_rx_data *rdata, pjsip_tx_data *tdata)
1213 if (!registered_authenticator) {
1214 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is successful\n");
1217 return registered_authenticator->check_authentication(endpoint, rdata, tdata);
1220 static struct ast_sip_outbound_authenticator *registered_outbound_authenticator;
1222 int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1224 if (registered_outbound_authenticator) {
1225 ast_log(LOG_WARNING, "Outbound authenticator %p is already registered. Cannot register a new one\n", registered_outbound_authenticator);
1228 registered_outbound_authenticator = auth;
1229 ast_debug(1, "Registered SIP outbound authenticator module %p\n", auth);
1230 ast_module_ref(ast_module_info->self);
1234 void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1236 if (registered_outbound_authenticator != auth) {
1237 ast_log(LOG_WARNING, "Trying to unregister outbound authenticator %p but outbound authenticator %p registered\n",
1238 auth, registered_outbound_authenticator);
1241 registered_outbound_authenticator = NULL;
1242 ast_debug(1, "Unregistered SIP outbound authenticator %p\n", auth);
1243 ast_module_unref(ast_module_info->self);
1246 int ast_sip_create_request_with_auth(const struct ast_sip_auth_vector *auths, pjsip_rx_data *challenge,
1247 pjsip_transaction *tsx, pjsip_tx_data **new_request)
1249 if (!registered_outbound_authenticator) {
1250 ast_log(LOG_WARNING, "No SIP outbound authenticator registered. Cannot respond to authentication challenge\n");
1253 return registered_outbound_authenticator->create_request_with_auth(auths, challenge, tsx, new_request);
1256 struct endpoint_identifier_list {
1257 struct ast_sip_endpoint_identifier *identifier;
1258 AST_RWLIST_ENTRY(endpoint_identifier_list) list;
1261 static AST_RWLIST_HEAD_STATIC(endpoint_identifiers, endpoint_identifier_list);
1263 int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1265 struct endpoint_identifier_list *id_list_item;
1266 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1268 id_list_item = ast_calloc(1, sizeof(*id_list_item));
1269 if (!id_list_item) {
1270 ast_log(LOG_ERROR, "Unabled to add endpoint identifier. Out of memory.\n");
1273 id_list_item->identifier = identifier;
1275 AST_RWLIST_INSERT_TAIL(&endpoint_identifiers, id_list_item, list);
1276 ast_debug(1, "Registered endpoint identifier %p\n", identifier);
1278 ast_module_ref(ast_module_info->self);
1282 void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1284 struct endpoint_identifier_list *iter;
1285 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1286 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_identifiers, iter, list) {
1287 if (iter->identifier == identifier) {
1288 AST_RWLIST_REMOVE_CURRENT(list);
1290 ast_debug(1, "Unregistered endpoint identifier %p\n", identifier);
1291 ast_module_unref(ast_module_info->self);
1295 AST_RWLIST_TRAVERSE_SAFE_END;
1298 struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata)
1300 struct endpoint_identifier_list *iter;
1301 struct ast_sip_endpoint *endpoint = NULL;
1302 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1303 AST_RWLIST_TRAVERSE(&endpoint_identifiers, iter, list) {
1304 ast_assert(iter->identifier->identify_endpoint != NULL);
1305 endpoint = iter->identifier->identify_endpoint(rdata);
1313 AST_RWLIST_HEAD_STATIC(endpoint_formatters, ast_sip_endpoint_formatter);
1315 int ast_sip_register_endpoint_formatter(struct ast_sip_endpoint_formatter *obj)
1317 SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1318 AST_RWLIST_INSERT_TAIL(&endpoint_formatters, obj, next);
1319 ast_module_ref(ast_module_info->self);
1323 void ast_sip_unregister_endpoint_formatter(struct ast_sip_endpoint_formatter *obj)
1325 struct ast_sip_endpoint_formatter *i;
1326 SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1327 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_formatters, i, next) {
1329 AST_RWLIST_REMOVE_CURRENT(next);
1330 ast_module_unref(ast_module_info->self);
1334 AST_RWLIST_TRAVERSE_SAFE_END;
1337 int ast_sip_format_endpoint_ami(struct ast_sip_endpoint *endpoint,
1338 struct ast_sip_ami *ami, int *count)
1341 struct ast_sip_endpoint_formatter *i;
1342 SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1344 AST_RWLIST_TRAVERSE(&endpoint_formatters, i, next) {
1345 if (i->format_ami && ((res = i->format_ami(endpoint, ami)) < 0)) {
1356 pjsip_endpoint *ast_sip_get_pjsip_endpoint(void)
1358 return ast_pjsip_endpoint;
1361 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)
1363 pj_str_t tmp, local_addr;
1365 pjsip_sip_uri *sip_uri;
1366 pjsip_transport_type_e type = PJSIP_TRANSPORT_UNSPECIFIED;
1368 char uuid_str[AST_UUID_STR_LEN];
1370 if (ast_strlen_zero(user)) {
1371 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1375 user = ast_uuid_to_str(uuid, uuid_str, sizeof(uuid_str));
1378 /* Parse the provided target URI so we can determine what transport it will end up using */
1379 pj_strdup_with_null(pool, &tmp, target);
1381 if (!(uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0)) ||
1382 (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
1386 sip_uri = pjsip_uri_get_uri(uri);
1388 /* Determine the transport type to use */
1389 if (PJSIP_URI_SCHEME_IS_SIPS(sip_uri)) {
1390 type = PJSIP_TRANSPORT_TLS;
1391 } else if (!sip_uri->transport_param.slen) {
1392 type = PJSIP_TRANSPORT_UDP;
1394 type = pjsip_transport_get_type_from_name(&sip_uri->transport_param);
1397 if (type == PJSIP_TRANSPORT_UNSPECIFIED) {
1401 /* If the host is IPv6 turn the transport into an IPv6 version */
1402 if (pj_strchr(&sip_uri->host, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1403 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1406 if (!ast_strlen_zero(domain)) {
1407 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1408 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1412 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1413 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1417 /* Get the local bound address for the transport that will be used when communicating with the provided URI */
1418 if (pjsip_tpmgr_find_local_addr(pjsip_endpt_get_tpmgr(ast_sip_get_pjsip_endpoint()), pool, type, selector,
1419 &local_addr, &local_port) != PJ_SUCCESS) {
1421 /* If no local address can be retrieved using the transport manager use the host one */
1422 pj_strdup(pool, &local_addr, pj_gethostname());
1423 local_port = pjsip_transport_get_default_port_for_type(PJSIP_TRANSPORT_UDP);
1426 /* If IPv6 was specified in the transport, set the proper type */
1427 if (pj_strchr(&local_addr, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1428 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1431 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1432 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1433 "<sip:%s@%s%.*s%s:%d%s%s>",
1435 (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1436 (int)local_addr.slen,
1438 (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1440 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1441 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1446 static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpoint, pjsip_tpselector *selector)
1448 RAII_VAR(struct ast_sip_transport *, transport, NULL, ao2_cleanup);
1449 const char *transport_name = endpoint->transport;
1451 if (ast_strlen_zero(transport_name)) {
1455 transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_name);
1457 if (!transport || !transport->state) {
1458 ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport '%s' for endpoint '%s'\n",
1459 transport_name, ast_sorcery_object_get_id(endpoint));
1463 if (transport->state->transport) {
1464 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1465 selector->u.transport = transport->state->transport;
1466 } else if (transport->state->factory) {
1467 selector->type = PJSIP_TPSELECTOR_LISTENER;
1468 selector->u.listener = transport->state->factory;
1476 pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user)
1478 char enclosed_uri[PJSIP_MAX_URL_SIZE];
1479 pj_str_t local_uri = { "sip:temp@temp", 13 }, remote_uri, target_uri;
1480 pjsip_dialog *dlg = NULL;
1481 const char *outbound_proxy = endpoint->outbound_proxy;
1482 pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1483 static const pj_str_t HCONTACT = { "Contact", 7 };
1485 snprintf(enclosed_uri, sizeof(enclosed_uri), "<%s>", uri);
1486 pj_cstr(&remote_uri, enclosed_uri);
1488 pj_cstr(&target_uri, uri);
1490 if (pjsip_dlg_create_uac(pjsip_ua_instance(), &local_uri, NULL, &remote_uri, &target_uri, &dlg) != PJ_SUCCESS) {
1494 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1495 pjsip_dlg_terminate(dlg);
1499 if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
1500 pjsip_dlg_terminate(dlg);
1504 /* Update the dialog with the new local URI, we do it afterwards so we can use the dialog pool for construction */
1505 pj_strdup_with_null(dlg->pool, &dlg->local.info_str, &local_uri);
1506 dlg->local.info->uri = pjsip_parse_uri(dlg->pool, dlg->local.info_str.ptr, dlg->local.info_str.slen, 0);
1507 dlg->local.contact = pjsip_parse_hdr(dlg->pool, &HCONTACT, local_uri.ptr, local_uri.slen, NULL);
1509 /* If a request user has been specified and we are permitted to change it, do so */
1510 if (!ast_strlen_zero(request_user)) {
1511 pjsip_sip_uri *sip_uri;
1513 if (PJSIP_URI_SCHEME_IS_SIP(dlg->target) || PJSIP_URI_SCHEME_IS_SIPS(dlg->target)) {
1514 sip_uri = pjsip_uri_get_uri(dlg->target);
1515 pj_strdup2(dlg->pool, &sip_uri->user, request_user);
1517 if (PJSIP_URI_SCHEME_IS_SIP(dlg->remote.info->uri) || PJSIP_URI_SCHEME_IS_SIPS(dlg->remote.info->uri)) {
1518 sip_uri = pjsip_uri_get_uri(dlg->remote.info->uri);
1519 pj_strdup2(dlg->pool, &sip_uri->user, request_user);
1523 /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
1526 pjsip_dlg_set_transport(dlg, &selector);
1528 if (!ast_strlen_zero(outbound_proxy)) {
1529 pjsip_route_hdr route_set, *route;
1530 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1533 pj_list_init(&route_set);
1535 pj_strdup2_with_null(dlg->pool, &tmp, outbound_proxy);
1536 if (!(route = pjsip_parse_hdr(dlg->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1538 pjsip_dlg_terminate(dlg);
1541 pj_list_push_back(&route_set, route);
1543 pjsip_dlg_set_route_set(dlg, &route_set);
1551 pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1555 pjsip_transport_type_e type = rdata->tp_info.transport->key.type;
1558 contact.ptr = pj_pool_alloc(rdata->tp_info.pool, PJSIP_MAX_URL_SIZE);
1559 contact.slen = pj_ansi_snprintf(contact.ptr, PJSIP_MAX_URL_SIZE,
1560 "<sip:%s%.*s%s:%d%s%s>",
1561 (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1562 (int)rdata->tp_info.transport->local_name.host.slen,
1563 rdata->tp_info.transport->local_name.host.ptr,
1564 (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1565 rdata->tp_info.transport->local_name.port,
1566 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1567 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1569 status = pjsip_dlg_create_uas(pjsip_ua_instance(), rdata, &contact, &dlg);
1570 if (status != PJ_SUCCESS) {
1571 char err[PJ_ERR_MSG_SIZE];
1573 pj_strerror(status, err, sizeof(err));
1574 ast_log(LOG_ERROR, "Could not create dialog with endpoint %s. %s\n",
1575 ast_sorcery_object_get_id(endpoint), err);
1582 /* PJSIP doesn't know about the INFO method, so we have to define it ourselves */
1583 static const pjsip_method info_method = {PJSIP_OTHER_METHOD, {"INFO", 4} };
1584 static const pjsip_method message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
1588 const pjsip_method *pmethod;
1590 { "INVITE", &pjsip_invite_method },
1591 { "CANCEL", &pjsip_cancel_method },
1592 { "ACK", &pjsip_ack_method },
1593 { "BYE", &pjsip_bye_method },
1594 { "REGISTER", &pjsip_register_method },
1595 { "OPTIONS", &pjsip_options_method },
1596 { "SUBSCRIBE", &pjsip_subscribe_method },
1597 { "NOTIFY", &pjsip_notify_method },
1598 { "PUBLISH", &pjsip_publish_method },
1599 { "INFO", &info_method },
1600 { "MESSAGE", &message_method },
1603 static const pjsip_method *get_pjsip_method(const char *method)
1606 for (i = 0; i < ARRAY_LEN(methods); ++i) {
1607 if (!strcmp(method, methods[i].method)) {
1608 return methods[i].pmethod;
1614 static int create_in_dialog_request(const pjsip_method *method, struct pjsip_dialog *dlg, pjsip_tx_data **tdata)
1616 if (pjsip_dlg_create_request(dlg, method, -1, tdata) != PJ_SUCCESS) {
1617 ast_log(LOG_WARNING, "Unable to create in-dialog request.\n");
1624 static pj_bool_t supplement_on_rx_request(pjsip_rx_data *rdata);
1625 static pjsip_module supplement_module = {
1626 .name = { "Out of dialog supplement hook", 29 },
1628 .priority = PJSIP_MOD_PRIORITY_APPLICATION - 1,
1629 .on_rx_request = supplement_on_rx_request,
1632 static int create_out_of_dialog_request(const pjsip_method *method, struct ast_sip_endpoint *endpoint,
1633 const char *uri, struct ast_sip_contact *provided_contact, pjsip_tx_data **tdata)
1635 RAII_VAR(struct ast_sip_contact *, contact, ao2_bump(provided_contact), ao2_cleanup);
1636 pj_str_t remote_uri;
1639 pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1641 if (ast_strlen_zero(uri)) {
1642 if (!endpoint && !contact) {
1643 ast_log(LOG_ERROR, "An endpoint and/or uri must be specified\n");
1648 contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
1650 if (!contact || ast_strlen_zero(contact->uri)) {
1651 ast_log(LOG_ERROR, "Unable to retrieve contact for endpoint %s\n",
1652 ast_sorcery_object_get_id(endpoint));
1656 pj_cstr(&remote_uri, contact->uri);
1658 pj_cstr(&remote_uri, uri);
1662 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1663 ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport selector for endpoint %s\n",
1664 ast_sorcery_object_get_id(endpoint));
1669 pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "Outbound request", 256, 256);
1672 ast_log(LOG_ERROR, "Unable to create PJLIB memory pool\n");
1676 if (sip_dialog_create_from(pool, &from, endpoint ? endpoint->fromuser : NULL,
1677 endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) {
1678 ast_log(LOG_ERROR, "Unable to create From header for %.*s request to endpoint %s\n",
1679 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1680 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1684 if (pjsip_endpt_create_request(ast_sip_get_pjsip_endpoint(), method, &remote_uri,
1685 &from, &remote_uri, &from, NULL, -1, NULL, tdata) != PJ_SUCCESS) {
1686 ast_log(LOG_ERROR, "Unable to create outbound %.*s request to endpoint %s\n",
1687 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1688 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1692 /* If an outbound proxy is specified on the endpoint apply it to this request */
1693 if (endpoint && !ast_strlen_zero(endpoint->outbound_proxy) &&
1694 ast_sip_set_outbound_proxy((*tdata), endpoint->outbound_proxy)) {
1695 ast_log(LOG_ERROR, "Unable to apply outbound proxy on request %.*s to endpoint %s\n",
1696 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1697 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1701 ast_sip_mod_data_set((*tdata)->pool, (*tdata)->mod_data, supplement_module.id, MOD_DATA_CONTACT, ao2_bump(contact));
1703 /* We can release this pool since request creation copied all the necessary
1704 * data into the outbound request's pool
1706 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1710 int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
1711 struct ast_sip_endpoint *endpoint, const char *uri,
1712 struct ast_sip_contact *contact, pjsip_tx_data **tdata)
1714 const pjsip_method *pmethod = get_pjsip_method(method);
1717 ast_log(LOG_WARNING, "Unknown method '%s'. Cannot send request\n", method);
1722 return create_in_dialog_request(pmethod, dlg, tdata);
1724 return create_out_of_dialog_request(pmethod, endpoint, uri, contact, tdata);
1728 AST_RWLIST_HEAD_STATIC(supplements, ast_sip_supplement);
1730 int ast_sip_register_supplement(struct ast_sip_supplement *supplement)
1732 struct ast_sip_supplement *iter;
1734 SCOPED_LOCK(lock, &supplements, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1736 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&supplements, iter, next) {
1737 if (iter->priority > supplement->priority) {
1738 AST_RWLIST_INSERT_BEFORE_CURRENT(supplement, next);
1743 AST_RWLIST_TRAVERSE_SAFE_END;
1746 AST_RWLIST_INSERT_TAIL(&supplements, supplement, next);
1748 ast_module_ref(ast_module_info->self);
1752 void ast_sip_unregister_supplement(struct ast_sip_supplement *supplement)
1754 struct ast_sip_supplement *iter;
1755 SCOPED_LOCK(lock, &supplements, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1756 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&supplements, iter, next) {
1757 if (supplement == iter) {
1758 AST_RWLIST_REMOVE_CURRENT(next);
1759 ast_module_unref(ast_module_info->self);
1763 AST_RWLIST_TRAVERSE_SAFE_END;
1766 static int send_in_dialog_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg)
1768 if (pjsip_dlg_send_request(dlg, tdata, -1, NULL) != PJ_SUCCESS) {
1769 ast_log(LOG_WARNING, "Unable to send in-dialog request.\n");
1775 static pj_bool_t does_method_match(const pj_str_t *message_method, const char *supplement_method)
1779 if (ast_strlen_zero(supplement_method)) {
1783 pj_cstr(&method, supplement_method);
1785 return pj_stristr(&method, message_method) ? PJ_TRUE : PJ_FALSE;
1788 /*! \brief Structure to hold information about an outbound request */
1789 struct send_request_data {
1790 struct ast_sip_endpoint *endpoint; /*! The endpoint associated with this request */
1791 void *token; /*! Information to be provided to the callback upon receipt of a response */
1792 void (*callback)(void *token, pjsip_event *e); /*! The callback to be called upon receipt of a response */
1795 static void send_request_data_destroy(void *obj)
1797 struct send_request_data *req_data = obj;
1798 ao2_cleanup(req_data->endpoint);
1801 static struct send_request_data *send_request_data_alloc(struct ast_sip_endpoint *endpoint,
1802 void *token, void (*callback)(void *token, pjsip_event *e))
1804 struct send_request_data *req_data = ao2_alloc(sizeof(*req_data), send_request_data_destroy);
1810 req_data->endpoint = ao2_bump(endpoint);
1811 req_data->token = token;
1812 req_data->callback = callback;
1817 static void send_request_cb(void *token, pjsip_event *e)
1819 RAII_VAR(struct send_request_data *, req_data, token, ao2_cleanup);
1820 pjsip_transaction *tsx = e->body.tsx_state.tsx;
1821 pjsip_rx_data *challenge = e->body.tsx_state.src.rdata;
1822 pjsip_tx_data *tdata;
1823 struct ast_sip_supplement *supplement;
1825 AST_RWLIST_RDLOCK(&supplements);
1826 AST_LIST_TRAVERSE(&supplements, supplement, next) {
1827 if (supplement->incoming_response && does_method_match(&challenge->msg_info.cseq->method.name, supplement->method)) {
1828 supplement->incoming_response(req_data->endpoint, challenge);
1831 AST_RWLIST_UNLOCK(&supplements);
1833 if (tsx->status_code == 401 || tsx->status_code == 407) {
1834 if (!ast_sip_create_request_with_auth(&req_data->endpoint->outbound_auths, challenge, tsx, &tdata)) {
1835 pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, req_data->token, req_data->callback);
1840 if (req_data->callback) {
1841 req_data->callback(req_data->token, e);
1845 static int send_out_of_dialog_request(pjsip_tx_data *tdata, struct ast_sip_endpoint *endpoint,
1846 void *token, void (*callback)(void *token, pjsip_event *e))
1848 struct ast_sip_supplement *supplement;
1849 struct send_request_data *req_data = send_request_data_alloc(endpoint, token, callback);
1850 struct ast_sip_contact *contact = ast_sip_mod_data_get(tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT);
1856 AST_RWLIST_RDLOCK(&supplements);
1857 AST_LIST_TRAVERSE(&supplements, supplement, next) {
1858 if (supplement->outgoing_request && does_method_match(&tdata->msg->line.req.method.name, supplement->method)) {
1859 supplement->outgoing_request(endpoint, contact, tdata);
1862 AST_RWLIST_UNLOCK(&supplements);
1864 ast_sip_mod_data_set(tdata->pool, tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT, NULL);
1865 ao2_cleanup(contact);
1867 if (pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, req_data, send_request_cb) != PJ_SUCCESS) {
1868 ast_log(LOG_ERROR, "Error attempting to send outbound %.*s request to endpoint %s\n",
1869 (int) pj_strlen(&tdata->msg->line.req.method.name),
1870 pj_strbuf(&tdata->msg->line.req.method.name),
1871 ast_sorcery_object_get_id(endpoint));
1872 ao2_cleanup(req_data);
1879 int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg,
1880 struct ast_sip_endpoint *endpoint, void *token,
1881 void (*callback)(void *token, pjsip_event *e))
1883 ast_assert(tdata->msg->type == PJSIP_REQUEST_MSG);
1886 return send_in_dialog_request(tdata, dlg);
1888 return send_out_of_dialog_request(tdata, endpoint, token, callback);
1892 int ast_sip_set_outbound_proxy(pjsip_tx_data *tdata, const char *proxy)
1894 pjsip_route_hdr *route;
1895 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1898 pj_strdup2_with_null(tdata->pool, &tmp, proxy);
1899 if (!(route = pjsip_parse_hdr(tdata->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1903 pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr*)route);
1908 int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value)
1912 pjsip_generic_string_hdr *hdr;
1914 pj_cstr(&hdr_name, name);
1915 pj_cstr(&hdr_value, value);
1917 hdr = pjsip_generic_string_hdr_create(tdata->pool, &hdr_name, &hdr_value);
1919 pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *) hdr);
1923 static pjsip_msg_body *ast_body_to_pjsip_body(pj_pool_t *pool, const struct ast_sip_body *body)
1929 pj_cstr(&type, body->type);
1930 pj_cstr(&subtype, body->subtype);
1931 pj_cstr(&body_text, body->body_text);
1933 return pjsip_msg_body_create(pool, &type, &subtype, &body_text);
1936 int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body)
1938 pjsip_msg_body *pjsip_body = ast_body_to_pjsip_body(tdata->pool, body);
1939 tdata->msg->body = pjsip_body;
1943 int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies)
1946 /* NULL for type and subtype automatically creates "multipart/mixed" */
1947 pjsip_msg_body *body = pjsip_multipart_create(tdata->pool, NULL, NULL);
1949 for (i = 0; i < num_bodies; ++i) {
1950 pjsip_multipart_part *part = pjsip_multipart_create_part(tdata->pool);
1951 part->body = ast_body_to_pjsip_body(tdata->pool, bodies[i]);
1952 pjsip_multipart_add_part(tdata->pool, body, part);
1955 tdata->msg->body = body;
1959 int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text)
1961 size_t combined_size = strlen(body_text) + tdata->msg->body->len;
1962 struct ast_str *body_buffer = ast_str_alloca(combined_size);
1964 ast_str_set(&body_buffer, 0, "%.*s%s", (int) tdata->msg->body->len, (char *) tdata->msg->body->data, body_text);
1966 tdata->msg->body->data = pj_pool_alloc(tdata->pool, combined_size);
1967 pj_memcpy(tdata->msg->body->data, ast_str_buffer(body_buffer), combined_size);
1968 tdata->msg->body->len = combined_size;
1973 struct ast_taskprocessor *ast_sip_create_serializer(void)
1975 struct ast_taskprocessor *serializer;
1976 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1977 char name[AST_UUID_STR_LEN];
1983 ast_uuid_to_str(uuid, name, sizeof(name));
1985 serializer = ast_threadpool_serializer(name, sip_threadpool);
1992 int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1995 return ast_taskprocessor_push(serializer, sip_task, task_data);
1997 return ast_threadpool_push(sip_threadpool, sip_task, task_data);
2001 struct sync_task_data {
2006 int (*task)(void *);
2010 static int sync_task(void *data)
2012 struct sync_task_data *std = data;
2013 std->fail = std->task(std->task_data);
2015 ast_mutex_lock(&std->lock);
2017 ast_cond_signal(&std->cond);
2018 ast_mutex_unlock(&std->lock);
2022 int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
2024 /* This method is an onion */
2025 struct sync_task_data std;
2027 if (ast_sip_thread_is_servant()) {
2028 return sip_task(task_data);
2031 ast_mutex_init(&std.lock);
2032 ast_cond_init(&std.cond, NULL);
2033 std.fail = std.complete = 0;
2034 std.task = sip_task;
2035 std.task_data = task_data;
2038 if (ast_taskprocessor_push(serializer, sync_task, &std)) {
2042 if (ast_threadpool_push(sip_threadpool, sync_task, &std)) {
2047 ast_mutex_lock(&std.lock);
2048 while (!std.complete) {
2049 ast_cond_wait(&std.cond, &std.lock);
2051 ast_mutex_unlock(&std.lock);
2053 ast_mutex_destroy(&std.lock);
2054 ast_cond_destroy(&std.cond);
2058 void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size)
2060 size_t chars_to_copy = MIN(size - 1, pj_strlen(src));
2061 memcpy(dest, pj_strbuf(src), chars_to_copy);
2062 dest[chars_to_copy] = '\0';
2065 int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype)
2067 pjsip_media_type compare;
2069 if (!content_type) {
2073 pjsip_media_type_init2(&compare, type, subtype);
2075 return pjsip_media_type_cmp(content_type, &compare, 0) ? 0 : -1;
2078 pj_caching_pool caching_pool;
2079 pj_pool_t *memory_pool;
2080 pj_thread_t *monitor_thread;
2081 static int monitor_continue;
2083 static void *monitor_thread_exec(void *endpt)
2085 while (monitor_continue) {
2086 const pj_time_val delay = {0, 10};
2087 pjsip_endpt_handle_events(ast_pjsip_endpoint, &delay);
2092 static void stop_monitor_thread(void)
2094 monitor_continue = 0;
2095 pj_thread_join(monitor_thread);
2098 AST_THREADSTORAGE(pj_thread_storage);
2099 AST_THREADSTORAGE(servant_id_storage);
2100 #define SIP_SERVANT_ID 0x5E2F1D
2102 static void sip_thread_start(void)
2104 pj_thread_desc *desc;
2105 pj_thread_t *thread;
2106 uint32_t *servant_id;
2108 servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
2110 ast_log(LOG_ERROR, "Could not set SIP servant ID in thread-local storage.\n");
2113 *servant_id = SIP_SERVANT_ID;
2115 desc = ast_threadstorage_get(&pj_thread_storage, sizeof(pj_thread_desc));
2117 ast_log(LOG_ERROR, "Could not get thread desc from thread-local storage. Expect awful things to occur\n");
2120 pj_bzero(*desc, sizeof(*desc));
2122 if (pj_thread_register("Asterisk Thread", *desc, &thread) != PJ_SUCCESS) {
2123 ast_log(LOG_ERROR, "Couldn't register thread with PJLIB.\n");
2127 int ast_sip_thread_is_servant(void)
2129 uint32_t *servant_id;
2131 servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
2136 return *servant_id == SIP_SERVANT_ID;
2139 void *ast_sip_dict_get(void *ht, const char *key)
2141 unsigned int hval = 0;
2147 return pj_hash_get(ht, key, PJ_HASH_KEY_STRING, &hval);
2150 void *ast_sip_dict_set(pj_pool_t* pool, void *ht,
2151 const char *key, void *val)
2154 ht = pj_hash_create(pool, 11);
2157 pj_hash_set(pool, ht, key, PJ_HASH_KEY_STRING, 0, val);
2162 static pj_bool_t supplement_on_rx_request(pjsip_rx_data *rdata)
2164 struct ast_sip_supplement *supplement;
2166 if (pjsip_rdata_get_dlg(rdata)) {
2170 AST_RWLIST_RDLOCK(&supplements);
2171 AST_LIST_TRAVERSE(&supplements, supplement, next) {
2172 if (supplement->incoming_request && does_method_match(&rdata->msg_info.msg->line.req.method.name, supplement->method)) {
2173 supplement->incoming_request(ast_pjsip_rdata_get_endpoint(rdata), rdata);
2176 AST_RWLIST_UNLOCK(&supplements);
2181 int ast_sip_send_response(pjsip_response_addr *res_addr, pjsip_tx_data *tdata, struct ast_sip_endpoint *sip_endpoint)
2183 struct ast_sip_supplement *supplement;
2184 pjsip_cseq_hdr *cseq = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_CSEQ, NULL);
2185 struct ast_sip_contact *contact = ast_sip_mod_data_get(tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT);
2187 AST_RWLIST_RDLOCK(&supplements);
2188 AST_LIST_TRAVERSE(&supplements, supplement, next) {
2189 if (supplement->outgoing_response && does_method_match(&cseq->method.name, supplement->method)) {
2190 supplement->outgoing_response(sip_endpoint, contact, tdata);
2193 AST_RWLIST_UNLOCK(&supplements);
2195 ast_sip_mod_data_set(tdata->pool, tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT, NULL);
2196 ao2_cleanup(contact);
2198 return pjsip_endpt_send_response(ast_sip_get_pjsip_endpoint(), res_addr, tdata, NULL, NULL);
2201 int ast_sip_create_response(const pjsip_rx_data *rdata, int st_code,
2202 struct ast_sip_contact *contact, pjsip_tx_data **tdata)
2204 int res = pjsip_endpt_create_response(ast_sip_get_pjsip_endpoint(), rdata, st_code, NULL, tdata);
2207 ast_sip_mod_data_set((*tdata)->pool, (*tdata)->mod_data, supplement_module.id, MOD_DATA_CONTACT, ao2_bump(contact));
2213 static void remove_request_headers(pjsip_endpoint *endpt)
2215 const pjsip_hdr *request_headers = pjsip_endpt_get_request_headers(endpt);
2216 pjsip_hdr *iter = request_headers->next;
2218 while (iter != request_headers) {
2219 pjsip_hdr *to_erase = iter;
2221 pj_list_erase(to_erase);
2225 static int load_module(void)
2227 /* The third parameter is just copied from
2228 * example code from PJLIB. This can be adjusted
2232 struct ast_threadpool_options options;
2234 if (pj_init() != PJ_SUCCESS) {
2235 return AST_MODULE_LOAD_DECLINE;
2238 if (pjlib_util_init() != PJ_SUCCESS) {
2240 return AST_MODULE_LOAD_DECLINE;
2243 pj_caching_pool_init(&caching_pool, NULL, 1024 * 1024);
2244 if (pjsip_endpt_create(&caching_pool.factory, "SIP", &ast_pjsip_endpoint) != PJ_SUCCESS) {
2245 ast_log(LOG_ERROR, "Failed to create PJSIP endpoint structure. Aborting load\n");
2246 pj_caching_pool_destroy(&caching_pool);
2247 return AST_MODULE_LOAD_DECLINE;
2250 /* PJSIP will automatically try to add a Max-Forwards header. Since we want to control that,
2251 * we need to stop PJSIP from doing it automatically
2253 remove_request_headers(ast_pjsip_endpoint);
2255 memory_pool = pj_pool_create(&caching_pool.factory, "SIP", 1024, 1024, NULL);
2257 ast_log(LOG_ERROR, "Failed to create memory pool for SIP. Aborting load\n");
2258 pjsip_endpt_destroy(ast_pjsip_endpoint);
2259 ast_pjsip_endpoint = NULL;
2260 pj_caching_pool_destroy(&caching_pool);
2261 return AST_MODULE_LOAD_DECLINE;
2264 if (ast_sip_initialize_system()) {
2265 ast_log(LOG_ERROR, "Failed to initialize SIP 'system' configuration section. Aborting load\n");
2266 pj_pool_release(memory_pool);
2268 pjsip_endpt_destroy(ast_pjsip_endpoint);
2269 ast_pjsip_endpoint = NULL;
2270 pj_caching_pool_destroy(&caching_pool);
2271 return AST_MODULE_LOAD_DECLINE;
2274 sip_get_threadpool_options(&options);
2275 options.thread_start = sip_thread_start;
2276 sip_threadpool = ast_threadpool_create("SIP", NULL, &options);
2277 if (!sip_threadpool) {
2278 ast_log(LOG_ERROR, "Failed to create SIP threadpool. Aborting load\n");
2279 pj_pool_release(memory_pool);
2281 pjsip_endpt_destroy(ast_pjsip_endpoint);
2282 ast_pjsip_endpoint = NULL;
2283 pj_caching_pool_destroy(&caching_pool);
2284 return AST_MODULE_LOAD_DECLINE;
2287 pjsip_tsx_layer_init_module(ast_pjsip_endpoint);
2288 pjsip_ua_init_module(ast_pjsip_endpoint, NULL);
2290 monitor_continue = 1;
2291 status = pj_thread_create(memory_pool, "SIP", (pj_thread_proc *) &monitor_thread_exec,
2292 NULL, PJ_THREAD_DEFAULT_STACK_SIZE * 2, 0, &monitor_thread);
2293 if (status != PJ_SUCCESS) {
2294 ast_log(LOG_ERROR, "Failed to start SIP monitor thread. Aborting load\n");
2295 pj_pool_release(memory_pool);
2297 pjsip_endpt_destroy(ast_pjsip_endpoint);
2298 ast_pjsip_endpoint = NULL;
2299 pj_caching_pool_destroy(&caching_pool);
2300 return AST_MODULE_LOAD_DECLINE;
2303 ast_sip_initialize_global_headers();
2305 if (ast_res_pjsip_initialize_configuration(ast_module_info)) {
2306 ast_log(LOG_ERROR, "Failed to initialize SIP configuration. Aborting load\n");
2307 ast_sip_destroy_global_headers();
2308 stop_monitor_thread();
2309 pj_pool_release(memory_pool);
2311 pjsip_endpt_destroy(ast_pjsip_endpoint);
2312 ast_pjsip_endpoint = NULL;
2313 pj_caching_pool_destroy(&caching_pool);
2314 return AST_MODULE_LOAD_DECLINE;
2317 if (ast_sip_initialize_distributor()) {
2318 ast_log(LOG_ERROR, "Failed to register distributor module. Aborting load\n");
2319 ast_res_pjsip_destroy_configuration();
2320 ast_sip_destroy_global_headers();
2321 stop_monitor_thread();
2322 pj_pool_release(memory_pool);
2324 pjsip_endpt_destroy(ast_pjsip_endpoint);
2325 ast_pjsip_endpoint = NULL;
2326 pj_caching_pool_destroy(&caching_pool);
2327 return AST_MODULE_LOAD_DECLINE;
2330 if (ast_sip_register_service(&supplement_module)) {
2331 ast_log(LOG_ERROR, "Failed to initialize supplement hooks. Aborting load\n");
2332 ast_sip_destroy_distributor();
2333 ast_res_pjsip_destroy_configuration();
2334 ast_sip_destroy_global_headers();
2335 stop_monitor_thread();
2336 pj_pool_release(memory_pool);
2338 pjsip_endpt_destroy(ast_pjsip_endpoint);
2339 ast_pjsip_endpoint = NULL;
2340 pj_caching_pool_destroy(&caching_pool);
2341 return AST_MODULE_LOAD_DECLINE;
2344 if (ast_sip_initialize_outbound_authentication()) {
2345 ast_log(LOG_ERROR, "Failed to initialize outbound authentication. Aborting load\n");
2346 ast_sip_unregister_service(&supplement_module);
2347 ast_sip_destroy_distributor();
2348 ast_res_pjsip_destroy_configuration();
2349 ast_sip_destroy_global_headers();
2350 stop_monitor_thread();
2351 pj_pool_release(memory_pool);
2353 pjsip_endpt_destroy(ast_pjsip_endpoint);
2354 ast_pjsip_endpoint = NULL;
2355 pj_caching_pool_destroy(&caching_pool);
2356 return AST_MODULE_LOAD_DECLINE;
2359 ast_res_pjsip_init_options_handling(0);
2361 ast_module_ref(ast_module_info->self);
2363 return AST_MODULE_LOAD_SUCCESS;
2366 static int reload_module(void)
2368 if (ast_res_pjsip_reload_configuration()) {
2369 return AST_MODULE_LOAD_DECLINE;
2371 ast_res_pjsip_init_options_handling(1);
2375 static int unload_module(void)
2377 /* This will never get called as this module can't be unloaded */
2381 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Basic SIP resource",
2382 .load = load_module,
2383 .unload = unload_module,
2384 .reload = reload_module,
2385 .load_pri = AST_MODPRI_CHANNEL_DEPEND - 5,
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